Initial Commit

common/embeddedaudiosynthesis/arm-wt-22k/Android.mk

@@ -0,0 +1,104 @@
+LOCAL_PATH := $(call my-dir)
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES = \
+	lib_src/eas_chorus.c \
+	lib_src/eas_chorusdata.c \
+	lib_src/eas_data.c \
+	lib_src/eas_dlssynth.c \
+	lib_src/eas_flog.c \
+	lib_src/eas_ima_tables.c \
+	lib_src/eas_imaadpcm.c \
+	lib_src/eas_imelody.c \
+	lib_src/eas_imelodydata.c \
+	lib_src/eas_math.c \
+	lib_src/eas_mdls.c \
+	lib_src/eas_midi.c \
+	lib_src/eas_mididata.c \
+	lib_src/eas_mixbuf.c \
+	lib_src/eas_mixer.c \
+	lib_src/eas_ota.c \
+	lib_src/eas_otadata.c \
+	lib_src/eas_pan.c \
+	lib_src/eas_pcm.c \
+	lib_src/eas_pcmdata.c \
+	lib_src/eas_public.c \
+	lib_src/eas_reverb.c \
+	lib_src/eas_reverbdata.c \
+	lib_src/eas_rtttl.c \
+	lib_src/eas_rtttldata.c \
+	lib_src/eas_smf.c \
+	lib_src/eas_smfdata.c \
+	lib_src/eas_voicemgt.c \
+	lib_src/eas_wtengine.c \
+	lib_src/eas_wtsynth.c \
+	lib_src/eas_xmf.c \
+	lib_src/eas_xmfdata.c \
+	lib_src/wt_22khz.c \
+	lib_src/jet.c \
+	host_src/eas_config.c \
+	host_src/eas_hostmm.c \
+	host_src/eas_main.c \
+	host_src/eas_report.c \
+	host_src/eas_wave.c
+
+# not using these modules
+#	lib_src/eas_wavefile.c \
+#	lib_src/eas_wavefiledata.c \
+
+LOCAL_CFLAGS+= -O2 -D UNIFIED_DEBUG_MESSAGES -D EAS_WT_SYNTH \
+	-D _IMELODY_PARSER -D _RTTTL_PARSER -D _OTA_PARSER \
+	-D _XMF_PARSER -D NUM_OUTPUT_CHANNELS=2 \
+	-D _SAMPLE_RATE_22050 -D MAX_SYNTH_VOICES=64 \
+	-D _8_BIT_SAMPLES -D _FILTER_ENABLED \
+	-D DLS_SYNTHESIZER \
+	-D _REVERB_ENABLED
+	
+# not using these options
+# -D _WAVE_PARSER
+# -D _IMA_DECODER (needed for IMA-ADPCM wave files)
+# -D _CHORUS_ENABLED
+  	
+LOCAL_C_INCLUDES:= \
+	$(LOCAL_PATH)/host_src \
+	$(LOCAL_PATH)/lib_src
+
+LOCAL_ARM_MODE := arm
+
+LOCAL_MODULE := libsonivox
+
+LOCAL_COPY_HEADERS_TO := libsonivox
+LOCAL_COPY_HEADERS := \
+	host_src/eas.h \
+	host_src/eas_types.h \
+	host_src/eas_reverb.h \
+	host_src/jet.h
+
+ifeq ($(TARGET_ARCH),arm)
+LOCAL_SRC_FILES+= \
+	lib_src/ARM-E_filter_gnu.s \
+	lib_src/ARM-E_interpolate_loop_gnu.s \
+	lib_src/ARM-E_interpolate_noloop_gnu.s \
+	lib_src/ARM-E_mastergain_gnu.s \
+	lib_src/ARM-E_voice_gain_gnu.s
+
+asm_flags := \
+	-I $(LOCAL_PATH)/lib_src \
+	--defsym SAMPLE_RATE_22050=1 \
+	--defsym STEREO_OUTPUT=1 \
+	--defsym FILTER_ENABLED=1 \
+	--defsym SAMPLES_8_BIT=1
+
+LOCAL_CFLAGS+= -D NATIVE_EAS_KERNEL \
+	$(foreach f,$(asm_flags),-Wa,"$(f)")
+
+LOCAL_COPY_HEADERS += lib_src/ARM_synth_constants_gnu.inc
+endif
+
+LOCAL_SHARED_LIBRARIES := \
+	libutils libcutils
+
+LOCAL_LDLIBS := -lpthread
+
+include $(BUILD_SHARED_LIBRARY)
+

common/embeddedaudiosynthesis/arm-wt-22k/host_src/arm-wt-22k.mak

@@ -0,0 +1,25 @@
+#
+# Auto-generated sample makefile
+#
+# This makefile is intended for use with GNU make.
+# Set the paths to the tools (CC, AR, LD, etc.)
+#
+
+vpath %.c host_src
+
+CC = C:\Program Files\GNUARM\bin\arm-elf-gcc.exe
+AS = C:\Program Files\GNUARM\bin\arm-elf-as.exe
+LD = C:\Program Files\GNUARM\bin\arm-elf-gcc.exe
+AR = C:\Program Files\GNUARM\bin\arm-elf-ar.exe
+
+%.o: %.c
+	$(CC) -c -O2 -o $@ -I host_src -D UNIFIED_DEBUG_MESSAGES -D EAS_WT_SYNTH -D _IMELODY_PARSER -D _RTTTL_PARSER -D _OTA_PARSER -D _XMF_PARSER -D _WAVE_PARSER -D _REVERB_ENABLED -D _CHORUS_ENABLED -D MMAPI_SUPPORT -D JET_INTERFACE $<
+
+%.o: %.s
+	$(AS) -o $@ -EL -mcpu=arm946e-s -mfpu=softfpa -I lib_src --defsym CHECK_STACK=0 --defsym REVERB=0 --defsym CHORUS=0 --defsym STEREO_OUTPUT=1 --defsym SAMPLE_RATE_22050=1 --defsym SAMPLES_8_BIT=1 --defsym FILTER_ENABLED=1 $<
+
+OBJS = eas_main.o eas_report.o eas_wave.o eas_hostmm.o eas_config.o
+
+arm-wt-22k: $(OBJS)
+	$(LD) -o $@ $(OBJS) libarm-wt-22k.a -lm
+

common/embeddedaudiosynthesis/arm-wt-22k/host_src/eas_build.h

@@ -0,0 +1,36 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * host_src\eas_build.h
+ *
+ * Contents and purpose:
+ * This file contains the build configuration for this
+ * build. The buildGUIDStr is a GUID created during
+ * the build process and is guaranteed to be unique
+ * for each build.
+ *
+ * Copyright Sonic Network Inc. 2006
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ * This file was autogenerated by buildid.exe
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _GUID_1feda229b9a845e996f473c0a80e7220_
+#define _GUID_1feda229b9a845e996f473c0a80e7220_
+
+#define _BUILD_VERSION_ "1feda229-b9a8-45e9-96f4-73c0a80e7220"
+#define _BUILD_TIME_ 0x4743badd
+
+#endif /* _GUID_1feda229b9a845e996f473c0a80e7220_ */

common/embeddedaudiosynthesis/arm-wt-22k/host_src/eas_chorus.h

@@ -0,0 +1,53 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_chorus.h
+ *
+ * Contents and purpose:
+ * Contains parameter enumerations for the Chorus effect
+ *
+ *
+ * Copyright Sonic Network Inc. 2006
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 309 $
+ *   $Date: 2006-09-12 18:52:45 -0700 (Tue, 12 Sep 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef EAS_CHORUS_H
+#define EAS_CHORUS_H
+
+/* enumerated parameter settings for Chorus effect */
+typedef enum
+{
+    EAS_PARAM_CHORUS_BYPASS,
+    EAS_PARAM_CHORUS_PRESET,
+    EAS_PARAM_CHORUS_RATE,
+    EAS_PARAM_CHORUS_DEPTH,
+    EAS_PARAM_CHORUS_LEVEL
+} E_CHORUS_PARAMS;
+
+typedef enum
+{
+    EAS_PARAM_CHORUS_PRESET1,
+    EAS_PARAM_CHORUS_PRESET2,
+    EAS_PARAM_CHORUS_PRESET3,
+    EAS_PARAM_CHORUS_PRESET4
+} E_CHORUS_PRESETS;
+
+
+#endif

common/embeddedaudiosynthesis/arm-wt-22k/host_src/eas_config.c

@@ -0,0 +1,619 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_config.c
+ *
+ * Contents and purpose:
+ * This file contains the Configuration Module interface (CM). The CM
+ * is a module compiled external to the library that sets the configuration
+ * for this build. It allows the library to find optional components and
+ * links to static memory allocations (when used in a static configuration).
+ *
+ * DO NOT MODIFY THIS FILE!
+ *
+ * NOTE: This module is not intended to be modified by the customer. It
+ * needs to be included in the build process with the correct configuration
+ * defines (see the library documentation for information on how to configure
+ * the library).
+ *
+ * Copyright Sonic Network Inc. 2004-2006
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 796 $
+ *   $Date: 2007-08-01 00:15:25 -0700 (Wed, 01 Aug 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#include "eas.h"
+#include "eas_config.h"
+
+
+#ifdef _MFI_PARSER
+/*----------------------------------------------------------------------------
+ * Vendor/Device ID for MFi Extensions
+ *
+ * Define the preprocessor symbols to establish the vendor ID and
+ * device ID for the MFi PCM/ADPCM extensions.
+ *----------------------------------------------------------------------------
+*/
+const EAS_U8 eas_MFIVendorIDMSB = (MFI_VENDOR_ID >> 8) & 0xff;
+const EAS_U8 eas_MFIVendorIDLSB = MFI_VENDOR_ID & 0xff;
+const EAS_U8 eas_MFIDeviceID = MFI_DEVICE_ID;
+#endif
+
+/*----------------------------------------------------------------------------
+ *
+ * parserModules
+ *
+ * This structure is used by the EAS library to locate file parsing
+ * modules.
+ *----------------------------------------------------------------------------
+*/
+
+/* define the external file parsers */
+extern EAS_VOID_PTR EAS_SMF_Parser;
+
+#ifdef _XMF_PARSER
+extern EAS_VOID_PTR EAS_XMF_Parser;
+#endif
+
+#ifdef _SMAF_PARSER
+extern EAS_VOID_PTR EAS_SMAF_Parser;
+#endif
+
+#ifdef _WAVE_PARSER
+extern EAS_VOID_PTR EAS_Wave_Parser;
+#endif
+
+#ifdef _OTA_PARSER
+extern EAS_VOID_PTR EAS_OTA_Parser;
+#endif
+
+#ifdef _IMELODY_PARSER
+extern EAS_VOID_PTR EAS_iMelody_Parser;
+#endif
+
+#ifdef _RTTTL_PARSER
+extern EAS_VOID_PTR EAS_RTTTL_Parser;
+#endif
+
+#if defined (_CMX_PARSER) || defined(_MFI_PARSER)
+extern EAS_VOID_PTR EAS_CMF_Parser;
+#endif
+
+/* initalize pointers to parser interfaces */
+/*lint -e{605} not pretty, but it works */
+EAS_VOID_PTR const parserModules[] =
+{
+    &EAS_SMF_Parser,
+
+#ifdef _XMF_PARSER
+    &EAS_XMF_Parser,
+#endif
+
+#ifdef _WAVE_PARSER
+    &EAS_Wave_Parser,
+#endif
+
+#ifdef _SMAF_PARSER
+    &EAS_SMAF_Parser,
+#endif
+
+#ifdef _OTA_PARSER
+    &EAS_OTA_Parser,
+#endif
+
+#ifdef _IMELODY_PARSER
+    &EAS_iMelody_Parser,
+#endif
+
+#ifdef _RTTTL_PARSER
+    &EAS_RTTTL_Parser,
+#endif
+
+#if defined (_CMX_PARSER) || defined(_MFI_PARSER)
+    &EAS_CMF_Parser
+#endif
+};
+#define NUM_PARSER_MODULES (sizeof(parserModules) / sizeof(EAS_VOID_PTR))
+
+/*----------------------------------------------------------------------------
+ * Data Modules
+ *----------------------------------------------------------------------------
+*/
+
+#ifdef _STATIC_MEMORY
+extern EAS_VOID_PTR eas_SMFData;
+extern EAS_VOID_PTR eas_Data;
+extern EAS_VOID_PTR eas_MixBuffer;
+extern EAS_VOID_PTR eas_Synth;
+extern EAS_VOID_PTR eas_MIDI;
+extern EAS_VOID_PTR eas_PCMData;
+extern EAS_VOID_PTR eas_MIDIData;
+
+#ifdef _XMF_PARSER
+extern EAS_VOID_PTR eas_XMFData;
+#endif
+
+#ifdef _SMAF_PARSER
+extern EAS_VOID_PTR eas_SMAFData;
+#endif
+
+#ifdef _OTA_PARSER
+extern EAS_VOID_PTR eas_OTAData;
+#endif
+
+#ifdef _IMELODY_PARSER
+extern EAS_VOID_PTR eas_iMelodyData;
+#endif
+
+#ifdef _RTTTL_PARSER
+extern EAS_VOID_PTR eas_RTTTLData;
+#endif
+
+#ifdef _WAVE_PARSER
+extern EAS_VOID_PTR eas_WaveData;
+#endif
+
+#if defined (_CMX_PARSER) || defined(_MFI_PARSER)
+extern EAS_VOID_PTR eas_CMFData;
+#endif
+#endif
+
+/*----------------------------------------------------------------------------
+ *
+ * Effects Modules
+ *
+ * These declarations are used by the EAS library to locate
+ * effects modules.
+ *----------------------------------------------------------------------------
+*/
+
+#ifdef _ENHANCER_ENABLED
+extern EAS_VOID_PTR                 EAS_Enhancer;
+#define EAS_ENHANCER_INTERFACE      &EAS_Enhancer
+#ifdef _STATIC_MEMORY
+extern EAS_VOID_PTR                 eas_EnhancerData;
+#define EAS_ENHANCER_DATA           &eas_EnhancerData
+#else
+#define EAS_ENHANCER_DATA           NULL
+#endif
+#else
+#define EAS_ENHANCER_INTERFACE      NULL
+#define EAS_ENHANCER_DATA           NULL
+#endif
+
+#ifdef _COMPRESSOR_ENABLED
+extern EAS_VOID_PTR                 EAS_Compressor;
+#define EAS_COMPRESSOR_INTERFACE    &EAS_Compressor
+#ifdef _STATIC_MEMORY
+extern EAS_VOID_PTR                 eas_CompressorData;
+#define EAS_COMPRESSOR_DATA         &eas_CompressorData
+#else
+#define EAS_COMPRESSOR_DATA         NULL
+#endif
+#else
+#define EAS_COMPRESSOR_INTERFACE    NULL
+#define EAS_COMPRESSOR_DATA         NULL
+#endif
+
+#ifdef _MAXIMIZER_ENABLED
+extern EAS_VOID_PTR                 EAS_Maximizer;
+#define EAS_MAXIMIZER_INTERFACE     &EAS_Maximizer
+#ifdef _STATIC_MEMORY
+extern EAS_VOID_PTR                 eas_MaximizerData;
+#define EAS_MAXIMIZER_DATA          &eas_MaximizerData
+#else
+#define EAS_MAXIMIZER_DATA          NULL
+#endif
+#else
+#define EAS_MAXIMIZER_INTERFACE NULL
+#define EAS_MAXIMIZER_DATA          NULL
+#endif
+
+
+#ifdef _REVERB_ENABLED
+extern EAS_VOID_PTR                 EAS_Reverb;
+#define EAS_REVERB_INTERFACE        &EAS_Reverb
+#ifdef _STATIC_MEMORY
+extern EAS_VOID_PTR                 eas_ReverbData;
+#define EAS_REVERB_DATA             &eas_ReverbData
+#else
+#define EAS_REVERB_DATA             NULL
+#endif
+#else
+#define EAS_REVERB_INTERFACE        NULL
+#define EAS_REVERB_DATA             NULL
+#endif
+
+#ifdef _CHORUS_ENABLED
+extern EAS_VOID_PTR                 EAS_Chorus;
+#define EAS_CHORUS_INTERFACE        &EAS_Chorus
+#ifdef _STATIC_MEMORY
+extern EAS_VOID_PTR                 eas_ChorusData;
+#define EAS_CHORUS_DATA             &eas_ChorusData
+#else
+#define EAS_CHORUS_DATA             NULL
+#endif
+#else
+#define EAS_CHORUS_INTERFACE        NULL
+#define EAS_CHORUS_DATA             NULL
+#endif
+
+#ifdef _WIDENER_ENABLED
+extern EAS_VOID_PTR                 EAS_Widener;
+#define EAS_WIDENER_INTERFACE       &EAS_Widener
+#ifdef _STATIC_MEMORY
+extern EAS_VOID_PTR                 eas_WidenerData;
+#define EAS_WIDENER_DATA            &eas_WidenerData
+#else
+#define EAS_WIDENER_DATA            NULL
+#endif
+#else
+#define EAS_WIDENER_INTERFACE       NULL
+#define EAS_WIDENER_DATA            NULL
+#endif
+
+#ifdef _GRAPHIC_EQ_ENABLED
+extern EAS_VOID_PTR                 EAS_GraphicEQ;
+#define EAS_GRAPHIC_EQ_INTERFACE    &EAS_GraphicEQ
+#ifdef _STATIC_MEMORY
+extern EAS_VOID_PTR                 eas_GraphicEQData;
+#define EAS_GRAPHIC_EQ_DATA         &eas_GraphicEQData
+#else
+#define EAS_GRAPHIC_EQ_DATA         NULL
+#endif
+#else
+#define EAS_GRAPHIC_EQ_INTERFACE    NULL
+#define EAS_GRAPHIC_EQ_DATA         NULL
+#endif
+
+#ifdef _WOW_ENABLED
+extern EAS_VOID_PTR                 EAS_Wow;
+#define EAS_WOW_INTERFACE           &EAS_Wow
+#ifdef _STATIC_MEMORY
+#error "WOW module requires dynamic memory model"
+#else
+#define EAS_WOW_DATA                NULL
+#endif
+#else
+#define EAS_WOW_INTERFACE           NULL
+#define EAS_WOW_DATA                NULL
+#endif
+
+#ifdef _TONECONTROLEQ_ENABLED
+extern EAS_VOID_PTR                 EAS_ToneControlEQ;
+#define EAS_TONECONTROLEQ_INTERFACE &EAS_ToneControlEQ
+#ifdef _STATIC_MEMORY
+extern EAS_VOID_PTR                 eas_ToneControlEQData;
+#define EAS_TONECONTROLEQ_DATA      &eas_ToneControlEQData
+#else
+#define EAS_TONECONTROLEQ_DATA      NULL
+#endif
+#else
+#define EAS_TONECONTROLEQ_INTERFACE NULL
+#define EAS_TONECONTROLEQ_DATA      NULL
+#endif
+
+/*lint -e{605} not pretty, but it works */
+EAS_VOID_PTR const effectsModules[] =
+{
+    EAS_ENHANCER_INTERFACE,
+    EAS_COMPRESSOR_INTERFACE,
+    EAS_REVERB_INTERFACE,
+    EAS_CHORUS_INTERFACE,
+    EAS_WIDENER_INTERFACE,
+    EAS_GRAPHIC_EQ_INTERFACE,
+    EAS_WOW_INTERFACE,
+    EAS_MAXIMIZER_INTERFACE,
+    EAS_TONECONTROLEQ_INTERFACE
+};
+
+EAS_VOID_PTR const effectsData[] =
+{
+    EAS_ENHANCER_DATA,
+    EAS_COMPRESSOR_DATA,
+    EAS_REVERB_DATA,
+    EAS_CHORUS_DATA,
+    EAS_WIDENER_DATA,
+    EAS_GRAPHIC_EQ_DATA,
+    EAS_WOW_DATA,
+    EAS_MAXIMIZER_DATA,
+    EAS_TONECONTROLEQ_DATA
+};
+
+/*----------------------------------------------------------------------------
+ *
+ * Optional Modules
+ *
+ * These declarations are used by the EAS library to locate
+ * effects modules.
+ *----------------------------------------------------------------------------
+*/
+
+#ifdef _METRICS_ENABLED
+extern EAS_VOID_PTR                 EAS_Metrics;
+#define EAS_METRICS_INTERFACE       &EAS_Metrics
+#ifdef _STATIC_MEMORY
+extern EAS_VOID_PTR                 eas_MetricsData;
+#define EAS_METRICS_DATA            &eas_MetricsData
+#else
+#define EAS_METRICS_DATA            NULL
+#endif
+#else
+#define EAS_METRICS_INTERFACE       NULL
+#define EAS_METRICS_DATA            NULL
+#endif
+
+#ifdef MMAPI_SUPPORT
+extern EAS_VOID_PTR                 EAS_TC_Parser;
+#define EAS_TONE_CONTROL_PARSER     &EAS_TC_Parser
+#ifdef _STATIC_MEMORY
+extern EAS_VOID_PTR                 eas_TCData;
+#define EAS_TONE_CONTROL_DATA       &eas_TCData
+#else
+#define EAS_TONE_CONTROL_DATA       NULL
+#endif
+#else
+#define EAS_TONE_CONTROL_PARSER     NULL
+#define EAS_TONE_CONTROL_DATA       NULL
+#endif
+
+/*lint -e{605} not pretty, but it works */
+EAS_VOID_PTR const optionalModules[] =
+{
+    EAS_TONE_CONTROL_PARSER,
+    EAS_METRICS_INTERFACE
+};
+
+EAS_VOID_PTR const optionalData[] =
+{
+    EAS_TONE_CONTROL_DATA,
+    EAS_METRICS_DATA
+};
+
+/*----------------------------------------------------------------------------
+ * EAS_CMStaticMemoryModel()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * This function returns true if EAS has been configured for
+ * a static memory model. There are some limitations in the
+ * static memory model, see the documentation for more
+ * information.
+ *
+ * Outputs:
+ *  returns EAS_TRUE if a module is found
+ *----------------------------------------------------------------------------
+*/
+EAS_BOOL EAS_CMStaticMemoryModel (void)
+{
+#ifdef _STATIC_MEMORY
+    return EAS_TRUE;
+#else
+    return EAS_FALSE;
+#endif
+}
+
+/*----------------------------------------------------------------------------
+ * EAS_CMEnumModules()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * This function is used to find pointers to optional modules.
+ *
+ * Inputs:
+ *  module          - module number
+ *
+ * Outputs:
+ *  returns a pointer to the module function table or NULL if no module
+ *----------------------------------------------------------------------------
+*/
+EAS_VOID_PTR EAS_CMEnumModules (EAS_INT module)
+{
+
+    if (module >= (EAS_INT) NUM_PARSER_MODULES)
+        return NULL;
+    return parserModules[module];
+}
+
+/*----------------------------------------------------------------------------
+ * EAS_CMEnumData()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * This function is used to find pointers to static memory allocations.
+ *
+ * Inputs:
+ *  dataModule          - enumerated module number
+ *
+ * Outputs:
+ * Returns handle to data or NULL if not found
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, dataModule) used only when _STATIC_MEMORY is defined */
+EAS_VOID_PTR EAS_CMEnumData (EAS_INT dataModule)
+{
+
+#ifdef _STATIC_MEMORY
+    switch (dataModule)
+    {
+
+    /* main instance data for synthesizer */
+    case EAS_CM_EAS_DATA:
+        return &eas_Data;
+
+    /* mix buffer for mix engine */
+    case EAS_CM_MIX_BUFFER:
+        /*lint -e{545} lint doesn't like this because it sees the underlying type */
+        return &eas_MixBuffer;
+
+    /* instance data for synth */
+    case EAS_CM_SYNTH_DATA:
+        return &eas_Synth;
+
+    /* instance data for MIDI parser */
+    case EAS_CM_MIDI_DATA:
+        return &eas_MIDI;
+
+    /* instance data for SMF parser */
+    case EAS_CM_SMF_DATA:
+        return &eas_SMFData;
+
+#ifdef _XMF_PARSER
+    /* instance data for XMF parser */
+    case EAS_CM_XMF_DATA:
+        return &eas_XMFData;
+#endif
+
+#ifdef _SMAF_PARSER
+    /* instance data for SMAF parser */
+    case EAS_CM_SMAF_DATA:
+        return &eas_SMAFData;
+#endif
+
+    /* instance data for the PCM engine */
+    case EAS_CM_PCM_DATA:
+        /*lint -e{545} lint doesn't like this because it sees the underlying type */
+        return &eas_PCMData;
+
+    case EAS_CM_MIDI_STREAM_DATA:
+        return &eas_MIDIData;
+
+#ifdef _OTA_PARSER
+    /* instance data for OTA parser */
+    case EAS_CM_OTA_DATA:
+        return &eas_OTAData;
+#endif
+
+#ifdef _IMELODY_PARSER
+    /* instance data for iMelody parser */
+    case EAS_CM_IMELODY_DATA:
+        return &eas_iMelodyData;
+#endif
+
+#ifdef _RTTTL_PARSER
+    /* instance data for RTTTL parser */
+    case EAS_CM_RTTTL_DATA:
+        return &eas_RTTTLData;
+#endif
+
+#ifdef _WAVE_PARSER
+    /* instance data for WAVE parser */
+    case EAS_CM_WAVE_DATA:
+        return &eas_WaveData;
+#endif
+
+#if defined (_CMX_PARSER) || defined(_MFI_PARSER)
+    /* instance data for CMF parser */
+    case EAS_CM_CMF_DATA:
+        return &eas_CMFData;
+#endif
+
+    default:
+        return NULL;
+    }
+
+#else
+    return NULL;
+#endif
+}
+
+/*----------------------------------------------------------------------------
+ * EAS_CMEnumFXModules()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * This function is used to find pointers to optional effects modules.
+ *
+ * Inputs:
+ *  module          - enumerated module number
+ *  pModule         - pointer to module interface
+ *
+ * Outputs:
+ *  Returns pointer to function table or NULL if not found
+ *----------------------------------------------------------------------------
+*/
+EAS_VOID_PTR EAS_CMEnumFXModules (EAS_INT module)
+{
+
+    if (module >= NUM_EFFECTS_MODULES)
+        return NULL;
+    return effectsModules[module];
+}
+
+/*----------------------------------------------------------------------------
+ * EAS_CMEnumFXData()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * This function is used to find pointers to static memory allocations.
+ *
+ * Inputs:
+ *  dataModule          - enumerated module number
+ *  pData               - pointer to handle variable
+ *
+ * Outputs:
+ * Returns handle to data or NULL if not found
+ *----------------------------------------------------------------------------
+*/
+EAS_VOID_PTR EAS_CMEnumFXData (EAS_INT dataModule)
+{
+
+    if (dataModule >= NUM_EFFECTS_MODULES)
+        return NULL;
+    return effectsData[dataModule];
+}
+
+/*----------------------------------------------------------------------------
+ * EAS_CMEnumOptModules()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * This function is used to find pointers to optional modules.
+ *
+ * Inputs:
+ *  module          - enumerated module number
+ *
+ * Outputs:
+ *  returns pointer to function table or NULL if no module
+ *----------------------------------------------------------------------------
+*/
+EAS_VOID_PTR EAS_CMEnumOptModules (EAS_INT module)
+{
+
+    /* sanity check */
+    if (module >= NUM_OPTIONAL_MODULES)
+        return EAS_FALSE;
+    return optionalModules[module];
+}
+
+/*----------------------------------------------------------------------------
+ * EAS_CMEnumOptData()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * This function is used to find pointers to static memory allocations.
+ *
+ * Inputs:
+ *  dataModule          - enumerated module number
+ *
+ * Outputs:
+ * Returns handle to data or NULL if not found
+ *----------------------------------------------------------------------------
+*/
+EAS_VOID_PTR EAS_CMEnumOptData (EAS_INT dataModule)
+{
+
+    if (dataModule >= NUM_OPTIONAL_MODULES)
+        return NULL;
+    return optionalData[dataModule];
+}
+
+

common/embeddedaudiosynthesis/arm-wt-22k/host_src/eas_config.h

@@ -0,0 +1,191 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_config.h
+ *
+ * Contents and purpose:
+ * This header declares the Configuration Module interface (CM). The CM
+ * is a module compiled external to the library that sets the configuration
+ * for this build. It allows the library to find optional components and
+ * links to static memory allocations (when used in a static configuration).
+ *
+ * NOTE: This module is not intended to be modified by the customer. It
+ * needs to be included in the build process with the correct configuration
+ * defines (see the library documentation for information on how to configure
+ * the library).
+ *
+ * DO NOT MODIFY THIS FILE!
+ *
+ * Copyright 2005 Sonic Network Inc.
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 82 $
+ *   $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+// sentinel
+#ifndef _EAS_CONFIG_H
+#define _EAS_CONFIG_H
+
+#include "eas_types.h"
+
+/* list of enumerators for optional modules */
+typedef enum {
+    EAS_CM_FILE_PARSERS = 1
+} E_CM_ENUM_MODULES;
+
+/* list of enumerators for module and memory pointers */
+typedef enum {
+    EAS_CM_EAS_DATA = 1,
+    EAS_CM_MIX_BUFFER,
+    EAS_CM_SYNTH_DATA,
+    EAS_CM_MIDI_DATA,
+    EAS_CM_SMF_DATA,
+    EAS_CM_XMF_DATA,
+    EAS_CM_SMAF_DATA,
+    EAS_CM_PCM_DATA,
+    EAS_CM_MIDI_STREAM_DATA,
+    EAS_CM_METRICS_DATA,
+    EAS_CM_OTA_DATA,
+    EAS_CM_IMELODY_DATA,
+    EAS_CM_RTTTL_DATA,
+    EAS_CM_WAVE_DATA,
+    EAS_CM_CMF_DATA
+} E_CM_DATA_MODULES;
+
+typedef struct
+{
+    int maxSMFStreams;
+    void *pSMFData;
+    void *pSMFStream;
+} S_EAS_SMF_PTRS;
+
+typedef struct
+{
+    int maxSMAFStreams;
+    void *pSMAFData;
+    void *pSMAFStream;
+} S_EAS_SMAF_PTRS;
+
+/*----------------------------------------------------------------------------
+ * EAS_CMStaticMemoryModel()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * This function returns true if EAS has been configured for
+ * a static memory model. There are some limitations in the
+ * static memory model, see the documentation for more
+ * information.
+ *
+ * Outputs:
+ *  returns EAS_TRUE if a module is found
+ *----------------------------------------------------------------------------
+*/
+EAS_BOOL EAS_CMStaticMemoryModel (void);
+
+/*----------------------------------------------------------------------------
+ * EAS_CMEnumModules()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * This function is used to find pointers to optional modules.
+ *
+ * Inputs:
+ *  module          - module number
+ *
+ * Outputs:
+ *  returns a pointer to the module function table or NULL if no module
+ *----------------------------------------------------------------------------
+*/
+EAS_VOID_PTR EAS_CMEnumModules (EAS_INT module);
+
+/*----------------------------------------------------------------------------
+ * EAS_CMEnumData()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * This function is used to find pointers to static memory allocations.
+ *
+ * Inputs:
+ *  dataModule          - enumerated module number
+ *
+ * Outputs:
+ * Returns handle to data or NULL if not found
+ *----------------------------------------------------------------------------
+*/
+EAS_VOID_PTR EAS_CMEnumData (EAS_INT dataModule);
+
+/*----------------------------------------------------------------------------
+ * EAS_CMEnumFXModules()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * This function is used to find pointers to optional effects modules.
+ *
+ * Inputs:
+ *  module          - enumerated module number
+ *  pModule         - pointer to module interface
+ *
+ * Outputs:
+ *  Returns pointer to function table or NULL if not found
+ *----------------------------------------------------------------------------
+*/
+EAS_VOID_PTR EAS_CMEnumFXModules (EAS_INT module);
+
+/*----------------------------------------------------------------------------
+ * EAS_CMEnumFXData()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * This function is used to find pointers to static memory allocations.
+ *
+ * Inputs:
+ *  dataModule          - enumerated module number
+ *  pData               - pointer to handle variable
+ *
+ * Outputs:
+ * Returns handle to data or NULL if not found
+ *----------------------------------------------------------------------------
+*/
+EAS_VOID_PTR EAS_CMEnumFXData (EAS_INT dataModule);
+
+/*----------------------------------------------------------------------------
+ * EAS_CMEnumOptModules()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * This function is used to find pointers to optional modules.
+ *
+ * Inputs:
+ *  module          - enumerated module number
+ *
+ * Outputs:
+ *  returns pointer to function table or NULL if no module
+ *----------------------------------------------------------------------------
+*/
+EAS_VOID_PTR EAS_CMEnumOptModules (EAS_INT module);
+
+/*----------------------------------------------------------------------------
+ * EAS_CMEnumOptData()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * This function is used to find pointers to static memory allocations.
+ *
+ * Inputs:
+ *  dataModule          - enumerated module number
+ *
+ * Outputs:
+ * Returns handle to data or NULL if not found
+ *----------------------------------------------------------------------------
+*/
+EAS_VOID_PTR EAS_CMEnumOptData (EAS_INT dataModule);
+
+#endif /* end _EAS_CONFIG_H */

common/embeddedaudiosynthesis/arm-wt-22k/host_src/eas_debugmsgs.h

@@ -0,0 +1,87 @@
+/* Auto-generated from source file: eas_chorusdata.c */
+/* Auto-generated from source file: eas_xmfdata.c */
+/* Auto-generated from source file: eas_mdls.c */
+{ 0x19299ed4, 0x00000022, "eas_mdls.c[2561]: #Instruments: %u\n" },
+{ 0x19299ed4, 0x00000023, "eas_mdls.c[2562]: #Regions: %u\n" },
+{ 0x19299ed4, 0x00000024, "eas_mdls.c[2563]: #Articulations: %u\n" },
+{ 0x19299ed4, 0x00000025, "eas_mdls.c[2564]: #Waves: %u\n" },
+{ 0x19299ed4, 0x00000026, "eas_mdls.c[2569]: Locale: %lu:%lu:%lu\n" },
+{ 0x19299ed4, 0x00000027, "eas_mdls.c[2576]: \tRegion: %u\n" },
+{ 0x19299ed4, 0x00000028, "eas_mdls.c[2578]: \t\tm_nGain: %d\n" },
+{ 0x19299ed4, 0x00000029, "eas_mdls.c[2579]: \t\trangeLow:rangeHigh %u:%u\n" },
+{ 0x19299ed4, 0x0000002a, "eas_mdls.c[2580]: \t\tm_nKeyGroupAndFlags: %04x\n" },
+{ 0x19299ed4, 0x0000002b, "eas_mdls.c[2581]: \t\tm_nLoopStart: %u\n" },
+{ 0x19299ed4, 0x0000002c, "eas_mdls.c[2582]: \t\tm_nLoopEnd: %u\n" },
+{ 0x19299ed4, 0x0000002d, "eas_mdls.c[2583]: \t\tm_nNetTuningInCents: %d\n" },
+{ 0x19299ed4, 0x0000002e, "eas_mdls.c[2584]: \t\tArt Index: %u\n" },
+{ 0x19299ed4, 0x0000002f, "eas_mdls.c[2585]: \t\tWave Index: %u\n" },
+{ 0x19299ed4, 0x00000030, "eas_mdls.c[2597]: Articulation: %u\n" },
+{ 0x19299ed4, 0x00000031, "eas_mdls.c[2599]: \t\tm_nEG2toFilterDepth: %d\n" },
+{ 0x19299ed4, 0x00000032, "eas_mdls.c[2600]: \t\tm_nEG2toPitchDepth: %d\n" },
+{ 0x19299ed4, 0x00000033, "eas_mdls.c[2601]: \t\tm_nFilterCutoffFrequency: %d\n" },
+{ 0x19299ed4, 0x00000034, "eas_mdls.c[2602]: \t\tm_nFilterResonance: %u\n" },
+{ 0x19299ed4, 0x00000035, "eas_mdls.c[2603]: \t\tm_nLFOAmplitudeDepth: %d\n" },
+{ 0x19299ed4, 0x00000036, "eas_mdls.c[2604]: \t\tm_nLFODelayTime: %d\n" },
+{ 0x19299ed4, 0x00000037, "eas_mdls.c[2605]: \t\tm_nLFOFrequency: %d\n" },
+{ 0x19299ed4, 0x00000038, "eas_mdls.c[2606]: \t\tm_nLFOPitchDepth: %d\n" },
+{ 0x19299ed4, 0x00000039, "eas_mdls.c[2607]: \t\tm_nPan: %d\n" },
+{ 0x19299ed4, 0x0000003a, "eas_mdls.c[2610]: \t\tm_nAttack: %d\n" },
+{ 0x19299ed4, 0x0000003b, "eas_mdls.c[2611]: \t\tm_nDecay: %d\n" },
+{ 0x19299ed4, 0x0000003c, "eas_mdls.c[2612]: \t\tm_nSustain: %d\n" },
+{ 0x19299ed4, 0x0000003d, "eas_mdls.c[2613]: \t\tm_nRelease: %d\n" },
+{ 0x19299ed4, 0x0000003e, "eas_mdls.c[2616]: \t\tm_nAttack: %d\n" },
+{ 0x19299ed4, 0x0000003f, "eas_mdls.c[2617]: \t\tm_nDecay: %d\n" },
+{ 0x19299ed4, 0x00000040, "eas_mdls.c[2618]: \t\tm_nSustain: %d\n" },
+{ 0x19299ed4, 0x00000041, "eas_mdls.c[2619]: \t\tm_nRelease: %d\n" },
+{ 0x19299ed4, 0x00000042, "eas_mdls.c[2626]: Wave Index: %d\n" },
+{ 0x19299ed4, 0x00000043, "eas_mdls.c[2627]: \tSize %u\n" },
+{ 0x19299ed4, 0x00000044, "eas_mdls.c[2628]: \tPointer %08lx\n" },
+/* Auto-generated from source file: eas_mididata.c */
+/* Auto-generated from source file: eas_pan.c */
+/* Auto-generated from source file: eas_wavefiledata.c */
+/* Auto-generated from source file: eas_reverbdata.c */
+/* Auto-generated from source file: eas_imelodydata.c */
+/* Auto-generated from source file: eas_ota.c */
+/* Auto-generated from source file: eas_mixbuf.c */
+/* Auto-generated from source file: eas_tcdata.c */
+/* Auto-generated from source file: jet.c */
+/* Auto-generated from source file: eas_rtttl.c */
+/* Auto-generated from source file: eas_reverb.c */
+/* Auto-generated from source file: eas_pcmdata.c */
+/* Auto-generated from source file: eas_chorus.c */
+/* Auto-generated from source file: eas_math.c */
+/* Auto-generated from source file: eas_xmf.c */
+/* Auto-generated from source file: eas_smfdata.c */
+/* Auto-generated from source file: eas_imelody.c */
+/* Auto-generated from source file: eas_dlssynth.c */
+/* Auto-generated from source file: eas_public.c */
+/* Auto-generated from source file: eas_rtttldata.c */
+/* Auto-generated from source file: eas_voicemgt.c */
+/* Auto-generated from source file: eas_tonecontrol.c */
+/* Auto-generated from source file: eas_wtengine.c */
+/* Auto-generated from source file: eas_imaadpcm.c */
+{ 0x2380b977, 0x00000006, "eas_imaadpcm.c[305]: IMADecoderLocate: Time=%d, samples=%d\n" },
+{ 0x2380b977, 0x00000007, "eas_imaadpcm.c[328]: IMADecoderLocate: Looped sample, numBlocks=%d, samplesPerLoop=%d, samplesInLastBlock=%d, samples=%d\n" },
+{ 0x2380b977, 0x00000008, "eas_imaadpcm.c[335]: IMADecoderLocate: Byte location in audio = %d\n" },
+{ 0x2380b977, 0x00000009, "eas_imaadpcm.c[345]: IMADecoderLocate: bytesLeft = %d\n" },
+/* Auto-generated from source file: eas_flog.c */
+/* Auto-generated from source file: eas_midi.c */
+/* Auto-generated from source file: eas_otadata.c */
+/* Auto-generated from source file: eas_ima_tables.c */
+/* Auto-generated from source file: eas_data.c */
+/* Auto-generated from source file: eas_pcm.c */
+/* Auto-generated from source file: eas_mixer.c */
+/* Auto-generated from source file: eas_wavefile.c */
+/* Auto-generated from source file: eas_wtsynth.c */
+/* Auto-generated from source file: eas_smf.c */
+/* Auto-generated from source file: eas_wave.c */
+/* Auto-generated from source file: eas_hostmm.c */
+{ 0x1a54b6e8, 0x00000001, "eas_hostmm.c[586]: Vibrate state: %d\n" },
+{ 0x1a54b6e8, 0x00000002, "eas_hostmm.c[601]: LED state: %d\n" },
+{ 0x1a54b6e8, 0x00000003, "eas_hostmm.c[616]: Backlight state: %d\n" },
+{ 0x1a54b6e8, 0x00000004, "eas_hostmm.c[162]: HWMemCpy: bad amount: %d\n" },
+{ 0x1a54b6e8, 0x00000005, "eas_hostmm.c[179]: HWMemSet: bad amount: %d\n" },
+{ 0x1a54b6e8, 0x00000006, "eas_hostmm.c[196]: HWMemCmp: bad amount: %d\n" },
+/* Auto-generated from source file: eas_config.c */
+/* Auto-generated from source file: eas_main.c */
+{ 0xe624f4d9, 0x00000005, "eas_main.c[106]: Play length: %d.%03d (secs)\n" },

common/embeddedaudiosynthesis/arm-wt-22k/host_src/eas_host.h

@@ -0,0 +1,87 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_host.h
+ *
+ * Contents and purpose:
+ * This header defines the host wrapper functions for stdio, stdlib, etc.
+ * The host application must provide an abstraction layer for these functions
+ * to support certain features, such as SMAF and SMF-1 conversion.
+ *
+ * DO NOT MODIFY THIS FILE!
+ *
+ * Copyright 2005 Sonic Network Inc.
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 82 $
+ *   $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+// sentinel
+#ifndef _EAS_HOST_H
+#define _EAS_HOST_H
+
+#include "eas_types.h"
+
+/* for C++ linkage */
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/* initialization and shutdown routines */
+extern EAS_RESULT EAS_HWInit(EAS_HW_DATA_HANDLE *hwInstData);
+extern EAS_RESULT EAS_HWShutdown(EAS_HW_DATA_HANDLE hwInstData);
+
+/* threading */
+extern void* EAS_HWRegisterSignalHandler();
+extern EAS_RESULT EAS_HWUnRegisterSignalHandler(void *cookie);
+
+/* memory functions */
+extern void *EAS_HWMemSet(void *s, int c, EAS_I32 n);
+extern void *EAS_HWMemCpy(void *s1, const void *s2, EAS_I32 n);
+extern EAS_I32 EAS_HWMemCmp(const void *s1, const void *s2, EAS_I32 n);
+
+/* memory allocation */
+extern void *EAS_HWMalloc(EAS_HW_DATA_HANDLE hwInstData, EAS_I32 size);
+extern void EAS_HWFree(EAS_HW_DATA_HANDLE hwInstData, void *p);
+
+/* file I/O */
+extern EAS_RESULT EAS_HWOpenFile(EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_LOCATOR locator, EAS_FILE_HANDLE *pFile, EAS_FILE_MODE mode);
+extern EAS_RESULT EAS_HWReadFile(EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, void *pBuffer, EAS_I32 n, EAS_I32 *pBytesRead);
+extern EAS_RESULT EAS_HWGetByte(EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, void *p);
+extern EAS_RESULT EAS_HWGetWord (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, void *p, EAS_BOOL msbFirst);
+extern EAS_RESULT EAS_HWGetDWord (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, void *p, EAS_BOOL msbFirst);
+extern EAS_RESULT EAS_HWFilePos (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, EAS_I32 *pPosition);
+extern EAS_RESULT EAS_HWFileSeek (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, EAS_I32 position);
+extern EAS_RESULT EAS_HWFileSeekOfs (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, EAS_I32 position);
+extern EAS_RESULT EAS_HWFileLength (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, EAS_I32 *pLength);
+extern EAS_RESULT EAS_HWDupHandle (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, EAS_FILE_HANDLE* pFile);
+extern EAS_RESULT EAS_HWCloseFile (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file);
+
+/* vibrate, LED, and backlight functions */
+extern EAS_RESULT EAS_HWVibrate(EAS_HW_DATA_HANDLE hwInstData, EAS_BOOL state);
+extern EAS_RESULT EAS_HWLED(EAS_HW_DATA_HANDLE hwInstData, EAS_BOOL state);
+extern EAS_RESULT EAS_HWBackLight(EAS_HW_DATA_HANDLE hwInstData, EAS_BOOL state);
+
+#ifdef __cplusplus
+} /* end extern "C" */
+#endif
+
+
+/* host yield function */
+extern EAS_BOOL EAS_HWYield(EAS_HW_DATA_HANDLE hwInstData);
+#endif /* end _EAS_HOST_H */

common/embeddedaudiosynthesis/arm-wt-22k/host_src/eas_hostmm.c

@@ -0,0 +1,688 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_hostmm.c
+ *
+ * Contents and purpose:
+ * This file contains the host wrapper functions for stdio, stdlib, etc.
+ * This is a sample version that reads from a filedescriptor.
+ * The file locator (EAS_FILE_LOCATOR) handle passed to
+ * HWOpenFile is the same one that is passed to EAS_OpenFile.
+ *
+ * Modify this file to suit the needs of your particular system.
+ *
+ * EAS_MAX_FILE_HANDLES sets the maximum number of MIDI streams within
+ * a MIDI type 1 file that can be played.
+ *
+ * EAS_HW_FILE is a structure to support the file I/O functions. It
+ * comprises the file descriptor, the file read pointer, and
+ * the dup flag, which when set, indicates that the file handle has
+ * been duplicated, and offset and length within the file.
+ *
+ * Copyright 2005 Sonic Network Inc.
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 795 $
+ *   $Date: 2007-08-01 00:14:45 -0700 (Wed, 01 Aug 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifdef _lint
+#include "lint_stdlib.h"
+#else
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <unistd.h>
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <fcntl.h>
+#include <limits.h>
+#include <sys/mman.h>
+#include <errno.h>
+#include <signal.h>
+#define LOG_TAG "Sonivox"
+#endif
+
+#include "eas_host.h"
+
+/* Only for debugging LED, vibrate, and backlight functions */
+#include "eas_report.h"
+
+/* this module requires dynamic memory support */
+#ifdef _STATIC_MEMORY
+#error "eas_hostmm.c requires the dynamic memory model!\n"
+#endif
+
+#ifndef EAS_MAX_FILE_HANDLES
+// 100 max file handles == 3 * (nb tracks(32) + 1 for the segment) + 1 for jet file
+//                         3 == 1(playing segment) + 1(prepared segment)
+//                              + 1(after end of playing segment, before files closed)
+#define EAS_MAX_FILE_HANDLES    100
+#endif
+
+/*
+ * this structure and the related function are here
+ * to support the ability to create duplicate handles
+ * and buffering it in memory. If your system uses
+ * in-memory resources, you can eliminate the calls
+ * to malloc and free, the dup flag, and simply track
+ * the file size and read position.
+ */
+typedef struct eas_hw_file_tag
+{
+    EAS_I32 fileSize;
+    EAS_I32 filePos;
+    EAS_BOOL dup;
+    int fd;
+    EAS_I32 offset;
+} EAS_HW_FILE;
+
+typedef struct eas_hw_inst_data_tag
+{
+    EAS_HW_FILE files[EAS_MAX_FILE_HANDLES];
+} EAS_HW_INST_DATA;
+
+pthread_key_t EAS_sigbuskey;
+
+/*----------------------------------------------------------------------------
+ * EAS_HWInit
+ *
+ * Initialize host wrapper interface
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_HWInit (EAS_HW_DATA_HANDLE *pHWInstData)
+{
+    EAS_HW_FILE *file;
+    int i;
+
+    /* need to track file opens for duplicate handles */
+    *pHWInstData = malloc(sizeof(EAS_HW_INST_DATA));
+    if (!(*pHWInstData))
+        return EAS_ERROR_MALLOC_FAILED;
+
+    EAS_HWMemSet(*pHWInstData, 0, sizeof(EAS_HW_INST_DATA));
+
+    file = (*pHWInstData)->files;
+    for (i = 0; i < EAS_MAX_FILE_HANDLES; i++)
+    {
+        file->fd = -1;
+        file++;
+    }
+
+
+    return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * EAS_HWShutdown
+ *
+ * Shut down host wrapper interface
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_HWShutdown (EAS_HW_DATA_HANDLE hwInstData)
+{
+
+    free(hwInstData);
+    return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_HWMalloc
+ *
+ * Allocates dynamic memory
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, hwInstData) hwInstData available for customer use */
+void *EAS_HWMalloc (EAS_HW_DATA_HANDLE hwInstData, EAS_I32 size)
+{
+    /* Since this whole library loves signed sizes, let's not let
+     * negative or 0 values through */
+    if (size <= 0)
+      return NULL;
+    return malloc((size_t) size);
+}
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_HWFree
+ *
+ * Frees dynamic memory
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, hwInstData) hwInstData available for customer use */
+void EAS_HWFree (EAS_HW_DATA_HANDLE hwInstData, void *p)
+{
+    free(p);
+}
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_HWMemCpy
+ *
+ * Copy memory wrapper
+ *
+ *----------------------------------------------------------------------------
+*/
+void *EAS_HWMemCpy (void *dest, const void *src, EAS_I32 amount)
+{
+    if (amount < 0)  {
+      EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x1a54b6e8, 0x00000004 , amount);
+      exit(255);
+    }
+    return memcpy(dest, src, (size_t) amount);
+}
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_HWMemSet
+ *
+ * Set memory wrapper
+ *
+ *----------------------------------------------------------------------------
+*/
+void *EAS_HWMemSet (void *dest, int val, EAS_I32 amount)
+{
+    if (amount < 0)  {
+      EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x1a54b6e8, 0x00000005 , amount);
+      exit(255);
+    }
+    return memset(dest, val, (size_t) amount);
+}
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_HWMemCmp
+ *
+ * Compare memory wrapper
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_I32 EAS_HWMemCmp (const void *s1, const void *s2, EAS_I32 amount)
+{
+    if (amount < 0) {
+      EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x1a54b6e8, 0x00000006 , amount);
+      exit(255);
+    }
+    return (EAS_I32) memcmp(s1, s2, (size_t) amount);
+}
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_HWOpenFile
+ *
+ * Open a file for read or write
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_HWOpenFile (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_LOCATOR locator, EAS_FILE_HANDLE *pFile, EAS_FILE_MODE mode)
+{
+    EAS_HW_FILE *file;
+    int fd;
+    int i;
+
+    /* set return value to NULL */
+    *pFile = NULL;
+
+    /* only support read mode at this time */
+    if (mode != EAS_FILE_READ)
+        return EAS_ERROR_INVALID_FILE_MODE;
+
+    /* find an empty entry in the file table */
+    file = hwInstData->files;
+    for (i = 0; i < EAS_MAX_FILE_HANDLES; i++)
+    {
+        /* is this slot being used? */
+        if (file->fd < 0)
+        {
+            if (locator->path) {
+                /* open the file */
+                if ((fd = open(locator->path, O_RDONLY)) < 0) {
+                    return EAS_ERROR_FILE_OPEN_FAILED;
+                }
+            } else {
+                /* else file is already open */
+                fd = dup(locator->fd);
+            }
+
+            /* determine the file size */
+            if (locator->length == 0) {
+                if (lseek(fd, 0, SEEK_END) < 0) {
+                    close(fd);
+                    return EAS_ERROR_FILE_LENGTH;
+                }
+                if ((file->fileSize = (EAS_I32)lseek(fd, 0, SEEK_CUR)) == -1L) {
+                    close(fd);
+                    return EAS_ERROR_FILE_LENGTH;
+                }
+            }
+
+            // file size was passed in
+            else {
+                file->fileSize = (EAS_I32) locator->length;
+            }
+
+            file->fd = fd;
+            file->offset = (EAS_I32)locator->offset;
+
+            /* initialize some values */
+            file->filePos = 0;
+            file->dup = EAS_FALSE;
+
+            *pFile = file;
+            return EAS_SUCCESS;
+        }
+        file++;
+    }
+
+    /* too many open files */
+    return EAS_ERROR_MAX_FILES_OPEN;
+}
+
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_HWReadFile
+ *
+ * Read data from a file
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, hwInstData) hwInstData available for customer use */
+EAS_RESULT EAS_HWReadFile (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, void *pBuffer, EAS_I32 n, EAS_I32 *pBytesRead)
+{
+    EAS_I32 count;
+
+    /* make sure we have a valid handle */
+    if (file->fd < 0)
+        return EAS_ERROR_INVALID_HANDLE;
+
+    if (n < 0)
+      return EAS_EOF;
+
+    /* calculate the bytes to read */
+    count = file->fileSize - file->filePos;
+    if (n < count)
+        count = n;
+    if (count < 0)
+      return EAS_EOF;
+
+    /* copy the data to the requested location, and advance the pointer */
+    if (count) {
+        lseek(file->fd, file->filePos + file->offset, SEEK_SET);
+        count = read(file->fd, pBuffer, count);
+    }
+    file->filePos += count;
+    *pBytesRead = count;
+
+    /* were n bytes read? */
+    if (count!= n)
+        return EAS_EOF;
+    return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_HWGetByte
+ *
+ * Read a byte from a file
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, hwInstData) hwInstData available for customer use */
+EAS_RESULT EAS_HWGetByte (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, void *p)
+{
+    EAS_I32 numread;
+    return EAS_HWReadFile(hwInstData, file, p, 1, &numread);
+}
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_HWGetWord
+ *
+ * Read a 16 bit word from a file
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, hwInstData) hwInstData available for customer use */
+EAS_RESULT EAS_HWGetWord (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, void *p, EAS_BOOL msbFirst)
+{
+    EAS_RESULT result;
+    EAS_U8 c1, c2;
+
+    /* read 2 bytes from the file */
+    if ((result = EAS_HWGetByte(hwInstData, file, &c1)) != EAS_SUCCESS)
+        return result;
+    if ((result = EAS_HWGetByte(hwInstData, file, &c2)) != EAS_SUCCESS)
+        return result;
+
+    /* order them as requested */
+    if (msbFirst)
+        *((EAS_U16*) p) = ((EAS_U16) c1 << 8) | c2;
+    else
+        *((EAS_U16*) p) = ((EAS_U16) c2 << 8) | c1;
+
+    return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_HWGetDWord
+ *
+ * Returns the current location in the file
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, hwInstData) hwInstData available for customer use */
+EAS_RESULT EAS_HWGetDWord (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, void *p, EAS_BOOL msbFirst)
+{
+    EAS_RESULT result;
+    EAS_U8 c1, c2,c3,c4;
+
+    /* read 4 bytes from the file */
+    if ((result = EAS_HWGetByte(hwInstData, file, &c1)) != EAS_SUCCESS)
+        return result;
+    if ((result = EAS_HWGetByte(hwInstData, file, &c2)) != EAS_SUCCESS)
+        return result;
+    if ((result = EAS_HWGetByte(hwInstData, file, &c3)) != EAS_SUCCESS)
+        return result;
+    if ((result = EAS_HWGetByte(hwInstData, file, &c4)) != EAS_SUCCESS)
+        return result;
+
+    /* order them as requested */
+    if (msbFirst)
+        *((EAS_U32*) p) = ((EAS_U32) c1 << 24) | ((EAS_U32) c2 << 16) | ((EAS_U32) c3 << 8) | c4;
+    else
+        *((EAS_U32*) p)= ((EAS_U32) c4 << 24) | ((EAS_U32) c3 << 16) | ((EAS_U32) c2 << 8) | c1;
+
+    return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_HWFilePos
+ *
+ * Returns the current location in the file
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, hwInstData) hwInstData available for customer use */
+EAS_RESULT EAS_HWFilePos (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, EAS_I32 *pPosition)
+{
+
+    /* make sure we have a valid handle */
+    if (file->fd < 0)
+        return EAS_ERROR_INVALID_HANDLE;
+
+    *pPosition = file->filePos;
+    return EAS_SUCCESS;
+} /* end EAS_HWFilePos */
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_HWFileSeek
+ *
+ * Seek to a specific location in the file
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, hwInstData) hwInstData available for customer use */
+EAS_RESULT EAS_HWFileSeek (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, EAS_I32 position)
+{
+
+    /* make sure we have a valid handle */
+    if (file->fd < 0)
+        return EAS_ERROR_INVALID_HANDLE;
+
+    /* validate new position */
+    if ((position < 0) || (position > file->fileSize))
+        return EAS_ERROR_FILE_SEEK;
+
+    /* save new position */
+    file->filePos = position;
+    return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_HWFileSeekOfs
+ *
+ * Seek forward or back relative to the current position
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, hwInstData) hwInstData available for customer use */
+EAS_RESULT EAS_HWFileSeekOfs (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, EAS_I32 position)
+{
+
+    /* make sure we have a valid handle */
+    if (file->fd < 0)
+        return EAS_ERROR_INVALID_HANDLE;
+
+    /* determine the file position */
+    position += file->filePos;
+    if ((position < 0) || (position > file->fileSize))
+        return EAS_ERROR_FILE_SEEK;
+
+    /* save new position */
+    file->filePos = position;
+    return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_HWFileLength
+ *
+ * Return the file length
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, hwInstData) hwInstData available for customer use */
+EAS_RESULT EAS_HWFileLength (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, EAS_I32 *pLength)
+{
+
+    /* make sure we have a valid handle */
+    if (file->fd < 0)
+        return EAS_ERROR_INVALID_HANDLE;
+
+    *pLength = file->fileSize;
+    return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_HWDupHandle
+ *
+ * Duplicate a file handle
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_HWDupHandle (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, EAS_FILE_HANDLE *pDupFile)
+{
+    EAS_HW_FILE *dupFile;
+    int i;
+
+    /* make sure we have a valid handle */
+    if (file->fd < 0)
+        return EAS_ERROR_INVALID_HANDLE;
+
+    /* find an empty entry in the file table */
+    dupFile = hwInstData->files;
+    for (i = 0; i < EAS_MAX_FILE_HANDLES; i++)
+    {
+        /* is this slot being used? */
+        if (dupFile->fd < 0)
+        {
+            /* copy info from the handle to be duplicated */
+            dupFile->filePos = file->filePos;
+            dupFile->fileSize = file->fileSize;
+            dupFile->fd = file->fd;
+            dupFile->offset = file->offset;
+
+            /* set the duplicate handle flag */
+            dupFile->dup = file->dup = EAS_TRUE;
+
+            *pDupFile = dupFile;
+            return EAS_SUCCESS;
+        }
+        dupFile++;
+    }
+
+    /* too many open files */
+    return EAS_ERROR_MAX_FILES_OPEN;
+}
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_HWClose
+ *
+ * Wrapper for fclose function
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_HWCloseFile (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file1)
+{
+    EAS_HW_FILE *file2,*dupFile;
+    int i;
+
+
+    /* make sure we have a valid handle */
+    if (file1->fd < 0)
+        return EAS_ERROR_INVALID_HANDLE;
+
+    /* check for duplicate handle */
+    if (file1->dup)
+    {
+        dupFile = NULL;
+        file2 = hwInstData->files;
+        for (i = 0; i < EAS_MAX_FILE_HANDLES; i++)
+        {
+            /* check for duplicate */
+            if ((file1 != file2) && (file2->fd == file1->fd))
+            {
+                /* is there more than one duplicate? */
+                if (dupFile != NULL)
+                {
+                    /* clear this entry and return */
+                    file1->fd = -1;
+                    return EAS_SUCCESS;
+                }
+
+                /* this is the first duplicate found */
+                else
+                    dupFile = file2;
+            }
+            file2++;
+        }
+
+        /* there is only one duplicate, clear the dup flag */
+        if (dupFile)
+            dupFile->dup = EAS_FALSE;
+        else
+            /* if we get here, there's a serious problem */
+            return EAS_ERROR_HANDLE_INTEGRITY;
+
+        /* clear this entry and return */
+        file1->fd = -1;
+        return EAS_SUCCESS;
+    }
+
+    /* no duplicates - close the file */
+    close(file1->fd);
+    /* clear this entry and return */
+    file1->fd = -1;
+    return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_HWVibrate
+ *
+ * Turn on/off vibrate function
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, hwInstData) hwInstData available for customer use */
+EAS_RESULT EAS_HWVibrate (EAS_HW_DATA_HANDLE hwInstData, EAS_BOOL state)
+{
+    EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x1a54b6e8, 0x00000001 , state);
+    return EAS_SUCCESS;
+} /* end EAS_HWVibrate */
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_HWLED
+ *
+ * Turn on/off LED
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, hwInstData) hwInstData available for customer use */
+EAS_RESULT EAS_HWLED (EAS_HW_DATA_HANDLE hwInstData, EAS_BOOL state)
+{
+    EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x1a54b6e8, 0x00000002 , state);
+    return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_HWBackLight
+ *
+ * Turn on/off backlight
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, hwInstData) hwInstData available for customer use */
+EAS_RESULT EAS_HWBackLight (EAS_HW_DATA_HANDLE hwInstData, EAS_BOOL state)
+{
+    EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x1a54b6e8, 0x00000003 , state);
+    return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_HWYield
+ *
+ * This function is called periodically by the EAS library to give the
+ * host an opportunity to allow other tasks to run. There are two ways to
+ * use this call:
+ *
+ * If you have a multi-tasking OS, you can call the yield function in the
+ * OS to allow other tasks to run. In this case, return EAS_FALSE to tell
+ * the EAS library to continue processing when control returns from this
+ * function.
+ *
+ * If tasks run in a single thread by sequential function calls (sometimes
+ * call a "commutator loop"), return EAS_TRUE to cause the EAS Library to
+ * return to the caller. Be sure to check the number of bytes rendered
+ * before passing the audio buffer to the codec - it may not be filled.
+ * The next call to EAS_Render will continue processing until the buffer
+ * has been filled.
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, hwInstData) hwInstData available for customer use */
+EAS_BOOL EAS_HWYield (EAS_HW_DATA_HANDLE hwInstData)
+{
+    /* put your code here */
+    return EAS_FALSE;
+}
+

common/embeddedaudiosynthesis/arm-wt-22k/host_src/eas_main.c

@@ -0,0 +1,464 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_main.c
+ *
+ * Contents and purpose:
+ * The entry point and high-level functions for the EAS Synthesizer test
+ * harness.
+ *
+ * Copyright Sonic Network Inc. 2004
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 775 $
+ *   $Date: 2007-07-20 10:11:11 -0700 (Fri, 20 Jul 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifdef _lint
+#include "lint_stdlib.h"
+#else
+#include <stdio.h>
+#include <string.h>
+#include <stdlib.h>
+#include <time.h>
+#endif
+
+#include "eas.h"
+#include "eas_wave.h"
+#include "eas_report.h"
+
+/* determines how many EAS buffers to fill a host buffer */
+#define NUM_BUFFERS         8
+
+/* default file to play if no filename is specified on the command line */
+static const char defaultTestFile[] = "test.mid";
+
+EAS_I32 polyphony;
+
+/* prototypes for helper functions */
+static void StrCopy(char *dest, const char *src, EAS_I32 size);
+static EAS_BOOL ChangeFileExt(char *str, const char *ext, EAS_I32 size);
+static EAS_RESULT PlayFile (EAS_DATA_HANDLE easData, const char* filename, const char* outputFile, const S_EAS_LIB_CONFIG *pLibConfig, void *buffer, EAS_I32 bufferSize);
+static EAS_BOOL EASLibraryCheck (const S_EAS_LIB_CONFIG *pLibConfig);
+
+/* main is defined after playfile to avoid the need for two passes through lint */
+
+/*----------------------------------------------------------------------------
+ * PlayFile()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * This function plays the file requested by filename
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *
+ *----------------------------------------------------------------------------
+*/
+
+static EAS_RESULT PlayFile (EAS_DATA_HANDLE easData, const char* filename, const char* outputFile, const S_EAS_LIB_CONFIG *pLibConfig, void *buffer, EAS_I32 bufferSize)
+{
+    EAS_HANDLE handle;
+    EAS_RESULT result, reportResult;
+    EAS_I32 count;
+    EAS_STATE state;
+    EAS_I32 playTime;
+    char waveFilename[256];
+    WAVE_FILE *wFile;
+    EAS_INT i;
+    EAS_PCM *p;
+    EAS_FILE file;
+
+    /* determine the name of the output file */
+    wFile = NULL;
+    if (outputFile == NULL)
+    {
+        StrCopy(waveFilename, filename, sizeof(waveFilename));
+        if (!ChangeFileExt(waveFilename, "wav", sizeof(waveFilename)))
+        {
+            { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Error in output filename %s\n", waveFilename); */ }
+            return EAS_FAILURE;
+        }
+        outputFile = waveFilename;
+    }
+
+    /* call EAS library to open file */
+    file.path = filename;
+    file.fd = 0;
+    if ((reportResult = EAS_OpenFile(easData, &file, &handle)) != EAS_SUCCESS)
+    {
+        { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "EAS_OpenFile returned %ld\n", reportResult); */ }
+        return reportResult;
+    }
+
+    /* prepare to play the file */
+    if ((result = EAS_Prepare(easData, handle)) != EAS_SUCCESS)
+    {
+        { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "EAS_Prepare returned %ld\n", result); */ }
+        reportResult = result;
+    }
+
+    /* get play length */
+    if ((result = EAS_ParseMetaData(easData, handle, &playTime)) != EAS_SUCCESS)
+    {
+        { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "EAS_ParseMetaData returned %ld\n", result); */ }
+        return result;
+    }
+    EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0xe624f4d9, 0x00000005 , playTime / 1000, playTime % 1000);
+
+    if (reportResult == EAS_SUCCESS)
+    {
+        /* create the output file */
+        wFile = WaveFileCreate(outputFile, pLibConfig->numChannels, pLibConfig->sampleRate, sizeof(EAS_PCM) * 8);
+        if (!wFile)
+        {
+            { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Unable to create output file %s\n", waveFilename); */ }
+            reportResult = EAS_FAILURE;
+        }
+    }
+
+    /* rendering loop */
+    while (reportResult == EAS_SUCCESS)
+    {
+
+        /* we may render several buffers here to fill one host buffer */
+        for (i = 0, p = buffer; i < NUM_BUFFERS; i++, p+= pLibConfig->mixBufferSize * pLibConfig->numChannels)
+        {
+
+            /* get the current time */
+            if ((result = EAS_GetLocation(easData, handle, &playTime)) != EAS_SUCCESS)
+            {
+                { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "EAS_GetLocation returned %d\n",result); */ }
+                if (reportResult == EAS_SUCCESS)
+                    reportResult = result;
+                break;
+            }
+            { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Parser time: %d.%03d\n", playTime / 1000, playTime % 1000); */ }
+
+            /* render a buffer of audio */
+            if ((result = EAS_Render(easData, p, pLibConfig->mixBufferSize, &count)) != EAS_SUCCESS)
+            {
+                { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "EAS_Render returned %d\n",result); */ }
+                if (reportResult == EAS_SUCCESS)
+                    reportResult = result;
+            }
+        }
+
+        if (result == EAS_SUCCESS)
+        {
+            /* write it to the wave file */
+            if (WaveFileWrite(wFile, buffer, bufferSize) != bufferSize)
+            {
+                { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "WaveFileWrite failed\n"); */ }
+                reportResult = EAS_FAILURE;
+            }
+        }
+
+        if (reportResult == EAS_SUCCESS)
+        {
+            /* check stream state */
+            if ((result = EAS_State(easData, handle, &state)) != EAS_SUCCESS)
+            {
+                { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "EAS_State returned %d\n", result); */ }
+                reportResult = result;
+            }
+
+            /* is playback complete */
+            if ((state == EAS_STATE_STOPPED) || (state == EAS_STATE_ERROR))
+                break;
+        }
+    }
+
+    /* close the output file */
+    if (wFile)
+    {
+        if (!WaveFileClose(wFile))
+        {
+            { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Error closing wave file %s\n", waveFilename); */ }
+            if (reportResult == EAS_SUCCESS)
+                result = EAS_FAILURE;
+        }
+    }
+
+    /* close the input file */
+    if ((result = EAS_CloseFile(easData,handle)) != EAS_SUCCESS)
+    {
+        { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "EAS_Close returned %ld\n", result); */ }
+        if (reportResult == EAS_SUCCESS)
+            result = EAS_FAILURE;
+    }
+
+    return reportResult;
+} /* end PlayFile */
+
+/*----------------------------------------------------------------------------
+ * main()
+ *----------------------------------------------------------------------------
+ * Purpose: The entry point for the EAS sample application
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *
+ *----------------------------------------------------------------------------
+*/
+int main( int argc, char **argv )
+{
+    EAS_DATA_HANDLE easData;
+    const S_EAS_LIB_CONFIG *pLibConfig;
+    void *buffer;
+    EAS_RESULT result, playResult;
+    EAS_I32 bufferSize;
+    int i;
+    int temp;
+    FILE *debugFile;
+    char *outputFile = NULL;
+
+    /* set the error reporting level */
+    EAS_SetDebugLevel(_EAS_SEVERITY_INFO);
+    debugFile = NULL;
+
+    /* process command-line arguments */
+    for (i = 1; i < argc; i++)
+    {
+        /* check for switch */
+        if (argv[i][0] == '-')
+        {
+            switch (argv[i][1])
+            {
+            case 'd':
+                temp = argv[i][2];
+                if ((temp >= '0') || (temp <= '9'))
+                    EAS_SetDebugLevel(temp);
+                else
+                    { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Invalid debug level %d\n", temp); */ }
+                break;
+            case 'f':
+                if ((debugFile = fopen(&argv[i][2],"w")) == NULL)
+                    { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Unable to create debug file %s\n", &argv[i][2]); */ }
+                else
+                    EAS_SetDebugFile(debugFile, EAS_TRUE);
+                break;
+            case 'o':
+                outputFile = &argv[i][2];
+                break;
+            case 'p':
+                polyphony = atoi(&argv[i][2]);
+                if (polyphony < 1)
+                    polyphony = 1;
+                { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "Polyphony set to %d\n", polyphony); */ }
+                break;
+            default:
+                break;
+            }
+            continue;
+        }
+    }
+
+    /* assume success */
+    playResult = EAS_SUCCESS;
+
+    /* get the library configuration */
+    pLibConfig = EAS_Config();
+    if (!EASLibraryCheck(pLibConfig))
+        return -1;
+    if (polyphony > pLibConfig->maxVoices)
+        polyphony = pLibConfig->maxVoices;
+
+    /* calculate buffer size */
+    bufferSize = pLibConfig->mixBufferSize * pLibConfig->numChannels * (EAS_I32)sizeof(EAS_PCM) * NUM_BUFFERS;
+
+    /* allocate output buffer memory */
+    buffer = malloc((EAS_U32)bufferSize);
+    if (!buffer)
+    {
+        { /* dpp: EAS_ReportEx(_EAS_SEVERITY_FATAL, "Error allocating memory for audio buffer\n"); */ }
+        return EAS_FAILURE;
+    }
+
+    /* initialize the EAS library */
+    polyphony = pLibConfig->maxVoices;
+    if ((result = EAS_Init(&easData)) != EAS_SUCCESS)
+    {
+        { /* dpp: EAS_ReportEx(_EAS_SEVERITY_FATAL, "EAS_Init returned %ld - aborting!\n", result); */ }
+        free(buffer);
+        return result;
+    }
+
+    /*
+     * Some debugging environments don't allow for passed parameters.
+     * In this case, just play the default MIDI file "test.mid"
+     */
+    if (argc < 2)
+    {
+        { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "Playing '%s'\n", defaultTestFile); */ }
+        if ((playResult = PlayFile(easData, defaultTestFile, NULL, pLibConfig, buffer, bufferSize)) != EAS_SUCCESS)
+        {
+            { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Error %d playing file %s\n", playResult, defaultTestFile); */ }
+        }
+    }
+    /* iterate through the list of files to be played */
+    else
+    {
+        for (i = 1; i < argc; i++)
+        {
+            /* check for switch */
+            if (argv[i][0] != '-')
+            {
+
+                { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "Playing '%s'\n", argv[i]); */ }
+                if ((playResult = PlayFile(easData, argv[i], outputFile, pLibConfig, buffer, bufferSize)) != EAS_SUCCESS)
+                {
+                    { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Error %d playing file %s\n", playResult, argv[i]); */ }
+                    break;
+                }
+            }
+        }
+    }
+
+    /* shutdown the EAS library */
+    if ((result = EAS_Shutdown(easData)) != EAS_SUCCESS)
+    {
+        { /* dpp: EAS_ReportEx(_EAS_SEVERITY_FATAL, "EAS_Shutdown returned %ld\n", result); */ }
+    }
+
+    /* free the output buffer */
+    free(buffer);
+
+    /* close the debug file */
+    if (debugFile)
+        fclose(debugFile);
+
+    /* play errors take precedence over shutdown errors */
+    if (playResult != EAS_SUCCESS)
+        return playResult;
+    return result;
+} /* end main */
+
+/*----------------------------------------------------------------------------
+ * StrCopy()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Safe string copy
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *
+ *----------------------------------------------------------------------------
+*/
+static void StrCopy(char *dest, const char *src, EAS_I32 size)
+{
+    int len;
+
+    strncpy(dest, src, (size_t) size-1);
+    len = (int) strlen(src);
+    if (len < size)
+        dest[len] = 0;
+} /* end StrCopy */
+
+/*----------------------------------------------------------------------------
+ * ChangeFileExt()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Changes the file extension of a filename
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_BOOL ChangeFileExt(char *str, const char *ext, EAS_I32 size)
+{
+    char *p;
+
+    /* find the extension, if any */
+    p = strrchr(str,'.');
+    if (!p)
+    {
+        if ((EAS_I32)(strlen(str) + 5) > size)
+            return EAS_FALSE;
+        strcat(str,".");
+        strcat(str,ext);
+        return EAS_TRUE;
+    }
+
+    /* make sure there's room for the extension */
+    p++;
+    *p = 0;
+    if ((EAS_I32)(strlen(str) + 4) > size)
+        return EAS_FALSE;
+    strcat(str,ext);
+    return EAS_TRUE;
+} /* end ChangeFileExt */
+
+/*----------------------------------------------------------------------------
+ * EASLibraryCheck()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Displays the library version and checks it against the header
+ * file used to build this code.
+ *
+ * Inputs:
+ * pLibConfig       - library configuration retrieved from the library
+ *
+ * Outputs:
+ * returns EAS_TRUE if matched
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_BOOL EASLibraryCheck (const S_EAS_LIB_CONFIG *pLibConfig)
+{
+
+    /* display the library version */
+    { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "EAS Library Version %d.%d.%d.%d\n",
+        pLibConfig->libVersion >> 24,
+        (pLibConfig->libVersion >> 16) & 0x0f,
+        (pLibConfig->libVersion >> 8) & 0x0f,
+        pLibConfig->libVersion & 0x0f); */ }
+
+    /* display some info about the library build */
+    if (pLibConfig->checkedVersion)
+        { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "\tChecked library\n"); */ }
+    { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "\tMaximum polyphony: %d\n", pLibConfig->maxVoices); */ }
+    { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "\tNumber of channels: %d\n", pLibConfig->numChannels); */ }
+    { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "\tSample rate: %d\n", pLibConfig->sampleRate); */ }
+    { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "\tMix buffer size: %d\n", pLibConfig->mixBufferSize); */ }
+    if (pLibConfig->filterEnabled)
+        { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "\tFilter enabled\n"); */ }
+#ifndef _WIN32_WCE
+    { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "\tLibrary Build Timestamp: %s", ctime((time_t*)&pLibConfig->buildTimeStamp)); */ }
+#endif
+    { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "\tLibrary Build ID: %s\n", pLibConfig->buildGUID); */ }
+
+    /* check it against the header file used to build this code */
+    /*lint -e{778} constant expression used for display purposes may evaluate to zero */
+    if (LIB_VERSION != pLibConfig->libVersion)
+    {
+        { /* dpp: EAS_ReportEx(_EAS_SEVERITY_FATAL, "Library version does not match header files. EAS Header Version %d.%d.%d.%d\n",
+            LIB_VERSION >> 24,
+            (LIB_VERSION >> 16) & 0x0f,
+            (LIB_VERSION >> 8) & 0x0f,
+            LIB_VERSION & 0x0f); */ }
+        return EAS_FALSE;
+    }
+    return EAS_TRUE;
+} /* end EASLibraryCheck */
+

common/embeddedaudiosynthesis/arm-wt-22k/host_src/eas_report.c

@@ -0,0 +1,264 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_report.c
+ *
+ * Contents and purpose:
+ * This file contains the debug message handling routines for the EAS library.
+ * These routines should be modified as needed for your system.
+ *
+ * Copyright 2005 Sonic Network Inc.
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 659 $
+ *   $Date: 2007-04-24 13:36:35 -0700 (Tue, 24 Apr 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifdef _lint
+#include "lint_stdlib.h"
+#else
+#include <stdlib.h>
+#include <stdio.h>
+#include <stdarg.h>
+#endif
+
+#include "eas_report.h"
+
+static int severityLevel = 9999;
+
+/* debug file */
+static FILE *debugFile = NULL;
+int flush = 0;
+
+#ifndef _NO_DEBUG_PREPROCESSOR
+
+/* structure should have an #include for each error message header file */
+S_DEBUG_MESSAGES debugMessages[] =
+{
+#ifndef UNIFIED_DEBUG_MESSAGES
+#include "eas_config_msgs.h"
+
+
+#include "eas_host_msgs.h"
+#include "eas_hostmm_msgs.h"
+#include "eas_math_msgs.h"
+#include "eas_midi_msgs.h"
+#include "eas_mixer_msgs.h"
+#include "eas_pcm_msgs.h"
+#include "eas_public_msgs.h"
+#include "eas_smf_msgs.h"
+#include "eas_wave_msgs.h"
+#include "eas_voicemgt_msgs.h"
+
+#ifdef _FM_SYNTH
+#include "eas_fmsynth_msgs.h"
+#include "eas_fmengine_msgs.h"
+#endif
+
+#ifdef _WT_SYNTH
+#include "eas_wtsynth_msgs.h"
+#include "eas_wtengine_msgs.h"
+#endif
+
+#ifdef _ARM_TEST_MAIN
+#include "arm_main_msgs.h"
+#endif
+
+#ifdef _EAS_MAIN
+#include "eas_main_msgs.h"
+#endif
+
+#ifdef _EAS_MAIN_IPC
+#include "eas_main_ipc_msgs.h"
+#endif
+
+#ifdef _METRICS_ENABLED
+#include "eas_perf_msgs.h"
+#endif
+
+#ifdef _COMPRESSOR_ENABLED
+#include "eas_compressor_msgs.h"
+#endif
+
+#ifdef _ENHANCER_ENABLED
+#include "eas_enhancer_msgs.h"
+#endif
+
+#ifdef _WOW_ENABLED
+#include "eas_wow_msgs.h"
+#endif
+
+#ifdef _SMAF_PARSER
+#include "eas_smaf_msgs.h"
+#endif
+
+#ifdef _OTA_PARSER
+#include "eas_ota_msgs.h"
+#endif
+
+#ifdef _IMELODY_PARSER
+#include "eas_imelody_msgs.h"
+#endif
+
+#ifdef _WAVE_PARSER
+#include "eas_wavefile_msgs.h"
+#endif
+
+#if defined(_CMX_PARSER) || defined(_MFI_PARSER)
+#include "eas_cmf_msgs.h"
+#endif
+
+#if defined(_CMX_PARSER) || defined(_MFI_PARSER) || defined(_WAVE_PARSER)
+#include "eas_imaadpcm_msgs.h"
+#endif
+
+#else
+#include "eas_debugmsgs.h"
+#endif
+
+/* denotes end of error messages */
+{ 0,0,0 }
+};
+
+/*----------------------------------------------------------------------------
+ * EAS_ReportEx()
+ *
+ * This is the error message handler. The default handler outputs error
+ * messages to stdout. Modify this as needed for your system.
+ *----------------------------------------------------------------------------
+*/
+void EAS_ReportEx (int severity, unsigned long hashCode, int serialNum, ...)
+{
+    va_list vargs;
+    int i;
+
+    /* check severity level */
+    if (severity > severityLevel)
+        return;
+
+    /* find the error message and output to stdout */
+    /*lint -e{661} we check for NULL pointer - no fence post error here */
+    for (i = 0; debugMessages[i].m_pDebugMsg; i++)
+    {
+        if ((debugMessages[i].m_nHashCode == hashCode) &&
+        (debugMessages[i].m_nSerialNum == serialNum))
+        {
+            /*lint -e{826} <allow variable args> */
+            va_start(vargs, serialNum);
+            if (debugFile)
+            {
+                vfprintf(debugFile, debugMessages[i].m_pDebugMsg, vargs);
+                if (flush)
+                    fflush(debugFile);
+            }
+            else
+            {
+                vprintf(debugMessages[i].m_pDebugMsg, vargs);
+            }
+            va_end(vargs);
+            return;
+        }
+    }
+    printf("Unrecognized error: Severity=%d; HashCode=%lu; SerialNum=%d\n", severity, hashCode, serialNum);
+} /* end EAS_ReportEx */
+
+#else
+/*----------------------------------------------------------------------------
+ * EAS_Report()
+ *
+ * This is the error message handler. The default handler outputs error
+ * messages to stdout. Modify this as needed for your system.
+ *----------------------------------------------------------------------------
+*/
+void EAS_Report (int severity, const char *fmt, ...)
+{
+    va_list vargs;
+
+    /* check severity level */
+    if (severity > severityLevel)
+        return;
+
+    /*lint -e{826} <allow variable args> */
+    va_start(vargs, fmt);
+    if (debugFile)
+    {
+        vfprintf(debugFile, fmt, vargs);
+        if (flush)
+            fflush(debugFile);
+    }
+    else
+    {
+        vprintf(fmt, vargs);
+    }
+    va_end(vargs);
+} /* end EAS_Report */
+
+/*----------------------------------------------------------------------------
+ * EAS_ReportX()
+ *
+ * This is the error message handler. The default handler outputs error
+ * messages to stdout. Modify this as needed for your system.
+ *----------------------------------------------------------------------------
+*/
+void EAS_ReportX (int severity, const char *fmt, ...)
+{
+    va_list vargs;
+
+    /* check severity level */
+    if (severity > severityLevel)
+        return;
+
+    /*lint -e{826} <allow variable args> */
+    va_start(vargs, fmt);
+    if (debugFile)
+    {
+        vfprintf(debugFile, fmt, vargs);
+        if (flush)
+            fflush(debugFile);
+    }
+    else
+    {
+        vprintf(fmt, vargs);
+    }
+    va_end(vargs);
+} /* end EAS_ReportX */
+#endif
+
+/*----------------------------------------------------------------------------
+ * EAS_SetDebugLevel()
+ *
+ * Sets the level for debug message output
+ *----------------------------------------------------------------------------
+*/
+
+void EAS_SetDebugLevel (int severity)
+{
+    severityLevel = severity;
+} /* end EAS_SetDebugLevel */
+
+/*----------------------------------------------------------------------------
+ * EAS_SetDebugFile()
+ *
+ * Redirect debugger output to the specified file.
+ *----------------------------------------------------------------------------
+*/
+void EAS_SetDebugFile (void *file, int flushAfterWrite)
+{
+    debugFile = (FILE*) file;
+    flush = flushAfterWrite;
+} /* end EAS_SetDebugFile */
+

common/embeddedaudiosynthesis/arm-wt-22k/host_src/eas_report.h

@@ -0,0 +1,77 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_report.h
+ *
+ * Contents and purpose:
+ * This file contains the debug message handling routines for the EAS library.
+ * These routines should be modified as needed for your system.
+ *
+ * DO NOT MODIFY THIS FILE!
+ *
+ * Copyright 2005 Sonic Network Inc.
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 82 $
+ *   $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+/* sentinel */
+#ifndef _EAS_REPORT_H
+#define _EAS_REPORT_H
+
+#define _EAS_SEVERITY_NOFILTER 0
+#define _EAS_SEVERITY_FATAL 1
+#define _EAS_SEVERITY_ERROR 2
+#define _EAS_SEVERITY_WARNING 3
+#define _EAS_SEVERITY_INFO 4
+#define _EAS_SEVERITY_DETAIL 5
+
+/* for C++ linkage */
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#ifndef _NO_DEBUG_PREPROCESSOR
+
+/* structure for included debug message header files */
+typedef struct
+{
+    unsigned long m_nHashCode;
+    int m_nSerialNum;
+    char *m_pDebugMsg;
+} S_DEBUG_MESSAGES;
+
+/* debug message handling prototypes */
+extern void EAS_ReportEx (int severity, unsigned long hashCode, int serialNum, ...);
+
+#else
+
+/* these prototypes are used if the debug preprocessor is not used */
+extern void EAS_Report (int severity, const char* fmt, ...);
+extern void EAS_ReportX (int severity, const char* fmt, ...);
+
+#endif
+
+extern void EAS_SetDebugLevel (int severity);
+extern void EAS_SetDebugFile (void *file, int flushAfterWrite);
+
+#ifdef __cplusplus
+} /* end extern "C" */
+#endif
+
+#endif

common/embeddedaudiosynthesis/arm-wt-22k/host_src/eas_reverb.h

@@ -0,0 +1,54 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_reverb.h
+ *
+ * Contents and purpose:
+ * Contains parameter enumerations for the Reverb effect
+ *
+ *
+ * Copyright Sonic Network Inc. 2006
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 300 $
+ *   $Date: 2006-09-11 17:37:20 -0700 (Mon, 11 Sep 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_REVERB_H
+#define _EAS_REVERB_H
+
+
+/* enumerated parameter settings for Reverb effect */
+typedef enum
+{
+    EAS_PARAM_REVERB_BYPASS,
+    EAS_PARAM_REVERB_PRESET,
+    EAS_PARAM_REVERB_WET,
+    EAS_PARAM_REVERB_DRY
+} E_REVERB_PARAMS;
+
+
+typedef enum
+{
+    EAS_PARAM_REVERB_LARGE_HALL,
+    EAS_PARAM_REVERB_HALL,
+    EAS_PARAM_REVERB_CHAMBER,
+    EAS_PARAM_REVERB_ROOM,
+} E_REVERB_PRESETS;
+
+
+#endif /* _REVERB_H */

common/embeddedaudiosynthesis/arm-wt-22k/host_src/eas_types.h

@@ -0,0 +1,273 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_types.h
+ *
+ * Contents and purpose:
+ * The public interface header for the EAS synthesizer.
+ *
+ * This header only contains declarations that are specific
+ * to this implementation.
+ *
+ * DO NOT MODIFY THIS FILE!
+ *
+ * Copyright Sonic Network Inc. 2004
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 726 $
+ *   $Date: 2007-06-14 23:10:46 -0700 (Thu, 14 Jun 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_TYPES_H
+#define _EAS_TYPES_H
+
+/* EAS_RESULT return codes */
+typedef long EAS_RESULT;
+#define EAS_SUCCESS                         0
+#define EAS_FAILURE                         -1
+#define EAS_ERROR_INVALID_MODULE            -2
+#define EAS_ERROR_MALLOC_FAILED             -3
+#define EAS_ERROR_FILE_POS                  -4
+#define EAS_ERROR_INVALID_FILE_MODE         -5
+#define EAS_ERROR_FILE_SEEK                 -6
+#define EAS_ERROR_FILE_LENGTH               -7
+#define EAS_ERROR_NOT_IMPLEMENTED           -8
+#define EAS_ERROR_CLOSE_FAILED              -9
+#define EAS_ERROR_FILE_OPEN_FAILED          -10
+#define EAS_ERROR_INVALID_HANDLE            -11
+#define EAS_ERROR_NO_MIX_BUFFER             -12
+#define EAS_ERROR_PARAMETER_RANGE           -13
+#define EAS_ERROR_MAX_FILES_OPEN            -14
+#define EAS_ERROR_UNRECOGNIZED_FORMAT       -15
+#define EAS_BUFFER_SIZE_MISMATCH            -16
+#define EAS_ERROR_FILE_FORMAT               -17
+#define EAS_ERROR_SMF_NOT_INITIALIZED       -18
+#define EAS_ERROR_LOCATE_BEYOND_END         -19
+#define EAS_ERROR_INVALID_PCM_TYPE          -20
+#define EAS_ERROR_MAX_PCM_STREAMS           -21
+#define EAS_ERROR_NO_VOICE_ALLOCATED        -22
+#define EAS_ERROR_INVALID_CHANNEL           -23
+#define EAS_ERROR_ALREADY_STOPPED           -24
+#define EAS_ERROR_FILE_READ_FAILED          -25
+#define EAS_ERROR_HANDLE_INTEGRITY          -26
+#define EAS_ERROR_MAX_STREAMS_OPEN          -27
+#define EAS_ERROR_INVALID_PARAMETER         -28
+#define EAS_ERROR_FEATURE_NOT_AVAILABLE     -29
+#define EAS_ERROR_SOUND_LIBRARY             -30
+#define EAS_ERROR_NOT_VALID_IN_THIS_STATE   -31
+#define EAS_ERROR_NO_VIRTUAL_SYNTHESIZER    -32
+#define EAS_ERROR_FILE_ALREADY_OPEN         -33
+#define EAS_ERROR_FILE_ALREADY_CLOSED       -34
+#define EAS_ERROR_INCOMPATIBLE_VERSION      -35
+#define EAS_ERROR_QUEUE_IS_FULL             -36
+#define EAS_ERROR_QUEUE_IS_EMPTY            -37
+#define EAS_ERROR_FEATURE_ALREADY_ACTIVE    -38
+
+/* special return codes */
+#define EAS_EOF                             3
+#define EAS_STREAM_BUFFERING                4
+#define EAS_BUFFER_FULL                     5
+
+/* EAS_STATE return codes */
+typedef long EAS_STATE;
+typedef enum
+{
+    EAS_STATE_READY = 0,
+    EAS_STATE_PLAY,
+    EAS_STATE_STOPPING,
+    EAS_STATE_PAUSING,
+    EAS_STATE_STOPPED,
+    EAS_STATE_PAUSED,
+    EAS_STATE_OPEN,
+    EAS_STATE_ERROR,
+    EAS_STATE_EMPTY
+} E_EAS_STATE;
+
+/* constants */
+#ifndef EAS_CONST
+#define EAS_CONST const
+#endif
+
+/* definition for public interface functions */
+#ifndef EAS_PUBLIC
+#define EAS_PUBLIC
+#endif
+
+/* boolean values */
+typedef unsigned EAS_BOOL;
+typedef unsigned char EAS_BOOL8;
+
+#define EAS_FALSE   0
+#define EAS_TRUE    1
+
+/* scalar variable definitions */
+typedef unsigned char EAS_U8;
+typedef signed char EAS_I8;
+typedef char EAS_CHAR;
+
+typedef unsigned short EAS_U16;
+typedef short EAS_I16;
+
+typedef unsigned long EAS_U32;
+typedef long EAS_I32;
+
+typedef unsigned EAS_UINT;
+typedef int EAS_INT;
+typedef long EAS_LONG;
+
+/* audio output type */
+typedef short EAS_PCM;
+
+/* file open modes */
+typedef EAS_I32 EAS_FILE_MODE;
+#define EAS_FILE_READ   1
+#define EAS_FILE_WRITE  2
+
+/* file locator e.g. filename or memory pointer */
+typedef struct s_eas_file_tag {
+    const char* path;
+    int         fd;
+    long long   offset;
+    long long   length;
+} EAS_FILE, *EAS_FILE_LOCATOR;
+
+/* handle to stream */
+typedef struct s_eas_stream_tag *EAS_HANDLE;
+
+/* handle to file */
+typedef struct eas_hw_file_tag *EAS_FILE_HANDLE;
+
+/* handle for synthesizer data */
+typedef struct s_eas_data_tag *EAS_DATA_HANDLE;
+
+/* handle to persistent data for host wrapper interface */
+typedef struct eas_hw_inst_data_tag *EAS_HW_DATA_HANDLE;
+
+/* handle to sound library */
+typedef struct s_eas_sndlib_tag *EAS_SNDLIB_HANDLE;
+typedef struct s_eas_dls_tag *EAS_DLSLIB_HANDLE;
+
+/* pointer to frame buffer - used in split architecture only */
+typedef struct s_eas_frame_buffer_tag *EAS_FRAME_BUFFER_HANDLE;
+
+/* untyped pointer for instance data */
+typedef void *EAS_VOID_PTR;
+
+/* inline functions */
+#ifndef EAS_INLINE
+#if defined (__XCC__)
+#define EAS_INLINE __inline__
+#elif defined (__GNUC__)
+#define EAS_INLINE inline static
+#else
+#define EAS_INLINE __inline
+#endif
+#endif
+
+/* define NULL value */
+#ifndef NULL
+#define NULL 0
+#endif
+
+/* metadata types for metadata return codes */
+typedef enum
+{
+    EAS_METADATA_UNKNOWN = 0,
+    EAS_METADATA_TITLE,
+    EAS_METADATA_AUTHOR,
+    EAS_METADATA_COPYRIGHT,
+    EAS_METADATA_LYRIC,
+    EAS_METADATA_TEXT
+} E_EAS_METADATA_TYPE;
+
+/* metadata callback function */
+typedef void (*EAS_METADATA_CBFUNC) (E_EAS_METADATA_TYPE metaDataType, char *metaDataBuf, EAS_VOID_PTR pUserData);
+
+/* file types for metadata return codes */
+typedef enum
+{
+    EAS_FILE_UNKNOWN = 0,
+    EAS_FILE_SMF0,
+    EAS_FILE_SMF1,
+    EAS_FILE_SMAF_UNKNOWN,
+    EAS_FILE_SMAF_MA2,
+    EAS_FILE_SMAF_MA3,
+    EAS_FILE_SMAF_MA5,
+    EAS_FILE_CMX,
+    EAS_FILE_MFI,
+    EAS_FILE_OTA,
+    EAS_FILE_IMELODY,
+    EAS_FILE_RTTTL,
+    EAS_FILE_XMF0,
+    EAS_FILE_XMF1,
+    EAS_FILE_WAVE_PCM,
+    EAS_FILE_WAVE_IMA_ADPCM,
+    EAS_FILE_MMAPI_TONE_CONTROL
+} E_EAS_FILE_TYPE;
+
+/* enumeration for synthesizers */
+typedef enum
+{
+    EAS_MCU_SYNTH = 0,
+    EAS_DSP_SYNTH
+} E_SYNTHESIZER;
+
+/* external audio callback program change */
+typedef struct s_ext_audio_prg_chg_tag
+{
+    EAS_U16     bank;
+    EAS_U8      program;
+    EAS_U8      channel;
+} S_EXT_AUDIO_PRG_CHG;
+
+/* external audio callback event */
+typedef struct s_ext_audio_event_tag
+{
+    EAS_U8      channel;
+    EAS_U8      note;
+    EAS_U8      velocity;
+    EAS_BOOL8   noteOn;
+} S_EXT_AUDIO_EVENT;
+
+typedef struct s_midi_controllers_tag
+{
+    EAS_U8      modWheel;           /* CC1 */
+    EAS_U8      volume;             /* CC7 */
+    EAS_U8      pan;                /* CC10 */
+    EAS_U8      expression;         /* CC11 */
+    EAS_U8      channelPressure;    /* MIDI channel pressure */
+
+#ifdef  _REVERB
+    EAS_U8      reverbSend;         /* CC91 */
+#endif
+
+#ifdef  _CHORUS
+    EAS_U8      chorusSend;         /* CC93 */
+#endif
+} S_MIDI_CONTROLLERS;
+
+/* iMode play modes enumeration for EAS_SetPlayMode */
+typedef enum
+{
+    IMODE_PLAY_ALL = 0,
+    IMODE_PLAY_PARTIAL
+} E_I_MODE_PLAY_MODE;
+
+typedef EAS_BOOL (*EAS_EXT_PRG_CHG_FUNC) (EAS_VOID_PTR pInstData, S_EXT_AUDIO_PRG_CHG *pPrgChg);
+typedef EAS_BOOL (*EAS_EXT_EVENT_FUNC) (EAS_VOID_PTR pInstData, S_EXT_AUDIO_EVENT *pEvent);
+
+#endif /* #ifndef _EAS_TYPES_H */

common/embeddedaudiosynthesis/arm-wt-22k/host_src/eas_wave.c

@@ -0,0 +1,423 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_wave.c
+ *
+ * Contents and purpose:
+ * This module contains .WAV file functions for the EAS synthesizer
+ * test harness.
+ *
+ * Copyright Sonic Network Inc. 2005
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 658 $
+ *   $Date: 2007-04-24 13:35:49 -0700 (Tue, 24 Apr 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+/* lint complaints about most C library headers, so we use our own during lint step */
+#ifdef _lint
+#include "lint_stdlib.h"
+#else
+#include <stdio.h>
+#include <stdlib.h>
+#endif
+
+#include "eas_wave.h"
+
+/* .WAV file format tags */
+const EAS_U32 riffTag = 0x46464952;
+const EAS_U32 waveTag = 0x45564157;
+const EAS_U32 fmtTag = 0x20746d66;
+const EAS_U32 dataTag = 0x61746164;
+
+#ifdef _BIG_ENDIAN
+/*----------------------------------------------------------------------------
+ * FlipDWord()
+ *----------------------------------------------------------------------------
+ * Purpose: Endian flip a DWORD for big-endian processors
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *
+ *----------------------------------------------------------------------------
+*/
+static void FlipDWord (EAS_U32 *pValue)
+{
+    EAS_U8 *p;
+    EAS_U32 temp;
+
+    p = (EAS_U8*) pValue;
+    temp = (((((p[3] << 8) | p[2]) << 8) | p[1]) << 8) | p[0];
+    *pValue = temp;
+}
+
+/*----------------------------------------------------------------------------
+ * FlipWord()
+ *----------------------------------------------------------------------------
+ * Purpose: Endian flip a WORD for big-endian processors
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *
+ *----------------------------------------------------------------------------
+*/
+static void FlipWord (EAS_U16 *pValue)
+{
+    EAS_U8 *p;
+    EAS_U16 temp;
+
+    p = (EAS_U8*) pValue;
+    temp = (p[1] << 8) | p[0];
+    *pValue = temp;
+}
+
+/*----------------------------------------------------------------------------
+ * FlipWaveHeader()
+ *----------------------------------------------------------------------------
+ * Purpose: Endian flip the wave header for big-endian processors
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *
+ *----------------------------------------------------------------------------
+*/
+static void FlipWaveHeader (WAVE_HEADER *p)
+{
+
+    FlipDWord(&p->nRiffTag);
+    FlipDWord(&p->nRiffSize);
+    FlipDWord(&p->nWaveTag);
+    FlipDWord(&p->nFmtTag);
+    FlipDWord(&p->nFmtSize);
+    FlipDWord(&p->nDataTag);
+    FlipDWord(&p->nDataSize);
+    FlipWord(&p->fc.wFormatTag);
+    FlipWord(&p->fc.nChannels);
+    FlipDWord(&p->fc.nSamplesPerSec);
+    FlipDWord(&p->fc.nAvgBytesPerSec);
+    FlipWord(&p->fc.nBlockAlign);
+    FlipWord(&p->fc.wBitsPerSample);
+
+}
+#endif
+
+/*----------------------------------------------------------------------------
+ * WaveFileCreate()
+ *----------------------------------------------------------------------------
+ * Purpose: Opens a wave file for writing and writes the header
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *
+ *----------------------------------------------------------------------------
+*/
+
+WAVE_FILE *WaveFileCreate (const char *filename, EAS_I32 nChannels, EAS_I32 nSamplesPerSec, EAS_I32 wBitsPerSample)
+{
+    WAVE_FILE *wFile;
+
+    /* allocate memory */
+    wFile = malloc(sizeof(WAVE_FILE));
+    if (!wFile)
+        return NULL;
+    wFile->write = EAS_TRUE;
+
+    /* create the file */
+    wFile->file = fopen(filename,"wb");
+    if (!wFile->file)
+    {
+        free(wFile);
+        return NULL;
+    }
+
+    /* initialize PCM format .WAV file header */
+    wFile->wh.nRiffTag = riffTag;
+    wFile->wh.nRiffSize = sizeof(WAVE_HEADER) - 8;
+    wFile->wh.nWaveTag = waveTag;
+    wFile->wh.nFmtTag = fmtTag;
+    wFile->wh.nFmtSize = sizeof(FMT_CHUNK);
+
+    /* initalize 'fmt' chunk */
+    wFile->wh.fc.wFormatTag = 1;
+    wFile->wh.fc.nChannels = (EAS_U16) nChannels;
+    wFile->wh.fc.nSamplesPerSec = (EAS_U32) nSamplesPerSec;
+    wFile->wh.fc.wBitsPerSample = (EAS_U16) wBitsPerSample;
+    wFile->wh.fc.nBlockAlign = (EAS_U16) (nChannels * (EAS_U16) (wBitsPerSample / 8));
+    wFile->wh.fc.nAvgBytesPerSec = wFile->wh.fc.nBlockAlign * (EAS_U32) nSamplesPerSec;
+
+    /* initialize 'data' chunk */
+    wFile->wh.nDataTag = dataTag;
+    wFile->wh.nDataSize = 0;
+
+#ifdef _BIG_ENDIAN
+    FlipWaveHeader(&wFile->wh);
+#endif
+
+    /* write the header */
+    if (fwrite(&wFile->wh, sizeof(WAVE_HEADER), 1, wFile->file) != 1)
+    {
+        fclose(wFile->file);
+        free(wFile);
+        return NULL;
+    }
+
+#ifdef _BIG_ENDIAN
+    FlipWaveHeader(&wFile->wh);
+#endif
+
+    /* return the file handle */
+    return wFile;
+} /* end WaveFileCreate */
+
+/*----------------------------------------------------------------------------
+ * WaveFileWrite()
+ *----------------------------------------------------------------------------
+ * Purpose: Writes data to the wave file
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_I32 WaveFileWrite (WAVE_FILE *wFile, void *buffer, EAS_I32 n)
+{
+    EAS_I32 count;
+
+    /* make sure we have an open file */
+    if (wFile == NULL)
+    {
+        return 0;
+    }
+
+#ifdef _BIG_ENDIAN
+    {
+        EAS_I32 i;
+        EAS_U16 *p;
+        p = buffer;
+        i = n >> 1;
+        while (i--)
+            FlipWord(p++);
+    }
+#endif
+
+    /* write the data */
+    count = (EAS_I32) fwrite(buffer, 1, (size_t) n, wFile->file);
+
+    /* add the number of bytes written */
+    wFile->wh.nRiffSize += (EAS_U32) count;
+    wFile->wh.nDataSize += (EAS_U32) count;
+
+    /* return the count of bytes written */
+    return count;
+} /* end WriteWaveHeader */
+
+/*----------------------------------------------------------------------------
+ * WaveFileClose()
+ *----------------------------------------------------------------------------
+ * Purpose: Opens a wave file for writing and writes the header
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *
+ *----------------------------------------------------------------------------
+*/
+
+EAS_BOOL WaveFileClose (WAVE_FILE *wFile)
+{
+    EAS_I32 count = 1;
+
+    /* return to beginning of file and write the header */
+    if (wFile->write)
+    {
+        if (fseek(wFile->file, 0L, SEEK_SET) == 0)
+        {
+
+#ifdef _BIG_ENDIAN
+            FlipWaveHeader(&wFile->wh);
+#endif
+            count = (EAS_I32) fwrite(&wFile->wh, sizeof(WAVE_HEADER), 1, wFile->file);
+#ifdef _BIG_ENDIAN
+            FlipWaveHeader(&wFile->wh);
+#endif
+        }
+    }
+
+    /* close the file */
+    if (fclose(wFile->file) != 0)
+        count = 0;
+
+    /* free the memory */
+    free(wFile);
+
+    /* return the file handle */
+    return (count == 1 ? EAS_TRUE : EAS_FALSE);
+} /* end WaveFileClose */
+
+#ifdef _WAVE_FILE_READ
+#ifdef _BIG_ENDIAN
+#error "WaveFileOpen not currently supported on big-endian processors"
+#endif
+/*----------------------------------------------------------------------------
+ * WaveFileOpen()
+ *----------------------------------------------------------------------------
+ * Purpose: Opens a wave file for reading and reads the header
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *
+ *----------------------------------------------------------------------------
+*/
+
+WAVE_FILE *WaveFileOpen (const char *filename)
+{
+    WAVE_FILE *wFile;
+    struct
+    {
+        EAS_U32 tag;
+        EAS_U32 size;
+    } chunk;
+    EAS_U32 tag;
+    EAS_I32 startChunkPos;
+    EAS_INT state;
+    EAS_BOOL done;
+
+    /* allocate memory */
+    wFile = malloc(sizeof(WAVE_FILE));
+    if (!wFile)
+        return NULL;
+
+    /* open the file */
+    wFile->write = EAS_FALSE;
+    wFile->file = fopen(filename,"rb");
+    if (!wFile->file)
+    {
+        free(wFile);
+        return NULL;
+    }
+
+    /* make lint happy */
+    chunk.tag = chunk.size = 0;
+    startChunkPos = 0;
+
+    /* read the RIFF tag and file size */
+    state = 0;
+    done = EAS_FALSE;
+    while (!done)
+    {
+
+        switch(state)
+        {
+            /* read the RIFF tag */
+            case 0:
+                if (fread(&chunk, sizeof(chunk), 1, wFile->file) != 1)
+                    done = EAS_TRUE;
+                else
+                {
+                    if (chunk.tag != riffTag)
+                        done = EAS_TRUE;
+                    else
+                        state++;
+                }
+                break;
+
+            /* read the WAVE tag */
+            case 1:
+                if (fread(&tag, sizeof(tag), 1, wFile->file) != 1)
+                    done = EAS_TRUE;
+                else
+                {
+                    if (tag != waveTag)
+                        done = EAS_TRUE;
+                    else
+                        state++;
+                }
+                break;
+
+            /* looking for fmt chunk */
+            case 2:
+                if (fread(&chunk, sizeof(chunk), 1, wFile->file) != 1)
+                    done = EAS_TRUE;
+                else
+                {
+                    startChunkPos = ftell(wFile->file);
+
+                    /* not fmt tag, skip it */
+                    if (chunk.tag != fmtTag)
+                        fseek(wFile->file, startChunkPos + (EAS_I32) chunk.size, SEEK_SET);
+                    else
+                        state++;
+                }
+                break;
+
+            /* read fmt chunk */
+            case 3:
+                if (fread(&wFile->wh.fc, sizeof(FMT_CHUNK), 1, wFile->file) != 1)
+                    done = EAS_TRUE;
+                else
+                {
+                    fseek(wFile->file, startChunkPos + (EAS_I32) chunk.size, SEEK_SET);
+                    state++;
+                }
+                break;
+
+            /* looking for data chunk */
+            case 4:
+                if (fread(&chunk, sizeof(chunk), 1, wFile->file) != 1)
+                    done = EAS_TRUE;
+                else
+                {
+                    startChunkPos = ftell(wFile->file);
+
+                    /* not data tag, skip it */
+                    if (chunk.tag != dataTag)
+                        fseek(wFile->file, startChunkPos + (EAS_I32) chunk.size, SEEK_SET);
+                    else
+                    {
+                        wFile->dataSize = chunk.size;
+                        state++;
+                        done = EAS_TRUE;
+                    }
+                }
+                break;
+
+            default:
+                done = EAS_TRUE;
+                break;
+        }
+    }
+
+    /* if not final state, an error occurred */
+    if (state != 5)
+    {
+        fclose(wFile->file);
+        free(wFile);
+        return NULL;
+    }
+
+    /* return the file handle */
+    return wFile;
+} /* end WaveFileOpen */
+#endif
+
+
+

common/embeddedaudiosynthesis/arm-wt-22k/host_src/eas_wave.h

@@ -0,0 +1,74 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_wave.h
+ *
+ * Contents and purpose:
+ * Writes output to a .WAV file
+ *
+ * DO NOT MODIFY THIS FILE!
+ *
+ * Copyright Sonic Network Inc. 2005
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 82 $
+ *   $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+#include "eas_types.h"
+
+/* sentinel */
+#ifndef _EAS_WAVE_H
+#define _EAS_WAVE_H
+
+/* .WAV file format chunk */
+typedef struct {
+    EAS_U16 wFormatTag;
+    EAS_U16 nChannels;
+    EAS_U32 nSamplesPerSec;
+    EAS_U32 nAvgBytesPerSec;
+    EAS_U16 nBlockAlign;
+    EAS_U16 wBitsPerSample;
+} FMT_CHUNK;
+
+/* .WAV file header */
+typedef struct {
+    EAS_U32 nRiffTag;
+    EAS_U32 nRiffSize;
+    EAS_U32 nWaveTag;
+    EAS_U32 nFmtTag;
+    EAS_U32 nFmtSize;
+    FMT_CHUNK fc;
+    EAS_U32 nDataTag;
+    EAS_U32 nDataSize;
+} WAVE_HEADER;
+
+typedef struct {
+    WAVE_HEADER wh;
+    FILE *file;
+    EAS_BOOL write;
+    EAS_U32 dataSize;
+} WAVE_FILE;
+
+WAVE_FILE *WaveFileCreate (const char *filename, EAS_I32 nChannels, EAS_I32 nSamplesPerSec, EAS_I32 wBitsPerSample);
+EAS_I32 WaveFileWrite (WAVE_FILE *wFile, void *buffer, EAS_I32 n);
+EAS_BOOL WaveFileClose (WAVE_FILE *wFile);
+WAVE_FILE *WaveFileOpen (const char *filename);
+
+#endif /* end #ifndef _EAS_WAVE_H */
+
+
+

common/embeddedaudiosynthesis/arm-wt-22k/host_src/jet.h

@@ -0,0 +1,199 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * jet.h
+ *
+ * Contents and purpose:
+ * Public interface for JET sound engine
+ *
+ * Copyright (c) 2006 Sonic Network Inc.
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 554 $
+ *   $Date: 2007-02-02 11:06:10 -0800 (Fri, 02 Feb 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _JET_H
+#define _JET_H
+
+#include "eas_types.h"
+#include "eas.h"
+
+/* for C++ linkage */
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/* opaque handle types for JET interface */
+typedef struct s_jet_data_tag *JET_DATA_HANDLE;
+
+typedef struct s_jet_config_tag
+{
+    EAS_U8      appEventRangeLow;
+    EAS_U8      appEventRangeHigh;
+} S_JET_CONFIG;
+
+typedef struct s_jet_status_tag
+{
+    EAS_INT     currentUserID;
+    EAS_INT     segmentRepeatCount;
+    EAS_INT     numQueuedSegments;
+    EAS_BOOL    paused;
+    EAS_I32     location;
+    EAS_U8      currentPlayingSegment;
+    EAS_U8      currentQueuedSegment;
+} S_JET_STATUS;
+
+typedef struct s_jet_event_tag
+{
+    EAS_U8      segment;
+    EAS_U8      channel;
+    EAS_U8      track;
+    EAS_U8      controller;
+    EAS_U8      value;
+} S_JET_EVENT;
+
+/*----------------------------------------------------------------------------
+ * JET_Init()
+ *----------------------------------------------------------------------------
+ * Initializes the JET library, allocates memory, etc. Call
+ * JET_Shutdown to de-allocate memory. Pass NULL for pConfig
+ * to use defaults. If passing config data, configSize should be
+ * sizeof(S_JET_CONFIG). This allows for future expansion of the
+ * config structure while maintaining compatibility.
+ *----------------------------------------------------------------------------
+*/
+EAS_PUBLIC EAS_RESULT JET_Init (EAS_DATA_HANDLE easHandle, const S_JET_CONFIG *pConfig, EAS_INT configSize);
+
+/*----------------------------------------------------------------------------
+ * JET_Shutdown()
+ *----------------------------------------------------------------------------
+ * Frees any memory used by the JET library
+ *----------------------------------------------------------------------------
+*/
+EAS_PUBLIC EAS_RESULT JET_Shutdown (EAS_DATA_HANDLE easHandle);
+
+/*----------------------------------------------------------------------------
+ * JET_OpenFile()
+ *----------------------------------------------------------------------------
+ * Opens a JET content file for playback
+ *----------------------------------------------------------------------------
+*/
+EAS_PUBLIC EAS_RESULT JET_OpenFile (EAS_DATA_HANDLE easHandle, EAS_FILE_LOCATOR locator);
+
+/*----------------------------------------------------------------------------
+ * JET_GetAppData()
+ *----------------------------------------------------------------------------
+ * Returns location and size of application data in the JET file
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT JET_GetAppData (EAS_DATA_HANDLE easHandle, EAS_I32 *pAppDataOffset, EAS_I32 *pAppDataSize);
+
+/*----------------------------------------------------------------------------
+ * JET_CloseFile()
+ *----------------------------------------------------------------------------
+ * Closes a JET content file and releases associated resources
+ *----------------------------------------------------------------------------
+*/
+EAS_PUBLIC EAS_RESULT JET_CloseFile (EAS_DATA_HANDLE easHandle);
+
+/*----------------------------------------------------------------------------
+ * JET_Status()
+ *----------------------------------------------------------------------------
+ * Returns current status
+ *----------------------------------------------------------------------------
+*/
+EAS_PUBLIC EAS_RESULT JET_Status (EAS_DATA_HANDLE easHandle, S_JET_STATUS *pStatus);
+
+/*----------------------------------------------------------------------------
+ * JET_GetEvent()
+ *----------------------------------------------------------------------------
+ * Checks for application events
+ *----------------------------------------------------------------------------
+*/
+EAS_PUBLIC EAS_BOOL JET_GetEvent (EAS_DATA_HANDLE easHandle, EAS_U32 *pEventRaw, S_JET_EVENT *pEvent);
+
+/*----------------------------------------------------------------------------
+ * JET_ParseEvent()
+ *----------------------------------------------------------------------------
+ * Returns current status
+ *----------------------------------------------------------------------------
+*/
+EAS_PUBLIC void JET_ParseEvent (EAS_U32 event, S_JET_EVENT *pEvent);
+
+/*----------------------------------------------------------------------------
+ * JET_QueueSegment()
+ *----------------------------------------------------------------------------
+ * Queue a segment for playback
+ *----------------------------------------------------------------------------
+*/
+EAS_PUBLIC EAS_RESULT JET_QueueSegment (EAS_DATA_HANDLE easHandle, EAS_INT segmentNum, EAS_INT libNum, EAS_INT repeatCount, EAS_INT transpose, EAS_U32 muteFlags, EAS_U8 userID);
+
+/*----------------------------------------------------------------------------
+ * JET_Play()
+ *----------------------------------------------------------------------------
+ * Starts playback of the file
+ *----------------------------------------------------------------------------
+*/
+EAS_PUBLIC EAS_RESULT JET_Play (EAS_DATA_HANDLE easHandle);
+
+/*----------------------------------------------------------------------------
+ * JET_Pause()
+ *----------------------------------------------------------------------------
+ * Pauses playback of the file
+ *----------------------------------------------------------------------------
+*/
+EAS_PUBLIC EAS_RESULT JET_Pause (EAS_DATA_HANDLE easHandle);
+
+/*----------------------------------------------------------------------------
+ * JET_SetMuteFlags()
+ *----------------------------------------------------------------------------
+ * Change the state of the mute flags
+ *----------------------------------------------------------------------------
+*/
+EAS_PUBLIC EAS_RESULT JET_SetMuteFlags (EAS_DATA_HANDLE easHandle, EAS_U32 muteFlags, EAS_BOOL sync);
+
+/*----------------------------------------------------------------------------
+ * JET_SetMuteFlag()
+ *----------------------------------------------------------------------------
+ * Change the state of a single mute flag
+ *----------------------------------------------------------------------------
+*/
+EAS_PUBLIC EAS_RESULT JET_SetMuteFlag (EAS_DATA_HANDLE easHandle, EAS_INT trackNum, EAS_BOOL muteFlag, EAS_BOOL sync);
+
+/*----------------------------------------------------------------------------
+ * JET_TriggerClip()
+ *----------------------------------------------------------------------------
+ * Unmute a track and then mute it when it is complete
+ *----------------------------------------------------------------------------
+*/
+EAS_PUBLIC EAS_RESULT JET_TriggerClip (EAS_DATA_HANDLE easHandle, EAS_INT clipID);
+
+/*----------------------------------------------------------------------------
+ * JET_Clear_Queue()
+ *----------------------------------------------------------------------------
+ * Clears all segments in the queue
+ *----------------------------------------------------------------------------
+*/
+EAS_PUBLIC EAS_RESULT JET_Clear_Queue (EAS_DATA_HANDLE easHandle);
+
+#ifdef __cplusplus
+} /* end extern "C" */
+#endif
+
+
+#endif
+

common/embeddedaudiosynthesis/arm-wt-22k/jetcreator_lib_src/darwin-x86/EASLIb.xcodeproj/project.pbxproj

@@ -0,0 +1,361 @@
+// !$*UTF8*$!
+{
+	archiveVersion = 1;
+	classes = {
+	};
+	objectVersion = 44;
+	objects = {
+
+/* Begin PBXBuildFile section */
+		9A56ACEA0F72B84D00D115A7 /* libeaswt_vst_lib.a in Frameworks */ = {isa = PBXBuildFile; fileRef = 9A56ACD80F72B7D600D115A7 /* libeaswt_vst_lib.a */; };
+		C5AB01090E3400DC0051F183 /* EASLib.h in Headers */ = {isa = PBXBuildFile; fileRef = C5AB01070E3400DC0051F183 /* EASLib.h */; };
+		C5AB010A0E3400DC0051F183 /* EASLib.c in Sources */ = {isa = PBXBuildFile; fileRef = C5AB01080E3400DC0051F183 /* EASLib.c */; };
+		C5AB01240E3406090051F183 /* eas_config.c in Sources */ = {isa = PBXBuildFile; fileRef = C5AB01230E3406090051F183 /* eas_config.c */; };
+		C5AB017A0E340DD60051F183 /* AudioUnit.framework in Frameworks */ = {isa = PBXBuildFile; fileRef = C5AB01790E340DD60051F183 /* AudioUnit.framework */; };
+		C5AB018C0E340EAA0051F183 /* AudioToolbox.framework in Frameworks */ = {isa = PBXBuildFile; fileRef = C5AB018B0E340EAA0051F183 /* AudioToolbox.framework */; };
+		C5AB01910E340EC50051F183 /* CoreAudio.framework in Frameworks */ = {isa = PBXBuildFile; fileRef = C5AB01900E340EC50051F183 /* CoreAudio.framework */; };
+		C5AB02F60E341AAC0051F183 /* Carbon.framework in Frameworks */ = {isa = PBXBuildFile; fileRef = C5AB02F50E341AAC0051F183 /* Carbon.framework */; };
+/* End PBXBuildFile section */
+
+/* Begin PBXContainerItemProxy section */
+		9A56ACD70F72B7D600D115A7 /* PBXContainerItemProxy */ = {
+			isa = PBXContainerItemProxy;
+			containerPortal = 9A56ACD30F72B7D600D115A7 /* easwt_vst_lib.xcodeproj */;
+			proxyType = 2;
+			remoteGlobalIDString = D2AAC046055464E500DB518D;
+			remoteInfo = easwt_vst_lib;
+		};
+		9A56ACFC0F72B8B200D115A7 /* PBXContainerItemProxy */ = {
+			isa = PBXContainerItemProxy;
+			containerPortal = 9A56ACD30F72B7D600D115A7 /* easwt_vst_lib.xcodeproj */;
+			proxyType = 1;
+			remoteGlobalIDString = D2AAC045055464E500DB518D;
+			remoteInfo = easwt_vst_lib;
+		};
+/* End PBXContainerItemProxy section */
+
+/* Begin PBXFileReference section */
+		9A56ACD30F72B7D600D115A7 /* easwt_vst_lib.xcodeproj */ = {isa = PBXFileReference; lastKnownFileType = "wrapper.pb-project"; path = easwt_vst_lib.xcodeproj; sourceTree = "<group>"; };
+		C5AB01070E3400DC0051F183 /* EASLib.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = EASLib.h; sourceTree = "<group>"; };
+		C5AB01080E3400DC0051F183 /* EASLib.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = EASLib.c; sourceTree = "<group>"; };
+		C5AB01230E3406090051F183 /* eas_config.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = eas_config.c; path = ../../host_src/eas_config.c; sourceTree = SOURCE_ROOT; };
+		C5AB01790E340DD60051F183 /* AudioUnit.framework */ = {isa = PBXFileReference; lastKnownFileType = wrapper.framework; name = AudioUnit.framework; path = /System/Library/Frameworks/AudioUnit.framework; sourceTree = "<absolute>"; };
+		C5AB018B0E340EAA0051F183 /* AudioToolbox.framework */ = {isa = PBXFileReference; lastKnownFileType = wrapper.framework; name = AudioToolbox.framework; path = /System/Library/Frameworks/AudioToolbox.framework; sourceTree = "<absolute>"; };
+		C5AB01900E340EC50051F183 /* CoreAudio.framework */ = {isa = PBXFileReference; lastKnownFileType = wrapper.framework; name = CoreAudio.framework; path = /System/Library/Frameworks/CoreAudio.framework; sourceTree = "<absolute>"; };
+		C5AB02F50E341AAC0051F183 /* Carbon.framework */ = {isa = PBXFileReference; lastKnownFileType = wrapper.framework; name = Carbon.framework; path = /System/Library/Frameworks/Carbon.framework; sourceTree = "<absolute>"; };
+		D2AAC0630554660B00DB518D /* libEASLIb.dylib */ = {isa = PBXFileReference; explicitFileType = "compiled.mach-o.dylib"; includeInIndex = 0; path = libEASLIb.dylib; sourceTree = BUILT_PRODUCTS_DIR; };
+/* End PBXFileReference section */
+
+/* Begin PBXFrameworksBuildPhase section */
+		D289988505E68E00004EDB86 /* Frameworks */ = {
+			isa = PBXFrameworksBuildPhase;
+			buildActionMask = 2147483647;
+			files = (
+				9A56ACEA0F72B84D00D115A7 /* libeaswt_vst_lib.a in Frameworks */,
+				C5AB017A0E340DD60051F183 /* AudioUnit.framework in Frameworks */,
+				C5AB018C0E340EAA0051F183 /* AudioToolbox.framework in Frameworks */,
+				C5AB01910E340EC50051F183 /* CoreAudio.framework in Frameworks */,
+				C5AB02F60E341AAC0051F183 /* Carbon.framework in Frameworks */,
+			);
+			runOnlyForDeploymentPostprocessing = 0;
+		};
+/* End PBXFrameworksBuildPhase section */
+
+/* Begin PBXGroup section */
+		08FB7794FE84155DC02AAC07 /* EASLIb */ = {
+			isa = PBXGroup;
+			children = (
+				9A56ACD30F72B7D600D115A7 /* easwt_vst_lib.xcodeproj */,
+				C5AB02F50E341AAC0051F183 /* Carbon.framework */,
+				C5AB01900E340EC50051F183 /* CoreAudio.framework */,
+				C5AB018B0E340EAA0051F183 /* AudioToolbox.framework */,
+				C5AB01790E340DD60051F183 /* AudioUnit.framework */,
+				08FB7795FE84155DC02AAC07 /* Source */,
+				1AB674ADFE9D54B511CA2CBB /* Products */,
+			);
+			name = EASLIb;
+			sourceTree = "<group>";
+		};
+		08FB7795FE84155DC02AAC07 /* Source */ = {
+			isa = PBXGroup;
+			children = (
+				C5AB01230E3406090051F183 /* eas_config.c */,
+				C5AB01070E3400DC0051F183 /* EASLib.h */,
+				C5AB01080E3400DC0051F183 /* EASLib.c */,
+			);
+			name = Source;
+			sourceTree = "<group>";
+		};
+		1AB674ADFE9D54B511CA2CBB /* Products */ = {
+			isa = PBXGroup;
+			children = (
+				D2AAC0630554660B00DB518D /* libEASLIb.dylib */,
+			);
+			name = Products;
+			sourceTree = "<group>";
+		};
+		9A56ACD40F72B7D600D115A7 /* Products */ = {
+			isa = PBXGroup;
+			children = (
+				9A56ACD80F72B7D600D115A7 /* libeaswt_vst_lib.a */,
+			);
+			name = Products;
+			sourceTree = "<group>";
+		};
+/* End PBXGroup section */
+
+/* Begin PBXHeadersBuildPhase section */
+		D2AAC0600554660B00DB518D /* Headers */ = {
+			isa = PBXHeadersBuildPhase;
+			buildActionMask = 2147483647;
+			files = (
+				C5AB01090E3400DC0051F183 /* EASLib.h in Headers */,
+			);
+			runOnlyForDeploymentPostprocessing = 0;
+		};
+/* End PBXHeadersBuildPhase section */
+
+/* Begin PBXNativeTarget section */
+		D2AAC0620554660B00DB518D /* EASLIb */ = {
+			isa = PBXNativeTarget;
+			buildConfigurationList = 1DEB914A08733D8E0010E9CD /* Build configuration list for PBXNativeTarget "EASLIb" */;
+			buildPhases = (
+				D2AAC0600554660B00DB518D /* Headers */,
+				D2AAC0610554660B00DB518D /* Sources */,
+				D289988505E68E00004EDB86 /* Frameworks */,
+			);
+			buildRules = (
+			);
+			dependencies = (
+				9A56ACFD0F72B8B200D115A7 /* PBXTargetDependency */,
+			);
+			name = EASLIb;
+			productName = EASLIb;
+			productReference = D2AAC0630554660B00DB518D /* libEASLIb.dylib */;
+			productType = "com.apple.product-type.library.dynamic";
+		};
+/* End PBXNativeTarget section */
+
+/* Begin PBXProject section */
+		08FB7793FE84155DC02AAC07 /* Project object */ = {
+			isa = PBXProject;
+			buildConfigurationList = 1DEB914E08733D8E0010E9CD /* Build configuration list for PBXProject "EASLIb" */;
+			compatibilityVersion = "Xcode 3.0";
+			developmentRegion = English;
+			hasScannedForEncodings = 1;
+			knownRegions = (
+				en,
+			);
+			mainGroup = 08FB7794FE84155DC02AAC07 /* EASLIb */;
+			projectDirPath = "";
+			projectReferences = (
+				{
+					ProductGroup = 9A56ACD40F72B7D600D115A7 /* Products */;
+					ProjectRef = 9A56ACD30F72B7D600D115A7 /* easwt_vst_lib.xcodeproj */;
+				},
+			);
+			projectRoot = "";
+			targets = (
+				D2AAC0620554660B00DB518D /* EASLIb */,
+			);
+		};
+/* End PBXProject section */
+
+/* Begin PBXReferenceProxy section */
+		9A56ACD80F72B7D600D115A7 /* libeaswt_vst_lib.a */ = {
+			isa = PBXReferenceProxy;
+			fileType = archive.ar;
+			path = libeaswt_vst_lib.a;
+			remoteRef = 9A56ACD70F72B7D600D115A7 /* PBXContainerItemProxy */;
+			sourceTree = BUILT_PRODUCTS_DIR;
+		};
+/* End PBXReferenceProxy section */
+
+/* Begin PBXSourcesBuildPhase section */
+		D2AAC0610554660B00DB518D /* Sources */ = {
+			isa = PBXSourcesBuildPhase;
+			buildActionMask = 2147483647;
+			files = (
+				C5AB010A0E3400DC0051F183 /* EASLib.c in Sources */,
+				C5AB01240E3406090051F183 /* eas_config.c in Sources */,
+			);
+			runOnlyForDeploymentPostprocessing = 0;
+		};
+/* End PBXSourcesBuildPhase section */
+
+/* Begin PBXTargetDependency section */
+		9A56ACFD0F72B8B200D115A7 /* PBXTargetDependency */ = {
+			isa = PBXTargetDependency;
+			name = easwt_vst_lib;
+			targetProxy = 9A56ACFC0F72B8B200D115A7 /* PBXContainerItemProxy */;
+		};
+/* End PBXTargetDependency section */
+
+/* Begin XCBuildConfiguration section */
+		1DEB914B08733D8E0010E9CD /* Debug */ = {
+			isa = XCBuildConfiguration;
+			buildSettings = {
+				COPY_PHASE_STRIP = NO;
+				EXECUTABLE_PREFIX = lib;
+				GCC_DYNAMIC_NO_PIC = NO;
+				GCC_ENABLE_FIX_AND_CONTINUE = YES;
+				GCC_MODEL_TUNING = G5;
+				GCC_OPTIMIZATION_LEVEL = 0;
+				GCC_PREPROCESSOR_DEFINITIONS = (
+					"EAS_MAX_FILE_HANDLES=100",
+					"_DEBUG_CMF=1",
+					"JET_INTERFACE=1",
+					"MAX_SMF_STREAMS=32",
+					"NUM_OUTPUT_CHANNELS=2",
+					"_SAMPLE_RATE_44100=1",
+					"_8_BIT_SAMPLES=1",
+					"_FILTER_ENABLED=1",
+					"_NO_DEBUG_PREPROCESSOR=1",
+					"_CORE_CYCLES_PER_MEMORY_CYCLE=1",
+					"_IMA_DECODER=1",
+					"_XMF_PARSER=1",
+					"_DLS_PARSER=1",
+					"MAX_DLS_MEMORY=16777216",
+					"EAS_WT_SYNTH=1",
+					"MAX_SYNTH_VOICES=64",
+					"X_ENHANCER_ENABLED=1",
+					"_REVERB_ENABLED=1",
+					"_CHORUS_ENABLED=1",
+					"DLS_SYNTHESIZER=1",
+					"EXTERNAL_AUDIO=1",
+					"TEST_HARNESS=1",
+					"MMAPI_SUPPORT=1",
+					"X_MAXIMIZER_ENABLED=1",
+					"X_COMPRESSOR_ENABLED=1",
+					"X_STATS=1",
+					"X_DEBUG_DLS=1",
+					"X_OPTIMIZED_MONO=1",
+					"X_CHECKED_BUILD=1",
+					"_NO_ALIGN=1",
+					"_C_REFERENCE=1",
+					"XMAXIMIZER_USE_FLOATING_POINT=1",
+					"BUFFERED_FILE_ACCESS=1",
+				);
+				HEADER_SEARCH_PATHS = (
+					"${TARGET_BUILD_DIR}/**",
+					"../../Common/**",
+					"../../Parsers/**",
+					"../../Effects/**",
+					"../../WTSynth/**",
+					"../../FMSynth/**",
+					"../../SplitArch/**",
+					"../../Jet/**",
+				);
+				INSTALL_PATH = /usr/local/lib;
+				MACOSX_DEPLOYMENT_TARGET = 10.4;
+				PRODUCT_NAME = EASLIb;
+				ZERO_LINK = YES;
+			};
+			name = Debug;
+		};
+		1DEB914C08733D8E0010E9CD /* Release */ = {
+			isa = XCBuildConfiguration;
+			buildSettings = {
+				ARCHS = i386;
+				DEBUG_INFORMATION_FORMAT = "dwarf-with-dsym";
+				EXECUTABLE_PREFIX = lib;
+				GCC_AUTO_VECTORIZATION = YES;
+				GCC_MODEL_TUNING = G5;
+				GCC_PREPROCESSOR_DEFINITIONS = (
+					"BUFFERED_FILE_ACCESS=1",
+					"EAS_FILE_BUFFER_SIZE=1024",
+					"EAS_MAX_FILE_HANDLES=100",
+					"JET_INTERFACE=1",
+					"MAX_SMF_STREAMS=32",
+					"EAS_WT_SYNTH=1",
+					"NUM_OUTPUT_CHANNELS=2",
+					"_SAMPLE_RATE_44100=1",
+					"_8_BIT_SAMPLES=1",
+					"_FILTER_ENABLED=1",
+					"_NO_DEBUG_PREPROCESSOR=1",
+					"_CORE_CYCLES_PER_MEMORY_CYCLE=1",
+					"_IMA_DECODER=1",
+					"_XMF_PARSER=1",
+					"_DLS_PARSER=1",
+					"MAX_DLS_MEMORY=16777216",
+					"MAX_SYNTH_VOICES=64",
+					"X_ENHANCER_ENABLED=1",
+					"_REVERB_ENABLED=1",
+					"_CHORUS_ENABLED=1",
+					"DLS_SYNTHESIZER=1",
+					"EXTERNAL_AUDIO=1",
+					"TEST_HARNESS=1",
+					"MMAPI_SUPPORT=1",
+					"X_MAXIMIZER_ENABLED=1",
+					"X_COMPRESSOR_ENABLED=1",
+					"X_STATS=1",
+					"X_DEBUG_DLS=1",
+					"X_OPTIMIZED_MONO=1",
+					"X_CHECKED_BUILD=1",
+					"_NO_ALIGN=1",
+					"_C_REFERENCE=1",
+					"XMAXIMIZER_USE_FLOATING_POINT=1",
+				);
+				GCC_UNROLL_LOOPS = YES;
+				HEADER_SEARCH_PATHS = (
+					"${TARGET_BUILD_DIR}/**",
+					"../../Common/**",
+					"../../Parsers/**",
+					"../../Effects/**",
+					"../../WTSynth/**",
+					"../../FMSynth/**",
+					"../../SplitArch/**",
+					"../../Jet/**",
+				);
+				INSTALL_PATH = /usr/local/lib;
+				MACOSX_DEPLOYMENT_TARGET = 10.4;
+				PRODUCT_NAME = EASLIb;
+			};
+			name = Release;
+		};
+		1DEB914F08733D8E0010E9CD /* Debug */ = {
+			isa = XCBuildConfiguration;
+			buildSettings = {
+				GCC_WARN_ABOUT_RETURN_TYPE = YES;
+				GCC_WARN_UNUSED_VARIABLE = YES;
+				PREBINDING = NO;
+				SDKROOT = "$(DEVELOPER_SDK_DIR)/MacOSX10.5.sdk";
+			};
+			name = Debug;
+		};
+		1DEB915008733D8E0010E9CD /* Release */ = {
+			isa = XCBuildConfiguration;
+			buildSettings = {
+				ARCHS = i386;
+				GCC_WARN_ABOUT_RETURN_TYPE = YES;
+				GCC_WARN_UNUSED_VARIABLE = YES;
+				PREBINDING = NO;
+				SDKROOT = "$(DEVELOPER_SDK_DIR)/MacOSX10.5.sdk";
+			};
+			name = Release;
+		};
+/* End XCBuildConfiguration section */
+
+/* Begin XCConfigurationList section */
+		1DEB914A08733D8E0010E9CD /* Build configuration list for PBXNativeTarget "EASLIb" */ = {
+			isa = XCConfigurationList;
+			buildConfigurations = (
+				1DEB914B08733D8E0010E9CD /* Debug */,
+				1DEB914C08733D8E0010E9CD /* Release */,
+			);
+			defaultConfigurationIsVisible = 0;
+			defaultConfigurationName = Release;
+		};
+		1DEB914E08733D8E0010E9CD /* Build configuration list for PBXProject "EASLIb" */ = {
+			isa = XCConfigurationList;
+			buildConfigurations = (
+				1DEB914F08733D8E0010E9CD /* Debug */,
+				1DEB915008733D8E0010E9CD /* Release */,
+			);
+			defaultConfigurationIsVisible = 0;
+			defaultConfigurationName = Release;
+		};
+/* End XCConfigurationList section */
+	};
+	rootObject = 08FB7793FE84155DC02AAC07 /* Project object */;
+}

common/embeddedaudiosynthesis/arm-wt-22k/jetcreator_lib_src/darwin-x86/EASLIb.xcodeproj/xcuserdata/jeff.farrand.xcuserdatad/xcschemes/EASLIb.xcscheme

@@ -0,0 +1,57 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<Scheme
+   version = "1.3">
+   <BuildAction
+      parallelizeBuildables = "YES"
+      buildImplicitDependencies = "YES">
+      <BuildActionEntries>
+         <BuildActionEntry
+            buildForTesting = "YES"
+            buildForRunning = "YES"
+            buildForProfiling = "YES"
+            buildForArchiving = "YES"
+            buildForAnalyzing = "YES">
+            <BuildableReference
+               BuildableIdentifier = "primary"
+               BlueprintIdentifier = "D2AAC0620554660B00DB518D"
+               BuildableName = "libEASLIb.dylib"
+               BlueprintName = "EASLIb"
+               ReferencedContainer = "container:EASLIb.xcodeproj">
+            </BuildableReference>
+         </BuildActionEntry>
+      </BuildActionEntries>
+   </BuildAction>
+   <TestAction
+      selectedDebuggerIdentifier = "Xcode.DebuggerFoundation.Debugger.GDB"
+      selectedLauncherIdentifier = "Xcode.DebuggerFoundation.Launcher.GDB"
+      shouldUseLaunchSchemeArgsEnv = "YES"
+      buildConfiguration = "Debug">
+      <Testables>
+      </Testables>
+   </TestAction>
+   <LaunchAction
+      selectedDebuggerIdentifier = "Xcode.DebuggerFoundation.Debugger.GDB"
+      selectedLauncherIdentifier = "Xcode.DebuggerFoundation.Launcher.GDB"
+      launchStyle = "0"
+      useCustomWorkingDirectory = "NO"
+      buildConfiguration = "Debug"
+      debugDocumentVersioning = "YES"
+      allowLocationSimulation = "YES">
+      <AdditionalOptions>
+      </AdditionalOptions>
+   </LaunchAction>
+   <ProfileAction
+      shouldUseLaunchSchemeArgsEnv = "YES"
+      savedToolIdentifier = ""
+      useCustomWorkingDirectory = "NO"
+      buildConfiguration = "Release"
+      debugDocumentVersioning = "YES">
+   </ProfileAction>
+   <AnalyzeAction
+      buildConfiguration = "Debug">
+   </AnalyzeAction>
+   <ArchiveAction
+      buildConfiguration = "Release"
+      revealArchiveInOrganizer = "YES">
+   </ArchiveAction>
+</Scheme>

common/embeddedaudiosynthesis/arm-wt-22k/jetcreator_lib_src/darwin-x86/EASLIb.xcodeproj/xcuserdata/jeff.farrand.xcuserdatad/xcschemes/xcschememanagement.plist

@@ -0,0 +1,22 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<!DOCTYPE plist PUBLIC "-//Apple//DTD PLIST 1.0//EN" "http://www.apple.com/DTDs/PropertyList-1.0.dtd">
+<plist version="1.0">
+<dict>
+	<key>SchemeUserState</key>
+	<dict>
+		<key>EASLIb.xcscheme</key>
+		<dict>
+			<key>orderHint</key>
+			<integer>13</integer>
+		</dict>
+	</dict>
+	<key>SuppressBuildableAutocreation</key>
+	<dict>
+		<key>D2AAC0620554660B00DB518D</key>
+		<dict>
+			<key>primary</key>
+			<true/>
+		</dict>
+	</dict>
+</dict>
+</plist>

common/embeddedaudiosynthesis/arm-wt-22k/jetcreator_lib_src/darwin-x86/EASLIb.xcodeproj/xcuserdata/ryan.gerleve.xcuserdatad/xcschemes/EASLIb.xcscheme

@@ -0,0 +1,57 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<Scheme
+   version = "1.3">
+   <BuildAction
+      parallelizeBuildables = "YES"
+      buildImplicitDependencies = "YES">
+      <BuildActionEntries>
+         <BuildActionEntry
+            buildForTesting = "YES"
+            buildForRunning = "YES"
+            buildForProfiling = "YES"
+            buildForArchiving = "YES"
+            buildForAnalyzing = "YES">
+            <BuildableReference
+               BuildableIdentifier = "primary"
+               BlueprintIdentifier = "D2AAC0620554660B00DB518D"
+               BuildableName = "libEASLIb.dylib"
+               BlueprintName = "EASLIb"
+               ReferencedContainer = "container:EASLIb.xcodeproj">
+            </BuildableReference>
+         </BuildActionEntry>
+      </BuildActionEntries>
+   </BuildAction>
+   <TestAction
+      selectedDebuggerIdentifier = "Xcode.DebuggerFoundation.Debugger.GDB"
+      selectedLauncherIdentifier = "Xcode.DebuggerFoundation.Launcher.GDB"
+      shouldUseLaunchSchemeArgsEnv = "YES"
+      buildConfiguration = "Debug">
+      <Testables>
+      </Testables>
+   </TestAction>
+   <LaunchAction
+      selectedDebuggerIdentifier = "Xcode.DebuggerFoundation.Debugger.GDB"
+      selectedLauncherIdentifier = "Xcode.DebuggerFoundation.Launcher.GDB"
+      launchStyle = "0"
+      useCustomWorkingDirectory = "NO"
+      buildConfiguration = "Debug"
+      debugDocumentVersioning = "YES"
+      allowLocationSimulation = "YES">
+      <AdditionalOptions>
+      </AdditionalOptions>
+   </LaunchAction>
+   <ProfileAction
+      shouldUseLaunchSchemeArgsEnv = "YES"
+      savedToolIdentifier = ""
+      useCustomWorkingDirectory = "NO"
+      buildConfiguration = "Release"
+      debugDocumentVersioning = "YES">
+   </ProfileAction>
+   <AnalyzeAction
+      buildConfiguration = "Debug">
+   </AnalyzeAction>
+   <ArchiveAction
+      buildConfiguration = "Release"
+      revealArchiveInOrganizer = "YES">
+   </ArchiveAction>
+</Scheme>

common/embeddedaudiosynthesis/arm-wt-22k/jetcreator_lib_src/darwin-x86/EASLIb.xcodeproj/xcuserdata/ryan.gerleve.xcuserdatad/xcschemes/xcschememanagement.plist

@@ -0,0 +1,22 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<!DOCTYPE plist PUBLIC "-//Apple//DTD PLIST 1.0//EN" "http://www.apple.com/DTDs/PropertyList-1.0.dtd">
+<plist version="1.0">
+<dict>
+	<key>SchemeUserState</key>
+	<dict>
+		<key>EASLIb.xcscheme</key>
+		<dict>
+			<key>orderHint</key>
+			<integer>4</integer>
+		</dict>
+	</dict>
+	<key>SuppressBuildableAutocreation</key>
+	<dict>
+		<key>D2AAC0620554660B00DB518D</key>
+		<dict>
+			<key>primary</key>
+			<true/>
+		</dict>
+	</dict>
+</dict>
+</plist>

common/embeddedaudiosynthesis/arm-wt-22k/jetcreator_lib_src/darwin-x86/EASLib.h

@@ -0,0 +1,31 @@
+/*
+ *  EASLib.h
+ *  EASLIb
+ *
+ * Copyright (C) 2008 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ */
+
+#include "eas_types.h"
+
+extern EAS_RESULT EAS_LibInit (EAS_DATA_HANDLE *pHandle)
+{
+    return EAS_Init(pEASData);
+}
+
+EAS_EXPORT EAS_LibShutdown (EAS_DATA_HANDLE handle)
+{
+    EAS_Shutdown(handle);
+}

common/embeddedaudiosynthesis/arm-wt-22k/jetcreator_lib_src/darwin-x86/README.txt

@@ -0,0 +1,9 @@
+OS X JetCreator Library Build Instructions
+
+1/ Open easwt_vst_lib.xcodeproj and build
+
+2/ Open EASLIb.xcodeproj and build
+
+3/ Copy build/Release/libEASLIb.dylib to the JetCreator folder
+
+

common/embeddedaudiosynthesis/arm-wt-22k/jetcreator_lib_src/darwin-x86/eas_host_debug.h

@@ -0,0 +1,47 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_host_debug.h
+ *
+ * Contents and purpose:
+ * This header defines the host wrapper functions for simulating
+ * error conditions.
+ *
+ * Copyright (C) 2008 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+*/
+
+// sentinel
+#ifndef _EAS_HOST_DEBUG_H
+#define _EAS_HOST_DEBUG_H
+
+#include "eas_types.h"
+
+typedef enum
+{
+    eInitError,
+    eShutdownError,
+    eMallocError,
+    eOpenError,
+    eReadError,
+    ePosError,
+    eSeekError,
+    eLengthError,
+    eDupError,
+    eCloseError,
+    eNumErrorConditions,
+} E_ERROR_CONDITIONS;
+
+#endif
+

common/embeddedaudiosynthesis/arm-wt-22k/jetcreator_lib_src/darwin-x86/eastestv37.c

@@ -0,0 +1,999 @@
+/*----------------------------------------------------------------------------
+ *
+ * Copyright (C) 2008 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+*/
+
+#include "eas_sndlib.h"
+
+/*----------------------------------------------------------------------------
+ * Articulations
+ *----------------------------------------------------------------------------
+*/
+const S_ARTICULATION testArticulations[] =
+{
+    { /* articulation 0 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 1 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        100, 0, 19, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 2 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        100, 34, 951, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 3 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        100, 86, 951, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 4 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        100, 172, 951, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 5 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        100, 345, 951, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 6 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        100, 517, 951, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 7 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        100, 689, 951, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 8 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        100, 861, 951, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 9 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        100, 1723, 951, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 10 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        100, 0, 191, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 11 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        100, 0, 382, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 12 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        100, 0, 951, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 13 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        100, 0, 1903, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 14 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        100, 0, 3804, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 15 */
+        { 1902, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 16 */
+        { 380, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 17 */
+        { 190, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 18 */
+        { 38, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 19 */
+        { 19, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 20 */
+        { 10, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 21 */
+        { 5, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 22 */
+        { 32767, 17213, 0, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 23 */
+        { 32767, 28809, 0, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 24 */
+        { 32767, 30725, 0, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 25 */
+        { 32767, 32349, 0, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 26 */
+        { 32767, 32558, 0, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 27 */
+        { 32767, 32663, 0, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 28 */
+        { 32767, 32715, 0, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 29 */
+        { 32767, 30725, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 30 */
+        { 32767, 30725, 3566, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 31 */
+        { 32767, 30725, 42, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 32 */
+        { 32767, 30725, 5, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 33 */
+        { 32767, 30725, 2, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 34 */
+        { 32767, 0, 32767, 17213 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 35 */
+        { 32767, 0, 32767, 28809 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 36 */
+        { 32767, 0, 32767, 30725 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 37 */
+        { 32767, 0, 32767, 32349 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 38 */
+        { 32767, 0, 32767, 32558 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 39 */
+        { 32767, 0, 32767, 32663 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 40 */
+        { 32767, 0, 32767, 32715 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 41 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 190, 0, 0, 0, 1, 0, 0
+    },
+    { /* articulation 42 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 189, 0, 0, 0, 3, 0, 0
+    },
+    { /* articulation 43 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 190, 0, 0, 0, 4, 0, 0
+    },
+    { /* articulation 44 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 190, 0, 0, 0, 6, 0, 0
+    },
+    { /* articulation 45 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        -1200, 0, 190, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 46 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        -600, 0, 190, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 47 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        -100, 0, 190, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 48 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        -50, 0, 190, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 49 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        50, 0, 190, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 50 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        100, 0, 190, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 51 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        600, 0, 190, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 52 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        1200, 0, 190, 0, 0, 0, 0, 0, 0
+    },
+    { /* articulation 53 */
+        { 32767, 0, 32767, 0 },
+        { 190, 190, 0, 0 },
+        0, 0, 951, -1200, 0, 0, 0, 0, 0
+    },
+    { /* articulation 54 */
+        { 32767, 0, 32767, 0 },
+        { 190, 190, 0, 0 },
+        0, 0, 951, -600, 0, 0, 0, 0, 0
+    },
+    { /* articulation 55 */
+        { 32767, 0, 32767, 0 },
+        { 190, 190, 0, 0 },
+        0, 0, 951, -100, 0, 0, 0, 0, 0
+    },
+    { /* articulation 56 */
+        { 32767, 0, 32767, 0 },
+        { 190, 190, 0, 0 },
+        0, 0, 951, -50, 0, 0, 0, 0, 0
+    },
+    { /* articulation 57 */
+        { 32767, 0, 32767, 0 },
+        { 190, 190, 0, 0 },
+        0, 0, 951, 50, 0, 0, 0, 0, 0
+    },
+    { /* articulation 58 */
+        { 32767, 0, 32767, 0 },
+        { 190, 190, 0, 0 },
+        0, 0, 951, 100, 0, 0, 0, 0, 0
+    },
+    { /* articulation 59 */
+        { 32767, 0, 32767, 0 },
+        { 190, 190, 0, 0 },
+        0, 0, 951, 600, 0, 0, 0, 0, 0
+    },
+    { /* articulation 60 */
+        { 32767, 0, 32767, 0 },
+        { 190, 190, 0, 0 },
+        0, 0, 951, 1200, 0, 0, 0, 0, 0
+    },
+    { /* articulation 61 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 5535, 0, 0, 0
+    },
+    { /* articulation 62 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 7121, 0, 0, 0
+    },
+    { /* articulation 63 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 8321, 0, 0, 0
+    },
+    { /* articulation 64 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 9906, 0, 0, 0
+    },
+    { /* articulation 65 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 11106, 0, 0, 0
+    },
+    { /* articulation 66 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 9521, 0, 0, 0
+    },
+    { /* articulation 67 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 9521, 0, 8, 0
+    },
+    { /* articulation 68 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 9521, 0, 16, 0
+    },
+    { /* articulation 69 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 9521, 0, 24, 0
+    },
+    { /* articulation 70 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 9521, 0, 30, 0
+    },
+    { /* articulation 71 */
+        { 32767, 0, 32767, 0 },
+        { 190, 190, 0, 0 },
+        0, 0, 951, 0, -6400, 9521, 0, 0, 0
+    },
+    { /* articulation 72 */
+        { 32767, 0, 32767, 0 },
+        { 190, 190, 0, 0 },
+        0, 0, 951, 0, -3200, 9521, 0, 0, 0
+    },
+    { /* articulation 73 */
+        { 32767, 0, 32767, 0 },
+        { 190, 190, 0, 0 },
+        0, 0, 951, 0, -1600, 9521, 0, 0, 0
+    },
+    { /* articulation 74 */
+        { 32767, 0, 32767, 0 },
+        { 190, 190, 0, 0 },
+        0, 0, 951, 0, -800, 9521, 0, 0, 0
+    },
+    { /* articulation 75 */
+        { 32767, 0, 32767, 0 },
+        { 190, 190, 0, 0 },
+        0, 0, 951, 0, 800, 7121, 0, 0, 0
+    },
+    { /* articulation 76 */
+        { 190, 30725, 32767, 0 },
+        { 190, 190, 0, 0 },
+        0, 0, 951, 0, 1600, 7121, 0, 0, 0
+    },
+    { /* articulation 77 */
+        { 32767, 0, 32767, 0 },
+        { 190, 190, 0, 0 },
+        0, 0, 951, 0, 3200, 7121, 0, 0, 0
+    },
+    { /* articulation 78 */
+        { 32767, 0, 32767, 0 },
+        { 190, 190, 0, 0 },
+        0, 0, 951, 0, 6400, 7121, 0, 0, 0
+    },
+    { /* articulation 79 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 190, 0, 0, 11106, 0, 0, 0
+    },
+    { /* articulation 80 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 191, 0, 0, 11106, 0, 0, 0
+    },
+    { /* articulation 81 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 190, 0, 0, 7121, 0, 0, 0
+    },
+    { /* articulation 82 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 0, 0, 0, -63
+    },
+    { /* articulation 83 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 0, 0, 0, -50
+    },
+    { /* articulation 84 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 0, 0, 0, -37
+    },
+    { /* articulation 85 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 0, 0, 0, -25
+    },
+    { /* articulation 86 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 0, 0, 0, -12
+    },
+    { /* articulation 87 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 0, 0, 0, 12
+    },
+    { /* articulation 88 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 0, 0, 0, 25
+    },
+    { /* articulation 89 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 0, 0, 0, 37
+    },
+    { /* articulation 90 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 0, 0, 0, 50
+    },
+    { /* articulation 91 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 0, 0, 0, 63
+    },
+    { /* articulation 92 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 9907, 0, 0, 0
+    },
+    { /* articulation 93 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 10574, 0, 0, 0
+    },
+    { /* articulation 94 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 11373, 0, 0, 0
+    },
+    { /* articulation 95 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        0, 0, 951, 0, 0, 11376, 0, 0, 0
+    },
+    { /* articulation 96 */
+        { 32767, 0, 32767, 0 },
+        { 32767, 32767, 32767, 0 },
+        100, 0, 949, 0, 0, 0, 0, 0, 0
+    }
+}; /*end Articulations */
+
+/*----------------------------------------------------------------------------
+ * Regions
+ *----------------------------------------------------------------------------
+*/
+const S_WT_REGION testRegions[] =
+{
+    { { 32769, 0, 127 }, -6000, 32767, 101, 301, 4, 0 }, /* region 0 */
+    { { 1, 0, 60 }, -6000, 32767, 101, 301, 4, 2 }, /* region 1 */
+    { { 1, 61, 61 }, -6000, 32767, 101, 301, 4, 3 }, /* region 2 */
+    { { 1, 62, 62 }, -6000, 32767, 101, 301, 4, 4 }, /* region 3 */
+    { { 1, 63, 63 }, -6000, 32767, 101, 301, 4, 5 }, /* region 4 */
+    { { 1, 64, 64 }, -6000, 32767, 101, 301, 4, 6 }, /* region 5 */
+    { { 1, 65, 65 }, -6000, 32767, 101, 301, 4, 7 }, /* region 6 */
+    { { 1, 66, 66 }, -6000, 32767, 101, 301, 4, 8 }, /* region 7 */
+    { { 32769, 67, 127 }, -6000, 32767, 101, 301, 4, 9 }, /* region 8 */
+    { { 32769, 0, 127 }, -6005, 32767, 3, 171, 5, 0 }, /* region 9 */
+    { { 32768, 0, 127 }, -6555, 32767, 0, 0, 2, 0 }, /* region 10 */
+    { { 32770, 0, 127 }, -6000, 32767, 0, 0, 0, 0 }, /* region 11 */
+    { { 1, 60, 60 }, -6000, 32767, 101, 301, 4, 0 }, /* region 12 */
+    { { 1, 61, 61 }, -6100, 16422, 101, 151, 4, 0 }, /* region 13 */
+    { { 1, 62, 62 }, -6200, 8231, 101, 151, 4, 0 }, /* region 14 */
+    { { 1, 63, 63 }, -6300, 2067, 101, 151, 4, 0 }, /* region 15 */
+    { { 1, 64, 64 }, -6400, 130, 101, 151, 4, 0 }, /* region 16 */
+    { { 32769, 65, 65 }, -6500, 1, 101, 151, 4, 0 }, /* region 17 */
+    { { 1, 60, 60 }, -6000, 32767, 101, 301, 4, 0 }, /* region 18 */
+    { { 1, 61, 61 }, -6200, 32767, 101, 151, 4, 0 }, /* region 19 */
+    { { 1, 62, 62 }, -6400, 32767, 101, 151, 4, 0 }, /* region 20 */
+    { { 1, 63, 63 }, -6600, 32767, 101, 151, 4, 0 }, /* region 21 */
+    { { 1, 64, 64 }, -6800, 32767, 101, 151, 4, 0 }, /* region 22 */
+    { { 1, 65, 65 }, -7000, 32767, 101, 151, 4, 0 }, /* region 23 */
+    { { 1, 66, 66 }, -7200, 32767, 101, 151, 4, 0 }, /* region 24 */
+    { { 1, 67, 67 }, -7400, 32767, 101, 151, 4, 0 }, /* region 25 */
+    { { 1, 68, 68 }, -7600, 32767, 101, 151, 4, 0 }, /* region 26 */
+    { { 1, 69, 69 }, -7800, 32767, 101, 151, 4, 0 }, /* region 27 */
+    { { 1, 70, 70 }, -8000, 32767, 101, 151, 4, 0 }, /* region 28 */
+    { { 1, 71, 71 }, -8200, 32767, 101, 151, 4, 0 }, /* region 29 */
+    { { 32769, 72, 72 }, -8400, 32767, 101, 151, 4, 0 }, /* region 30 */
+    { { 1, 60, 60 }, -6000, 32767, 101, 301, 4, 0 }, /* region 31 */
+    { { 1, 61, 61 }, -6099, 32767, 101, 151, 4, 0 }, /* region 32 */
+    { { 1, 62, 62 }, -6190, 32767, 101, 151, 4, 0 }, /* region 33 */
+    { { 1, 63, 63 }, -6250, 32767, 101, 151, 4, 0 }, /* region 34 */
+    { { 1, 64, 64 }, -6300, 32767, 101, 151, 4, 0 }, /* region 35 */
+    { { 1, 65, 65 }, -6500, 32767, 101, 151, 4, 0 }, /* region 36 */
+    { { 1, 66, 66 }, -6601, 32767, 101, 151, 4, 0 }, /* region 37 */
+    { { 1, 67, 67 }, -6710, 32767, 101, 151, 4, 0 }, /* region 38 */
+    { { 1, 68, 68 }, -6850, 32767, 101, 151, 4, 0 }, /* region 39 */
+    { { 32769, 69, 69 }, -7000, 32767, 101, 151, 4, 0 }, /* region 40 */
+    { { 1, 0, 0 }, 0, 32767, 101, 151, 4, 1 }, /* region 41 */
+    { { 1, 1, 1 }, -100, 32767, 101, 151, 4, 10 }, /* region 42 */
+    { { 1, 2, 2 }, -200, 32767, 101, 151, 4, 11 }, /* region 43 */
+    { { 1, 3, 3 }, -300, 32767, 101, 151, 4, 12 }, /* region 44 */
+    { { 1, 4, 4 }, -400, 32767, 101, 151, 4, 13 }, /* region 45 */
+    { { 1, 5, 5 }, -500, 32767, 101, 151, 4, 14 }, /* region 46 */
+    { { 1, 6, 6 }, -600, 32767, 101, 151, 4, 0 }, /* region 47 */
+    { { 1, 7, 7 }, -700, 32767, 101, 151, 4, 15 }, /* region 48 */
+    { { 1, 8, 8 }, -800, 32767, 101, 151, 4, 16 }, /* region 49 */
+    { { 1, 9, 9 }, -900, 32767, 101, 151, 4, 17 }, /* region 50 */
+    { { 1, 10, 10 }, -1000, 32767, 101, 151, 4, 18 }, /* region 51 */
+    { { 1, 11, 11 }, -1100, 32767, 101, 151, 4, 19 }, /* region 52 */
+    { { 1, 12, 12 }, -1200, 32767, 101, 151, 4, 20 }, /* region 53 */
+    { { 1, 13, 13 }, -1300, 32767, 101, 151, 4, 21 }, /* region 54 */
+    { { 1, 14, 14 }, -1400, 32767, 101, 151, 4, 22 }, /* region 55 */
+    { { 1, 15, 15 }, -1500, 32767, 101, 151, 4, 23 }, /* region 56 */
+    { { 1, 16, 16 }, -1600, 32767, 101, 151, 4, 24 }, /* region 57 */
+    { { 1, 17, 17 }, -1700, 32767, 101, 151, 4, 25 }, /* region 58 */
+    { { 1, 18, 18 }, -1800, 32767, 101, 151, 4, 26 }, /* region 59 */
+    { { 1, 19, 19 }, -1900, 32767, 101, 151, 4, 27 }, /* region 60 */
+    { { 1, 20, 20 }, -2000, 32767, 101, 151, 4, 28 }, /* region 61 */
+    { { 1, 21, 21 }, -2100, 32767, 101, 151, 4, 29 }, /* region 62 */
+    { { 1, 22, 22 }, -2200, 32767, 101, 151, 4, 30 }, /* region 63 */
+    { { 1, 23, 23 }, -2300, 32767, 101, 151, 4, 31 }, /* region 64 */
+    { { 1, 24, 24 }, -2400, 32767, 101, 151, 4, 32 }, /* region 65 */
+    { { 1, 25, 25 }, -2500, 32767, 101, 151, 4, 33 }, /* region 66 */
+    { { 1, 26, 26 }, -2600, 32767, 101, 151, 4, 24 }, /* region 67 */
+    { { 1, 27, 27 }, -2700, 32767, 101, 151, 4, 0 }, /* region 68 */
+    { { 1, 28, 28 }, -2800, 32767, 101, 151, 4, 34 }, /* region 69 */
+    { { 1, 29, 29 }, -2900, 32767, 101, 151, 4, 35 }, /* region 70 */
+    { { 1, 30, 30 }, -3000, 32767, 101, 151, 4, 36 }, /* region 71 */
+    { { 1, 31, 31 }, -3100, 32767, 101, 151, 4, 37 }, /* region 72 */
+    { { 1, 32, 32 }, -3200, 32767, 101, 151, 4, 38 }, /* region 73 */
+    { { 1, 33, 33 }, -3300, 32767, 101, 151, 4, 39 }, /* region 74 */
+    { { 1, 34, 34 }, -3400, 32767, 101, 151, 4, 40 }, /* region 75 */
+    { { 1, 35, 35 }, -3500, 32767, 101, 151, 4, 41 }, /* region 76 */
+    { { 1, 36, 36 }, -3600, 32767, 101, 151, 4, 42 }, /* region 77 */
+    { { 1, 37, 37 }, -3700, 32767, 101, 151, 4, 43 }, /* region 78 */
+    { { 1, 38, 38 }, -3800, 32767, 101, 151, 4, 44 }, /* region 79 */
+    { { 1, 39, 39 }, -3900, 32767, 101, 151, 4, 45 }, /* region 80 */
+    { { 1, 40, 40 }, -4000, 32767, 101, 151, 4, 46 }, /* region 81 */
+    { { 1, 41, 41 }, -4100, 32767, 101, 151, 4, 47 }, /* region 82 */
+    { { 1, 42, 42 }, -4200, 32767, 101, 151, 4, 48 }, /* region 83 */
+    { { 1, 43, 43 }, -4300, 32767, 101, 151, 4, 49 }, /* region 84 */
+    { { 1, 44, 44 }, -4400, 32767, 101, 151, 4, 50 }, /* region 85 */
+    { { 1, 45, 45 }, -4500, 32767, 101, 151, 4, 51 }, /* region 86 */
+    { { 1, 46, 46 }, -4600, 32767, 101, 151, 4, 52 }, /* region 87 */
+    { { 1, 47, 47 }, -4700, 32767, 101, 151, 4, 53 }, /* region 88 */
+    { { 1, 48, 48 }, -4800, 32767, 101, 151, 4, 54 }, /* region 89 */
+    { { 1, 49, 49 }, -4900, 32767, 101, 151, 4, 55 }, /* region 90 */
+    { { 1, 50, 50 }, -5000, 32767, 101, 151, 4, 56 }, /* region 91 */
+    { { 1, 51, 51 }, -5100, 32767, 101, 151, 4, 57 }, /* region 92 */
+    { { 1, 52, 52 }, -5200, 32767, 101, 151, 4, 58 }, /* region 93 */
+    { { 1, 53, 53 }, -5300, 32767, 101, 151, 4, 59 }, /* region 94 */
+    { { 1, 54, 54 }, -5400, 32767, 101, 151, 4, 60 }, /* region 95 */
+    { { 2, 55, 55 }, -5500, 32767, 0, 0, 0, 61 }, /* region 96 */
+    { { 2, 56, 56 }, -5600, 32767, 0, 0, 0, 62 }, /* region 97 */
+    { { 2, 57, 57 }, -5700, 32767, 0, 0, 0, 63 }, /* region 98 */
+    { { 2, 58, 58 }, -5800, 32767, 0, 0, 0, 64 }, /* region 99 */
+    { { 2, 59, 59 }, -5900, 32767, 0, 0, 0, 65 }, /* region 100 */
+    { { 2, 60, 60 }, -6000, 32767, 0, 0, 0, 0 }, /* region 101 */
+    { { 2, 61, 61 }, -6100, 32767, 0, 0, 0, 66 }, /* region 102 */
+    { { 2, 62, 62 }, -6200, 32767, 0, 0, 0, 67 }, /* region 103 */
+    { { 2, 63, 63 }, -6300, 32767, 0, 0, 0, 68 }, /* region 104 */
+    { { 2, 64, 64 }, -6400, 32767, 0, 0, 0, 69 }, /* region 105 */
+    { { 2, 65, 65 }, -6500, 32767, 0, 0, 0, 70 }, /* region 106 */
+    { { 2, 66, 66 }, -6600, 32767, 0, 0, 0, 71 }, /* region 107 */
+    { { 2, 67, 67 }, -6700, 32767, 0, 0, 0, 72 }, /* region 108 */
+    { { 2, 68, 68 }, -6800, 32767, 0, 0, 0, 73 }, /* region 109 */
+    { { 2, 69, 69 }, -6900, 32767, 0, 0, 0, 74 }, /* region 110 */
+    { { 2, 70, 70 }, -7000, 32767, 0, 0, 0, 75 }, /* region 111 */
+    { { 2, 71, 71 }, -7100, 32767, 0, 0, 0, 76 }, /* region 112 */
+    { { 2, 72, 72 }, -7200, 32767, 0, 0, 0, 77 }, /* region 113 */
+    { { 2, 73, 73 }, -7300, 32767, 0, 0, 0, 78 }, /* region 114 */
+    { { 2, 74, 74 }, -7400, 32767, 0, 0, 0, 79 }, /* region 115 */
+    { { 2, 75, 75 }, -7500, 32767, 0, 0, 0, 79 }, /* region 116 */
+    { { 2, 76, 76 }, -7600, 32767, 0, 0, 0, 79 }, /* region 117 */
+    { { 2, 77, 77 }, -7700, 32767, 0, 0, 0, 80 }, /* region 118 */
+    { { 2, 78, 78 }, -7800, 32767, 0, 0, 0, 81 }, /* region 119 */
+    { { 2, 79, 79 }, -7900, 32767, 0, 0, 0, 81 }, /* region 120 */
+    { { 2, 80, 80 }, -8000, 32767, 0, 0, 0, 81 }, /* region 121 */
+    { { 2, 81, 81 }, -8100, 32767, 0, 0, 0, 81 }, /* region 122 */
+    { { 2, 82, 82 }, -8200, 32767, 0, 0, 0, 0 }, /* region 123 */
+    { { 257, 83, 83 }, -8300, 32767, 101, 151, 4, 0 }, /* region 124 */
+    { { 257, 84, 84 }, -8405, 32767, 0, 171, 5, 0 }, /* region 125 */
+    { { 0, 85, 85 }, -9055, 32767, 0, 0, 2, 82 }, /* region 126 */
+    { { 0, 86, 86 }, -9155, 32767, 0, 0, 2, 83 }, /* region 127 */
+    { { 0, 87, 87 }, -9255, 32767, 0, 0, 2, 84 }, /* region 128 */
+    { { 0, 88, 88 }, -9355, 32767, 0, 0, 2, 85 }, /* region 129 */
+    { { 0, 89, 89 }, -9455, 32767, 0, 0, 2, 86 }, /* region 130 */
+    { { 0, 90, 90 }, -9555, 32767, 0, 0, 2, 0 }, /* region 131 */
+    { { 0, 91, 91 }, -9655, 32767, 0, 0, 2, 87 }, /* region 132 */
+    { { 0, 92, 92 }, -9755, 32767, 0, 0, 2, 88 }, /* region 133 */
+    { { 0, 93, 93 }, -9855, 32767, 0, 0, 2, 89 }, /* region 134 */
+    { { 0, 94, 94 }, -9955, 32767, 0, 0, 2, 90 }, /* region 135 */
+    { { 0, 95, 95 }, -10055, 32767, 0, 0, 2, 91 }, /* region 136 */
+    { { 2, 96, 96 }, -9600, 32767, 0, 0, 0, 63 }, /* region 137 */
+    { { 2, 97, 97 }, -9700, 32767, 0, 0, 0, 92 }, /* region 138 */
+    { { 2, 98, 98 }, -9800, 32767, 0, 0, 0, 93 }, /* region 139 */
+    { { 2, 99, 99 }, -9900, 32767, 0, 0, 0, 94 }, /* region 140 */
+    { { 2, 100, 100 }, -10000, 32767, 0, 0, 0, 95 }, /* region 141 */
+    { { 32770, 101, 101 }, -10100, 32767, 0, 0, 0, 0 }, /* region 142 */
+    { { 1, 36, 60 }, -6000, 32767, 1481, 1565, 0, 0 }, /* region 143 */
+    { { 1, 61, 61 }, -7300, 32767, 740, 782, 1, 0 }, /* region 144 */
+    { { 32769, 62, 62 }, -8599, 32767, 370, 391, 3, 0 }, /* region 145 */
+    { { 32769, 60, 60 }, -6000, 32767, 101, 301, 4, 1 }, /* region 146 */
+    { { 32769, 60, 60 }, -6000, 32767, 101, 301, 4, 50 }, /* region 147 */
+    { { 32769, 60, 60 }, -6000, 32767, 101, 301, 4, 11 }, /* region 148 */
+    { { 32769, 60, 60 }, -6000, 32767, 101, 301, 4, 96 }, /* region 149 */
+    { { 32769, 60, 60 }, -6000, 32767, 101, 301, 4, 13 }, /* region 150 */
+    { { 32769, 60, 60 }, -6000, 32767, 101, 301, 4, 14 } /* region 151 */
+}; /* end Regions */
+
+/*----------------------------------------------------------------------------
+ * Programs
+ *----------------------------------------------------------------------------
+*/
+const S_PROGRAM testPrograms[] =
+{
+    { 0, 41 } /* program 0 */,
+    { 1, 10 } /* program 1 */,
+    { 2, 11 } /* program 2 */,
+    { 3, 12 } /* program 3 */,
+    { 4, 18 } /* program 4 */,
+    { 5, 31 } /* program 5 */,
+    { 6, 143 } /* program 6 */,
+    { 7, 146 } /* program 7 */,
+    { 8, 147 } /* program 8 */,
+    { 9, 148 } /* program 9 */,
+    { 10, 149 } /* program 10 */,
+    { 11, 150 } /* program 11 */,
+    { 12, 151 } /* program 12 */,
+    { 13, 0 } /* program 13 */,
+    { 14, 9 } /* program 14 */,
+    { 15, 1 } /* program 15 */
+}; /* end Programs */
+
+/*----------------------------------------------------------------------------
+ * Banks
+ *----------------------------------------------------------------------------
+*/
+#define testBanks NULL
+
+/*----------------------------------------------------------------------------
+ * Samples
+ *----------------------------------------------------------------------------
+*/
+
+const EAS_SAMPLE testSamples[] =
+{
+      -1,   -1,   -1,   -1,    0,    1,    2,    3,    4,    5,    6,    7,    9,   10,   11,   12,
+      13,   13,   14,   13,   13,   13,   12,   12,   10,    9,    8,    6,    5,    3,    2,    0,
+      -1,   -3,   -5,   -8,  -10,  -12,  -14,  -15,  -16,  -17,  -17,  -17,  -17,  -16,  -14,  -13,
+     -11,  -10,   -8,   -6,   -5,   -4,   -4,   -3,   -3,   -2,   -1,   -1,   -1,   -1,   -1,   -1,
+      -1,   -2,   -3,   -3,   -3,   -3,   -3,   -2,   -2,   -2,   -1,   -1,   -1,   -1,    0,    0,
+       0,    0,    0,    0,    0,    1,    2,    4,    6,    9,   11,   12,   13,   15,   16,   17,
+      19,   20,   22,   23,   25,   28,   32,   34,   36,   36,   35,   31,   25,   18,   11,    3,
+      -5,  -13,  -20,  -26,  -31,  -34,  -35,  -36,  -36,  -35,  -34,  -32,  -29,  -26,  -23,  -19,
+     -16,  -12,   -8,   -5,   -4,   -4,   -4,   -6,   -8,   -9,  -11,  -11,  -12,  -13,  -13,  -13,
+     -12,  -11,  -10,   -9,   -7,   -5,   -4,   -3,   -2,   -1,   -1,    0,    0,    2,    4,    5,
+       7,    7,    7,    7,    6,    6,    6,    6,    6,    6,    7,    9,   12,   14,   17,   20,
+      24,   27,   30,   33,   36,   39,   41,   43,   46,   49,   51,   51,   50,   48,   43,   35,
+      25,   14,    1,  -12,  -25,  -37,  -48,  -56,  -61,  -63,  -62,  -60,  -57,  -53,  -48,  -42,
+     -35,  -29,  -23,  -17,  -12,   -8,   -5,   -3,   -3,   -4,   -5,   -7,   -9,  -11,  -12,  -14,
+     -14,  -14,  -14,  -13,  -11,   -9,   -7,   -5,   -3,   -1,    0,    1,    1,    1,    2,    3,
+       4,    5,    6,    6,    7,    8,    9,   10,   11,   11,   11,   11,   10,    9,    9,   10,
+      12,   14,   17,   20,   23,   26,   30,   34,   38,   40,   43,   46,   49,   52,   55,   57,
+      58,   56,   50,   42,   30,   16,    1,  -15,  -29,  -43,  -54,  -63,  -68,  -69,  -68,  -65,
+     -60,  -55,  -49,  -42,  -35,  -29,  -22,  -16,  -10,   -6,   -2,   -1,    0,   -1,   -3,   -6,
+      -9,  -12,  -16,  -18,  -20,  -21,  -21,  -21,  -19,  -17,  -14,  -11,   -8,   -5,   -2,    1,
+       3,    5,    7,    9,   11,   12,   12,   12,   11,   10,    9,    8,    7,    7,    7,    7,
+       8,    9,   10,   12,   15,   17,   19,   22,   24,   26,   28,   30,   32,   34,   36,   38,
+      42,   44,   47,   49,   53,   56,   55,   52,   45,   35,   22,    8,   -8,  -23,  -37,  -50,
+     -60,  -67,  -71,  -71,  -69,  -63,  -56,  -48,  -40,  -33,  -27,  -21,  -16,  -12,   -8,   -6,
+      -5,   -5,   -6,   -8,  -11,  -14,  -17,  -19,  -21,  -22,  -22,  -22,  -20,  -18,  -16,  -13,
+      -9,   -6,   -2,    1,    5,    7,    8,    9,   10,   11,   11,   11,   10,   10,    9,    9,
+       9,    8,    8,    9,    9,   10,   10,   12,   13,   14,   16,   18,   19,   21,   24,   26,
+      29,   32,   35,   39,   42,   46,   49,   51,   53,   56,   59,   59,   55,   48,   37,   22,
+       5,  -12,  -29,  -45,  -58,  -68,  -74,  -77,  -77,  -75,  -69,  -62,  -53,  -44,  -35,  -28,
+     -21,  -15,  -11,   -8,   -5,   -4,   -4,   -4,   -6,   -8,  -10,  -13,  -15,  -17,  -19,  -19,
+     -19,  -18,  -16,  -13,  -10,   -8,   -4,   -1,    1,    4,    5,    7,    8,   10,   11,   12,
+      13,   14,   14,   13,   12,   12,   11,    9,    8,    7,    6,    6,    7,    8,   10,   12,
+      15,   18,   21,   25,   29,   34,   39,   45,   51,   57,   61,   67,   73,   77,   82,   84,
+      81,   70,   52,   29,    3,  -24,  -49,  -72,  -90, -103, -110, -112, -109, -102,  -91,  -78,
+     -64,  -49,  -35,  -23,  -13,   -5,    1,    5,    8,    9,    9,    7,    4,    0,   -4,   -9,
+     -12,  -15,  -18,  -18,  -18,  -17,  -14,  -12,   -9,   -6,   -3,    0,    3,    6,    8,   10,
+      11,   12,   11,   11,   10,    8,    7,    5,    4,    2,    2,    1,    2,    3,    4,    6,
+       8,   11,   13,   15,   18,   21,   26,   31,   36,   42,   48,   53,   60,   66,   72,   77,
+      82,   87,   91,   91,   86,   73,   54,   28,   -2,  -32,  -61,  -86, -105, -119, -126, -126,
+    -121, -111,  -98,  -82,  -65,  -48,  -33,  -19,   -7,    1,    8,   12,   14,   14,   12,   10,
+       6,    1,   -4,   -9,  -14,  -18,  -20,  -21,  -22,  -21,  -19,  -17,  -14,  -11,   -8,   -4,
+       0,    3,    7,   10,   12,   13,   14,   14,   13,   12,   11,   10,    9,    8,    7,    5,
+       4,    3,    2,    2,    4,    6,    9,   12,   16,   20,   26,   31,   36,   42,   48,   53,
+      58,   64,   69,   74,   79,   82,   84,   83,   80,   72,   59,   39,   14,  -14,  -43,  -70,
+     -92, -109, -119, -123, -120, -113, -102,  -88,  -73,  -56,  -40,  -25,  -12,   -2,    5,   10,
+      12,   12,   11,    9,    5,    1,   -3,   -8,  -12,  -15,  -18,  -20,  -21,  -21,  -20,  -18,
+     -16,  -13,   -9,   -6,   -2,    1,    5,    8,   11,   13,   15,   16,   16,   16,   15,   14,
+      12,   10,    7,    5,    4,    3,    3,    5,    6,    8,    9,   12,   15,   19,   23,   27,
+      31,   35,   38,   42,   46,   50,   54,   59,   65,   71,   75,   77,   78,   76,   68,   54,
+      34,   10,  -16,  -43,  -68,  -88, -103, -112, -114, -112, -105,  -94,  -80,  -65,  -50,  -35,
+     -22,  -10,   -2,    5,    9,   11,   10,    9,    6,    2,   -3,   -7,  -12,  -15,  -18,  -20,
+     -21,  -22,  -21,  -20,  -17,  -14,  -11,   -7,   -3,    0,    4,    7,    9,   11,   13,   15,
+      17,   17,   17,   16,   14,   12,   11,   10,    8,    7,    5,    5,    5,    6,    7,    8,
+      10,   11,   14,   17,   20,   24,   28,   32,   37,   43,   49,   57,   63,   70,   77,   82,
+      83,   80,   73,   60,   42,   18,   -8,  -34,  -59,  -80,  -95, -105, -110, -109, -103,  -94,
+     -81,  -67,  -53,  -38,  -24,  -13,   -3,    4,    9,   11,   11,    9,    6,    3,   -2,   -6,
+     -10,  -14,  -17,  -19,  -20,  -20,  -20,  -19,  -17,  -15,  -12,   -8,   -5,   -1,    2,    6,
+       9,   11,   13,   15,   15,   16,   16,   15,   14,   12,   11,    9,    8,    6,    6,    5,
+       5,    5,    6,    7,    8,   10,   12,   15,   19,   23,   28,   32,   38,   44,   50,   58,
+      66,   74,   82,   87,   90,   89,   83,   69,   48,   21,   -9,  -41,  -69,  -93, -111, -121,
+    -124, -121, -113, -100,  -85,  -68,  -50,  -33,  -18,   -5,    5,   12,   16,   17,   16,   13,
+       9,    4,   -1,   -6,  -11,  -15,  -18,  -20,  -21,  -21,  -21,  -19,  -17,  -15,  -12,   -8,
+      -4,   -1,    3,    6,    9,   12,   14,   15,   15,   14,   13,   12,   11,   10,    9,    9,
+       8,    8,    6,    6,    5,    5,    6,    7,    8,   11,   15,   18,   23,   27,   31,   36,
+      41,   46,   51,   58,   66,   74,   82,   88,   91,   90,   83,   68,   46,   16,  -16,  -48,
+     -77, -101, -118, -127, -128, -124, -114, -101,  -84,  -66,  -47,  -30,  -14,   -1,    9,   16,
+      19,   20,   19,   15,   11,    5,   -1,   -6,  -11,  -16,  -19,  -21,  -23,  -23,  -23,  -21,
+     -19,  -16,  -12,   -8,   -4,    0,    4,    8,   11,   13,   14,   15,   15,   14,   13,   12,
+      11,   10,    9,    8,    7,    7,    7,    7,    7,    8,    8,    9,   11,   12,   14,   16,
+      19,   23,   27,   31,   35,   40,   45,   51,   58,   66,   74,   82,   88,   90,   88,   77,
+      58,   32,    1,  -31,  -62,  -87, -107, -119, -125, -123, -117, -105,  -90,  -73,  -56,  -38,
+     -22,   -8,    2,   10,   15,   17,   17,   15,   11,    7,    2,   -4,   -9,  -14,  -17,  -20,
+     -22,  -23,  -22,  -21,  -19,  -17,  -14,  -10,   -6,   -2,    2,    5,    9,   12,   14,   15,
+      17,   18,   18,   18,   17,   15,   12,   10,    8,    6,    4,    4,    4,    5,    6,    7,
+       8,   10,   13,   16,   19,   22,   25,   28,   32,   35,   40,   45,   52,   59,   67,   75,
+      81,   85,   86,   80,   66,   45,   18,  -12,  -42,  -69,  -91, -107, -116, -119, -115, -107,
+     -95,  -80,  -64,  -47,  -31,  -17,   -5,    4,   10,   13,   14,   14,   11,    7,    3,   -2,
+      -7,  -12,  -15,  -18,  -20,  -21,  -21,  -20,  -18,  -16,  -14,  -11,   -7,   -4,    0,    4,
+       7,   10,   12,   14,   16,   16,   16,   16,   16,   15,   14,   12,   10,    9,    7,    6,
+       5,    4,    4,    5,    6,    7,    8,   10,   12,   15,   19,   23,   27,   31,   36,   41,
+      47,   55,   64,   73,   81,   87,   90,   86,   75,   56,   30,    1,  -30,  -58,  -82, -100,
+    -112, -117, -115, -109,  -98,  -84,  -69,  -52,  -37,  -22,  -10,    0,    8,   12,   14,   14,
+      13,   10,    6,    2,   -3,   -8,  -12,  -16,  -18,  -20,  -20,  -20,  -19,  -17,  -15,  -12,
+      -9,   -5,   -1,    2,    6,    9,   12,   14,   16,   17,   17,   17,   16,   14,   13,   11,
+       8,    6,    4,    3,    2,    2,    2,    3,    5,    6,    8,   10,   12,   15,   18,   22,
+      26,   30,   35,   40,   47,   55,   63,   71,   80,   87,   90,   88,   78,   59,   34,    4,
+     -27,  -57,  -82, -101, -113, -118, -117, -110,  -99,  -85,  -69,  -53,  -37,  -22,   -9,    2,
+       9,   14,   17,   17,   15,   12,    8,    3,   -2,   -7,  -12,  -16,  -19,  -20,  -21,  -21,
+     -19,  -18,  -15,  -12,   -9,   -6,   -2,    2,    6,    9,   12,   14,   16,   17,   17,   16,
+      14,   12,   10,    8,    6,    5,    4,    3,    3,    2,    3,    4,    6,    7,    9,   11,
+      13,   16,   19,   22,   26,   30,   34,   39,   46,   53,   61,   70,   80,   88,   93,   91,
+      80,   61,   34,    3,  -30,  -59,  -84, -102, -114, -119, -117, -110,  -99,  -84,  -68,  -51,
+     -35,  -20,   -7,    3,   11,   15,   17,   18,   16,   13,    9,    4,   -2,   -7,  -11,  -15,
+     -18,  -20,  -21,  -21,  -20,  -19,  -16,  -14,  -11,   -8,   -4,    0,    3,    7,   10,   13,
+      15,   16,   17,   16,   14,   12,   10,    9,    7,    6,    5,    4,    4,    4,    5,    6,
+       7,    8,    9,   11,   13,   15,   18,   21,   24,   27,   31,   35,   41,   48,   55,   64,
+      73,   82,   89,   91,   85,   71,   48,   19,  -13,  -44,  -71,  -93, -108, -117, -118, -114,
+    -105,  -92,  -77,  -60,  -43,  -27,  -13,   -1,    8,   14,   17,   18,   17,   14,    8,   13,
+      -1,   -1,    0,    2,    4,    6,    9,   11,   13,   14,   13,   12,   11,    8,    5,    2,
+      -1,   -5,  -10,  -14,  -16,  -17,  -16,  -14,  -11,   -8,   -5,   -4,   -3,   -1,   -1,   -1,
+      -1,   -3,   -3,   -3,   -2,   -1,   -1,    0,    0,    0,    0,    2,    6,   11,   14,   16,
+      19,   22,   25,   32,   36,   34,   25,   11,   -5,  -20,  -31,  -35,  -36,  -34,  -29,  -23,
+     -16,   -8,   -4,   -4,   -8,  -11,  -12,  -13,  -12,  -10,   -7,   -4,   -2,   -1,    1,    4,
+       7,    7,    6,    6,    6,    7,   12,   17,   24,   30,   36,   41,   46,   51,   50,   43,
+      25,    2,  -25,  -48,  -61,  -62,  -57,  -48,  -35,  -23,  -12,   -5,   -3,   -5,   -9,  -12,
+     -14,  -14,  -12,   -7,   -3,    0,    1,    2,    4,    6,    7,    9,   11,   11,   10,    9,
+      12,   17,   23,   30,   37,   43,   49,   55,   58,   50,   30,    1,  -29,  -54,  -68,  -68,
+     -60,  -49,  -35,  -22,  -10,   -2,    0,   -3,   -9,  -16,  -20,  -21,  -19,  -14,   -8,   -2,
+       3,    7,   11,   12,   11,    9,    7,    7,    8,   10,   15,   19,   24,   28,   33,   36,
+      42,   47,   53,   56,   45,   23,   -8,  -37,  -60,  -71,  -69,  -56,  -40,  -27,  -16,   -8,
+      -5,   -6,  -11,  -17,  -21,  -22,  -20,  -16,   -9,   -2,    5,    8,   10,   11,   10,    9,
+       8,    8,    9,   10,   13,   16,   19,   24,   29,   35,   42,   49,   53,   59,   55,   37,
+       5,  -29,  -58,  -74,  -77,  -69,  -53,  -35,  -21,  -11,   -5,   -4,   -6,  -10,  -15,  -19,
+     -19,  -16,  -10,   -5,    1,    5,    8,   11,   13,   14,   12,   10,    8,    6,    7,   10,
+      15,   21,   29,   39,   51,   62,   72,   82,   81,   52,    3,  -50,  -90, -110, -109,  -91,
+     -64,  -35,  -13,    1,    8,    9,    4,   -4,  -12,  -18,  -18,  -14,   -9,   -3,    3,    8,
+      11,   11,   10,    7,    4,    2,    2,    4,    8,   13,   18,   26,   36,   48,   60,   72,
+      82,   91,   86,   54,   -2,  -61, -105, -126, -121,  -98,  -65,  -33,   -7,    8,   14,   12,
+       6,   -4,  -14,  -20,  -22,  -19,  -14,   -8,    0,    7,   12,   14,   13,   11,    9,    7,
+       4,    2,    4,    9,   16,   26,   36,   48,   58,   69,   79,   84,   80,   59,   14,  -43,
+     -92, -119, -120, -102,  -73,  -40,  -12,    5,   12,   11,    5,   -3,  -12,  -18,  -21,  -20,
+     -16,   -9,   -2,    5,   11,   15,   16,   15,   12,    7,    4,    4,    6,    9,   15,   23,
+      31,   38,   46,   54,   65,   75,   78,   68,   34,  -16,  -68, -103, -114, -105,  -80,  -50,
+     -22,   -1,    9,   10,    6,   -3,  -12,  -18,  -21,  -21,  -17,  -11,   -3,    4,    9,   13,
+      16,   17,   14,   11,    8,    5,    5,    7,   10,   14,   20,   28,   37,   49,   64,   77,
+      83,   73,   42,   -8,  -59,  -95, -110, -103,  -81,  -53,  -24,   -3,    9,   11,    6,   -2,
+     -10,  -17,  -20,  -20,  -17,  -12,   -5,    2,    9,   13,   15,   16,   14,   11,    8,    6,
+       5,    6,    8,   12,   19,   28,   38,   50,   66,   82,   90,   83,   48,   -9,  -69, -111,
+    -124, -113,  -85,  -50,  -18,    5,   16,   16,    9,   -1,  -11,  -18,  -21,  -21,  -17,  -12,
+      -5,    3,    9,   14,   15,   13,   11,    9,    8,    6,    5,    6,    8,   15,   23,   31,
+      41,   52,   66,   82,   91,   83,   46,  -16,  -77, -118, -128, -114,  -84,  -47,  -14,    9,
+      19,   19,   10,   -1,  -11,  -19,  -23,  -23,  -19,  -12,   -4,    4,   11,   14,   15,   13,
+      11,    9,    7,    7,    7,    8,   11,   14,   19,   27,   35,   45,   58,   74,   88,   88,
+      58,    1,  -62, -107, -125, -116,  -90,  -56,  -22,    2,   15,   17,   11,    2,   -9,  -17,
+     -22,  -22,  -20,  -14,   -6,    2,    9,   14,   17,   18,   17,   12,    8,    4,    4,    6,
+       8,   13,   19,   25,   32,   40,   52,   67,   81,   86,   66,   18,  -42,  -91, -116, -115,
+     -95,  -64,  -31,   -5,   10,   14,   11,    3,   -7,  -15,  -20,  -21,  -18,  -14,   -7,    0,
+       7,   12,   16,   17,   16,   14,   10,    7,    5,    4,    6,    8,   12,   19,   27,   36,
+      47,   64,   81,   90,   75,   30,  -30,  -82, -112, -115,  -98,  -69,  -37,  -10,    8,   14,
+      13,    6,   -3,  -12,  -18,  -20,  -19,  -15,   -9,   -1,    6,   12,   16,   17,   16,   13,
+       8,    4,    2,    2,    5,    8,   12,   18,   26,   35,   47,   63,   80,   90,   78,   34,
+     -27,  -82, -113, -117,  -99,  -69,  -37,   -9,    9,   17,   15,    8,   -2,  -12,  -19,  -21,
+     -19,  -15,   -9,   -2,    6,   12,   16,   17,   14,   10,    6,    4,    3,    3,    6,    9,
+      13,   19,   26,   34,   46,   61,   80,   93,   80,   34,  -30,  -84, -114, -117,  -99,  -68,
+     -35,   -7,   11,   17,   16,    9,   -2,  -11,  -18,  -21,  -20,  -16,  -11,   -4,    3,   10,
+      15,   17,   14,   10,    7,    5,    4,    5,    7,    9,   13,   18,   24,   31,   41,   55,
+      73,   89,   85,   48,  -12,  -71, -108, -118, -104,  -77,  -43,  -13,    8,   16,   17,   16,
+       0,   -3,   -4,   -5,   -4,    0,   -4,   -8,  -20,  -25,   51,   11,  -55,    9,   39,   55,
+     -76,  -19,   92,  -23,  -58,    2,  -15,   57,   71,   34,  -41,   20,   51,   22,   63,  -21,
+      63,   74,   32,   36,   99,  -14,   27,  102,   66,   90,   79,   77,   58,   66,   65,  114,
+      69,   26,   30,   92,   90,   53,   78,   97,   77,   66,   39,   -4,   60,   57,   64,   68,
+     -16,   36,   49,   12,   19,   12,  -12,   21,   11,  -32,  -19,  -41,  -44,  -12,  -36,  -44,
+     -45,  -51,  -55,  -70,  -69,  -73,  -85, -102,  -86,  -99,  -92, -105, -108, -103, -100, -107,
+    -112, -104, -113, -112, -104, -119, -124, -115,  -87, -100, -128, -106,  -83, -105, -108, -109,
+     -91,  -95,  -90,  -83,  -80,  -79,  -83,  -68,  -59,  -53,  -63,  -69,  -57,  -28,  -11,  -38,
+     -43,  -23,  -10,   -6,    3,    1,    6,   14,   21,   23,   34,   40,   42,   48,   57,   61,
+      63,   65,   73,   82,   90,   82,   89,   98,  100,  110,  111,  109,  115,  120,  121,  121,
+     123,  123,  123,  122,  122,  122,  122,  122,  122,  122,  122,  122,  122,  122,  121,  121,
+     121,  121,  121,  121,  121,  121,  121,  121,  121,  121,  120,  120,  120,  120,  120,  120,
+     120,  120,  120,  120,  113,  104,  103,  113,  110,   94,   80,   88,   96,   89,   81,   70,
+      68,   64,   59,   59,   54,   44,   44,   39,   31,   31,   21,   18,   16,    8,    6,    2,
+      -3,   -6,  -13,  -19,  -27,  -31,  -34,  -30,  -38,  -47,  -48,  -51,  -56,  -64,  -68,  -67,
+     -68,  -76,  -73,  -79,  -85,  -94,  -89,  -90,  -94,  -92,  -99, -100, -107, -107, -104, -107,
+    -106, -112, -114, -113, -112, -111, -112, -116, -116, -115, -115, -115, -115, -115, -115, -115,
+    -114, -114, -114, -114, -114, -114, -114, -114, -113, -113, -113, -113, -113, -108, -106, -107,
+    -111, -110, -105, -108, -107, -102, -100, -103, -103, -100, -100,  -97,  -97, -100,  -99,  -97,
+     -94,  -90,  -94,  -94,  -93,  -93,  -92,  -92,  -90,  -89,  -89,  -90,  -91,  -87,  -87,  -88,
+     -89,  -88,  -84,  -85,  -88,  -87,  -86,  -85,  -85,  -84,  -82,  -83,  -83,  -82,  -81,  -80,
+     -79,  -79,  -79,  -78,  -75,  -74,  -75,  -73,  -74,  -71,  -69,  -67,  -69,  -66,  -64,  -66,
+     -65,  -64,  -62,  -58,  -57,  -57,  -57,  -56,  -52,  -51,  -50,  -48,  -46,  -43,  -43,  -42,
+     -39,  -36,  -35,  -34,  -32,  -31,  -29,  -26,  -23,  -23,  -21,  -15,  -18,  -15,  -11,   -8,
+      -6,   -7,    0,    2,    0,    2,    7,   15,   14,   15,   16,   19,   23,   27,   29,   30,
+      31,   34,   41,   41,   42,   44,   45,   51,   55,   54,   56,   57,   60,   65,   67,   69,
+      69,   71,   74,   77,   79,   78,   80,   82,   86,   86,   84,   86,   88,   87,   87,   87,
+      87,   87,   86,   86,   86,   86,   86,   85,   85,   85,   85,   85,   84,   84,   84,   84,
+      84,   83,   83,   83,   83,   83,   83,   82,   82,   82,   82,   82,   81,   81,   81,   81,
+      81,   80,   80,   80,   80,   80,   79,   79,   79,   79,   79,   78,   78,   78,   78,   76,
+      75,   72,   69,   67,   65,   64,   63,   59,   55,   54,   52,   50,   48,   45,   42,   38,
+      36,   34,   31,   30,   27,   24,   21,   18,   16,   14,   12,    9,    7,    4,    1,   -2,
+      -4,   -7,  -10,  -11,  -13,  -16,  -17,  -18,  -21,  -24,  -26,  -27,  -28,  -30,  -32,  -33,
+     -35,  -36,  -38,  -39,  -41,  -42,  -43,  -45,  -45,  -46,  -47,  -48,  -49,  -50,  -51,  -50,
+     -51,  -52,  -53,  -53,  -53,  -53,  -53,  -53,  -53,  -54,  -54,  -54,  -54,  -53,  -53,  -52,
+     -52,  -52,  -51,  -51,  -51,  -50,  -50,  -50,  -48,  -49,  -48,  -47,  -46,  -45,  -45,  -44,
+     -43,  -42,  -41,  -41,  -40,  -39,  -38,  -37,  -36,  -35,  -34,  -33,  -31,  -30,  -29,  -28,
+     -27,  -26,  -24,  -24,  -23,  -22,  -21,  -20,  -19,  -18,  -17,  -16,  -15,  -14,  -13,  -13,
+     -12,  -11,  -11,  -10,   -9,   -9,   -8,   -8,   -7,   -6,   -7,   -6,   -5,   -5,   -5,   -4,
+      -4,   -3,   -3,   -3,   -3,   -3,   -2,   -2,   -2,   -2,   -2,   -2,   -2,   -2,    0,    0,
+       0,    0,   -1,    0,    4,    9,   13,   13,   11,    5,   -1,  -10,  -16,  -17,  -11,   -5,
+      -2,   -1,   -2,   -3,   -2,   -1,    0,    0,    6,   14,   19,   26,   36,   25,   -5,  -30,
+     -36,  -29,  -16,   -4,   -7,  -13,  -12,   -7,   -2,    1,    6,    7,    6,   11,   24,   36,
+      46,   50,   26,  -25,  -60,  -57,  -36,  -12,   -3,   -9,  -14,  -12,   -3,    1,    4,    7,
+      11,   10,   12,   23,   37,   49,   58,   31,  -29,  -67,  -61,  -36,  -10,    0,   -9,  -20,
+     -19,   -8,    3,   11,   11,    7,    8,   15,   24,   32,   41,   53,   45,   -7,  -60,  -68,
+     -41,  -16,   -5,  -11,  -21,  -20,  -10,    4,   10,   10,    8,    9,   13,   19,   29,   42,
+      54,   55,    6,  -57,  -77,  -54,  -21,   -6,   -6,  -15,  -19,  -11,    1,    8,   13,   12,
+       8,    7,   15,   29,   50,   73,   80,    5,  -90, -109,  -65,  -13,    8,    4,  -13,  -18,
+     -10,    3,   11,   10,    4,    2,    8,   18,   36,   60,   82,   86,    1, -105, -121,  -66,
+      -8,   13,    6,  -14,  -22,  -15,   -1,   12,   13,    9,    4,    3,   16,   36,   58,   78,
+      81,   15,  -91, -121,  -74,  -13,   12,    5,  -11,  -21,  -16,   -2,   11,   16,   12,    4,
+       6,   15,   31,   46,   64,   79,   36,  -66, -115,  -81,  -23,    9,    6,  -11,  -21,  -17,
+      -4,    9,   16,   15,    8,    5,    9,   20,   37,   63,   83,   43,  -57, -110,  -83,  -25,
+       8,    7,  -10,  -20,  -17,   -5,    9,   15,   14,    8,    5,    8,   18,   37,   65,   91,
+      50,  -67, -124,  -86,  -19,   16,    9,  -10,  -21,  -18,   -5,    9,   15,   11,    8,    5,
+       8,   22,   40,   65,   92,   48,  -74, -128,  -85,  -16,   19,   11,  -11,  -23,  -19,   -4,
+      10,   15,   11,    8,    7,   11,   19,   35,   57,   88,   60,  -58, -125,  -92,  -24,   15,
+      12,   -8,  -22,  -20,   -7,    9,   17,   17,    8,    4,    8,   18,   31,   51,   81,   67,
+     -38, -116,  -96,  -33,    9,   11,   -6,  -20,  -19,   -8,    7,   16,   16,   11,    5,    5,
+      12,   26,   47,   80,   76,  -26, -111,  -99,  -39,    7,   13,   -3,  -18,  -19,   -9,    5,
+      16,   16,    8,    2,    4,   12,   25,   46,   79,   79,  -23, -112, -101,  -39,    9,   15,
+      -2,  -18,  -20,   -9,    5,   16,   14,    7,    3,    5,   13,   25,   44,   79,   81,  -25,
+    -113, -100,  -37,   10,   16,   -1,  -18,  -20,  -11,    3,   15,   15,    7,    4,    6,   13,
+      23,   40,   72,   86,   -7, -107, -106,  -46,    7,   16,    0,   16,   31,   47,   61,   75,
+      87,   98,  107,  115,  121,  125,  127,  127,  125,  121,  116,  108,   99,   88,   75,   62,
+      47,   32,   16,    0,  -16,  -31,  -47,  -61,  -75,  -87,  -98, -108, -116, -122, -126, -128,
+    -128, -126, -123, -117, -109, -100,  -89,  -77,  -64,  -49,  -34,  -18,   -2,   14,   29,   45,
+      59,   73,   86,   97,  106,  114,  121,  125,  127,  127,  126,  122,  116,  109,  100,   89,
+      77,   63,   49,   34,   18,    2,  -14,  -30,  -45,  -60,  -73,  -86,  -97, -107, -115, -121,
+    -126, -128, -128, -127, -123, -118, -110, -101,  -91,  -79,  -65,  -51,  -36,  -20,   -4,   12,
+      28,   43,   58,   72,   84,   96,  106,  114,  120,  124,  127,  127,  126,  122,  117,  109,
+     100,   90,   78,   64,   50,   35,   19,    3,  -13,  -29,  -44,  -59,  -73,  -85,  -97, -107,
+    -115, -121, -125, -128, -128, -127, -123, -118, -110, -101,  -91,  -79,  -65,  -51,  -36,  -20,
+      -4,   12,   28,   43,   58,   72,   84,   96,  106,  114,  120,  124,  127,  127,  126,  122,
+     117,  109,  100,   90,   78,   64,   50,   35,   19,    3,  -13,  -29,  -44,  -59,  -73,  -85,
+     -97, -107, -115, -121, -125, -128, -128, -127, -123, -118, -110, -101,  -91,  -79,  -65,  -51,
+     -36,  -20,   -4,   12,   28,   43,   58,   72,   84,   96,  106,  114,  120,  124,  127,  127,
+     126,  122,  117,  109,  100,   90,   78,   64,   50,   35,   19,    3,  -13,  -29,  -44,  -59,
+     -73,  -85,  -97, -107, -115, -121, -125, -128, -128, -127, -123, -118, -110, -101,  -91,  -79,
+     -65,  -51,  -36,  -20,   -4,   12,   28,   43,   58,   72,   84,   96,  106,  114,  120,  124,
+     127,  127,  126,  122,  117,  109,  100,   90,   78,   64,   50,   35,   19,    3,  -13,  -29,
+     -44,  -59,  -73,  -85,  -97, -107, -115, -121, -125, -128, -128, -127, -123, -118, -110, -101,
+     -91,  -79,  -65,  -51,  -36,  -20,   -4,   12,    0, -104,  -55,  -11,   24,   33,   30,    7,
+     -15,  -31,  -30,  -25,  -23,  -20,   -9,   10,   31,   59,   91,  111,  115,   92,   51,    7,
+     -33,  -64,  -81,  -81,  -71,  -51,  -22,   16,   52,   74,   82,   81,   68,   38,    0,  -40,
+     -81, -112, -124, -102,  -57,  -11,   24,   33,   30,    7,  -15,  -31,  -30,  -25,  -23,  -20,
+      -9,   10,   31,   59,   91,  111,  115,   92,   51,    7,  -33,  -64,  -81,  -81,  -71,  -51,
+     -22,   16,   52,   74,   82,   81,   68,   38,    0,  -40,  -81, -112, -124, -102,  -57,  -11,
+      24,   33,   30,    7,  -15,  -31,  -30,  -25,  -23,  -20,   -9,   10,   31,   59,   91,  111,
+     115,   92,   51,    7,  -33,  -64,  -81,  -81,  -71,  -51,  -22,   16,   52,   74,   82,   81,
+      68,   38,    0,  -40,  -81, -112, -124, -102,  -57,  -11,   24,   33,   30,    7,  -15,  -31,
+     -30,  -25,  -23,  -20,   -9,   10,   31,   59,   91,  111,  115,   92,   51,    7,  -33,  -64,
+     -81,  -81,  -71,  -51,  -22,   16,   52,   74,   82,   81,   68,   38,    0,  -40,  -81, -112,
+    -124, -102,  -57,  -11,    0
+};
+
+const EAS_U32 testSampleLengths[] =
+{
+    1568, 784, 642, 392, 302, 172
+};
+
+const EAS_U32 testSampleOffsets[] =
+{
+    0x00000000, 0x00000620, 0x00000930, 0x00000bb2, 0x00000d3a, 0x00000e68
+};
+
+/*----------------------------------------------------------------------------
+ * S_EAS
+ *----------------------------------------------------------------------------
+*/
+const S_EAS easTestLib =
+{
+    0x01534145,
+    0x00105622,
+    testBanks,
+    testPrograms,
+    testRegions,
+    testArticulations,
+    testSampleLengths,
+    testSampleOffsets,
+    testSamples,
+    0,
+    0,
+    16,
+    152,
+    97,
+    6,
+    0
+}; /* end S_EAS */
+
+/*----------------------------------------------------------------------------
+ * Statistics
+ *
+ * Number of banks: 0
+ * Number of programs: 16
+ * Number of regions: 152
+ * Number of articulations: 97
+ * Number of samples: 6
+ * Size of sample pool: 3861
+ *----------------------------------------------------------------------------
+*/
+/* end ..\..\EASLib\WTLibrary\eastestv37.c */

common/embeddedaudiosynthesis/arm-wt-22k/jetcreator_lib_src/darwin-x86/easwt_vst_lib.xcodeproj/project.pbxproj

@@ -0,0 +1,645 @@
+// !$*UTF8*$!
+{
+	archiveVersion = 1;
+	classes = {
+	};
+	objectVersion = 44;
+	objects = {
+
+/* Begin PBXBuildFile section */
+		9A56AB8F0F7197AE00D115A7 /* wt_44khz.c in Sources */ = {isa = PBXBuildFile; fileRef = 9A56AB8E0F7197AE00D115A7 /* wt_44khz.c */; };
+		9A56ABD00F71998600D115A7 /* eas_fmengine.h in Headers */ = {isa = PBXBuildFile; fileRef = 9A56ABCF0F71998600D115A7 /* eas_fmengine.h */; };
+		9A56ABDB0F719D4600D115A7 /* eas_xmf.c in Sources */ = {isa = PBXBuildFile; fileRef = 9A56ABD70F719D4600D115A7 /* eas_xmf.c */; };
+		9A56ABDC0F719D4600D115A7 /* eas_xmf.h in Headers */ = {isa = PBXBuildFile; fileRef = 9A56ABD80F719D4600D115A7 /* eas_xmf.h */; };
+		9A56ABDD0F719D4600D115A7 /* eas_xmfdata.c in Sources */ = {isa = PBXBuildFile; fileRef = 9A56ABD90F719D4600D115A7 /* eas_xmfdata.c */; };
+		9A56ABDE0F719D4600D115A7 /* eas_xmfdata.h in Headers */ = {isa = PBXBuildFile; fileRef = 9A56ABDA0F719D4600D115A7 /* eas_xmfdata.h */; };
+		9A56AC050F71BCFE00D115A7 /* eas_build.h in Headers */ = {isa = PBXBuildFile; fileRef = 9A56AC040F71BCFE00D115A7 /* eas_build.h */; };
+		C55B113D0E2D33B4006357C1 /* eas.h in Headers */ = {isa = PBXBuildFile; fileRef = C55B11290E2D33B4006357C1 /* eas.h */; };
+		C55B113E0E2D33B4006357C1 /* eas_audioconst.h in Headers */ = {isa = PBXBuildFile; fileRef = C55B112A0E2D33B4006357C1 /* eas_audioconst.h */; };
+		C55B11400E2D33B4006357C1 /* eas_config.h in Headers */ = {isa = PBXBuildFile; fileRef = C55B112C0E2D33B4006357C1 /* eas_config.h */; };
+		C55B11410E2D33B4006357C1 /* eas_data.h in Headers */ = {isa = PBXBuildFile; fileRef = C55B112D0E2D33B4006357C1 /* eas_data.h */; };
+		C55B11420E2D33B4006357C1 /* eas_host.h in Headers */ = {isa = PBXBuildFile; fileRef = C55B112E0E2D33B4006357C1 /* eas_host.h */; };
+		C55B11430E2D33B4006357C1 /* eas_math.h in Headers */ = {isa = PBXBuildFile; fileRef = C55B112F0E2D33B4006357C1 /* eas_math.h */; };
+		C55B11440E2D33B4006357C1 /* eas_midi.h in Headers */ = {isa = PBXBuildFile; fileRef = C55B11300E2D33B4006357C1 /* eas_midi.h */; };
+		C55B11450E2D33B4006357C1 /* eas_midictrl.h in Headers */ = {isa = PBXBuildFile; fileRef = C55B11310E2D33B4006357C1 /* eas_midictrl.h */; };
+		C55B11460E2D33B4006357C1 /* eas_mixer.c in Sources */ = {isa = PBXBuildFile; fileRef = C55B11320E2D33B4006357C1 /* eas_mixer.c */; };
+		C55B11470E2D33B4006357C1 /* eas_pan.h in Headers */ = {isa = PBXBuildFile; fileRef = C55B11330E2D33B4006357C1 /* eas_pan.h */; };
+		C55B11480E2D33B4006357C1 /* eas_pcm.h in Headers */ = {isa = PBXBuildFile; fileRef = C55B11340E2D33B4006357C1 /* eas_pcm.h */; };
+		C55B11490E2D33B4006357C1 /* eas_pcmdata.h in Headers */ = {isa = PBXBuildFile; fileRef = C55B11350E2D33B4006357C1 /* eas_pcmdata.h */; };
+		C55B114B0E2D33B4006357C1 /* eas_report.h in Headers */ = {isa = PBXBuildFile; fileRef = C55B11370E2D33B4006357C1 /* eas_report.h */; };
+		C55B114C0E2D33B4006357C1 /* eas_sndlib.h in Headers */ = {isa = PBXBuildFile; fileRef = C55B11380E2D33B4006357C1 /* eas_sndlib.h */; };
+		C55B114D0E2D33B4006357C1 /* eas_synth.h in Headers */ = {isa = PBXBuildFile; fileRef = C55B11390E2D33B4006357C1 /* eas_synth.h */; };
+		C55B114E0E2D33B4006357C1 /* eas_synth_protos.h in Headers */ = {isa = PBXBuildFile; fileRef = C55B113A0E2D33B4006357C1 /* eas_synth_protos.h */; };
+		C55B114F0E2D33B4006357C1 /* eas_types.h in Headers */ = {isa = PBXBuildFile; fileRef = C55B113B0E2D33B4006357C1 /* eas_types.h */; };
+		C55B11500E2D33B4006357C1 /* eas_vm_protos.h in Headers */ = {isa = PBXBuildFile; fileRef = C55B113C0E2D33B4006357C1 /* eas_vm_protos.h */; };
+		C55B115B0E2D3796006357C1 /* eas_effects.h in Headers */ = {isa = PBXBuildFile; fileRef = C55B11530E2D3796006357C1 /* eas_effects.h */; };
+		C55B11640E2D37FC006357C1 /* eas_imelodydata.h in Headers */ = {isa = PBXBuildFile; fileRef = C55B11620E2D37FC006357C1 /* eas_imelodydata.h */; };
+		C55B11760E2D38CE006357C1 /* eas_mdls.h in Headers */ = {isa = PBXBuildFile; fileRef = C55B116B0E2D38CE006357C1 /* eas_mdls.h */; };
+		C55B11770E2D38CE006357C1 /* eas_otadata.h in Headers */ = {isa = PBXBuildFile; fileRef = C55B116C0E2D38CE006357C1 /* eas_otadata.h */; };
+		C55B11780E2D38CE006357C1 /* eas_parser.h in Headers */ = {isa = PBXBuildFile; fileRef = C55B116D0E2D38CE006357C1 /* eas_parser.h */; };
+		C55B11790E2D38CE006357C1 /* eas_rtttldata.h in Headers */ = {isa = PBXBuildFile; fileRef = C55B116E0E2D38CE006357C1 /* eas_rtttldata.h */; };
+		C55B117B0E2D38CE006357C1 /* eas_smf.h in Headers */ = {isa = PBXBuildFile; fileRef = C55B11700E2D38CE006357C1 /* eas_smf.h */; };
+		C55B117C0E2D38CE006357C1 /* eas_smfdata.h in Headers */ = {isa = PBXBuildFile; fileRef = C55B11710E2D38CE006357C1 /* eas_smfdata.h */; };
+		C55B117D0E2D38CE006357C1 /* eas_tcdata.h in Headers */ = {isa = PBXBuildFile; fileRef = C55B11720E2D38CE006357C1 /* eas_tcdata.h */; };
+		C55B117E0E2D38CE006357C1 /* eas_wavefile.h in Headers */ = {isa = PBXBuildFile; fileRef = C55B11730E2D38CE006357C1 /* eas_wavefile.h */; };
+		C55B11830E2D38FB006357C1 /* jet.h in Headers */ = {isa = PBXBuildFile; fileRef = C55B11810E2D38FB006357C1 /* jet.h */; };
+		C55B11840E2D38FB006357C1 /* jet_data.h in Headers */ = {isa = PBXBuildFile; fileRef = C55B11820E2D38FB006357C1 /* jet_data.h */; };
+		C55B11860E2D395F006357C1 /* eas_wtengine.h in Headers */ = {isa = PBXBuildFile; fileRef = C55B11850E2D395F006357C1 /* eas_wtengine.h */; };
+		C55B11900E2D39ED006357C1 /* eas_reverb.h in Headers */ = {isa = PBXBuildFile; fileRef = C55B118E0E2D39ED006357C1 /* eas_reverb.h */; };
+		C55B11910E2D39ED006357C1 /* eas_reverbdata.h in Headers */ = {isa = PBXBuildFile; fileRef = C55B118F0E2D39ED006357C1 /* eas_reverbdata.h */; };
+		C55B11940E2D39FE006357C1 /* eas_chorus.h in Headers */ = {isa = PBXBuildFile; fileRef = C55B11920E2D39FE006357C1 /* eas_chorus.h */; };
+		C55B11950E2D39FE006357C1 /* eas_chorusdata.h in Headers */ = {isa = PBXBuildFile; fileRef = C55B11930E2D39FE006357C1 /* eas_chorusdata.h */; };
+		C55B11AE0E2D3B1B006357C1 /* eas_data.c in Sources */ = {isa = PBXBuildFile; fileRef = C55B119F0E2D3B1B006357C1 /* eas_data.c */; };
+		C55B11AF0E2D3B1B006357C1 /* eas_flog.c in Sources */ = {isa = PBXBuildFile; fileRef = C55B11A00E2D3B1B006357C1 /* eas_flog.c */; };
+		C55B11B10E2D3B1B006357C1 /* eas_ima_tables.c in Sources */ = {isa = PBXBuildFile; fileRef = C55B11A20E2D3B1B006357C1 /* eas_ima_tables.c */; };
+		C55B11B20E2D3B1B006357C1 /* eas_imaadpcm.c in Sources */ = {isa = PBXBuildFile; fileRef = C55B11A30E2D3B1B006357C1 /* eas_imaadpcm.c */; };
+		C55B11B30E2D3B1B006357C1 /* eas_math.c in Sources */ = {isa = PBXBuildFile; fileRef = C55B11A40E2D3B1B006357C1 /* eas_math.c */; };
+		C55B11B40E2D3B1B006357C1 /* eas_midi.c in Sources */ = {isa = PBXBuildFile; fileRef = C55B11A50E2D3B1B006357C1 /* eas_midi.c */; };
+		C55B11B50E2D3B1B006357C1 /* eas_mididata.c in Sources */ = {isa = PBXBuildFile; fileRef = C55B11A60E2D3B1B006357C1 /* eas_mididata.c */; };
+		C55B11B60E2D3B1B006357C1 /* eas_mixbuf.c in Sources */ = {isa = PBXBuildFile; fileRef = C55B11A70E2D3B1B006357C1 /* eas_mixbuf.c */; };
+		C55B11B70E2D3B1B006357C1 /* eas_pan.c in Sources */ = {isa = PBXBuildFile; fileRef = C55B11A80E2D3B1B006357C1 /* eas_pan.c */; };
+		C55B11B80E2D3B1B006357C1 /* eas_pcm.c in Sources */ = {isa = PBXBuildFile; fileRef = C55B11A90E2D3B1B006357C1 /* eas_pcm.c */; };
+		C55B11B90E2D3B1B006357C1 /* eas_pcmdata.c in Sources */ = {isa = PBXBuildFile; fileRef = C55B11AA0E2D3B1B006357C1 /* eas_pcmdata.c */; };
+		C55B11BA0E2D3B1B006357C1 /* eas_public.c in Sources */ = {isa = PBXBuildFile; fileRef = C55B11AB0E2D3B1B006357C1 /* eas_public.c */; };
+		C55B11BB0E2D3B1B006357C1 /* eas_voicemgt.c in Sources */ = {isa = PBXBuildFile; fileRef = C55B11AC0E2D3B1B006357C1 /* eas_voicemgt.c */; };
+		C55B11BD0E2D3B55006357C1 /* jet.c in Sources */ = {isa = PBXBuildFile; fileRef = C55B11BC0E2D3B55006357C1 /* jet.c */; };
+		C55B11C10E2D3BFD006357C1 /* eas_chorus.c in Sources */ = {isa = PBXBuildFile; fileRef = C55B11C00E2D3BFD006357C1 /* eas_chorus.c */; };
+		C55B11E40E2D3C62006357C1 /* eas_imelody.c in Sources */ = {isa = PBXBuildFile; fileRef = C55B11D00E2D3C62006357C1 /* eas_imelody.c */; };
+		C55B11E50E2D3C62006357C1 /* eas_imelodydata.c in Sources */ = {isa = PBXBuildFile; fileRef = C55B11D10E2D3C62006357C1 /* eas_imelodydata.c */; };
+		C55B11E60E2D3C62006357C1 /* eas_mdls.c in Sources */ = {isa = PBXBuildFile; fileRef = C55B11D20E2D3C62006357C1 /* eas_mdls.c */; };
+		C55B11E70E2D3C62006357C1 /* eas_ota.c in Sources */ = {isa = PBXBuildFile; fileRef = C55B11D30E2D3C62006357C1 /* eas_ota.c */; };
+		C55B11E80E2D3C62006357C1 /* eas_otadata.c in Sources */ = {isa = PBXBuildFile; fileRef = C55B11D40E2D3C62006357C1 /* eas_otadata.c */; };
+		C55B11E90E2D3C62006357C1 /* eas_rtttl.c in Sources */ = {isa = PBXBuildFile; fileRef = C55B11D50E2D3C62006357C1 /* eas_rtttl.c */; };
+		C55B11EA0E2D3C62006357C1 /* eas_rtttldata.c in Sources */ = {isa = PBXBuildFile; fileRef = C55B11D60E2D3C62006357C1 /* eas_rtttldata.c */; };
+		C55B11EE0E2D3C62006357C1 /* eas_smf.c in Sources */ = {isa = PBXBuildFile; fileRef = C55B11DA0E2D3C62006357C1 /* eas_smf.c */; };
+		C55B11EF0E2D3C62006357C1 /* eas_smfdata.c in Sources */ = {isa = PBXBuildFile; fileRef = C55B11DB0E2D3C62006357C1 /* eas_smfdata.c */; };
+		C55B11F00E2D3C62006357C1 /* eas_tcdata.c in Sources */ = {isa = PBXBuildFile; fileRef = C55B11DC0E2D3C62006357C1 /* eas_tcdata.c */; };
+		C55B11F10E2D3C62006357C1 /* eas_tonecontrol.c in Sources */ = {isa = PBXBuildFile; fileRef = C55B11DD0E2D3C62006357C1 /* eas_tonecontrol.c */; };
+		C55B11F20E2D3C62006357C1 /* eas_wavefile.c in Sources */ = {isa = PBXBuildFile; fileRef = C55B11DE0E2D3C62006357C1 /* eas_wavefile.c */; };
+		C55B11F30E2D3C62006357C1 /* eas_wavefiledata.c in Sources */ = {isa = PBXBuildFile; fileRef = C55B11DF0E2D3C62006357C1 /* eas_wavefiledata.c */; };
+		C55B11F70E2D3C7B006357C1 /* eas_reverb.c in Sources */ = {isa = PBXBuildFile; fileRef = C55B11F60E2D3C7B006357C1 /* eas_reverb.c */; };
+		C55B11FA0E2D3CAC006357C1 /* eastestv37.c in Sources */ = {isa = PBXBuildFile; fileRef = C55B11F80E2D3CAC006357C1 /* eastestv37.c */; };
+		C55B12050E2D3D56006357C1 /* eas_dlssynth.c in Sources */ = {isa = PBXBuildFile; fileRef = C55B12020E2D3D56006357C1 /* eas_dlssynth.c */; };
+		C55B12060E2D3D56006357C1 /* eas_wtengine.c in Sources */ = {isa = PBXBuildFile; fileRef = C55B12030E2D3D56006357C1 /* eas_wtengine.c */; };
+		C55B12070E2D3D56006357C1 /* eas_wtsynth.c in Sources */ = {isa = PBXBuildFile; fileRef = C55B12040E2D3D56006357C1 /* eas_wtsynth.c */; };
+/* End PBXBuildFile section */
+
+/* Begin PBXFileReference section */
+		9A56AB8E0F7197AE00D115A7 /* wt_44khz.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = wt_44khz.c; sourceTree = "<group>"; };
+		9A56ABCF0F71998600D115A7 /* eas_fmengine.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = eas_fmengine.h; path = "../../../arm-fm-22k/lib_src/eas_fmengine.h"; sourceTree = SOURCE_ROOT; };
+		9A56ABD70F719D4600D115A7 /* eas_xmf.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = eas_xmf.c; path = ../../lib_src/eas_xmf.c; sourceTree = SOURCE_ROOT; };
+		9A56ABD80F719D4600D115A7 /* eas_xmf.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = eas_xmf.h; path = ../../lib_src/eas_xmf.h; sourceTree = SOURCE_ROOT; };
+		9A56ABD90F719D4600D115A7 /* eas_xmfdata.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = eas_xmfdata.c; path = ../../lib_src/eas_xmfdata.c; sourceTree = SOURCE_ROOT; };
+		9A56ABDA0F719D4600D115A7 /* eas_xmfdata.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = eas_xmfdata.h; path = ../../lib_src/eas_xmfdata.h; sourceTree = SOURCE_ROOT; };
+		9A56AC040F71BCFE00D115A7 /* eas_build.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = eas_build.h; path = ../../host_src/eas_build.h; sourceTree = SOURCE_ROOT; };
+		C55B11290E2D33B4006357C1 /* eas.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = eas.h; path = ../../host_src/eas.h; sourceTree = SOURCE_ROOT; };
+		C55B112A0E2D33B4006357C1 /* eas_audioconst.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = eas_audioconst.h; path = ../../lib_src/eas_audioconst.h; sourceTree = SOURCE_ROOT; };
+		C55B112C0E2D33B4006357C1 /* eas_config.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = eas_config.h; path = ../../host_src/eas_config.h; sourceTree = SOURCE_ROOT; };
+		C55B112D0E2D33B4006357C1 /* eas_data.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = eas_data.h; path = ../../lib_src/eas_data.h; sourceTree = SOURCE_ROOT; };
+		C55B112E0E2D33B4006357C1 /* eas_host.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = eas_host.h; path = ../../host_src/eas_host.h; sourceTree = SOURCE_ROOT; };
+		C55B112F0E2D33B4006357C1 /* eas_math.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = eas_math.h; path = ../../lib_src/eas_math.h; sourceTree = SOURCE_ROOT; };
+		C55B11300E2D33B4006357C1 /* eas_midi.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = eas_midi.h; path = ../../lib_src/eas_midi.h; sourceTree = SOURCE_ROOT; };
+		C55B11310E2D33B4006357C1 /* eas_midictrl.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = eas_midictrl.h; path = ../../lib_src/eas_midictrl.h; sourceTree = SOURCE_ROOT; };
+		C55B11320E2D33B4006357C1 /* eas_mixer.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = eas_mixer.c; path = ../../lib_src/eas_mixer.c; sourceTree = SOURCE_ROOT; };
+		C55B11330E2D33B4006357C1 /* eas_pan.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = eas_pan.h; path = ../../lib_src/eas_pan.h; sourceTree = SOURCE_ROOT; };
+		C55B11340E2D33B4006357C1 /* eas_pcm.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = eas_pcm.h; path = ../../lib_src/eas_pcm.h; sourceTree = SOURCE_ROOT; };
+		C55B11350E2D33B4006357C1 /* eas_pcmdata.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = eas_pcmdata.h; path = ../../lib_src/eas_pcmdata.h; sourceTree = SOURCE_ROOT; };
+		C55B11370E2D33B4006357C1 /* eas_report.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = eas_report.h; path = ../../host_src/eas_report.h; sourceTree = SOURCE_ROOT; };
+		C55B11380E2D33B4006357C1 /* eas_sndlib.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = eas_sndlib.h; path = ../../lib_src/eas_sndlib.h; sourceTree = SOURCE_ROOT; };
+		C55B11390E2D33B4006357C1 /* eas_synth.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = eas_synth.h; path = ../../lib_src/eas_synth.h; sourceTree = SOURCE_ROOT; };
+		C55B113A0E2D33B4006357C1 /* eas_synth_protos.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = eas_synth_protos.h; path = ../../lib_src/eas_synth_protos.h; sourceTree = SOURCE_ROOT; };
+		C55B113B0E2D33B4006357C1 /* eas_types.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = eas_types.h; path = ../../host_src/eas_types.h; sourceTree = SOURCE_ROOT; };
+		C55B113C0E2D33B4006357C1 /* eas_vm_protos.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = eas_vm_protos.h; path = ../../lib_src/eas_vm_protos.h; sourceTree = SOURCE_ROOT; };
+		C55B11530E2D3796006357C1 /* eas_effects.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = eas_effects.h; path = ../../lib_src/eas_effects.h; sourceTree = SOURCE_ROOT; };
+		C55B11620E2D37FC006357C1 /* eas_imelodydata.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = eas_imelodydata.h; path = ../../lib_src/eas_imelodydata.h; sourceTree = SOURCE_ROOT; };
+		C55B116B0E2D38CE006357C1 /* eas_mdls.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = eas_mdls.h; path = ../../lib_src/eas_mdls.h; sourceTree = SOURCE_ROOT; };
+		C55B116C0E2D38CE006357C1 /* eas_otadata.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = eas_otadata.h; path = ../../lib_src/eas_otadata.h; sourceTree = SOURCE_ROOT; };
+		C55B116D0E2D38CE006357C1 /* eas_parser.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = eas_parser.h; path = ../../lib_src/eas_parser.h; sourceTree = SOURCE_ROOT; };
+		C55B116E0E2D38CE006357C1 /* eas_rtttldata.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = eas_rtttldata.h; path = ../../lib_src/eas_rtttldata.h; sourceTree = SOURCE_ROOT; };
+		C55B11700E2D38CE006357C1 /* eas_smf.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = eas_smf.h; path = ../../lib_src/eas_smf.h; sourceTree = SOURCE_ROOT; };
+		C55B11710E2D38CE006357C1 /* eas_smfdata.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = eas_smfdata.h; path = ../../lib_src/eas_smfdata.h; sourceTree = SOURCE_ROOT; };
+		C55B11720E2D38CE006357C1 /* eas_tcdata.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = eas_tcdata.h; path = ../../lib_src/eas_tcdata.h; sourceTree = SOURCE_ROOT; };
+		C55B11730E2D38CE006357C1 /* eas_wavefile.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = eas_wavefile.h; path = ../../lib_src/eas_wavefile.h; sourceTree = SOURCE_ROOT; };
+		C55B11810E2D38FB006357C1 /* jet.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = jet.h; path = ../../host_src/jet.h; sourceTree = SOURCE_ROOT; };
+		C55B11820E2D38FB006357C1 /* jet_data.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = jet_data.h; path = ../../lib_src/jet_data.h; sourceTree = SOURCE_ROOT; };
+		C55B11850E2D395F006357C1 /* eas_wtengine.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = eas_wtengine.h; path = ../../lib_src/eas_wtengine.h; sourceTree = SOURCE_ROOT; };
+		C55B118E0E2D39ED006357C1 /* eas_reverb.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = eas_reverb.h; path = ../../host_src/eas_reverb.h; sourceTree = SOURCE_ROOT; };
+		C55B118F0E2D39ED006357C1 /* eas_reverbdata.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = eas_reverbdata.h; path = ../../lib_src/eas_reverbdata.h; sourceTree = SOURCE_ROOT; };
+		C55B11920E2D39FE006357C1 /* eas_chorus.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = eas_chorus.h; path = ../../host_src/eas_chorus.h; sourceTree = SOURCE_ROOT; };
+		C55B11930E2D39FE006357C1 /* eas_chorusdata.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = eas_chorusdata.h; path = ../../lib_src/eas_chorusdata.h; sourceTree = SOURCE_ROOT; };
+		C55B119F0E2D3B1B006357C1 /* eas_data.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = eas_data.c; path = ../../lib_src/eas_data.c; sourceTree = SOURCE_ROOT; };
+		C55B11A00E2D3B1B006357C1 /* eas_flog.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = eas_flog.c; path = ../../lib_src/eas_flog.c; sourceTree = SOURCE_ROOT; };
+		C55B11A20E2D3B1B006357C1 /* eas_ima_tables.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = eas_ima_tables.c; path = ../../lib_src/eas_ima_tables.c; sourceTree = SOURCE_ROOT; };
+		C55B11A30E2D3B1B006357C1 /* eas_imaadpcm.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = eas_imaadpcm.c; path = ../../lib_src/eas_imaadpcm.c; sourceTree = SOURCE_ROOT; };
+		C55B11A40E2D3B1B006357C1 /* eas_math.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = eas_math.c; path = ../../lib_src/eas_math.c; sourceTree = SOURCE_ROOT; };
+		C55B11A50E2D3B1B006357C1 /* eas_midi.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = eas_midi.c; path = ../../lib_src/eas_midi.c; sourceTree = SOURCE_ROOT; };
+		C55B11A60E2D3B1B006357C1 /* eas_mididata.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = eas_mididata.c; path = ../../lib_src/eas_mididata.c; sourceTree = SOURCE_ROOT; };
+		C55B11A70E2D3B1B006357C1 /* eas_mixbuf.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = eas_mixbuf.c; path = ../../lib_src/eas_mixbuf.c; sourceTree = SOURCE_ROOT; };
+		C55B11A80E2D3B1B006357C1 /* eas_pan.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = eas_pan.c; path = ../../lib_src/eas_pan.c; sourceTree = SOURCE_ROOT; };
+		C55B11A90E2D3B1B006357C1 /* eas_pcm.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = eas_pcm.c; path = ../../lib_src/eas_pcm.c; sourceTree = SOURCE_ROOT; };
+		C55B11AA0E2D3B1B006357C1 /* eas_pcmdata.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = eas_pcmdata.c; path = ../../lib_src/eas_pcmdata.c; sourceTree = SOURCE_ROOT; };
+		C55B11AB0E2D3B1B006357C1 /* eas_public.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = eas_public.c; path = ../../lib_src/eas_public.c; sourceTree = "<group>"; };
+		C55B11AC0E2D3B1B006357C1 /* eas_voicemgt.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = eas_voicemgt.c; path = ../../lib_src/eas_voicemgt.c; sourceTree = SOURCE_ROOT; };
+		C55B11BC0E2D3B55006357C1 /* jet.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = jet.c; path = ../../lib_src/jet.c; sourceTree = SOURCE_ROOT; };
+		C55B11C00E2D3BFD006357C1 /* eas_chorus.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = eas_chorus.c; path = ../../lib_src/eas_chorus.c; sourceTree = SOURCE_ROOT; };
+		C55B11D00E2D3C62006357C1 /* eas_imelody.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = eas_imelody.c; path = ../../lib_src/eas_imelody.c; sourceTree = SOURCE_ROOT; };
+		C55B11D10E2D3C62006357C1 /* eas_imelodydata.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = eas_imelodydata.c; path = ../../lib_src/eas_imelodydata.c; sourceTree = SOURCE_ROOT; };
+		C55B11D20E2D3C62006357C1 /* eas_mdls.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = eas_mdls.c; path = ../../lib_src/eas_mdls.c; sourceTree = SOURCE_ROOT; };
+		C55B11D30E2D3C62006357C1 /* eas_ota.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = eas_ota.c; path = ../../lib_src/eas_ota.c; sourceTree = SOURCE_ROOT; };
+		C55B11D40E2D3C62006357C1 /* eas_otadata.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = eas_otadata.c; path = ../../lib_src/eas_otadata.c; sourceTree = SOURCE_ROOT; };
+		C55B11D50E2D3C62006357C1 /* eas_rtttl.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = eas_rtttl.c; path = ../../lib_src/eas_rtttl.c; sourceTree = SOURCE_ROOT; };
+		C55B11D60E2D3C62006357C1 /* eas_rtttldata.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = eas_rtttldata.c; path = ../../lib_src/eas_rtttldata.c; sourceTree = SOURCE_ROOT; };
+		C55B11DA0E2D3C62006357C1 /* eas_smf.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = eas_smf.c; path = ../../lib_src/eas_smf.c; sourceTree = SOURCE_ROOT; };
+		C55B11DB0E2D3C62006357C1 /* eas_smfdata.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = eas_smfdata.c; path = ../../lib_src/eas_smfdata.c; sourceTree = SOURCE_ROOT; };
+		C55B11DC0E2D3C62006357C1 /* eas_tcdata.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = eas_tcdata.c; path = ../../lib_src/eas_tcdata.c; sourceTree = SOURCE_ROOT; };
+		C55B11DD0E2D3C62006357C1 /* eas_tonecontrol.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = eas_tonecontrol.c; path = ../../lib_src/eas_tonecontrol.c; sourceTree = SOURCE_ROOT; };
+		C55B11DE0E2D3C62006357C1 /* eas_wavefile.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = eas_wavefile.c; path = ../../lib_src/eas_wavefile.c; sourceTree = SOURCE_ROOT; };
+		C55B11DF0E2D3C62006357C1 /* eas_wavefiledata.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = eas_wavefiledata.c; path = ../../lib_src/eas_wavefiledata.c; sourceTree = SOURCE_ROOT; };
+		C55B11F60E2D3C7B006357C1 /* eas_reverb.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = eas_reverb.c; path = ../../lib_src/eas_reverb.c; sourceTree = SOURCE_ROOT; };
+		C55B11F80E2D3CAC006357C1 /* eastestv37.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = eastestv37.c; sourceTree = "<group>"; };
+		C55B12020E2D3D56006357C1 /* eas_dlssynth.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = eas_dlssynth.c; path = ../../lib_src/eas_dlssynth.c; sourceTree = SOURCE_ROOT; };
+		C55B12030E2D3D56006357C1 /* eas_wtengine.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = eas_wtengine.c; path = ../../lib_src/eas_wtengine.c; sourceTree = SOURCE_ROOT; };
+		C55B12040E2D3D56006357C1 /* eas_wtsynth.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = eas_wtsynth.c; path = ../../lib_src/eas_wtsynth.c; sourceTree = SOURCE_ROOT; };
+		D2AAC046055464E500DB518D /* libeaswt_vst_lib.a */ = {isa = PBXFileReference; explicitFileType = archive.ar; includeInIndex = 0; path = libeaswt_vst_lib.a; sourceTree = BUILT_PRODUCTS_DIR; };
+/* End PBXFileReference section */
+
+/* Begin PBXFrameworksBuildPhase section */
+		D289987405E68DCB004EDB86 /* Frameworks */ = {
+			isa = PBXFrameworksBuildPhase;
+			buildActionMask = 2147483647;
+			files = (
+			);
+			runOnlyForDeploymentPostprocessing = 0;
+		};
+/* End PBXFrameworksBuildPhase section */
+
+/* Begin PBXGroup section */
+		08FB7794FE84155DC02AAC07 /* easwt_vst_lib */ = {
+			isa = PBXGroup;
+			children = (
+				C55B11280E2D31B6006357C1 /* Header Files */,
+				08FB7795FE84155DC02AAC07 /* Source */,
+				C6A0FF2B0290797F04C91782 /* Documentation */,
+				1AB674ADFE9D54B511CA2CBB /* Products */,
+			);
+			name = easwt_vst_lib;
+			sourceTree = "<group>";
+		};
+		08FB7795FE84155DC02AAC07 /* Source */ = {
+			isa = PBXGroup;
+			children = (
+				C55B119F0E2D3B1B006357C1 /* eas_data.c */,
+				C55B11A00E2D3B1B006357C1 /* eas_flog.c */,
+				C55B11A20E2D3B1B006357C1 /* eas_ima_tables.c */,
+				C55B11A30E2D3B1B006357C1 /* eas_imaadpcm.c */,
+				C55B11A40E2D3B1B006357C1 /* eas_math.c */,
+				C55B11A50E2D3B1B006357C1 /* eas_midi.c */,
+				C55B11A60E2D3B1B006357C1 /* eas_mididata.c */,
+				C55B11A70E2D3B1B006357C1 /* eas_mixbuf.c */,
+				C55B11320E2D33B4006357C1 /* eas_mixer.c */,
+				C55B11A80E2D3B1B006357C1 /* eas_pan.c */,
+				C55B11A90E2D3B1B006357C1 /* eas_pcm.c */,
+				C55B11AA0E2D3B1B006357C1 /* eas_pcmdata.c */,
+				C55B11AB0E2D3B1B006357C1 /* eas_public.c */,
+				C55B11AC0E2D3B1B006357C1 /* eas_voicemgt.c */,
+				C55B11BC0E2D3B55006357C1 /* jet.c */,
+				C55B11960E2D3A56006357C1 /* Chorus */,
+				C55B119A0E2D3A7F006357C1 /* Parsers */,
+				C55B11970E2D3A5C006357C1 /* Reverb */,
+				C55B119B0E2D3A99006357C1 /* SoundLibs */,
+				C55B119D0E2D3AB6006357C1 /* Wavetable */,
+			);
+			name = Source;
+			sourceTree = "<group>";
+		};
+		1AB674ADFE9D54B511CA2CBB /* Products */ = {
+			isa = PBXGroup;
+			children = (
+				D2AAC046055464E500DB518D /* libeaswt_vst_lib.a */,
+			);
+			name = Products;
+			sourceTree = "<group>";
+		};
+		C55B11280E2D31B6006357C1 /* Header Files */ = {
+			isa = PBXGroup;
+			children = (
+				9A56AC040F71BCFE00D115A7 /* eas_build.h */,
+				9A56ABCF0F71998600D115A7 /* eas_fmengine.h */,
+				C55B11880E2D39B0006357C1 /* Reverb */,
+				C55B11870E2D39A9006357C1 /* Chorus */,
+				C55B11850E2D395F006357C1 /* eas_wtengine.h */,
+				C55B11810E2D38FB006357C1 /* jet.h */,
+				C55B11820E2D38FB006357C1 /* jet_data.h */,
+				C55B116B0E2D38CE006357C1 /* eas_mdls.h */,
+				C55B116C0E2D38CE006357C1 /* eas_otadata.h */,
+				C55B116D0E2D38CE006357C1 /* eas_parser.h */,
+				C55B116E0E2D38CE006357C1 /* eas_rtttldata.h */,
+				C55B11700E2D38CE006357C1 /* eas_smf.h */,
+				C55B11710E2D38CE006357C1 /* eas_smfdata.h */,
+				C55B11720E2D38CE006357C1 /* eas_tcdata.h */,
+				C55B11730E2D38CE006357C1 /* eas_wavefile.h */,
+				C55B11620E2D37FC006357C1 /* eas_imelodydata.h */,
+				C55B11530E2D3796006357C1 /* eas_effects.h */,
+				C55B11290E2D33B4006357C1 /* eas.h */,
+				C55B112A0E2D33B4006357C1 /* eas_audioconst.h */,
+				C55B112C0E2D33B4006357C1 /* eas_config.h */,
+				C55B112D0E2D33B4006357C1 /* eas_data.h */,
+				C55B112E0E2D33B4006357C1 /* eas_host.h */,
+				C55B112F0E2D33B4006357C1 /* eas_math.h */,
+				C55B11300E2D33B4006357C1 /* eas_midi.h */,
+				C55B11310E2D33B4006357C1 /* eas_midictrl.h */,
+				C55B11330E2D33B4006357C1 /* eas_pan.h */,
+				C55B11340E2D33B4006357C1 /* eas_pcm.h */,
+				C55B11350E2D33B4006357C1 /* eas_pcmdata.h */,
+				C55B11370E2D33B4006357C1 /* eas_report.h */,
+				C55B11380E2D33B4006357C1 /* eas_sndlib.h */,
+				C55B11390E2D33B4006357C1 /* eas_synth.h */,
+				C55B113A0E2D33B4006357C1 /* eas_synth_protos.h */,
+				C55B113B0E2D33B4006357C1 /* eas_types.h */,
+				C55B113C0E2D33B4006357C1 /* eas_vm_protos.h */,
+			);
+			name = "Header Files";
+			sourceTree = "<group>";
+		};
+		C55B11870E2D39A9006357C1 /* Chorus */ = {
+			isa = PBXGroup;
+			children = (
+				C55B11920E2D39FE006357C1 /* eas_chorus.h */,
+				C55B11930E2D39FE006357C1 /* eas_chorusdata.h */,
+			);
+			name = Chorus;
+			sourceTree = "<group>";
+		};
+		C55B11880E2D39B0006357C1 /* Reverb */ = {
+			isa = PBXGroup;
+			children = (
+				C55B118E0E2D39ED006357C1 /* eas_reverb.h */,
+				C55B118F0E2D39ED006357C1 /* eas_reverbdata.h */,
+			);
+			name = Reverb;
+			sourceTree = "<group>";
+		};
+		C55B11960E2D3A56006357C1 /* Chorus */ = {
+			isa = PBXGroup;
+			children = (
+				C55B11C00E2D3BFD006357C1 /* eas_chorus.c */,
+			);
+			name = Chorus;
+			sourceTree = "<group>";
+		};
+		C55B11970E2D3A5C006357C1 /* Reverb */ = {
+			isa = PBXGroup;
+			children = (
+				C55B11F60E2D3C7B006357C1 /* eas_reverb.c */,
+			);
+			name = Reverb;
+			sourceTree = "<group>";
+		};
+		C55B119A0E2D3A7F006357C1 /* Parsers */ = {
+			isa = PBXGroup;
+			children = (
+				9A56ABD70F719D4600D115A7 /* eas_xmf.c */,
+				9A56ABD80F719D4600D115A7 /* eas_xmf.h */,
+				9A56ABD90F719D4600D115A7 /* eas_xmfdata.c */,
+				9A56ABDA0F719D4600D115A7 /* eas_xmfdata.h */,
+				C55B11D00E2D3C62006357C1 /* eas_imelody.c */,
+				C55B11D10E2D3C62006357C1 /* eas_imelodydata.c */,
+				C55B11D20E2D3C62006357C1 /* eas_mdls.c */,
+				C55B11D30E2D3C62006357C1 /* eas_ota.c */,
+				C55B11D40E2D3C62006357C1 /* eas_otadata.c */,
+				C55B11D50E2D3C62006357C1 /* eas_rtttl.c */,
+				C55B11D60E2D3C62006357C1 /* eas_rtttldata.c */,
+				C55B11DA0E2D3C62006357C1 /* eas_smf.c */,
+				C55B11DB0E2D3C62006357C1 /* eas_smfdata.c */,
+				C55B11DC0E2D3C62006357C1 /* eas_tcdata.c */,
+				C55B11DD0E2D3C62006357C1 /* eas_tonecontrol.c */,
+				C55B11DE0E2D3C62006357C1 /* eas_wavefile.c */,
+				C55B11DF0E2D3C62006357C1 /* eas_wavefiledata.c */,
+			);
+			name = Parsers;
+			sourceTree = "<group>";
+		};
+		C55B119B0E2D3A99006357C1 /* SoundLibs */ = {
+			isa = PBXGroup;
+			children = (
+				9A56AB8E0F7197AE00D115A7 /* wt_44khz.c */,
+				C55B11F80E2D3CAC006357C1 /* eastestv37.c */,
+			);
+			name = SoundLibs;
+			sourceTree = "<group>";
+		};
+		C55B119D0E2D3AB6006357C1 /* Wavetable */ = {
+			isa = PBXGroup;
+			children = (
+				C55B12020E2D3D56006357C1 /* eas_dlssynth.c */,
+				C55B12030E2D3D56006357C1 /* eas_wtengine.c */,
+				C55B12040E2D3D56006357C1 /* eas_wtsynth.c */,
+			);
+			name = Wavetable;
+			sourceTree = "<group>";
+		};
+		C6A0FF2B0290797F04C91782 /* Documentation */ = {
+			isa = PBXGroup;
+			children = (
+			);
+			name = Documentation;
+			sourceTree = "<group>";
+		};
+/* End PBXGroup section */
+
+/* Begin PBXHeadersBuildPhase section */
+		D2AAC043055464E500DB518D /* Headers */ = {
+			isa = PBXHeadersBuildPhase;
+			buildActionMask = 2147483647;
+			files = (
+				C55B113D0E2D33B4006357C1 /* eas.h in Headers */,
+				C55B113E0E2D33B4006357C1 /* eas_audioconst.h in Headers */,
+				C55B11400E2D33B4006357C1 /* eas_config.h in Headers */,
+				C55B11410E2D33B4006357C1 /* eas_data.h in Headers */,
+				C55B11420E2D33B4006357C1 /* eas_host.h in Headers */,
+				C55B11430E2D33B4006357C1 /* eas_math.h in Headers */,
+				C55B11440E2D33B4006357C1 /* eas_midi.h in Headers */,
+				C55B11450E2D33B4006357C1 /* eas_midictrl.h in Headers */,
+				C55B11470E2D33B4006357C1 /* eas_pan.h in Headers */,
+				C55B11480E2D33B4006357C1 /* eas_pcm.h in Headers */,
+				C55B11490E2D33B4006357C1 /* eas_pcmdata.h in Headers */,
+				C55B114B0E2D33B4006357C1 /* eas_report.h in Headers */,
+				C55B114C0E2D33B4006357C1 /* eas_sndlib.h in Headers */,
+				C55B114D0E2D33B4006357C1 /* eas_synth.h in Headers */,
+				C55B114E0E2D33B4006357C1 /* eas_synth_protos.h in Headers */,
+				C55B114F0E2D33B4006357C1 /* eas_types.h in Headers */,
+				C55B11500E2D33B4006357C1 /* eas_vm_protos.h in Headers */,
+				C55B115B0E2D3796006357C1 /* eas_effects.h in Headers */,
+				C55B11640E2D37FC006357C1 /* eas_imelodydata.h in Headers */,
+				C55B11760E2D38CE006357C1 /* eas_mdls.h in Headers */,
+				C55B11770E2D38CE006357C1 /* eas_otadata.h in Headers */,
+				C55B11780E2D38CE006357C1 /* eas_parser.h in Headers */,
+				C55B11790E2D38CE006357C1 /* eas_rtttldata.h in Headers */,
+				C55B117B0E2D38CE006357C1 /* eas_smf.h in Headers */,
+				C55B117C0E2D38CE006357C1 /* eas_smfdata.h in Headers */,
+				C55B117D0E2D38CE006357C1 /* eas_tcdata.h in Headers */,
+				C55B117E0E2D38CE006357C1 /* eas_wavefile.h in Headers */,
+				C55B11830E2D38FB006357C1 /* jet.h in Headers */,
+				C55B11840E2D38FB006357C1 /* jet_data.h in Headers */,
+				C55B11860E2D395F006357C1 /* eas_wtengine.h in Headers */,
+				C55B11900E2D39ED006357C1 /* eas_reverb.h in Headers */,
+				C55B11910E2D39ED006357C1 /* eas_reverbdata.h in Headers */,
+				C55B11940E2D39FE006357C1 /* eas_chorus.h in Headers */,
+				C55B11950E2D39FE006357C1 /* eas_chorusdata.h in Headers */,
+				9A56ABD00F71998600D115A7 /* eas_fmengine.h in Headers */,
+				9A56ABDC0F719D4600D115A7 /* eas_xmf.h in Headers */,
+				9A56ABDE0F719D4600D115A7 /* eas_xmfdata.h in Headers */,
+				9A56AC050F71BCFE00D115A7 /* eas_build.h in Headers */,
+			);
+			runOnlyForDeploymentPostprocessing = 0;
+		};
+/* End PBXHeadersBuildPhase section */
+
+/* Begin PBXNativeTarget section */
+		D2AAC045055464E500DB518D /* easwt_vst_lib */ = {
+			isa = PBXNativeTarget;
+			buildConfigurationList = 1DEB91EB08733DB70010E9CD /* Build configuration list for PBXNativeTarget "easwt_vst_lib" */;
+			buildPhases = (
+				D2AAC043055464E500DB518D /* Headers */,
+				D2AAC044055464E500DB518D /* Sources */,
+				D289987405E68DCB004EDB86 /* Frameworks */,
+			);
+			buildRules = (
+			);
+			dependencies = (
+			);
+			name = easwt_vst_lib;
+			productName = easwt_vst_lib;
+			productReference = D2AAC046055464E500DB518D /* libeaswt_vst_lib.a */;
+			productType = "com.apple.product-type.library.static";
+		};
+/* End PBXNativeTarget section */
+
+/* Begin PBXProject section */
+		08FB7793FE84155DC02AAC07 /* Project object */ = {
+			isa = PBXProject;
+			buildConfigurationList = 1DEB91EF08733DB70010E9CD /* Build configuration list for PBXProject "easwt_vst_lib" */;
+			compatibilityVersion = "Xcode 3.0";
+			hasScannedForEncodings = 1;
+			mainGroup = 08FB7794FE84155DC02AAC07 /* easwt_vst_lib */;
+			projectDirPath = "";
+			projectRoot = "";
+			targets = (
+				D2AAC045055464E500DB518D /* easwt_vst_lib */,
+			);
+		};
+/* End PBXProject section */
+
+/* Begin PBXSourcesBuildPhase section */
+		D2AAC044055464E500DB518D /* Sources */ = {
+			isa = PBXSourcesBuildPhase;
+			buildActionMask = 2147483647;
+			files = (
+				C55B11460E2D33B4006357C1 /* eas_mixer.c in Sources */,
+				C55B11AE0E2D3B1B006357C1 /* eas_data.c in Sources */,
+				C55B11AF0E2D3B1B006357C1 /* eas_flog.c in Sources */,
+				C55B11B10E2D3B1B006357C1 /* eas_ima_tables.c in Sources */,
+				C55B11B20E2D3B1B006357C1 /* eas_imaadpcm.c in Sources */,
+				C55B11B30E2D3B1B006357C1 /* eas_math.c in Sources */,
+				C55B11B40E2D3B1B006357C1 /* eas_midi.c in Sources */,
+				C55B11B50E2D3B1B006357C1 /* eas_mididata.c in Sources */,
+				C55B11B60E2D3B1B006357C1 /* eas_mixbuf.c in Sources */,
+				C55B11B70E2D3B1B006357C1 /* eas_pan.c in Sources */,
+				C55B11B80E2D3B1B006357C1 /* eas_pcm.c in Sources */,
+				C55B11B90E2D3B1B006357C1 /* eas_pcmdata.c in Sources */,
+				C55B11BA0E2D3B1B006357C1 /* eas_public.c in Sources */,
+				C55B11BB0E2D3B1B006357C1 /* eas_voicemgt.c in Sources */,
+				C55B11BD0E2D3B55006357C1 /* jet.c in Sources */,
+				C55B11C10E2D3BFD006357C1 /* eas_chorus.c in Sources */,
+				C55B11E40E2D3C62006357C1 /* eas_imelody.c in Sources */,
+				C55B11E50E2D3C62006357C1 /* eas_imelodydata.c in Sources */,
+				C55B11E60E2D3C62006357C1 /* eas_mdls.c in Sources */,
+				C55B11E70E2D3C62006357C1 /* eas_ota.c in Sources */,
+				C55B11E80E2D3C62006357C1 /* eas_otadata.c in Sources */,
+				C55B11E90E2D3C62006357C1 /* eas_rtttl.c in Sources */,
+				C55B11EA0E2D3C62006357C1 /* eas_rtttldata.c in Sources */,
+				C55B11EE0E2D3C62006357C1 /* eas_smf.c in Sources */,
+				C55B11EF0E2D3C62006357C1 /* eas_smfdata.c in Sources */,
+				C55B11F00E2D3C62006357C1 /* eas_tcdata.c in Sources */,
+				C55B11F10E2D3C62006357C1 /* eas_tonecontrol.c in Sources */,
+				C55B11F20E2D3C62006357C1 /* eas_wavefile.c in Sources */,
+				C55B11F30E2D3C62006357C1 /* eas_wavefiledata.c in Sources */,
+				C55B11F70E2D3C7B006357C1 /* eas_reverb.c in Sources */,
+				C55B11FA0E2D3CAC006357C1 /* eastestv37.c in Sources */,
+				C55B12050E2D3D56006357C1 /* eas_dlssynth.c in Sources */,
+				C55B12060E2D3D56006357C1 /* eas_wtengine.c in Sources */,
+				C55B12070E2D3D56006357C1 /* eas_wtsynth.c in Sources */,
+				9A56AB8F0F7197AE00D115A7 /* wt_44khz.c in Sources */,
+				9A56ABDB0F719D4600D115A7 /* eas_xmf.c in Sources */,
+				9A56ABDD0F719D4600D115A7 /* eas_xmfdata.c in Sources */,
+			);
+			runOnlyForDeploymentPostprocessing = 0;
+		};
+/* End PBXSourcesBuildPhase section */
+
+/* Begin XCBuildConfiguration section */
+		1DEB91EC08733DB70010E9CD /* Debug */ = {
+			isa = XCBuildConfiguration;
+			buildSettings = {
+				COPY_PHASE_STRIP = NO;
+				GCC_DYNAMIC_NO_PIC = NO;
+				GCC_ENABLE_FIX_AND_CONTINUE = YES;
+				GCC_MODEL_TUNING = G5;
+				GCC_OPTIMIZATION_LEVEL = 0;
+				GCC_PREPROCESSOR_DEFINITIONS = (
+					"DEBUG_JET=1",
+					"_DEBUG_SMAF=1",
+					"_DEBUG_CMF=1",
+					"JET_INTERFACE=1",
+					"MAX_SMF_STREAMS=32",
+					"NUM_OUTPUT_CHANNELS=2",
+					"_SAMPLE_RATE_44100=1",
+					"_8_BIT_SAMPLES=1",
+					"_FILTER_ENABLED=1",
+					"_NO_DEBUG_PREPROCESSOR=1",
+					"_CORE_CYCLES_PER_MEMORY_CYCLE=1",
+					"_IMA_DECODER=1",
+					"_XMF_PARSER=1",
+					"_DLS_PARSER=1",
+					"MAX_DLS_MEMORY=16777216",
+					"EAS_WT_SYNTH=1",
+					"MAX_SYNTH_VOICES=64",
+					"X_ENHANCER_ENABLED=1",
+					"_REVERB_ENABLED=1",
+					"_CHORUS_ENABLED=1",
+					"DLS_SYNTHESIZER=1",
+					"EXTERNAL_AUDIO=1",
+					"TEST_HARNESS=1",
+					"MMAPI_SUPPORT=1",
+					"X_MAXIMIZER_ENABLED=1",
+					"X_COMPRESSOR_ENABLED=1",
+					"X_STATS=1",
+					"X_DEBUG_DLS=1",
+					"X_OPTIMIZED_MONO=1",
+					"X_CHECKED_BUILD=1",
+					"_NO_ALIGN=1",
+					"_C_REFERENCE=1",
+					"XMAXIMIZER_USE_FLOATING_POINT=1",
+				);
+				HEADER_SEARCH_PATHS = (
+					"${TARGET_BUILD_DIR}/**",
+					"../../Common/**",
+					"../../Parsers/**",
+					"../../Effects/**",
+					"../../WTSynth/**",
+					"../../FMSynth/**",
+					"../../SplitArch/**",
+					"../../Jet/**",
+				);
+				INSTALL_PATH = /usr/local/lib;
+				MACOSX_DEPLOYMENT_TARGET = 10.4;
+				PRODUCT_NAME = easwt_vst_lib;
+				ZERO_LINK = YES;
+			};
+			name = Debug;
+		};
+		1DEB91ED08733DB70010E9CD /* Release */ = {
+			isa = XCBuildConfiguration;
+			buildSettings = {
+				ARCHS = i386;
+				DEBUG_INFORMATION_FORMAT = "dwarf-with-dsym";
+				GCC_AUTO_VECTORIZATION = YES;
+				GCC_GENERATE_DEBUGGING_SYMBOLS = NO;
+				GCC_MODEL_TUNING = G5;
+				GCC_OPTIMIZATION_LEVEL = 3;
+				GCC_PREPROCESSOR_DEFINITIONS = (
+					"JET_INTERFACE=1",
+					"MAX_SMF_STREAMS=32",
+					"NUM_OUTPUT_CHANNELS=2",
+					"_SAMPLE_RATE_44100=1",
+					"_8_BIT_SAMPLES=1",
+					"_FILTER_ENABLED=1",
+					"_NO_DEBUG_PREPROCESSOR=1",
+					"_CORE_CYCLES_PER_MEMORY_CYCLE=1",
+					"_IMA_DECODER=1",
+					"_XMF_PARSER=1",
+					"_DLS_PARSER=1",
+					"MAX_DLS_MEMORY=16777216",
+					"EAS_WT_SYNTH=1",
+					"MAX_SYNTH_VOICES=64",
+					"X_ENHANCER_ENABLED=1",
+					"_REVERB_ENABLED=1",
+					"_CHORUS_ENABLED=1",
+					"DLS_SYNTHESIZER=1",
+					"EXTERNAL_AUDIO=1",
+					"TEST_HARNESS=1",
+					"MMAPI_SUPPORT=1",
+					"X_MAXIMIZER_ENABLED=1",
+					"X_COMPRESSOR_ENABLED=1",
+					"X_STATS=1",
+					"X_DEBUG_DLS=1",
+					"X_OPTIMIZED_MONO=1",
+					"X_CHECKED_BUILD=1",
+					"_NO_ALIGN=1",
+					"_C_REFERENCE=1",
+					"XMAXIMIZER_USE_FLOATING_POINT=1",
+				);
+				GCC_UNROLL_LOOPS = YES;
+				HEADER_SEARCH_PATHS = (
+					"${TARGET_BUILD_DIR}/**",
+					"../../Common/**",
+					"../../Parsers/**",
+					"../../Effects/**",
+					"../../WTSynth/**",
+					"../../FMSynth/**",
+					"../../SplitArch/**",
+					"../../Jet/**",
+				);
+				INSTALL_PATH = /usr/local/lib;
+				MACOSX_DEPLOYMENT_TARGET = 10.4;
+				PRODUCT_NAME = easwt_vst_lib;
+			};
+			name = Release;
+		};
+		1DEB91F008733DB70010E9CD /* Debug */ = {
+			isa = XCBuildConfiguration;
+			buildSettings = {
+				GCC_WARN_ABOUT_RETURN_TYPE = YES;
+				GCC_WARN_UNUSED_VARIABLE = YES;
+				PREBINDING = NO;
+				SDKROOT = "$(DEVELOPER_SDK_DIR)/MacOSX10.5.sdk";
+			};
+			name = Debug;
+		};
+		1DEB91F108733DB70010E9CD /* Release */ = {
+			isa = XCBuildConfiguration;
+			buildSettings = {
+				ARCHS = i386;
+				GCC_WARN_ABOUT_RETURN_TYPE = YES;
+				GCC_WARN_UNUSED_VARIABLE = YES;
+				PREBINDING = NO;
+				PRECOMPS_INCLUDE_HEADERS_FROM_BUILT_PRODUCTS_DIR = NO;
+				SDKROOT = "$(DEVELOPER_SDK_DIR)/MacOSX10.5.sdk";
+			};
+			name = Release;
+		};
+/* End XCBuildConfiguration section */
+
+/* Begin XCConfigurationList section */
+		1DEB91EB08733DB70010E9CD /* Build configuration list for PBXNativeTarget "easwt_vst_lib" */ = {
+			isa = XCConfigurationList;
+			buildConfigurations = (
+				1DEB91EC08733DB70010E9CD /* Debug */,
+				1DEB91ED08733DB70010E9CD /* Release */,
+			);
+			defaultConfigurationIsVisible = 0;
+			defaultConfigurationName = Release;
+		};
+		1DEB91EF08733DB70010E9CD /* Build configuration list for PBXProject "easwt_vst_lib" */ = {
+			isa = XCConfigurationList;
+			buildConfigurations = (
+				1DEB91F008733DB70010E9CD /* Debug */,
+				1DEB91F108733DB70010E9CD /* Release */,
+			);
+			defaultConfigurationIsVisible = 0;
+			defaultConfigurationName = Release;
+		};
+/* End XCConfigurationList section */
+	};
+	rootObject = 08FB7793FE84155DC02AAC07 /* Project object */;
+}

common/embeddedaudiosynthesis/arm-wt-22k/jetcreator_lib_src/darwin-x86/easwt_vst_lib.xcodeproj/xcuserdata/jeff.farrand.xcuserdatad/xcschemes/easwt_vst_lib.xcscheme

@@ -0,0 +1,57 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<Scheme
+   version = "1.3">
+   <BuildAction
+      parallelizeBuildables = "YES"
+      buildImplicitDependencies = "YES">
+      <BuildActionEntries>
+         <BuildActionEntry
+            buildForTesting = "YES"
+            buildForRunning = "YES"
+            buildForProfiling = "YES"
+            buildForArchiving = "YES"
+            buildForAnalyzing = "YES">
+            <BuildableReference
+               BuildableIdentifier = "primary"
+               BlueprintIdentifier = "D2AAC045055464E500DB518D"
+               BuildableName = "libeaswt_vst_lib.a"
+               BlueprintName = "easwt_vst_lib"
+               ReferencedContainer = "container:easwt_vst_lib.xcodeproj">
+            </BuildableReference>
+         </BuildActionEntry>
+      </BuildActionEntries>
+   </BuildAction>
+   <TestAction
+      selectedDebuggerIdentifier = "Xcode.DebuggerFoundation.Debugger.GDB"
+      selectedLauncherIdentifier = "Xcode.DebuggerFoundation.Launcher.GDB"
+      shouldUseLaunchSchemeArgsEnv = "YES"
+      buildConfiguration = "Debug">
+      <Testables>
+      </Testables>
+   </TestAction>
+   <LaunchAction
+      selectedDebuggerIdentifier = "Xcode.DebuggerFoundation.Debugger.GDB"
+      selectedLauncherIdentifier = "Xcode.DebuggerFoundation.Launcher.GDB"
+      launchStyle = "0"
+      useCustomWorkingDirectory = "NO"
+      buildConfiguration = "Debug"
+      debugDocumentVersioning = "YES"
+      allowLocationSimulation = "YES">
+      <AdditionalOptions>
+      </AdditionalOptions>
+   </LaunchAction>
+   <ProfileAction
+      shouldUseLaunchSchemeArgsEnv = "YES"
+      savedToolIdentifier = ""
+      useCustomWorkingDirectory = "NO"
+      buildConfiguration = "Release"
+      debugDocumentVersioning = "YES">
+   </ProfileAction>
+   <AnalyzeAction
+      buildConfiguration = "Debug">
+   </AnalyzeAction>
+   <ArchiveAction
+      buildConfiguration = "Release"
+      revealArchiveInOrganizer = "YES">
+   </ArchiveAction>
+</Scheme>

common/embeddedaudiosynthesis/arm-wt-22k/jetcreator_lib_src/darwin-x86/easwt_vst_lib.xcodeproj/xcuserdata/jeff.farrand.xcuserdatad/xcschemes/xcschememanagement.plist

@@ -0,0 +1,22 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<!DOCTYPE plist PUBLIC "-//Apple//DTD PLIST 1.0//EN" "http://www.apple.com/DTDs/PropertyList-1.0.dtd">
+<plist version="1.0">
+<dict>
+	<key>SchemeUserState</key>
+	<dict>
+		<key>easwt_vst_lib.xcscheme</key>
+		<dict>
+			<key>orderHint</key>
+			<integer>11</integer>
+		</dict>
+	</dict>
+	<key>SuppressBuildableAutocreation</key>
+	<dict>
+		<key>D2AAC045055464E500DB518D</key>
+		<dict>
+			<key>primary</key>
+			<true/>
+		</dict>
+	</dict>
+</dict>
+</plist>

common/embeddedaudiosynthesis/arm-wt-22k/jetcreator_lib_src/darwin-x86/easwt_vst_lib.xcodeproj/xcuserdata/ryan.gerleve.xcuserdatad/xcschemes/easwt_vst_lib.xcscheme

@@ -0,0 +1,57 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<Scheme
+   version = "1.3">
+   <BuildAction
+      parallelizeBuildables = "YES"
+      buildImplicitDependencies = "YES">
+      <BuildActionEntries>
+         <BuildActionEntry
+            buildForTesting = "YES"
+            buildForRunning = "YES"
+            buildForProfiling = "YES"
+            buildForArchiving = "YES"
+            buildForAnalyzing = "YES">
+            <BuildableReference
+               BuildableIdentifier = "primary"
+               BlueprintIdentifier = "D2AAC045055464E500DB518D"
+               BuildableName = "libeaswt_vst_lib.a"
+               BlueprintName = "easwt_vst_lib"
+               ReferencedContainer = "container:easwt_vst_lib.xcodeproj">
+            </BuildableReference>
+         </BuildActionEntry>
+      </BuildActionEntries>
+   </BuildAction>
+   <TestAction
+      selectedDebuggerIdentifier = "Xcode.DebuggerFoundation.Debugger.GDB"
+      selectedLauncherIdentifier = "Xcode.DebuggerFoundation.Launcher.GDB"
+      shouldUseLaunchSchemeArgsEnv = "YES"
+      buildConfiguration = "Debug">
+      <Testables>
+      </Testables>
+   </TestAction>
+   <LaunchAction
+      selectedDebuggerIdentifier = "Xcode.DebuggerFoundation.Debugger.GDB"
+      selectedLauncherIdentifier = "Xcode.DebuggerFoundation.Launcher.GDB"
+      launchStyle = "0"
+      useCustomWorkingDirectory = "NO"
+      buildConfiguration = "Debug"
+      debugDocumentVersioning = "YES"
+      allowLocationSimulation = "YES">
+      <AdditionalOptions>
+      </AdditionalOptions>
+   </LaunchAction>
+   <ProfileAction
+      shouldUseLaunchSchemeArgsEnv = "YES"
+      savedToolIdentifier = ""
+      useCustomWorkingDirectory = "NO"
+      buildConfiguration = "Release"
+      debugDocumentVersioning = "YES">
+   </ProfileAction>
+   <AnalyzeAction
+      buildConfiguration = "Debug">
+   </AnalyzeAction>
+   <ArchiveAction
+      buildConfiguration = "Release"
+      revealArchiveInOrganizer = "YES">
+   </ArchiveAction>
+</Scheme>

common/embeddedaudiosynthesis/arm-wt-22k/jetcreator_lib_src/darwin-x86/easwt_vst_lib.xcodeproj/xcuserdata/ryan.gerleve.xcuserdatad/xcschemes/xcschememanagement.plist

@@ -0,0 +1,22 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<!DOCTYPE plist PUBLIC "-//Apple//DTD PLIST 1.0//EN" "http://www.apple.com/DTDs/PropertyList-1.0.dtd">
+<plist version="1.0">
+<dict>
+	<key>SchemeUserState</key>
+	<dict>
+		<key>easwt_vst_lib.xcscheme</key>
+		<dict>
+			<key>orderHint</key>
+			<integer>3</integer>
+		</dict>
+	</dict>
+	<key>SuppressBuildableAutocreation</key>
+	<dict>
+		<key>D2AAC045055464E500DB518D</key>
+		<dict>
+			<key>primary</key>
+			<true/>
+		</dict>
+	</dict>
+</dict>
+</plist>

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/ARM-E_filter_gnu.s

@@ -0,0 +1,134 @@
+@***********************************************************
+@ Function:    WT_VoiceFilter
+@ Processor:   ARM
+@ Description:
+@ Implements a 2-pole low-pass filter with resonanance
+@
+@ Usage: 
+@	void WT_VoiceFilter(
+@		S_FILTER CONTROL *pFilter,
+@		S_WT_FRAME *pWTFrame);
+@
+@ Copyright 2005 Sonic Network, Inc.
+@****************************************************************
+@ Revision Control:
+@   $Revision: 496 $
+@   $Date: 2006-12-11 14:33:26 -0800 (Mon, 11 Dec 2006) $
+@****************************************************************
+@
+@   where:
+@	S_FILTER_CONTROL *pFilter
+@	PASSED IN: r0
+@
+@	S_WT_FRAME *pWTFrame
+@	PASSED IN: r1
+@****************************************************************
+
+	.include	"ARM_synth_constants_gnu.inc"
+
+	.arm
+	.text
+
+
+	.global	WT_VoiceFilter
+	
+
+@ Register usage
+@ --------------
+pFilter	.req	r0
+pWTFrame	.req	r1
+pBuffer	.req	r2
+numSamples	.req	r3
+
+z1	.req	r4
+z2	.req	r5
+b1	.req	r6
+b2	.req	r7
+K	.req	r8
+
+tmp0	.req	r1	@ reuse register
+tmp1	.req	r9
+tmp2	.req	r10
+
+
+@SaveRegs	RLIST	{r4-r10, lr}
+@RestoreRegs	RLIST	{r4-r10, pc}
+
+
+	.func	WT_VoiceFilter
+WT_VoiceFilter:
+
+	STMFD	sp!, {r4-r10, lr}
+
+@
+@ Setup passed parameters in their destination registers
+@----------------------------------------------------------------
+
+	LDR		pBuffer, [pWTFrame, #m_pAudioBuffer]
+	LDR		numSamples, [pWTFrame, #m_numSamples]
+
+	@load state variables from pFilter structure
+	LDRSH	z1, [pFilter, #m_z1]
+	LDRSH	z2, [pFilter, #m_z2]
+
+	@load coefficients from pWTFrame structure
+	LDR		K, [pWTFrame, #m_k]
+	LDR		b1, [pWTFrame, #m_b1]
+	LDR		b2, [pWTFrame, #m_b2]
+	
+	RSB	b1, b1, #0						@ b1 = -b1
+	RSB	b2, b2, #0						@ b2 = -b2
+	MOV	b2, b2, ASR #1					@ b2 = b2 >> 1
+	MOV	K, K, ASR #1					@ K = K >> 1
+
+@
+@ Start processing
+@----------------------------------------------------------------
+
+	LDRSH	tmp0, [pBuffer]					@ fetch sample
+
+FilterLoop:
+	SMULBB	tmp2, z1, b1					@ tmp2 = z1 * -b1
+	SMLABB	tmp2, z2, b2, tmp2				@ tmp2 = (-b1 * z1) + (-b2 * z2)
+
+	MOV		z2, z1							@ delay line
+
+	SMLABB	tmp0, tmp0, K, tmp2				@ tmp1 = (K * x[n]) + (-b1 * z1) + (-b2 * z2)
+	
+	LDRSH	tmp1, [pBuffer, #NEXT_OUTPUT_PCM]	@ fetch next sample
+	
+	MOV		z1, tmp0, ASR #14				@ shift result to low word
+	STRH	z1, [pBuffer], #NEXT_OUTPUT_PCM	@ write back to buffer
+
+	SMULBB	tmp2, z1, b1					@ tmp2 = z1 * -b1
+	
+	SUBS	numSamples, numSamples, #2		@ unroll loop once
+
+	SMLABB	tmp2, z2, b2, tmp2				@ tmp2 = (-b1 * z1) + (-b2 * z2)
+
+	SMLABB	tmp1, tmp1, K, tmp2				@ tmp1 = (K * x[n]) + (-b1 * z1) + (-b2 * z2)
+
+	MOV		z2, z1							@ delay line
+
+	MOV		z1, tmp1, ASR #14				@ shift result to low word
+	
+	LDRGTSH	tmp0, [pBuffer, #NEXT_OUTPUT_PCM]	@ fetch next sample
+
+	STRH	z1, [pBuffer], #NEXT_OUTPUT_PCM	@ write back to buffer
+
+	BGT		FilterLoop
+@ save z terms
+@----------------------------------------------------------------
+
+	STRH	z1, [pFilter, #m_z1]
+	STRH	z2, [pFilter, #m_z2]
+
+@ Return to calling function
+@----------------------------------------------------------------
+
+	LDMFD	sp!,{r4-r10, lr}
+	BX		lr
+
+	.endfunc
+	.end
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/ARM-E_interpolate_loop_gnu.s

@@ -0,0 +1,131 @@
+@***********************************************************
+@ Function:    WT_Interpolate
+@ Processor:   ARM-E
+@ Description: the main synthesis function when fetching 
+@			   wavetable samples.
+@              C-callable.
+@
+@ Usage: 
+@	void WT_Interpolate(
+@		S_WT_VOICE *pWTVoice,
+@		S_WT_FRAME *pWTFrame);
+@
+@ Copyright Sonic Network Inc. 2004
+@****************************************************************
+@ Revision Control:
+@   $Revision: 496 $
+@   $Date: 2006-12-11 14:33:26 -0800 (Mon, 11 Dec 2006) $
+@****************************************************************
+@
+@   where:
+@	S_WT_VOICE *pWTVoice
+@	PASSED IN: r0
+@
+@	S_WT_FRAME *pWTFrame;
+@	PASSED IN: r1
+@****************************************************************
+
+	.include	"ARM_synth_constants_gnu.inc"
+
+	.arm
+	.text
+
+	.global	WT_Interpolate
+
+
+@ Register usage
+@ --------------
+pWTVoice	.req	r0
+pWTFrame	.req	r1
+
+numSamples	.req	r2
+phaseIncrement	.req	r3
+pOutputBuffer	.req	r4
+
+tmp0	.req	r1	@reuse register
+tmp1	.req	r5
+tmp2	.req	r6
+
+pLoopEnd	.req	r7
+pLoopStart	.req	r8
+
+pPhaseAccum	.req	r9
+phaseFrac	.req	r10
+phaseFracMask	.req	r11
+
+@SaveRegs	RLIST	{r4-r11,lr}
+@RestoreRegs	RLIST	{r4-r11,pc}
+
+	.func	WT_Interpolate
+WT_Interpolate:
+
+	STMFD	sp!,{r4-r11,lr}
+
+@
+@ Fetch parameters from structures
+@----------------------------------------------------------------
+
+	LDR		pOutputBuffer, [pWTFrame, #m_pAudioBuffer]
+	LDR		numSamples, [pWTFrame, #m_numSamples]
+
+	LDR		phaseIncrement, [pWTFrame, #m_phaseIncrement]
+	LDR		pPhaseAccum, [pWTVoice, #m_pPhaseAccum]
+	LDR		phaseFrac, [pWTVoice, #m_phaseFrac]
+	LDR		phaseFracMask,=PHASE_FRAC_MASK
+
+	LDR		pLoopStart, [pWTVoice, #m_pLoopStart]
+	LDR		pLoopEnd, [pWTVoice, #m_pLoopEnd]
+	ADD		pLoopEnd, pLoopEnd, #1					@ need loop end to equal last sample + 1
+
+InterpolationLoop:
+	SUBS	tmp0, pPhaseAccum, pLoopEnd		@ check for loop end
+	ADDGE	pPhaseAccum, pLoopStart, tmp0	@ loop back to start
+
+	.ifdef	SAMPLES_8_BIT
+	LDRSB	tmp0, [pPhaseAccum]				@ tmp0 = x0
+	LDRSB	tmp1, [pPhaseAccum, #1]			@ tmp1 = x1
+	.else
+	LDRSH	tmp0, [pPhaseAccum]				@ tmp0 = x0
+	LDRSH	tmp1, [pPhaseAccum, #2]			@ tmp1 = x1
+	.endif
+
+	ADD		tmp2, phaseIncrement, phaseFrac	@ increment pointer here to avoid pipeline stall
+
+	SUB		tmp1, tmp1, tmp0						@ tmp1 = x1 - x0
+	SMULBB	tmp1, phaseFrac, tmp1			@ tmp1 = phaseFrac * tmp2
+
+@ This section performs a gain adjustment of -12dB for 16-bit samples
+@ or +36dB for 8-bit samples. For a high quality synthesizer, the output
+@ can be set to full scale, however if the filter is used, it can overflow
+@ with certain coefficients and signal sources. In this case, either a
+@ saturation operation should take in the filter before scaling back to
+@ 16 bits or the signal path should be increased to 18 bits or more.
+
+	.ifdef	SAMPLES_8_BIT
+	MOV		tmp0, tmp0, LSL #6							@ boost 8-bit signal by 36dB
+	.else
+	MOV		tmp0, tmp0, ASR #2							@ reduce 16-bit signal by 12dB
+	.endif															
+	
+	ADD		tmp1, tmp0, tmp1, ASR #(NUM_EG1_FRAC_BITS-6)	@ tmp1 = tmp0 + (tmp1 >> (15-6))
+															@	   = x0 + f * (x1 - x0) == interpolated result
+	
+	STRH	tmp1, [pOutputBuffer], #NEXT_OUTPUT_PCM	@ *pOutputBuffer++ = interpolated result
+	
+@ carry overflow from fraction to integer portion
+	ADD	pPhaseAccum, pPhaseAccum, tmp2, LSR #(NUM_PHASE_FRAC_BITS - NEXT_INPUT_PCM_SHIFT)
+	AND	phaseFrac, tmp2, phaseFracMask		@ nphaseFrac = frac part
+
+	SUBS	numSamples, numSamples, #1
+	BGT		InterpolationLoop
+	
+@ update and store phase
+	STR		pPhaseAccum, [pWTVoice, #m_pPhaseAccum]
+	STR		phaseFrac, [pWTVoice, #m_phaseFrac]
+
+	LDMFD	sp!,{r4-r11,lr}
+	BX		lr
+
+	.endfunc
+	.end
+	

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/ARM-E_interpolate_noloop_gnu.s

@@ -0,0 +1,130 @@
+@***********************************************************
+@ Function:    WT_InterpolateNoLoop
+@ Processor:   ARM-E
+@ Description: the main synthesis function when fetching 
+@			   wavetable samples.
+@              C-callable.
+@
+@ Usage: 
+@	void WT_InterpolateNoLoop(
+@		S_WT_VOICE *pWTVoice,
+@		S_WT_FRAME *pWTFrame);
+@
+@ Copyright Sonic Network Inc. 2004
+@****************************************************************
+@ Revision Control:
+@   $Revision: 496 $
+@   $Date: 2006-12-11 14:33:26 -0800 (Mon, 11 Dec 2006) $
+@****************************************************************
+@
+@   where:
+@	S_WT_VOICE *pWTVoice
+@	PASSED IN: r0
+@
+@	S_WT_FRAME *pWTFrame;
+@	PASSED IN: r1
+@****************************************************************
+
+	.include	"ARM_synth_constants_gnu.inc"
+
+	.arm
+	.text
+
+	
+	.global	WT_InterpolateNoLoop
+	
+
+@ Register usage
+@ --------------
+pWTVoice	.req	r0
+pWTFrame	.req	r1
+pOutputBuffer	.req	r2
+numSamples	.req	r3
+
+phaseIncrement	.req	r4
+pPhaseAccum	.req	r5
+phaseFrac	.req	r6
+phaseFracMask	.req	r7
+
+tmp0	.req	r1	@ reuse register
+tmp1	.req	r8
+tmp2	.req	r9
+
+
+@SaveRegs	RLIST	{r4-r9,lr}
+@RestoreRegs	RLIST	{r4-r9,pc}
+
+	.func	WT_InterpolateNoLoop
+WT_InterpolateNoLoop:
+
+	STMFD	sp!, {r4-r9,lr}
+
+@
+@ Fetch parameters from structures
+@----------------------------------------------------------------
+
+	LDR		pOutputBuffer, [pWTFrame, #m_pAudioBuffer]
+	LDR		numSamples, [pWTFrame, #m_numSamples]
+	
+	LDR		phaseIncrement, [pWTFrame, #m_phaseIncrement]
+	LDR		pPhaseAccum, [pWTVoice, #m_pPhaseAccum]
+	LDR		phaseFrac, [pWTVoice, #m_phaseFrac]
+	LDR		phaseFracMask,=PHASE_FRAC_MASK
+
+InterpolationLoop:
+
+	.ifdef	SAMPLES_8_BIT
+	LDRSB	tmp0, [pPhaseAccum]				@ tmp0 = x0
+	LDRSB	tmp1, [pPhaseAccum, #1]			@ tmp1 = x1
+	.else
+	LDRSH	tmp0, [pPhaseAccum]				@ tmp0 = x0
+	LDRSH	tmp1, [pPhaseAccum, #2]			@ tmp1 = x1
+	.endif
+
+	ADD		tmp2, phaseIncrement, phaseFrac	@ increment pointer here to avoid pipeline stall
+
+	SUB		tmp1, tmp1, tmp0				@ tmp1 = x1 - x0
+	SMULBB	tmp1, phaseFrac, tmp1			@ tmp1 = phaseFrac * tmp2
+
+@ This section performs a gain adjustment of -12dB for 16-bit samples
+@ or +36dB for 8-bit samples. For a high quality synthesizer, the output
+@ can be set to full scale, however if the filter is used, it can overflow
+@ with certain coefficients and signal sources. In this case, either a
+@ saturation operation should take in the filter before scaling back to
+@ 16 bits or the signal path should be increased to 18 bits or more.
+
+	.ifdef	SAMPLES_8_BIT
+	MOV		tmp0, tmp0, LSL #6							@ boost 8-bit signal by 36dB
+	.else
+	MOV		tmp0, tmp0, ASR #2							@ reduce 16-bit signal by 12dB
+	.endif															
+	
+	ADD		tmp1, tmp0, tmp1, ASR #(NUM_EG1_FRAC_BITS-6)	@ tmp1 = tmp0 + (tmp1 >> (15-6))
+															@	   = x0 + f * (x1 - x0) == interpolated result
+	
+	STRH	tmp1, [pOutputBuffer], #NEXT_OUTPUT_PCM	@ *pOutputBuffer++ = interpolated result
+	
+@ carry overflow from fraction to integer portion
+	ADD	pPhaseAccum, pPhaseAccum, tmp2, LSR #(NUM_PHASE_FRAC_BITS - NEXT_INPUT_PCM_SHIFT)
+	AND	phaseFrac, tmp2, phaseFracMask		@ nphaseFrac = frac part
+
+	SUBS	numSamples, numSamples, #1
+	BGT		InterpolationLoop
+
+@ Clean up and store any changes that were caused during the loop
+@----------------------------------------------------------------
+
+	@ update and store phase
+	STR		pPhaseAccum, [pWTVoice, #m_pPhaseAccum]
+	STR		phaseFrac, [pWTVoice, #m_phaseFrac]
+
+@
+@ Return to calling function
+@----------------------------------------------------------------
+
+	LDMFD	sp!,{r4-r9,lr}
+	BX		lr
+
+	.endfunc
+	.end
+	

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/ARM-E_mastergain_gnu.s

@@ -0,0 +1,109 @@
+@***********************************************************
+@ Function:    SynthMasterGain
+@ Processor:   ARM-E
+@ Description: Copies 32-bit synth output to 16-bit buffer
+@              with saturated gain control
+@              C-callable.
+@
+@ Usage: 
+@	SynthMasterGain(
+@			pInputBuffer
+@			pOutputBuffer,
+@			nGain,
+@			nNumLoopSamples
+@	);
+@
+@ Copyright Sonic Network Inc. 2004
+@****************************************************************
+@ Revision Control:
+@   $Revision: 496 $
+@   $Date: 2006-12-11 14:33:26 -0800 (Mon, 11 Dec 2006) $
+@****************************************************************
+@
+@   where:
+@	long *pInputBuffer
+@	PASSED IN: r0
+@
+@	EAS_PCM *pOutputBuffer
+@	PASSED IN: r1
+@
+@	short nGain
+@	PASSED IN: r2
+@
+@	EAS_U16	nNumLoopSamples
+@	PASSED IN: r3
+@
+@****************************************************************
+
+	.include	"ARM_synth_constants_gnu.inc"
+
+	.arm
+	.text
+
+	.func	SynthMasterGain
+SynthMasterGain:
+
+	.global	SynthMasterGain	@ allow other files to use this function
+	
+
+
+
+
+@ Stack frame
+@ -----------
+	.equ	RET_ADDR_SZ, 0			@return address
+	.equ	REG_SAVE_SZ, 0			@save-on-entry registers saved
+	.equ	FRAME_SZ, (8)			@local variables
+	.equ	ARG_BLK_SZ, 0			@argument block
+
+	.equ	PARAM_OFFSET, (ARG_BLK_SZ + FRAME_SZ + REG_SAVE_SZ + RET_ADDR_SZ)
+
+@ Register usage
+@ --------------
+pnInputBuffer	.req	r0
+pnOutputBuffer	.req	r1
+nGain	.req	r2
+nNumLoopSamples	.req	r3
+
+	STMFD	sp!,{r4-r6,r14}					@Save any save-on-entry registers that are used
+
+	LDR		r6, =0x7fff						@constant for saturation tests
+
+loop:
+	LDR		r4, [pnInputBuffer], #4			@fetch 1st output sample
+	
+	LDR		r5, [pnInputBuffer], #4			@fetch 2nd output sample
+
+	SMULWB	r4, r4, nGain					@output = gain * input
+	
+	CMP		r4, r6							@check for positive saturation
+	MOVGT	r4, r6							@saturate
+	CMN		r4, r6							@check for negative saturation
+	MVNLT	r4, r6							@saturate
+	
+	SMULWB	r5, r5, nGain					@output = gain * input
+	
+	STRH	r4, [pnOutputBuffer], #NEXT_OUTPUT_PCM	@save 1st output sample
+	
+	CMP		r5, r6							@check for positive saturation
+	MOVGT	r5, r6							@saturate
+	CMN		r5, r6							@check for negative saturation
+	MVNLT	r5, r6							@saturate
+	STRH	r5, [pnOutputBuffer], #NEXT_OUTPUT_PCM	@save 2nd output sample
+
+	SUBS	nNumLoopSamples, nNumLoopSamples, #2
+	BGT		loop
+
+@
+@ Return to calling function
+@----------------------------------------------------------------
+
+	LDMFD	sp!,{r4-r6, lr}				@ return to calling function
+	BX		lr
+
+@*****************************************************************************
+
+	.endfunc		@ end of function/procedure
+
+	.end		@ end of assembly code
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/ARM-E_voice_gain_gnu.s

@@ -0,0 +1,166 @@
+@***********************************************************
+@ Function:    WT_VoiceGain
+@ Processor:   ARM-E
+@ Description: the main synthesis function when fetching 
+@			   wavetable samples.
+@              C-callable.
+@
+@ Usage: 
+@ Usage: 
+@	WT_VoiceGain(
+@			S_WT_VOICE *pWTVoice,
+@			S_WT_FRAME *pWTFrame);
+@
+@ Copyright 2004, 2005 Sonic Network, Inc.
+@****************************************************************
+@ Revision Control:
+@   $Revision: 814 $
+@   $Date: 2007-08-02 10:34:53 -0700 (Thu, 02 Aug 2007) $
+@****************************************************************
+@
+@   where:
+@	S_WT_VOICE *psVoice
+@	PASSED IN: r0
+@
+@	S_WT_FRAME *pWTFrame
+@	PASSED IN: r1
+@****************************************************************
+
+
+
+	.include	"ARM_synth_constants_gnu.inc"
+
+	.arm
+	.text
+
+	.global	WT_VoiceGain
+	
+@ Register usage
+@ --------------
+pWTVoice	.req	r0
+pWTFrame	.req	r1
+pInputBuffer	.req	r2
+pMixBuffer	.req	r3
+
+tmp0	.req	r4
+tmp1	.req	r5
+tmp2	.req	r1	@ reuse register
+tmp3	.req	r6
+
+numSamples	.req	r9
+
+	.if	STEREO_OUTPUT
+gainIncLeft	.req	r7
+gainIncRight	.req	r8
+gainLeft	.req	r10
+gainRight	.req	r11
+	.else
+gainIncrement	.req	r7
+gain	.req	r8
+	.endif
+
+
+@ register context for local variables
+@SaveRegs	RLIST	{r4-r11,lr}
+@RestoreRegs	RLIST	{r4-r11,pc}
+
+	.func	WT_VoiceGain
+WT_VoiceGain:
+
+	STMFD	sp!, {r4-r11,lr}
+
+	LDR		pInputBuffer, [pWTFrame, #m_pAudioBuffer]
+	LDR		pMixBuffer, [pWTFrame, #m_pMixBuffer]
+	LDR		numSamples, [pWTFrame, #m_numSamples]
+	
+@----------------------------------------------------------------
+@ Stereo version
+@----------------------------------------------------------------
+@ NOTE: instructions are reordered to reduce the effect of latency 
+@ due to storage and computational dependencies.
+@----------------------------------------------------------------
+
+	.if	STEREO_OUTPUT
+
+	LDR		tmp0, [pWTFrame, #m_prevGain]
+	LDR		tmp1, [pWTFrame, #m_gainTarget]
+	
+	LDRSH	gainLeft, [pWTVoice, #m_gainLeft]
+	LDRSH	gainRight, [pWTVoice, #m_gainRight]
+	
+	MOV		gainIncLeft, gainLeft
+	SMULBB	gainLeft, tmp0, gainLeft
+
+	SMULBB	gainIncLeft, tmp1, gainIncLeft
+	SUB		gainIncLeft, gainIncLeft, gainLeft
+	MOV		gainLeft, gainLeft, ASR #(NUM_MIXER_GUARD_BITS - 2)
+	MOV		gainIncLeft, gainIncLeft, ASR #(SYNTH_UPDATE_PERIOD_IN_BITS + NUM_MIXER_GUARD_BITS - 2)
+
+	MOV		gainIncRight, gainRight
+	SMULBB	gainRight, tmp0, gainRight
+
+	SMULBB	gainIncRight, tmp1, gainIncRight
+	SUB		gainIncRight, gainIncRight, gainRight
+	MOV		gainRight, gainRight, ASR #(NUM_MIXER_GUARD_BITS - 2)
+	MOV		gainIncRight, gainIncRight, ASR #(SYNTH_UPDATE_PERIOD_IN_BITS + NUM_MIXER_GUARD_BITS - 2)
+
+	LDRSH		tmp0, [pInputBuffer], #2
+	
+StereoGainLoop:
+	LDR		tmp1, [pMixBuffer]
+
+	ADD		gainLeft, gainLeft, gainIncLeft
+
+	SMLAWB	tmp1, gainLeft, tmp0, tmp1
+
+	LDR		tmp2, [pMixBuffer, #4]
+
+	ADD		gainRight, gainRight, gainIncRight
+
+	STR		tmp1, [pMixBuffer], #4
+
+	SMLAWB	tmp2, gainRight, tmp0, tmp2
+
+	SUBS	numSamples, numSamples, #1
+
+	LDRGTSH	tmp0, [pInputBuffer], #2
+	
+	STR		tmp2, [pMixBuffer], #4
+
+	BGT		StereoGainLoop
+
+@----------------------------------------------------------------
+@ Mono version
+@----------------------------------------------------------------
+	.else
+
+	LDR		gain, [pWTFrame, #m_prevGain]
+	MOV		gain, gain, LSL #(NUM_MIXER_GUARD_BITS + 4)
+	LDR		gainIncrement, [pWTFrame, #m_gainTarget]
+	MOV		gainIncrement, gainIncrement, LSL #(NUM_MIXER_GUARD_BITS + 4)
+	SUB		gainIncrement, gainIncrement, gain
+	MOV		gainIncrement, gainIncrement, ASR #SYNTH_UPDATE_PERIOD_IN_BITS
+	
+MonoGainLoop:
+
+	LDRSH	tmp0, [pInputBuffer], #NEXT_OUTPUT_PCM	@ fetch voice output
+	
+	LDR		tmp1, [pMixBuffer]						@ get left channel output sample
+	ADD		gain, gain, gainIncrement				@ gain step to eliminate zipper noise
+	SMULWB	tmp0, gain, tmp0 						@ sample * local gain
+
+	MOV		tmp0, tmp0, ASR #1						@ add 6dB headroom
+	ADD 	tmp1, tmp0, tmp1
+	STR		tmp1, [pMixBuffer], #4					@ save and bump pointer
+	
+	SUBS	numSamples, numSamples, #1
+	BGT		MonoGainLoop
+
+	.endif	@end Mono version
+
+	LDMFD	sp!,{r4-r11,lr}
+	BX		lr
+	
+	.endfunc
+	.end
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/ARM_synth_constants_gnu.inc

@@ -0,0 +1,153 @@
+@***********************************************************
+@ File:         ARM_synth_constants.inc
+@ Processor:    ARM
+@ Description:  Contains constants and defines, most of which
+@               are mirrored in synth.h
+@
+@ Copyright Sonic Network Inc. 2004
+@****************************************************************
+@ Revision Control:
+@   $Revision: 741 $
+@   $Date: 2007-06-22 16:39:21 -0700 (Fri, 22 Jun 2007) $
+@****************************************************************
+
+
+    .ifdef  SAMPLE_RATE_8000
+    .equ    SYNTH_UPDATE_PERIOD_IN_BITS, 5
+    .equ    BUFFER_SIZE_IN_MONO_SAMPLES, 32
+    .endif
+
+    .ifdef  SAMPLE_RATE_16000
+    .equ    SYNTH_UPDATE_PERIOD_IN_BITS, 6
+    .equ    BUFFER_SIZE_IN_MONO_SAMPLES, 64
+    .endif
+
+    .ifdef  SAMPLE_RATE_20000
+    .equ    SYNTH_UPDATE_PERIOD_IN_BITS, 7
+    .equ    BUFFER_SIZE_IN_MONO_SAMPLES, 128
+    .endif
+
+    .ifdef  SAMPLE_RATE_22050
+    .equ    SYNTH_UPDATE_PERIOD_IN_BITS, 7
+    .equ    BUFFER_SIZE_IN_MONO_SAMPLES, 128
+    .endif
+
+    .ifdef  SAMPLE_RATE_24000
+    .equ    SYNTH_UPDATE_PERIOD_IN_BITS, 7
+    .equ    BUFFER_SIZE_IN_MONO_SAMPLES, 128
+    .endif
+
+    .ifdef  SAMPLE_RATE_32000
+    .equ    SYNTH_UPDATE_PERIOD_IN_BITS, 7
+    .equ    BUFFER_SIZE_IN_MONO_SAMPLES, 128
+    .endif
+
+    .ifdef  SAMPLE_RATE_44100
+    .equ    SYNTH_UPDATE_PERIOD_IN_BITS, 8
+    .equ    BUFFER_SIZE_IN_MONO_SAMPLES, 256
+    .endif
+
+    .ifdef  SAMPLE_RATE_48000
+    .equ    SYNTH_UPDATE_PERIOD_IN_BITS, 8
+    .equ    BUFFER_SIZE_IN_MONO_SAMPLES, 256
+    .endif
+
+
+@ if the OUTPUT PCM sample is 16-bits, then when using indexed addressing,
+@ the next sample is this many bytes away
+    .equ    NEXT_OUTPUT_PCM, 2
+
+@****************************************************************************
+@/* macros for fractional phase accumulator */
+    .equ    NUM_PHASE_FRAC_BITS, 15
+
+    .equ    PHASE_FRAC_MASK, 0x7FFF
+
+@ shift for phase accumulator when fraction carries over
+    .ifdef  SAMPLES_8_BIT
+    .equ    NEXT_INPUT_PCM_SHIFT, 0
+    .endif
+
+    .ifdef  SAMPLES_16_BIT
+    .equ    NEXT_INPUT_PCM_SHIFT, 1
+    .endif
+
+@****************************************************************************
+    .equ    NUM_MIXER_GUARD_BITS, 4
+
+@****************************************************************************
+@/* Envelope 1 (EG1) calculation macros */
+    .equ    NUM_EG1_FRAC_BITS, 15
+
+@****************************************************************************
+
+    .equ    NUM_ENHANCER_FILTER_COEF_FRAC_BITS, 5
+
+@****************************************************************************
+
+@
+@ I've temporarily given up on the idea of getting ADS/RV and gcc to
+@ handle a struct in a compatible fashion. Switching to old fashion EQU
+@
+
+    .if FILTER_ENABLED
+@**************************************
+@ typedef struct s_filter_tag
+    .equ    m_z1, 0
+    .equ    m_z2, 2
+    .endif
+
+@**************************************
+@ typedef struct s_wt_frame_tag
+    .equ    m_gainTarget, 0
+    .equ    m_phaseIncrement, 4
+
+    .if FILTER_ENABLED
+    .equ    m_k, 8
+    .equ    m_b1, 12
+    .equ    m_b2, 16
+    .equ    m_pAudioBuffer, 20
+    .equ    m_pMixBuffer, 24
+    .equ    m_numSamples, 28
+    .equ    m_prevGain, 32
+    .else
+    .equ    m_pAudioBuffer, 8
+    .equ    m_pMixBuffer, 12
+    .equ    m_numSamples, 16
+    .equ    m_prevGain, 20
+    .endif
+
+
+@**************************************
+@ typedef struct s_wt_voice_tag
+    .equ    m_pLoopEnd, 0   @ /* points to last PCM sample (not 1 beyond last) */
+    .equ    m_pLoopStart, 4 @ /* points to first sample at start of loop */
+    .equ    m_pPhaseAccum, 8    @ /* points to first sample at start of loop */
+    .equ    m_phaseFrac, 12 @ /* points to first sample at start of loop */
+
+    .if STEREO_OUTPUT
+    .equ    m_gainLeft, 16  @ /* current gain, left ch  */
+    .equ    m_gainRight, 18 @ /* current gain, right ch */
+    .endif
+
+
+@****************************************************************************
+@ enhancer
+    .equ    m_nEnhancerFeedForward1, 0
+    .equ    m_nEnhancerFeedback1, 1
+    .equ    m_nDriveCoef, 2
+    .equ    m_nEnhancerFeedback2, 3
+    .equ    m_nWet, 4
+    .equ    m_nDry, 5
+
+    .equ    m_zF0L, 6   @ filter 1 zero state var, left
+    .equ    m_zF1L, 8   @ filter 1 pole state var, left
+    .equ    m_zF2L, 10  @ filter 2 zero state var, left
+    .equ    m_zF0R, 12  @ filter 1 zero state var, right
+    .equ    m_zF1R, 14  @ filter 1 pole state var, right
+    .equ    m_zF2R, 16  @ filter 2 zero state var, right
+
+@****************************************************************************
+
+
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/arm-wt-22k_lib.mak

@@ -0,0 +1,25 @@
+#
+# Auto-generated sample makefile
+#
+# This makefile is intended for use with GNU make.
+# Set the paths to the tools (CC, AR, LD, etc.)
+#
+
+vpath %.c lib_src
+
+CC = C:\Program Files\GNUARM\bin\arm-elf-gcc.exe
+AS = C:\Program Files\GNUARM\bin\arm-elf-as.exe
+LD = C:\Program Files\GNUARM\bin\arm-elf-gcc.exe
+AR = C:\Program Files\GNUARM\bin\arm-elf-ar.exe
+
+%.o: %.c
+	$(CC) -c -O2 -o $@ -I lib_src -I host_src -D NUM_OUTPUT_CHANNELS=2 -D _SAMPLE_RATE_22050 -D MAX_SYNTH_VOICES=64 -D EAS_WT_SYNTH -D _8_BIT_SAMPLES -D _FILTER_ENABLED -D _IMELODY_PARSER -D _RTTTL_PARSER -D _OTA_PARSER -D _XMF_PARSER -D _WAVE_PARSER -D _REVERB_ENABLED -D _CHORUS_ENABLED -D DLS_SYNTHESIZER -D _IMA_DECODER -D MMAPI_SUPPORT -D NATIVE_EAS_KERNEL -D JET_INTERFACE $<
+
+%.o: %.s
+	$(AS) -o $@ -EL -mcpu=arm946e-s -mfpu=softfpa -I lib_src --defsym CHECK_STACK=0 --defsym REVERB=0 --defsym CHORUS=0 --defsym STEREO_OUTPUT=1 --defsym SAMPLE_RATE_22050=1 --defsym SAMPLES_8_BIT=1 --defsym FILTER_ENABLED=1 $<
+
+OBJS = eas_mididata.o eas_pan.o eas_wavefiledata.o eas_imelody.o eas_xmfdata.o ARM-E_interpolate_noloop_gnu.o eas_chorusdata.o ARM-E_voice_gain_gnu.o eas_ota.o eas_reverbdata.o eas_rtttl.o eas_reverb.o jet.o eas_mdls.o eas_mixbuf.o eas_smf.o eas_tcdata.o eas_chorus.o eas_pcmdata.o eas_xmf.o eas_smfdata.o eas_math.o eas_tonecontrol.o eas_rtttldata.o eas_voicemgt.o eas_public.o eas_dlssynth.o ARM-E_interpolate_loop_gnu.o ARM-E_filter_gnu.o eas_midi.o eas_otadata.o eas_flog.o eas_wtengine.o eas_imaadpcm.o eas_wtsynth.o wt_22khz.o eas_pcm.o eas_mixer.o eas_wavefile.o eas_ima_tables.o eas_data.o ARM-E_mastergain_gnu.o eas_imelodydata.o
+
+arm-wt-22k.a: $(OBJS)
+	$(AR) rc lib$@ $(OBJS)
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/dls.h

@@ -0,0 +1,268 @@
+ /*
+
+     dls.h
+
+     Description:
+
+     Interface defines and structures for the Instrument Collection Form
+     RIFF DLS.
+
+     Written by Sonic Foundry 1996.  Released for public use.
+
+ */
+
+ #ifndef _INC_DLS
+ #define _INC_DLS
+
+ /*
+
+    Layout of an instrument collection:
+
+
+    RIFF [] 'DLS ' [colh,INSTLIST,WAVEPOOL,INFOLIST]
+
+    INSTLIST
+    LIST [] 'lins'
+                  LIST [] 'ins ' [insh,RGNLIST,ARTLIST,INFOLIST]
+                  LIST [] 'ins ' [insh,RGNLIST,ARTLIST,INFOLIST]
+                  LIST [] 'ins ' [insh,RGNLIST,ARTLIST,INFOLIST]
+
+    RGNLIST
+    LIST [] 'lrgn'
+                  LIST [] 'rgn '  [rgnh,wsmp,wlnk,ARTLIST]
+                  LIST [] 'rgn '  [rgnh,wsmp,wlnk,ARTLIST]
+                  LIST [] 'rgn '  [rgnh,wsmp,wlnk,ARTLIST]
+
+    ARTLIST
+    LIST [] 'lart'
+            'art1' level 1 Articulation connection graph
+            'art2' level 2 Articulation connection graph
+            '3rd1' Possible 3rd party articulation structure 1
+            '3rd2' Possible 3rd party articulation structure 2 .... and so on
+
+    WAVEPOOL
+    ptbl [] [pool table]
+    LIST [] 'wvpl'
+                  [path],
+                  [path],
+                  LIST [] 'wave',RIFFWAVE
+                  LIST [] 'wave',RIFFWAVE
+                  LIST [] 'wave',RIFFWAVE
+                  LIST [] 'wave',RIFFWAVE
+                  LIST [] 'wave',RIFFWAVE
+
+    INFOLIST
+    LIST [] 'INFO'
+                  'icmt' 'One of those crazy comments.'
+                  'icop' 'Copyright (C) 1996 Sonic Foundry'
+
+ */
+
+
+ /*
+    FOURCC's used in the DLS file
+ */
+/* shree */
+//#define FAR
+
+/* shree
+
+ #define FOURCC_DLS   mmioFOURCC('D','L','S',' ')
+ #define FOURCC_COLH  mmioFOURCC('c','o','l','h')
+ #define FOURCC_WVPL  mmioFOURCC('w','v','p','l')
+ #define FOURCC_PTBL  mmioFOURCC('p','t','b','l')
+ #define FOURCC_PATH  mmioFOURCC('p','a','t','h')
+ #define FOURCC_wave  mmioFOURCC('w','a','v','e')
+ #define FOURCC_LINS  mmioFOURCC('l','i','n','s')
+ #define FOURCC_INS   mmioFOURCC('i','n','s',' ')
+ #define FOURCC_INSH  mmioFOURCC('i','n','s','h')
+ #define FOURCC_LRGN  mmioFOURCC('l','r','g','n')
+ #define FOURCC_RGN   mmioFOURCC('r','g','n',' ')
+ #define FOURCC_RGNH  mmioFOURCC('r','g','n','h')
+ #define FOURCC_LART  mmioFOURCC('l','a','r','t')
+ #define FOURCC_ART1  mmioFOURCC('a','r','t','1')
+ #define FOURCC_WLNK  mmioFOURCC('w','l','n','k')
+ #define FOURCC_WSMP  mmioFOURCC('w','s','m','p')
+ #define FOURCC_VERS  mmioFOURCC('v','e','r','s')
+*/
+ /*
+    Articulation connection graph definitions
+ */
+
+ /* Generic Sources */
+ #define CONN_SRC_NONE              0x0000
+ #define CONN_SRC_LFO               0x0001
+ #define CONN_SRC_KEYONVELOCITY     0x0002
+ #define CONN_SRC_KEYNUMBER         0x0003
+ #define CONN_SRC_EG1               0x0004
+ #define CONN_SRC_EG2               0x0005
+ #define CONN_SRC_PITCHWHEEL        0x0006
+
+ /* Midi Controllers 0-127 */
+ #define CONN_SRC_CC1               0x0081
+ #define CONN_SRC_CC7               0x0087
+ #define CONN_SRC_CC10              0x008a
+ #define CONN_SRC_CC11              0x008b
+
+ /* Registered Parameter Numbers */
+ #define CONN_SRC_RPN0             0x0100
+ #define CONN_SRC_RPN1             0x0101
+ #define CONN_SRC_RPN2             0x0102
+
+ /* Generic Destinations */
+ #define CONN_DST_NONE              0x0000
+ #define CONN_DST_ATTENUATION       0x0001
+ #define CONN_DST_RESERVED          0x0002
+ #define CONN_DST_PITCH             0x0003
+ #define CONN_DST_PAN               0x0004
+
+ /* LFO Destinations */
+ #define CONN_DST_LFO_FREQUENCY     0x0104
+ #define CONN_DST_LFO_STARTDELAY    0x0105
+
+ /* EG1 Destinations */
+ #define CONN_DST_EG1_ATTACKTIME    0x0206
+ #define CONN_DST_EG1_DECAYTIME     0x0207
+ #define CONN_DST_EG1_RESERVED      0x0208
+ #define CONN_DST_EG1_RELEASETIME   0x0209
+ #define CONN_DST_EG1_SUSTAINLEVEL  0x020a
+
+ /* EG2 Destinations */
+ #define CONN_DST_EG2_ATTACKTIME    0x030a
+ #define CONN_DST_EG2_DECAYTIME     0x030b
+ #define CONN_DST_EG2_RESERVED      0x030c
+ #define CONN_DST_EG2_RELEASETIME   0x030d
+ #define CONN_DST_EG2_SUSTAINLEVEL  0x030e
+
+ #define CONN_TRN_NONE              0x0000
+ #define CONN_TRN_CONCAVE           0x0001
+
+ typedef struct _DLSVERSION {
+   DWORD    dwVersionMS;
+   DWORD    dwVersionLS;
+ }DLSVERSION, FAR *LPDLSVERSION;
+
+
+ typedef struct _CONNECTION {
+   USHORT   usSource;
+   USHORT   usControl;
+   USHORT   usDestination;
+   USHORT   usTransform;
+   LONG     lScale;
+   }CONNECTION, FAR *LPCONNECTION;
+
+
+ /* Level 1 Articulation Data */
+
+ typedef struct _CONNECTIONLIST {
+   ULONG    cbSize;            /* size of the connection list structure */
+   ULONG    cConnections;      /* count of connections in the list */
+   } CONNECTIONLIST, FAR *LPCONNECTIONLIST;
+
+
+
+ /*
+    Generic type defines for regions and instruments
+ */
+
+ typedef struct _RGNRANGE {
+   USHORT usLow;
+   USHORT usHigh;
+ }RGNRANGE, FAR * LPRGNRANGE;
+
+ #define F_INSTRUMENT_DRUMS      0x80000000
+
+ typedef struct _MIDILOCALE {
+   ULONG ulBank;
+   ULONG ulInstrument;
+ }MIDILOCALE, FAR *LPMIDILOCALE;
+
+ /*
+    Header structures found in an DLS file for collection, instruments, and
+    regions.
+ */
+
+ #define F_RGN_OPTION_SELFNONEXCLUSIVE  0x0001
+
+ typedef struct _RGNHEADER {
+   RGNRANGE RangeKey;            /* Key range */
+   RGNRANGE RangeVelocity;       /* Velocity Range */
+   USHORT   fusOptions;          /* Synthesis options for this range */
+   USHORT   usKeyGroup;          /* Key grouping for non simultaneous play
+                                    0 = no group, 1 up is group
+                                    for Level 1 only groups 1-15 are allowed */
+ }RGNHEADER, FAR *LPRGNHEADER;
+
+ typedef struct _INSTHEADER {
+   ULONG      cRegions;          /* Count of regions in this instrument */
+   MIDILOCALE Locale;            /* Intended MIDI locale of this instrument */
+ }INSTHEADER, FAR *LPINSTHEADER;
+
+ typedef struct _DLSHEADER {
+   ULONG      cInstruments;      /* Count of instruments in the collection */
+ }DLSHEADER, FAR *LPDLSHEADER;
+
+ /*
+    definitions for the Wave link structure
+ */
+
+ /*****  For level 1 only WAVELINK_CHANNEL_MONO is valid  ****
+   ulChannel allows for up to 32 channels of audio with each bit position
+   specifiying a channel of playback */
+
+ #define WAVELINK_CHANNEL_LEFT    0x0001
+ #define WAVELINK_CHANNEL_RIGHT   0x0002
+
+ #define F_WAVELINK_PHASE_MASTER  0x0001
+
+ typedef struct _WAVELINK { /* any paths or links are stored right after struct */
+   USHORT   fusOptions;     /* options flags for this wave */
+   USHORT   usPhaseGroup;   /* Phase grouping for locking channels */
+   ULONG    ulChannel;      /* channel placement */
+   ULONG    ulTableIndex;   /* index into the wave pool table, 0 based */
+ }WAVELINK, FAR *LPWAVELINK;
+
+ #define POOL_CUE_NULL  0xffffffff
+
+ typedef struct _POOLCUE {
+ //  ULONG    ulEntryIndex;   /* Index entry in the list */
+   ULONG    ulOffset;       /* Offset to the entry in the list */
+ }POOLCUE, FAR *LPPOOLCUE;
+
+ typedef struct _POOLTABLE {
+   ULONG    cbSize;            /* size of the pool table structure */
+   ULONG    cCues;             /* count of cues in the list */
+   } POOLTABLE, FAR *LPPOOLTABLE;
+
+ /*
+    Structures for the "wsmp" chunk
+ */
+
+ #define F_WSMP_NO_TRUNCATION     0x0001
+ #define F_WSMP_NO_COMPRESSION    0x0002
+
+
+ typedef struct _rwsmp {
+   ULONG   cbSize;
+   USHORT  usUnityNote;         /* MIDI Unity Playback Note */
+   SHORT   sFineTune;           /* Fine Tune in log tuning */
+   LONG    lAttenuation;        /* Overall Attenuation to be applied to data */
+   ULONG   fulOptions;          /* Flag options */
+   ULONG   cSampleLoops;        /* Count of Sample loops, 0 loops is one shot */
+   } WSMPL, FAR *LPWSMPL;
+
+
+ /* This loop type is a normal forward playing loop which is continually
+    played until the envelope reaches an off threshold in the release
+    portion of the volume envelope */
+
+ #define WLOOP_TYPE_FORWARD   0
+
+ typedef struct _rloop {
+   ULONG cbSize;
+   ULONG ulType;              /* Loop Type */
+   ULONG ulStart;             /* Start of loop in samples */
+   ULONG ulLength;            /* Length of loop in samples */
+ } WLOOP, FAR *LPWLOOP;
+
+ #endif /* _INC_DLS */

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/dls2.h

@@ -0,0 +1,122 @@
+/*
+
+    dls2.h
+
+    Description:
+
+    Interface defines and structures for the DLS2 extensions of DLS.
+
+
+     Written by Microsoft 1998.  Released for public use.
+
+*/
+
+#ifndef _INC_DLS2
+#define _INC_DLS2
+
+/*
+     FOURCC's used in the DLS2 file, in addition to DLS1 chunks
+*/
+
+#define FOURCC_RGN2  mmioFOURCC('r','g','n','2')
+#define FOURCC_LAR2  mmioFOURCC('l','a','r','2')
+#define FOURCC_ART2  mmioFOURCC('a','r','t','2')
+#define FOURCC_CDL   mmioFOURCC('c','d','l',' ')
+#define FOURCC_DLID  mmioFOURCC('d','l','i','d')
+
+/*
+     Articulation connection graph definitions. These are in addition to
+     the definitions in the DLS1 header.
+*/
+
+/* Generic Sources (in addition to DLS1 sources. */
+#define CONN_SRC_POLYPRESSURE       0x0007  /* Polyphonic Pressure */
+#define CONN_SRC_CHANNELPRESSURE        0x0008  /* Channel Pressure */
+#define CONN_SRC_VIBRATO            0x0009  /* Vibrato LFO */
+#define CONN_SRC_MONOPRESSURE           0x000a  /* MIDI Mono pressure */
+
+
+/* Midi Controllers */
+#define CONN_SRC_CC91           0x00db  /* Reverb Send */
+#define CONN_SRC_CC93           0x00dd  /* Chorus Send */
+
+
+/* Generic Destinations */
+#define CONN_DST_GAIN           0x0001  /* Same as CONN_DST_ ATTENUATION, but more appropriate terminology. */
+#define CONN_DST_KEYNUMBER 0x0005  /* Key Number Generator */
+
+/* Audio Channel Output Destinations */
+#define CONN_DST_LEFT           0x0010  /* Left Channel Send */
+#define CONN_DST_RIGHT          0x0011  /* Right Channel Send */
+#define CONN_DST_CENTER         0x0012  /* Center Channel Send */
+#define CONN_DST_LEFTREAR           0x0013  /* Left Rear Channel Send */
+#define CONN_DST_RIGHTREAR          0x0014  /* Right Rear Channel Send */
+#define CONN_DST_LFE_CHANNEL        0x0015  /* LFE Channel Send */
+#define CONN_DST_CHORUS         0x0080  /* Chorus Send */
+#define CONN_DST_REVERB         0x0081  /* Reverb Send */
+
+/* Vibrato LFO Destinations */
+#define CONN_DST_VIB_FREQUENCY      0x0114  /* Vibrato Frequency */
+#define CONN_DST_VIB_STARTDELAY     0x0115  /* Vibrato Start Delay */
+
+/* EG1 Destinations */
+#define CONN_DST_EG1_DELAYTIME      0x020B  /* EG1 Delay Time */
+#define CONN_DST_EG1_HOLDTIME       0x020C  /* EG1 Hold Time */
+#define CONN_DST_EG1_SHUTDOWNTIME       0x020D  /* EG1 Shutdown Time */
+
+
+/*  EG2 Destinations */
+#define CONN_DST_EG2_DELAYTIME      0x030F  /* EG2 Delay Time */
+#define CONN_DST_EG2_HOLDTIME       0x0310  /* EG2 Hold Time */
+
+
+/* Filter Destinations */
+#define CONN_DST_FILTER_CUTOFF      0x0500  /* Filter Cutoff Frequency */
+#define CONN_DST_FILTER_Q           0x0501  /* Filter Resonance */
+
+
+/* Transforms */
+#define CONN_TRN_CONVEX         0x0002  /* Convex Transform */
+#define CONN_TRN_SWITCH         0x0003  /* Switch Transform */
+
+
+/*  Conditional chunk operators */
+ #define DLS_CDL_AND            0x0001  /* X = X & Y */
+ #define DLS_CDL_OR         0x0002  /* X = X | Y */
+ #define DLS_CDL_XOR            0x0003  /* X = X ^ Y */
+ #define DLS_CDL_ADD            0x0004  /* X = X + Y */
+ #define DLS_CDL_SUBTRACT       0x0005  /* X = X - Y */
+ #define DLS_CDL_MULTIPLY       0x0006  /* X = X * Y */
+ #define DLS_CDL_DIVIDE     0x0007  /* X = X / Y */
+ #define DLS_CDL_LOGICAL_AND    0x0008  /* X = X && Y */
+ #define DLS_CDL_LOGICAL_OR     0x0009  /* X = X || Y */
+ #define DLS_CDL_LT         0x000A  /* X = (X < Y) */
+ #define DLS_CDL_LE         0x000B  /* X = (X <= Y) */
+ #define DLS_CDL_GT         0x000C  /* X = (X > Y) */
+ #define DLS_CDL_GE         0x000D  /* X = (X >= Y) */
+ #define DLS_CDL_EQ         0x000E  /* X = (X == Y) */
+ #define DLS_CDL_NOT            0x000F  /* X = !X */
+ #define DLS_CDL_CONST      0x0010  /* 32-bit constant */
+ #define DLS_CDL_QUERY      0x0011  /* 32-bit value returned from query */
+ #define DLS_CDL_QUERYSUPPORTED 0x0012  /* 32-bit value returned from query */
+
+/*
+  Loop and release
+*/
+
+#define WLOOP_TYPE_RELEASE 1
+
+/*
+  DLSID queries for <cdl-ck>
+*/
+DEFINE_DLSID(DLSID_GMInHardware, 0x178f2f24, 0xc364, 0x11d1, 0xa7, 0x60, 0x00, 0x00, 0xf8, 0x75, 0xac, 0x12);
+DEFINE_DLSID(DLSID_GSInHardware, 0x178f2f25, 0xc364, 0x11d1, 0xa7, 0x60, 0x00, 0x00, 0xf8, 0x75, 0xac, 0x12);
+DEFINE_DLSID(DLSID_XGInHardware, 0x178f2f26, 0xc364, 0x11d1, 0xa7, 0x60, 0x00, 0x00, 0xf8, 0x75, 0xac, 0x12);
+DEFINE_DLSID(DLSID_SupportsDLS1, 0x178f2f27, 0xc364, 0x11d1, 0xa7, 0x60, 0x00, 0x00, 0xf8, 0x75, 0xac, 0x12);
+DEFINE_DLSID(DLSID_SupportsDLS2, 0xf14599e5, 0x4689, 0x11d2, 0xaf, 0xa6, 0x0, 0xaa, 0x0, 0x24, 0xd8, 0xb6);
+DEFINE_DLSID(DLSID_SampleMemorySize, 0x178f2f28, 0xc364, 0x11d1, 0xa7, 0x60, 0x00, 0x00, 0xf8, 0x75, 0xac, 0x12);
+DEFINE_DLSID(DLSID_ManufacturersID, 0xb03e1181, 0x8095, 0x11d2, 0xa1, 0xef, 0x0, 0x60, 0x8, 0x33, 0xdb, 0xd8);
+DEFINE_DLSID(DLSID_ProductID, 0xb03e1182, 0x8095, 0x11d2, 0xa1, 0xef, 0x0, 0x60, 0x8, 0x33, 0xdb, 0xd8);
+DEFINE_DLSID(DLSID_SamplePlaybackRate, 0x2a91f713, 0xa4bf, 0x11d2, 0xbb, 0xdf, 0x0, 0x60, 0x8, 0x33, 0xdb, 0xd8);
+#endif  /* _INC_DLS2 */
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_audioconst.h

@@ -0,0 +1,97 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_audioconst.h
+ *
+ * Contents and purpose:
+ * Defines audio constants related to the sample rate, bit size, etc.
+ *
+ *
+ * Copyright Sonic Network Inc. 2005
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 82 $
+ *   $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_AUDIOCONST_H
+#define _EAS_AUDIOCONST_H
+
+/*----------------------------------------------------------------------------
+ * These macros define the various characteristics of the defined sample rates
+ *----------------------------------------------------------------------------
+ * BUFFER_SIZE_IN_MONO_SAMPLES  size of buffer in samples
+ * _OUTPUT_SAMPLE_RATE          compiled output sample rate
+ * AUDIO_FRAME_LENGTH           length of an audio frame in 256ths of a millisecond
+ * SYNTH_UPDATE_PERIOD_IN_BITS  length of an audio frame (2^x samples)
+ *----------------------------------------------------------------------------
+*/
+
+#if defined (_SAMPLE_RATE_8000)
+#define BUFFER_SIZE_IN_MONO_SAMPLES     32
+#define _OUTPUT_SAMPLE_RATE             8000
+#define AUDIO_FRAME_LENGTH              1024
+#define SYNTH_UPDATE_PERIOD_IN_BITS     5
+
+#elif defined (_SAMPLE_RATE_16000)
+#define BUFFER_SIZE_IN_MONO_SAMPLES     64
+#define _OUTPUT_SAMPLE_RATE             16000
+#define AUDIO_FRAME_LENGTH              1024
+#define SYNTH_UPDATE_PERIOD_IN_BITS     6
+
+#elif defined (_SAMPLE_RATE_20000)
+#define BUFFER_SIZE_IN_MONO_SAMPLES     128
+#define _OUTPUT_SAMPLE_RATE             20000
+#define AUDIO_FRAME_LENGTH              1638
+#define SYNTH_UPDATE_PERIOD_IN_BITS     7
+
+#elif defined (_SAMPLE_RATE_22050)
+#define BUFFER_SIZE_IN_MONO_SAMPLES     128
+#define _OUTPUT_SAMPLE_RATE             22050
+#define AUDIO_FRAME_LENGTH              1486
+#define SYNTH_UPDATE_PERIOD_IN_BITS     7
+
+#elif defined (_SAMPLE_RATE_24000)
+#define BUFFER_SIZE_IN_MONO_SAMPLES     128
+#define _OUTPUT_SAMPLE_RATE             24000
+#define AUDIO_FRAME_LENGTH              1365
+#define SYNTH_UPDATE_PERIOD_IN_BITS     7
+
+#elif defined (_SAMPLE_RATE_32000)
+#define BUFFER_SIZE_IN_MONO_SAMPLES     128
+#define _OUTPUT_SAMPLE_RATE             32000
+#define AUDIO_FRAME_LENGTH              1024
+#define SYNTH_UPDATE_PERIOD_IN_BITS     7
+
+#elif defined (_SAMPLE_RATE_44100)
+#define BUFFER_SIZE_IN_MONO_SAMPLES     256
+#define _OUTPUT_SAMPLE_RATE             44100
+#define AUDIO_FRAME_LENGTH              1486
+#define SYNTH_UPDATE_PERIOD_IN_BITS     8
+
+#elif defined (_SAMPLE_RATE_48000)
+#define BUFFER_SIZE_IN_MONO_SAMPLES     256
+#define _OUTPUT_SAMPLE_RATE             48000
+#define AUDIO_FRAME_LENGTH              1365
+#define SYNTH_UPDATE_PERIOD_IN_BITS     8
+
+#else
+#error "_SAMPLE_RATE_XXXXX must be defined to valid rate"
+#endif
+
+#endif /* #ifndef _EAS_AUDIOCONST_H */
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_chorus.c

@@ -0,0 +1,604 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_chorus.c
+ *
+ * Contents and purpose:
+ * Contains the implementation of the Chorus effect.
+ *
+ *
+ * Copyright Sonic Network Inc. 2006
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 499 $
+ *   $Date: 2006-12-11 16:07:20 -0800 (Mon, 11 Dec 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+#include "eas_data.h"
+#include "eas_effects.h"
+#include "eas_math.h"
+#include "eas_chorusdata.h"
+#include "eas_chorus.h"
+#include "eas_config.h"
+#include "eas_host.h"
+#include "eas_report.h"
+
+/* prototypes for effects interface */
+static EAS_RESULT ChorusInit (EAS_DATA_HANDLE pEASData, EAS_VOID_PTR *pInstData);
+static void ChorusProcess (EAS_VOID_PTR pInstData, EAS_PCM *pSrc, EAS_PCM *pDst, EAS_I32 numSamples);
+static EAS_RESULT ChorusShutdown (EAS_DATA_HANDLE pEASData, EAS_VOID_PTR pInstData);
+static EAS_RESULT ChorusGetParam (EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 *pValue);
+static EAS_RESULT ChorusSetParam (EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 value);
+
+/* common effects interface for configuration module */
+const S_EFFECTS_INTERFACE EAS_Chorus =
+{
+    ChorusInit,
+    ChorusProcess,
+    ChorusShutdown,
+    ChorusGetParam,
+    ChorusSetParam
+};
+
+
+
+//LFO shape table used by the chorus, larger table would sound better
+//this is a sine wave, where 32767 = 1.0
+static const EAS_I16 EAS_chorusShape[CHORUS_SHAPE_SIZE] = {
+    0, 1608, 3212, 4808, 6393, 7962, 9512, 11309, 12539, 14010, 15446, 16846, 18204, 19519, 20787, 22005, 23170,
+    24279, 25329, 26319, 27245, 28105, 28898, 29621, 30273, 30852, 31356, 31785, 32137, 32412, 32609, 32728,
+    32767, 32728, 32609, 32412, 32137, 31785, 31356, 30852, 30273, 29621, 28898, 28105, 27245, 26319, 25329,
+    24279, 23170, 22005, 20787, 19519, 18204, 16846, 15446, 14010, 12539, 11039, 9512, 7962, 6393, 4808, 3212,
+    1608, 0, -1608, -3212, -4808, -6393, -7962, -9512, -11309, -12539, -14010, -15446, -16846, -18204, -19519,
+    -20787, -22005, -23170, -24279, -25329, -26319, -27245, -28105, -28898, -29621, -30273, -30852, -31356, -31785,
+    -32137, -32412, -32609, -32728, -32767, -32728, -32609, -32412, -32137, -31785, -31356, -30852, -30273, -29621,
+    -28898, -28105, -27245, -26319, -25329, -24279, -23170, -22005, -20787, -19519, -18204, -16846, -15446, -14010,
+    -12539, -11039, -9512, -7962, -6393, -4808, -3212, -1608
+};
+
+/*----------------------------------------------------------------------------
+ * InitializeChorus()
+ *----------------------------------------------------------------------------
+ * Purpose: Initializes chorus parameters
+ *
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT ChorusInit (EAS_DATA_HANDLE pEASData, EAS_VOID_PTR *pInstData)
+{
+    S_CHORUS_OBJECT *pChorusData;
+    S_CHORUS_PRESET *pPreset;
+    EAS_I32 index;
+
+    /* check Configuration Module for data allocation */
+    if (pEASData->staticMemoryModel)
+        pChorusData = EAS_CMEnumFXData(EAS_MODULE_CHORUS);
+
+    /* allocate dynamic memory */
+    else
+        pChorusData = EAS_HWMalloc(pEASData->hwInstData, sizeof(S_CHORUS_OBJECT));
+
+    if (pChorusData == NULL)
+    {
+        { /* dpp: EAS_ReportEx(_EAS_SEVERITY_FATAL, "Failed to allocate Chorus memory\n"); */ }
+        return EAS_ERROR_MALLOC_FAILED;
+    }
+
+    /* clear the structure */
+    EAS_HWMemSet(pChorusData, 0, sizeof(S_CHORUS_OBJECT));
+
+    ChorusReadInPresets(pChorusData);
+
+    /* set some default values */
+    pChorusData->bypass =       EAS_CHORUS_BYPASS_DEFAULT;
+    pChorusData->preset =       EAS_CHORUS_PRESET_DEFAULT;
+    pChorusData->m_nLevel =     EAS_CHORUS_LEVEL_DEFAULT;
+    pChorusData->m_nRate =      EAS_CHORUS_RATE_DEFAULT;
+    pChorusData->m_nDepth =     EAS_CHORUS_DEPTH_DEFAULT;
+
+    //chorus rate and depth need some massaging from preset value (which is sample rate independent)
+
+    //convert rate from steps of .05 Hz to value which can be used as phase increment,
+    //with current CHORUS_SHAPE_SIZE and rate limits, this fits into 16 bits
+    //want to compute ((shapeSize * 65536) * (storedRate/20))/sampleRate;
+    //computing it as below allows rate steps to be evenly spaced
+    //uses 32 bit divide, but only once when new value is selected
+    pChorusData->m_nRate = (EAS_I16)
+        ((((EAS_I32)CHORUS_SHAPE_SIZE<<16)/(20*(EAS_I32)_OUTPUT_SAMPLE_RATE)) * pChorusData->m_nRate);
+
+    //convert depth from steps of .05 ms, to samples, with 16 bit whole part, discard fraction
+    //want to compute ((depth * sampleRate)/20000)
+    //use the following approximation since 105/32 is roughly 65536/20000
+    /*lint -e{704} use shift for performance */
+    pChorusData->m_nDepth = (EAS_I16)
+        (((((EAS_I32)pChorusData->m_nDepth * _OUTPUT_SAMPLE_RATE)>>5) * 105) >> 16);
+
+    pChorusData->m_nLevel = pChorusData->m_nLevel;
+
+    //zero delay memory for chorus
+    for (index = CHORUS_L_SIZE - 1; index >= 0; index--)
+    {
+        pChorusData->chorusDelayL[index] = 0;
+    }
+    for (index = CHORUS_R_SIZE - 1; index >= 0; index--)
+    {
+        pChorusData->chorusDelayR[index] = 0;
+    }
+
+    //init delay line index, these are used to implement circular delay buffer
+    pChorusData->chorusIndexL = 0;
+    pChorusData->chorusIndexR = 0;
+
+    //init LFO phase
+    //16 bit whole part, 16 bit fraction
+    pChorusData->lfoLPhase = 0;
+    pChorusData->lfoRPhase = (CHORUS_SHAPE_SIZE << 16) >> 2; // 1/4 of total, i.e. 90 degrees out of phase;
+
+    //init chorus delay position
+    //right now chorus delay is a compile-time value, as is sample rate
+    pChorusData->chorusTapPosition = (EAS_I16)((CHORUS_DELAY_MS * _OUTPUT_SAMPLE_RATE)/1000);
+
+    //now copy from the new preset into Chorus
+    pPreset = &pChorusData->m_sPreset.m_sPreset[pChorusData->m_nNextChorus];
+
+    pChorusData->m_nLevel = pPreset->m_nLevel;
+    pChorusData->m_nRate =  pPreset->m_nRate;
+    pChorusData->m_nDepth = pPreset->m_nDepth;
+
+    pChorusData->m_nRate = (EAS_I16)
+        ((((EAS_I32)CHORUS_SHAPE_SIZE<<16)/(20*(EAS_I32)_OUTPUT_SAMPLE_RATE)) * pChorusData->m_nRate);
+
+    /*lint -e{704} use shift for performance */
+    pChorusData->m_nDepth = (EAS_I16)
+        (((((EAS_I32)pChorusData->m_nDepth * _OUTPUT_SAMPLE_RATE)>>5) * 105) >> 16);
+
+    *pInstData = pChorusData;
+
+    return EAS_SUCCESS;
+} /* end ChorusInit */
+
+/*----------------------------------------------------------------------------
+ * WeightedTap()
+ *----------------------------------------------------------------------------
+ * Purpose: Does fractional array look-up using linear interpolation
+ *
+ * first convert indexDesired to actual desired index by taking into account indexReference
+ * then do linear interpolation between two actual samples using fractional part
+ *
+ * Inputs:
+ * array: pointer to array of signed 16 bit values, typically either PCM data or control data
+ * indexReference: the circular buffer relative offset
+ * indexDesired: the fractional index we are looking up (16 bits index + 16 bits fraction)
+ * indexLimit: the total size of the array, used to compute buffer wrap
+ *
+ * Outputs:
+ * Value from the input array, linearly interpolated between two actual data values
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_I16 WeightedTap(const EAS_I16 *array, EAS_I16 indexReference, EAS_I32 indexDesired, EAS_I16 indexLimit)
+{
+    EAS_I16 index;
+    EAS_I16 fraction;
+    EAS_I16 val1;
+    EAS_I16 val2;
+
+    //separate indexDesired into whole and fractional parts
+    /*lint -e{704} use shift for performance */
+    index = (EAS_I16)(indexDesired >> 16);
+    /*lint -e{704} use shift for performance */
+    fraction = (EAS_I16)((indexDesired>>1) & 0x07FFF); //just use 15 bits of fractional part
+
+    //adjust whole part by indexReference
+    index = indexReference - index;
+    //make sure we stay within array bounds, this implements circular buffer
+    while (index < 0)
+    {
+        index += indexLimit;
+    }
+
+    //get two adjacent values from the array
+    val1 = array[index];
+
+    //handle special case when index == 0, else typical case
+    if (index == 0)
+    {
+        val2 = array[indexLimit-1]; //get last value from array
+    }
+    else
+    {
+        val2 = array[index-1]; //get previous value from array
+    }
+
+    //compute linear interpolation as (val1 + ((val2-val1)*fraction))
+    return(val1 + (EAS_I16)MULT_EG1_EG1(val2-val1,fraction));
+}
+
+/*----------------------------------------------------------------------------
+ * ChorusProcess()
+ *----------------------------------------------------------------------------
+ * Purpose: compute the chorus on the input buffer, and mix into output buffer
+ *
+ *
+ * Inputs:
+ * src: pointer to input buffer of PCM values to be processed
+ * dst: pointer to output buffer of PCM values we are to sume the result with
+ * bufSize: the number of sample frames (i.e. stereo samples) in the buffer
+ *
+ * Outputs:
+ * None
+ *
+ *----------------------------------------------------------------------------
+*/
+//compute the chorus, and mix into output buffer
+static void ChorusProcess (EAS_VOID_PTR pInstData, EAS_PCM *pSrc, EAS_PCM *pDst, EAS_I32 numSamples)
+{
+    EAS_I32 ix;
+    EAS_I32 nChannelNumber;
+    EAS_I16 lfoValueLeft;
+    EAS_I16 lfoValueRight;
+    EAS_I32 positionOffsetL;
+    EAS_I32 positionOffsetR;
+    EAS_PCM tapL;
+    EAS_PCM tapR;
+    EAS_I32 tempValue;
+    EAS_PCM nInputSample;
+    EAS_I32 nOutputSample;
+    EAS_PCM *pIn;
+    EAS_PCM *pOut;
+
+    S_CHORUS_OBJECT *pChorusData;
+
+    pChorusData = (S_CHORUS_OBJECT*) pInstData;
+
+    //if the chorus is disabled or turned all the way down
+    if (pChorusData->bypass == EAS_TRUE || pChorusData->m_nLevel == 0)
+    {
+        if (pSrc != pDst)
+            EAS_HWMemCpy(pSrc, pDst, numSamples * NUM_OUTPUT_CHANNELS * (EAS_I32) sizeof(EAS_PCM));
+        return;
+    }
+
+    if (pChorusData->m_nNextChorus != pChorusData->m_nCurrentChorus)
+    {
+        ChorusUpdate(pChorusData);
+    }
+
+    for (nChannelNumber = 0; nChannelNumber < NUM_OUTPUT_CHANNELS; nChannelNumber++)
+    {
+
+        pIn = pSrc + nChannelNumber;
+        pOut = pDst + nChannelNumber;
+
+        if(nChannelNumber==0)
+        {
+            for (ix = 0; ix < numSamples; ix++)
+            {
+                nInputSample = *pIn;
+                pIn += NUM_OUTPUT_CHANNELS;
+
+                //feed input into chorus delay line
+                pChorusData->chorusDelayL[pChorusData->chorusIndexL] = nInputSample;
+
+                //compute chorus lfo value using phase as fractional index into chorus shape table
+                //resulting value is between -1.0 and 1.0, expressed as signed 16 bit number
+                lfoValueLeft = WeightedTap(EAS_chorusShape, 0, pChorusData->lfoLPhase, CHORUS_SHAPE_SIZE);
+
+                //scale chorus depth by lfo value to get relative fractional sample index
+                //index is expressed as 32 bit number with 16 bit fractional part
+                /*lint -e{703} use shift for performance */
+                positionOffsetL = pChorusData->m_nDepth * (((EAS_I32)lfoValueLeft) << 1);
+
+                //add fixed chorus delay to get actual fractional sample index
+                positionOffsetL += ((EAS_I32)pChorusData->chorusTapPosition) << 16;
+
+                //get tap value from chorus delay using fractional sample index
+                tapL = WeightedTap(pChorusData->chorusDelayL, pChorusData->chorusIndexL, positionOffsetL, CHORUS_L_SIZE);
+
+                //scale by chorus level, then sum with input buffer contents and saturate
+                tempValue = MULT_EG1_EG1(tapL, pChorusData->m_nLevel);
+                nOutputSample = SATURATE(tempValue + nInputSample);
+
+                *pOut = (EAS_I16)SATURATE(nOutputSample);
+                pOut += NUM_OUTPUT_CHANNELS;
+
+
+                //increment chorus delay index and make it wrap as needed
+                //this implements circular buffer
+                if ((pChorusData->chorusIndexL+=1) >= CHORUS_L_SIZE)
+                    pChorusData->chorusIndexL = 0;
+
+                //increment fractional lfo phase, and make it wrap as needed
+                pChorusData->lfoLPhase += pChorusData->m_nRate;
+                while (pChorusData->lfoLPhase >= (CHORUS_SHAPE_SIZE<<16))
+                {
+                    pChorusData->lfoLPhase -= (CHORUS_SHAPE_SIZE<<16);
+                }
+            }
+        }
+        else
+        {
+            for (ix = 0; ix < numSamples; ix++)
+            {
+                nInputSample = *pIn;
+                pIn += NUM_OUTPUT_CHANNELS;
+
+                //feed input into chorus delay line
+                pChorusData->chorusDelayR[pChorusData->chorusIndexR] = nInputSample;
+
+                //compute chorus lfo value using phase as fractional index into chorus shape table
+                //resulting value is between -1.0 and 1.0, expressed as signed 16 bit number
+                lfoValueRight = WeightedTap(EAS_chorusShape, 0, pChorusData->lfoRPhase, CHORUS_SHAPE_SIZE);
+
+                //scale chorus depth by lfo value to get relative fractional sample index
+                //index is expressed as 32 bit number with 16 bit fractional part
+                /*lint -e{703} use shift for performance */
+                positionOffsetR = pChorusData->m_nDepth * (((EAS_I32)lfoValueRight) << 1);
+
+                //add fixed chorus delay to get actual fractional sample index
+                positionOffsetR += ((EAS_I32)pChorusData->chorusTapPosition) << 16;
+
+                //get tap value from chorus delay using fractional sample index
+                tapR = WeightedTap(pChorusData->chorusDelayR, pChorusData->chorusIndexR, positionOffsetR, CHORUS_R_SIZE);
+
+                //scale by chorus level, then sum with output buffer contents and saturate
+                tempValue = MULT_EG1_EG1(tapR, pChorusData->m_nLevel);
+                nOutputSample = SATURATE(tempValue + nInputSample);
+
+                *pOut = (EAS_I16)SATURATE(nOutputSample);
+                pOut += NUM_OUTPUT_CHANNELS;
+
+                //increment chorus delay index and make it wrap as needed
+                //this implements circular buffer
+                if ((pChorusData->chorusIndexR+=1) >= CHORUS_R_SIZE)
+                    pChorusData->chorusIndexR = 0;
+
+                //increment fractional lfo phase, and make it wrap as needed
+                pChorusData->lfoRPhase += pChorusData->m_nRate;
+                while (pChorusData->lfoRPhase >= (CHORUS_SHAPE_SIZE<<16))
+                {
+                    pChorusData->lfoRPhase -= (CHORUS_SHAPE_SIZE<<16);
+                }
+            }
+        }
+
+    }
+}  /* end ChorusProcess */
+
+
+
+/*----------------------------------------------------------------------------
+ * ChorusShutdown()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Initializes the Chorus effect.
+ *
+ * Inputs:
+ * pInstData        - handle to instance data
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT ChorusShutdown (EAS_DATA_HANDLE pEASData, EAS_VOID_PTR pInstData)
+{
+    /* check Configuration Module for static memory allocation */
+    if (!pEASData->staticMemoryModel)
+        EAS_HWFree(pEASData->hwInstData, pInstData);
+    return EAS_SUCCESS;
+} /* end ChorusShutdown */
+
+/*----------------------------------------------------------------------------
+ * ChorusGetParam()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Get a Chorus parameter
+ *
+ * Inputs:
+ * pInstData        - handle to instance data
+ * param            - parameter index
+ * *pValue          - pointer to variable to hold retrieved value
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT ChorusGetParam (EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 *pValue)
+{
+    S_CHORUS_OBJECT *p;
+
+    p = (S_CHORUS_OBJECT*) pInstData;
+
+    switch (param)
+    {
+        case EAS_PARAM_CHORUS_BYPASS:
+            *pValue = (EAS_I32) p->bypass;
+            break;
+        case EAS_PARAM_CHORUS_PRESET:
+            *pValue = (EAS_I8) p->m_nCurrentChorus;
+            break;
+        case EAS_PARAM_CHORUS_RATE:
+            *pValue = (EAS_I32) p->m_nRate;
+            break;
+        case EAS_PARAM_CHORUS_DEPTH:
+            *pValue = (EAS_I32) p->m_nDepth;
+            break;
+        case EAS_PARAM_CHORUS_LEVEL:
+            *pValue = (EAS_I32) p->m_nLevel;
+            break;
+        default:
+            return EAS_ERROR_INVALID_PARAMETER;
+    }
+    return EAS_SUCCESS;
+} /* end ChorusGetParam */
+
+
+/*----------------------------------------------------------------------------
+ * ChorusSetParam()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Set a Chorus parameter
+ *
+ * Inputs:
+ * pInstData        - handle to instance data
+ * param            - parameter index
+ * *pValue          - new paramter value
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT ChorusSetParam (EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 value)
+{
+    S_CHORUS_OBJECT *p;
+
+    p = (S_CHORUS_OBJECT*) pInstData;
+
+    switch (param)
+    {
+        case EAS_PARAM_CHORUS_BYPASS:
+            p->bypass = (EAS_BOOL) value;
+            break;
+        case EAS_PARAM_CHORUS_PRESET:
+            if(value!=EAS_PARAM_CHORUS_PRESET1 && value!=EAS_PARAM_CHORUS_PRESET2 &&
+                value!=EAS_PARAM_CHORUS_PRESET3 && value!=EAS_PARAM_CHORUS_PRESET4)
+                return EAS_ERROR_INVALID_PARAMETER;
+            p->m_nNextChorus = (EAS_I8)value;
+            break;
+        case EAS_PARAM_CHORUS_RATE:
+            if(value<EAS_CHORUS_RATE_MIN || value>EAS_CHORUS_RATE_MAX)
+                return EAS_ERROR_INVALID_PARAMETER;
+            p->m_nRate = (EAS_I16) value;
+            break;
+        case EAS_PARAM_CHORUS_DEPTH:
+            if(value<EAS_CHORUS_DEPTH_MIN || value>EAS_CHORUS_DEPTH_MAX)
+                return EAS_ERROR_INVALID_PARAMETER;
+            p->m_nDepth = (EAS_I16) value;
+            break;
+        case EAS_PARAM_CHORUS_LEVEL:
+            if(value<EAS_CHORUS_LEVEL_MIN || value>EAS_CHORUS_LEVEL_MAX)
+                return EAS_ERROR_INVALID_PARAMETER;
+            p->m_nLevel = (EAS_I16) value;
+            break;
+
+        default:
+            return EAS_ERROR_INVALID_PARAMETER;
+    }
+    return EAS_SUCCESS;
+} /* end ChorusSetParam */
+
+
+/*----------------------------------------------------------------------------
+ * ChorusReadInPresets()
+ *----------------------------------------------------------------------------
+ * Purpose: sets global Chorus preset bank to defaults
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT ChorusReadInPresets(S_CHORUS_OBJECT *pChorusData)
+{
+
+    int preset = 0;
+    int defaultPreset = 0;
+
+    //now init any remaining presets to defaults
+    for (defaultPreset = preset; defaultPreset < CHORUS_MAX_TYPE; defaultPreset++)
+    {
+        S_CHORUS_PRESET *pPreset = &pChorusData->m_sPreset.m_sPreset[defaultPreset];
+        if (defaultPreset == 0 || defaultPreset > CHORUS_MAX_TYPE-1)
+        {
+            pPreset->m_nDepth = 39;
+            pPreset->m_nRate = 30;
+            pPreset->m_nLevel = 32767;
+        }
+        else if (defaultPreset == 1)
+        {
+            pPreset->m_nDepth = 21;
+            pPreset->m_nRate = 45;
+            pPreset->m_nLevel = 25000;
+        }
+        else if (defaultPreset == 2)
+        {
+            pPreset->m_nDepth = 53;
+            pPreset->m_nRate = 25;
+            pPreset->m_nLevel = 32000;
+        }
+        else if (defaultPreset == 3)
+        {
+            pPreset->m_nDepth = 32;
+            pPreset->m_nRate = 37;
+            pPreset->m_nLevel = 29000;
+        }
+    }
+
+    return EAS_SUCCESS;
+}
+
+
+/*----------------------------------------------------------------------------
+ * ChorusUpdate
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Update the Chorus preset parameters as required
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ * - chorus paramters will be changed
+ * - m_nCurrentRoom := m_nNextRoom
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT ChorusUpdate(S_CHORUS_OBJECT *pChorusData)
+{
+    S_CHORUS_PRESET *pPreset = &pChorusData->m_sPreset.m_sPreset[pChorusData->m_nNextChorus];
+
+    pChorusData->m_nLevel = pPreset->m_nLevel;
+    pChorusData->m_nRate =  pPreset->m_nRate;
+    pChorusData->m_nDepth = pPreset->m_nDepth;
+
+    pChorusData->m_nRate = (EAS_I16)
+        ((((EAS_I32)CHORUS_SHAPE_SIZE<<16)/(20*(EAS_I32)_OUTPUT_SAMPLE_RATE)) * pChorusData->m_nRate);
+
+    /*lint -e{704} use shift for performance */
+    pChorusData->m_nDepth = (EAS_I16)
+        (((((EAS_I32)pChorusData->m_nDepth * _OUTPUT_SAMPLE_RATE)>>5) * 105) >> 16);
+
+    pChorusData->m_nCurrentChorus = pChorusData->m_nNextChorus;
+
+    return EAS_SUCCESS;
+
+}   /* end ChorusUpdate */

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_chorusdata.c

@@ -0,0 +1,34 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_chorusdata.c
+ *
+ * Contents and purpose:
+ * Contains the static data allocation for the Chorus effect
+ *
+ *
+ * Copyright Sonic Network Inc. 2006
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 550 $
+ *   $Date: 2007-02-02 09:37:03 -0800 (Fri, 02 Feb 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#include "eas_chorusdata.h"
+
+S_CHORUS_OBJECT eas_ChorusData;
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_chorusdata.h

@@ -0,0 +1,160 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_chorusdata.h
+ *
+ * Contents and purpose:
+ * Contains the prototypes for the Chorus effect.
+ *
+ *
+ * Copyright Sonic Network Inc. 2006
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 309 $
+ *   $Date: 2006-09-12 18:52:45 -0700 (Tue, 12 Sep 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_CHORUS_H
+#define _EAS_CHORUS_H
+
+#include "eas_types.h"
+#include "eas_audioconst.h"
+
+//defines for chorus
+
+#define EAS_CHORUS_BYPASS_DEFAULT   1
+#define EAS_CHORUS_PRESET_DEFAULT   0
+#define EAS_CHORUS_RATE_DEFAULT     30
+#define EAS_CHORUS_DEPTH_DEFAULT    39
+#define EAS_CHORUS_LEVEL_DEFAULT    32767
+
+#define EAS_CHORUS_LEVEL_MIN        0
+#define EAS_CHORUS_LEVEL_MAX        32767
+
+#define EAS_CHORUS_RATE_MIN         10
+#define EAS_CHORUS_RATE_MAX         50
+
+#define EAS_CHORUS_DEPTH_MIN        15
+#define EAS_CHORUS_DEPTH_MAX        60
+
+#define CHORUS_SIZE_MS 20
+#define CHORUS_L_SIZE ((CHORUS_SIZE_MS*_OUTPUT_SAMPLE_RATE)/1000)
+#define CHORUS_R_SIZE CHORUS_L_SIZE
+#define CHORUS_SHAPE_SIZE 128
+#define CHORUS_DELAY_MS 10
+
+#define CHORUS_MAX_TYPE         4   // any Chorus numbers larger than this are invalid
+
+typedef struct
+{
+    EAS_I16             m_nRate;
+    EAS_I16             m_nDepth;
+    EAS_I16             m_nLevel;
+
+} S_CHORUS_PRESET;
+
+typedef struct
+{
+    S_CHORUS_PRESET     m_sPreset[CHORUS_MAX_TYPE]; //array of presets
+
+} S_CHORUS_PRESET_BANK;
+
+/* parameters for each Chorus */
+typedef struct
+{
+    EAS_I32 lfoLPhase;
+    EAS_I32 lfoRPhase;
+    EAS_I16 chorusIndexL;
+    EAS_I16 chorusIndexR;
+    EAS_U16 chorusTapPosition;
+
+    EAS_I16 m_nRate;
+    EAS_I16 m_nDepth;
+    EAS_I16 m_nLevel;
+
+    //delay lines used by the chorus, longer would sound better
+    EAS_PCM chorusDelayL[CHORUS_L_SIZE];
+    EAS_PCM chorusDelayR[CHORUS_R_SIZE];
+
+    EAS_BOOL    bypass;
+    EAS_I8      preset;
+
+    EAS_I16     m_nCurrentChorus;           // preset number for current Chorus
+    EAS_I16     m_nNextChorus;              // preset number for next Chorus
+
+    S_CHORUS_PRESET         pPreset;
+
+    S_CHORUS_PRESET_BANK    m_sPreset;
+
+} S_CHORUS_OBJECT;
+
+
+/*----------------------------------------------------------------------------
+ * WeightedTap()
+ *----------------------------------------------------------------------------
+ * Purpose: Does fractional array look-up using linear interpolation
+ *
+ * first convert indexDesired to actual desired index by taking into account indexReference
+ * then do linear interpolation between two actual samples using fractional part
+ *
+ * Inputs:
+ * array: pointer to array of signed 16 bit values, typically either PCM data or control data
+ * indexReference: the circular buffer relative offset
+ * indexDesired: the fractional index we are looking up (16 bits index + 16 bits fraction)
+ * indexLimit: the total size of the array, used to compute buffer wrap
+ *
+ * Outputs:
+ * Value from the input array, linearly interpolated between two actual data values
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_I16 WeightedTap(const EAS_I16 *array, EAS_I16 indexReference, EAS_I32 indexDesired, EAS_I16 indexLimit);
+
+/*----------------------------------------------------------------------------
+ * ChorusReadInPresets()
+ *----------------------------------------------------------------------------
+ * Purpose: sets global Chorus preset bank to defaults
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT ChorusReadInPresets(S_CHORUS_OBJECT *pChorusData);
+
+/*----------------------------------------------------------------------------
+ * ChorusUpdate
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Update the Chorus preset parameters as required
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ * - chorus paramters will be changed
+ * - m_nCurrentChorus := m_nNextChorus
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT ChorusUpdate(S_CHORUS_OBJECT* pChorusData);
+
+#endif /* #ifndef _EAS_CHORUSDATA_H */
+
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_ctype.h

@@ -0,0 +1,41 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_ctype.h
+ *
+ * Contents and purpose:
+ * This is a replacement for the CRT ctype.h functions. These
+ * functions are currently ASCII only, but eventually, we will want
+ * to support wide-characters for localization.
+ *
+ * Copyright (c) 2005 Sonic Network Inc.
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 429 $
+ *   $Date: 2006-10-19 23:50:15 -0700 (Thu, 19 Oct 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_CTYPE_H
+#define _EAS_CTYPE_H
+
+EAS_INLINE EAS_I8 IsDigit (EAS_I8 c) { return ((c >= '0') && (c <= '9')); }
+EAS_INLINE EAS_I8 IsSpace (EAS_I8 c) { return (((c >= 9) && (c <= 13)) || (c == ' ')); }
+EAS_INLINE EAS_I8 ToUpper (EAS_I8 c) { if ((c >= 'a') && (c <= 'z')) return c & ~0x20; else return c; }
+EAS_INLINE EAS_I8 ToLower (EAS_I8 c) { if ((c >= 'A') && (c <= 'Z')) return c |= 0x20; else return c; }
+
+#endif
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_data.c

@@ -0,0 +1,37 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_data.c
+ *
+ * Contents and purpose:
+ * Contains a data allocation for synthesizer
+ *
+ * Copyright Sonic Network Inc. 2004
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 547 $
+ *   $Date: 2007-01-31 16:30:17 -0800 (Wed, 31 Jan 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+// includes
+#include "eas_data.h"
+
+// globals
+S_EAS_DATA eas_Data;
+S_VOICE_MGR eas_Synth;
+S_SYNTH eas_MIDI;
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_data.h

@@ -0,0 +1,133 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_data.h
+ *
+ * Contents and purpose:
+ * This header defines all types, to support dynamic allocation of the
+ * memory resources needed for persistent EAS data.
+ *
+ * Copyright 2004 Sonic Network Inc.
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 842 $
+ *   $Date: 2007-08-23 14:32:31 -0700 (Thu, 23 Aug 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_DATA_H
+#define _EAS_DATA_H
+
+//#define JET_INTERFACE
+
+#include "eas_types.h"
+#include "eas_synthcfg.h"
+#include "eas.h"
+#include "eas_audioconst.h"
+#include "eas_sndlib.h"
+#include "eas_pcm.h"
+#include "eas_pcmdata.h"
+#include "eas_synth.h"
+#include "eas_miditypes.h"
+#include "eas_effects.h"
+
+#ifdef AUX_MIXER
+#include "eas_auxmixdata.h"
+#endif
+
+#ifdef JET_INTERFACE
+#include "jet.h"
+#endif
+
+#ifdef _METRICS_ENABLED
+#include "eas_perf.h"
+#endif
+
+#ifndef MAX_NUMBER_STREAMS
+#define MAX_NUMBER_STREAMS          4
+#endif
+
+/* flags for S_EAS_STREAM */
+#define STREAM_FLAGS_PARSED         1
+#define STREAM_FLAGS_PAUSE          2
+#define STREAM_FLAGS_LOCATE         4
+#define STREAM_FLAGS_RESUME         8
+
+/* structure for parsing a stream */
+typedef struct s_eas_stream_tag
+{
+    void                            *pParserModule;
+    EAS_U32                         time;
+    EAS_U32                         frameLength;
+    EAS_I32                         repeatCount;
+    EAS_VOID_PTR                    handle;
+    EAS_U8                          volume;
+    EAS_BOOL8                       streamFlags;
+} S_EAS_STREAM;
+
+/* default master volume is -10dB */
+#define DEFAULT_VOLUME              90
+#define DEFAULT_STREAM_VOLUME       100
+#define DEFAULT_STREAM_GAIN         14622
+
+/* 10 dB of boost available for individual parsers */
+#define STREAM_VOLUME_HEADROOM      10
+
+/* amalgamated persistent data type */
+typedef struct s_eas_data_tag
+{
+#ifdef _CHECKED_BUILD
+    EAS_U32                         handleCheck;
+#endif
+    EAS_HW_DATA_HANDLE              hwInstData;
+
+    S_EFFECTS_MODULE                effectsModules[NUM_EFFECTS_MODULES];
+
+#ifdef _METRICS_ENABLED
+    S_METRICS_INTERFACE             *pMetricsModule;
+    EAS_VOID_PTR                    pMetricsData;
+#endif
+
+    EAS_I32                         *pMixBuffer;
+    EAS_PCM                         *pOutputAudioBuffer;
+
+#ifdef AUX_MIXER
+    S_EAS_AUX_MIXER                 auxMixer;
+#endif
+
+#ifdef _MAXIMIZER_ENABLED
+    EAS_VOID_PTR                    pMaximizerData;
+#endif
+
+    S_EAS_STREAM                    streams[MAX_NUMBER_STREAMS];
+
+    S_PCM_STATE                     *pPCMStreams;
+
+    S_VOICE_MGR                     *pVoiceMgr;
+
+#ifdef JET_INTERFACE
+    JET_DATA_HANDLE                 jetHandle;
+#endif
+
+    EAS_U32                         renderTime;
+    EAS_I16                         masterGain;
+    EAS_U8                          masterVolume;
+    EAS_BOOL8                       staticMemoryModel;
+    EAS_BOOL8                       searchHeaderFlag;
+} S_EAS_DATA;
+
+#endif
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_dlssynth.c

@@ -0,0 +1,578 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_dlssynth.c
+ *
+ * Contents and purpose:
+ * Implements the Mobile DLS synthesizer.
+ *
+ * Copyright Sonic Network Inc. 2006
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 795 $
+ *   $Date: 2007-08-01 00:14:45 -0700 (Wed, 01 Aug 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+// includes
+#include "eas_data.h"
+#include "eas_report.h"
+#include "eas_host.h"
+#include "eas_math.h"
+#include "eas_synth_protos.h"
+#include "eas_wtsynth.h"
+#include "eas_pan.h"
+#include "eas_mdls.h"
+#include "eas_dlssynth.h"
+
+#ifdef _METRICS_ENABLED
+#include "eas_perf.h"
+#endif
+
+static void DLS_UpdateEnvelope (S_SYNTH_VOICE *pVoice, S_SYNTH_CHANNEL *pChannel,  const S_DLS_ENVELOPE *pEnvParams, EAS_I16 *pValue, EAS_I16 *pIncrement, EAS_U8 *pState);
+
+/*----------------------------------------------------------------------------
+ * DLS_MuteVoice()
+ *----------------------------------------------------------------------------
+ * Mute the voice using shutdown time from the DLS articulation data
+ *----------------------------------------------------------------------------
+*/
+void DLS_MuteVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum)
+{
+    S_WT_VOICE *pWTVoice;
+    const S_DLS_ARTICULATION *pDLSArt;
+
+    pWTVoice = &pVoiceMgr->wtVoices[voiceNum];
+    pDLSArt = &pSynth->pDLS->pDLSArticulations[pWTVoice->artIndex];
+
+    /* clear deferred action flags */
+    pVoice->voiceFlags &=
+        ~(VOICE_FLAG_DEFER_MIDI_NOTE_OFF |
+        VOICE_FLAG_SUSTAIN_PEDAL_DEFER_NOTE_OFF |
+        VOICE_FLAG_DEFER_MUTE);
+
+    /* set the envelope state */
+    pVoiceMgr->wtVoices[voiceNum].eg1State = eEnvelopeStateRelease;
+    pWTVoice->eg1Increment = pDLSArt->eg1ShutdownTime;
+    pVoiceMgr->wtVoices[voiceNum].eg2State = eEnvelopeStateRelease;
+    pWTVoice->eg2Increment = pDLSArt->eg2.releaseTime;
+}
+
+/*----------------------------------------------------------------------------
+ * DLS_ReleaseVoice()
+ *----------------------------------------------------------------------------
+ * Release the selected voice.
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pVoice) standard API, pVoice may be used by other synthesizers */
+void DLS_ReleaseVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum)
+{
+    S_WT_VOICE *pWTVoice;
+    const S_DLS_ARTICULATION *pDLSArt;
+
+    pWTVoice = &pVoiceMgr->wtVoices[voiceNum];
+    pDLSArt = &pSynth->pDLS->pDLSArticulations[pWTVoice->artIndex];
+
+    /* if still in attack phase, convert units to log */
+    /*lint -e{732} eg1Value is never negative */
+    /*lint -e{703} use shift for performance */
+    if (pWTVoice->eg1State == eEnvelopeStateAttack)
+        pWTVoice->eg1Value = (EAS_I16) ((EAS_flog2(pWTVoice->eg1Value) << 1) + 2048);
+
+    /* release EG1 */
+    pWTVoice->eg1State = eEnvelopeStateRelease;
+    pWTVoice->eg1Increment = pDLSArt->eg1.releaseTime;
+
+    /* release EG2 */
+    pWTVoice->eg2State = eEnvelopeStateRelease;
+    pWTVoice->eg2Increment = pDLSArt->eg2.releaseTime;
+}
+
+/*----------------------------------------------------------------------------
+ * DLS_SustainPedal()
+ *----------------------------------------------------------------------------
+ * The sustain pedal was just depressed. If the voice is still
+ * above the sustain level, catch the voice and continue holding.
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pChannel) pChannel reserved for future use */
+void DLS_SustainPedal (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, S_SYNTH_CHANNEL *pChannel, EAS_I32 voiceNum)
+{
+    S_WT_VOICE *pWTVoice;
+    const S_DLS_ARTICULATION *pDLSArt;
+
+    pWTVoice = &pVoiceMgr->wtVoices[voiceNum];
+    pDLSArt = &pSynth->pDLS->pDLSArticulations[pWTVoice->artIndex];
+
+    /* don't catch the voice if below the sustain level */
+    if (pWTVoice->eg1Value < pDLSArt->eg1.sustainLevel)
+        return;
+
+    /* defer releasing this note until the damper pedal is off */
+    pWTVoice->eg1State = eEnvelopeStateDecay;
+    pVoice->voiceState = eVoiceStatePlay;
+    pVoice->voiceFlags |= VOICE_FLAG_SUSTAIN_PEDAL_DEFER_NOTE_OFF;
+
+#ifdef _DEBUG_SYNTH
+    { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "DLS_SustainPedal: defer note off because sustain pedal is on\n"); */ }
+#endif
+}
+
+/*----------------------------------------------------------------------------
+ * DLS_UpdatePhaseInc()
+ *----------------------------------------------------------------------------
+ * Calculate the oscillator phase increment for the next frame
+ *----------------------------------------------------------------------------
+*/
+static EAS_I32 DLS_UpdatePhaseInc (S_WT_VOICE *pWTVoice, const S_DLS_ARTICULATION *pDLSArt, S_SYNTH_CHANNEL *pChannel, EAS_I32 pitchCents)
+{
+    EAS_I32 temp;
+
+    /* start with base mod LFO modulation */
+    temp = pDLSArt->modLFOToPitch;
+
+    /* add mod wheel effect */
+    /*lint -e{702} use shift for performance */
+    temp += ((pDLSArt->modLFOCC1ToPitch * pChannel->modWheel) >> 7);
+
+    /* add channel pressure effect */
+    /*lint -e{702} use shift for performance */
+    temp += ((pDLSArt->modLFOChanPressToPitch * pChannel->channelPressure) >> 7);
+
+    /* add total mod LFO effect */
+    pitchCents += FMUL_15x15(temp, pWTVoice->modLFO.lfoValue);
+
+    /* start with base vib LFO modulation */
+    temp = pDLSArt->vibLFOToPitch;
+
+    /* add mod wheel effect */
+    /*lint -e{702} use shift for performance */
+    temp += ((pDLSArt->vibLFOCC1ToPitch * pChannel->modWheel) >> 7);
+
+    /* add channel pressure effect */
+    /*lint -e{702} use shift for performance */
+    temp += ((pDLSArt->vibLFOChanPressToPitch * pChannel->channelPressure) >> 7);
+
+    /* add total vibrato LFO effect */
+    pitchCents += FMUL_15x15(temp, pWTVoice->vibLFO.lfoValue);
+
+    /* add EG2 effect */
+    pitchCents += FMUL_15x15(pDLSArt->eg2ToPitch, pWTVoice->eg2Value);
+
+    /* convert from cents to linear phase increment */
+    return EAS_Calculate2toX(pitchCents);
+}
+
+/*----------------------------------------------------------------------------
+ * DLS_UpdateGain()
+ *----------------------------------------------------------------------------
+ * Calculate the gain for the next frame
+ *----------------------------------------------------------------------------
+*/
+static EAS_I32 DLS_UpdateGain (S_WT_VOICE *pWTVoice, const S_DLS_ARTICULATION *pDLSArt, S_SYNTH_CHANNEL *pChannel, EAS_I32 gain, EAS_U8 velocity)
+{
+    EAS_I32 temp;
+
+    /* start with base mod LFO modulation */
+    temp = pDLSArt->modLFOToGain;
+
+    /* add mod wheel effect */
+    /*lint -e{702} use shift for performance */
+    temp += ((pDLSArt->modLFOCC1ToGain * pChannel->modWheel) >> 7);
+
+    /* add channel pressure effect */
+    /*lint -e{702} use shift for performance */
+    temp += ((pDLSArt->modLFOChanPressToGain * pChannel->channelPressure) >> 7);
+
+    /* add total mod LFO effect */
+    gain += FMUL_15x15(temp, pWTVoice->modLFO.lfoValue);
+    if (gain > 0)
+        gain = 0;
+
+    /* convert to linear gain including EG1 */
+    if (pWTVoice->eg1State != eEnvelopeStateAttack)
+    {
+        gain = (DLS_GAIN_FACTOR * gain) >> DLS_GAIN_SHIFT;
+        /*lint -e{702} use shift for performance */
+#if 1
+        gain += (pWTVoice->eg1Value - 32767) >> 1;
+        gain = EAS_LogToLinear16(gain);
+#else
+        gain = EAS_LogToLinear16(gain);
+        temp = EAS_LogToLinear16((pWTVoice->eg1Value - 32767) >> 1);
+        gain = FMUL_15x15(gain, temp);
+#endif
+    }
+    else
+    {
+        gain = (DLS_GAIN_FACTOR * gain) >> DLS_GAIN_SHIFT;
+        gain = EAS_LogToLinear16(gain);
+        gain = FMUL_15x15(gain, pWTVoice->eg1Value);
+    }
+
+    /* include MIDI channel gain */
+    gain = FMUL_15x15(gain, pChannel->staticGain);
+
+    /* include velocity */
+    if (pDLSArt->filterQandFlags & FLAG_DLS_VELOCITY_SENSITIVE)
+    {
+        temp = velocity << 8;
+        temp = FMUL_15x15(temp, temp);
+        gain = FMUL_15x15(gain, temp);
+    }
+
+    /* return gain */
+    return gain;
+}
+
+/*----------------------------------------------------------------------------
+ * DLS_UpdateFilter()
+ *----------------------------------------------------------------------------
+ * Update the Filter parameters
+ *----------------------------------------------------------------------------
+*/
+static void DLS_UpdateFilter (S_SYNTH_VOICE *pVoice, S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pIntFrame, S_SYNTH_CHANNEL *pChannel, const S_DLS_ARTICULATION *pDLSArt)
+{
+    EAS_I32 cutoff;
+    EAS_I32 temp;
+
+    /* no need to calculate filter coefficients if it is bypassed */
+    if (pDLSArt->filterCutoff == DEFAULT_DLS_FILTER_CUTOFF_FREQUENCY)
+    {
+        pIntFrame->frame.k = 0;
+        return;
+    }
+
+    /* start with base cutoff frequency */
+    cutoff = pDLSArt->filterCutoff;
+
+    /* get base mod LFO modulation */
+    temp = pDLSArt->modLFOToFc;
+
+    /* add mod wheel effect */
+    /*lint -e{702} use shift for performance */
+    temp += ((pDLSArt->modLFOCC1ToFc * pChannel->modWheel) >> 7);
+
+    /* add channel pressure effect */
+    /*lint -e{702} use shift for performance */
+    temp += ((pDLSArt->modLFOChanPressToFc* pChannel->channelPressure) >> 7);
+
+    /* add total mod LFO effect */
+    cutoff += FMUL_15x15(temp, pWTVoice->modLFO.lfoValue);
+
+    /* add EG2 effect */
+    cutoff += FMUL_15x15(pWTVoice->eg2Value, pDLSArt->eg2ToFc);
+
+    /* add velocity effect */
+    /*lint -e{702} use shift for performance */
+    cutoff += (pVoice->velocity * pDLSArt->velToFc) >> 7;
+
+    /* add velocity effect */
+    /*lint -e{702} use shift for performance */
+    cutoff += (pVoice->note * pDLSArt->keyNumToFc) >> 7;
+
+    /* subtract the A5 offset and the sampling frequency */
+    cutoff -= FILTER_CUTOFF_FREQ_ADJUST + A5_PITCH_OFFSET_IN_CENTS;
+
+    /* limit the cutoff frequency */
+    if (cutoff > FILTER_CUTOFF_MAX_PITCH_CENTS)
+        cutoff = FILTER_CUTOFF_MAX_PITCH_CENTS;
+    else if (cutoff < FILTER_CUTOFF_MIN_PITCH_CENTS)
+        cutoff = FILTER_CUTOFF_MIN_PITCH_CENTS;
+
+    WT_SetFilterCoeffs(pIntFrame, cutoff, pDLSArt->filterQandFlags & FILTER_Q_MASK);
+}
+
+/*----------------------------------------------------------------------------
+ * DLS_StartVoice()
+ *----------------------------------------------------------------------------
+ * Start up a DLS voice
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT DLS_StartVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum, EAS_U16 regionIndex)
+{
+    S_WT_VOICE *pWTVoice;
+    const S_DLS_REGION *pDLSRegion;
+    const S_DLS_ARTICULATION *pDLSArt;
+    S_SYNTH_CHANNEL *pChannel;
+
+#ifdef _DEBUG_SYNTH
+    { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "DLS_StartVoice: Voice %ld; Region %d\n", (EAS_I32) (pVoice - pVoiceMgr->voices), regionIndex); */ }
+#endif
+
+    pWTVoice = &pVoiceMgr->wtVoices[voiceNum];
+    pChannel = &pSynth->channels[pVoice->channel & 15];
+    pDLSRegion = &pSynth->pDLS->pDLSRegions[regionIndex & REGION_INDEX_MASK];
+    pWTVoice->artIndex = pDLSRegion->wtRegion.artIndex;
+    pDLSArt = &pSynth->pDLS->pDLSArticulations[pWTVoice->artIndex];
+
+    /* initialize the envelopes */
+    pWTVoice->eg1State = eEnvelopeStateInit;
+    DLS_UpdateEnvelope(pVoice, pChannel, &pDLSArt->eg1, &pWTVoice->eg1Value, &pWTVoice->eg1Increment, &pWTVoice->eg1State);
+    pWTVoice->eg2State = eEnvelopeStateInit;
+    DLS_UpdateEnvelope(pVoice, pChannel, &pDLSArt->eg2, &pWTVoice->eg2Value, &pWTVoice->eg2Increment, &pWTVoice->eg2State);
+
+    /* initialize the LFOs */
+    pWTVoice->modLFO.lfoValue = 0;
+    pWTVoice->modLFO.lfoPhase = pDLSArt->modLFO.lfoDelay;
+    pWTVoice->vibLFO.lfoValue = 0;
+    pWTVoice->vibLFO.lfoPhase = pDLSArt->vibLFO.lfoDelay;
+
+    /* initalize the envelopes and calculate initial gain */
+    DLS_UpdateEnvelope(pVoice, pChannel, &pDLSArt->eg1, &pWTVoice->eg1Value, &pWTVoice->eg1Increment, &pWTVoice->eg1State);
+    DLS_UpdateEnvelope(pVoice, pChannel, &pDLSArt->eg2, &pWTVoice->eg2Value, &pWTVoice->eg2Increment, &pWTVoice->eg2State);
+    pVoice->gain = (EAS_I16) DLS_UpdateGain(pWTVoice, pDLSArt, pChannel, pDLSRegion->wtRegion.gain, pVoice->velocity);
+
+#if (NUM_OUTPUT_CHANNELS == 2)
+    EAS_CalcPanControl((EAS_INT) pChannel->pan - 64 + (EAS_INT) pDLSArt->pan, &pWTVoice->gainLeft, &pWTVoice->gainRight);
+#endif
+
+    /* initialize the filter states */
+    pWTVoice->filter.z1 = 0;
+    pWTVoice->filter.z2 = 0;
+
+    /* initialize the oscillator */
+    pWTVoice->phaseAccum = (EAS_U32) pSynth->pDLS->pDLSSamples + pSynth->pDLS->pDLSSampleOffsets[pDLSRegion->wtRegion.waveIndex];
+    if (pDLSRegion->wtRegion.region.keyGroupAndFlags & REGION_FLAG_IS_LOOPED)
+    {
+        pWTVoice->loopStart = pWTVoice->phaseAccum + pDLSRegion->wtRegion.loopStart;
+        pWTVoice->loopEnd = pWTVoice->phaseAccum + pDLSRegion->wtRegion.loopEnd - 1;
+    }
+    else
+        pWTVoice->loopStart = pWTVoice->loopEnd = pWTVoice->phaseAccum + pSynth->pDLS->pDLSSampleLen[pDLSRegion->wtRegion.waveIndex] - 1;
+
+    return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * DLS_UpdateVoice()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Synthesize a block of samples for the given voice.
+ * Use linear interpolation.
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ * number of samples actually written to buffer
+ *
+ * Side Effects:
+ * - samples are added to the presently free buffer
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_BOOL DLS_UpdateVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum, EAS_I32 *pMixBuffer, EAS_I32 numSamples)
+{
+    S_WT_VOICE *pWTVoice;
+    S_SYNTH_CHANNEL *pChannel;
+    const S_DLS_REGION *pDLSRegion;
+    const S_DLS_ARTICULATION *pDLSArt;
+    S_WT_INT_FRAME intFrame;
+    EAS_I32 temp;
+    EAS_BOOL done = EAS_FALSE;
+
+    /* establish pointers to critical data */
+    pWTVoice = &pVoiceMgr->wtVoices[voiceNum];
+    pDLSRegion = &pSynth->pDLS->pDLSRegions[pVoice->regionIndex & REGION_INDEX_MASK];
+    pChannel = &pSynth->channels[pVoice->channel & 15];
+    pDLSArt = &pSynth->pDLS->pDLSArticulations[pWTVoice->artIndex];
+
+    /* update the envelopes */
+    DLS_UpdateEnvelope(pVoice, pChannel, &pDLSArt->eg1, &pWTVoice->eg1Value, &pWTVoice->eg1Increment, &pWTVoice->eg1State);
+    DLS_UpdateEnvelope(pVoice, pChannel, &pDLSArt->eg2, &pWTVoice->eg2Value, &pWTVoice->eg2Increment, &pWTVoice->eg2State);
+
+    /* update the LFOs using the EAS synth function */
+    WT_UpdateLFO(&pWTVoice->modLFO, pDLSArt->modLFO.lfoFreq);
+    WT_UpdateLFO(&pWTVoice->vibLFO, pDLSArt->vibLFO.lfoFreq);
+
+    /* calculate base frequency */
+    temp = pDLSArt->tuning + pChannel->staticPitch + pDLSRegion->wtRegion.tuning +
+        (((EAS_I32) pVoice->note * (EAS_I32) pDLSArt->keyNumToPitch) >> 7);
+
+    /* don't transpose rhythm channel */
+    if ((pChannel ->channelFlags & CHANNEL_FLAG_RHYTHM_CHANNEL) == 0)
+        temp += pSynth->globalTranspose * 100;
+
+    /* calculate phase increment including modulation effects */
+    intFrame.frame.phaseIncrement = DLS_UpdatePhaseInc(pWTVoice, pDLSArt, pChannel, temp);
+
+    /* calculate gain including modulation effects */
+    intFrame.frame.gainTarget = DLS_UpdateGain(pWTVoice, pDLSArt, pChannel, pDLSRegion->wtRegion.gain, pVoice->velocity);
+    intFrame.prevGain = pVoice->gain;
+
+    DLS_UpdateFilter(pVoice, pWTVoice, &intFrame, pChannel, pDLSArt);
+
+    /* call into engine to generate samples */
+    intFrame.pAudioBuffer = pVoiceMgr->voiceBuffer;
+    intFrame.pMixBuffer = pMixBuffer;
+    intFrame.numSamples = numSamples;
+    if (numSamples < 0)
+        return EAS_FALSE;
+
+    /* check for end of sample */
+    if ((pWTVoice->loopStart != WT_NOISE_GENERATOR) && (pWTVoice->loopStart == pWTVoice->loopEnd))
+        done = WT_CheckSampleEnd(pWTVoice, &intFrame, EAS_FALSE);
+
+    WT_ProcessVoice(pWTVoice, &intFrame);
+
+    /* clear flag */
+    pVoice->voiceFlags &= ~VOICE_FLAG_NO_SAMPLES_SYNTHESIZED_YET;
+
+    /* if the update interval has elapsed, then force the current gain to the next
+     * gain since we never actually reach the next gain when ramping -- we just get
+     * very close to the target gain.
+     */
+    pVoice->gain = (EAS_I16) intFrame.frame.gainTarget;
+
+    /* if voice has finished, set flag for voice manager */
+    if ((pVoice->voiceState != eVoiceStateStolen) && (pWTVoice->eg1State == eEnvelopeStateMuted))
+        done = EAS_TRUE;
+
+    return done;
+}
+
+/*----------------------------------------------------------------------------
+ * DLS_UpdateEnvelope()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Synthesize a block of samples for the given voice.
+ * Use linear interpolation.
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ * number of samples actually written to buffer
+ *
+ * Side Effects:
+ * - samples are added to the presently free buffer
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pChannel) pChannel not used in this instance */
+static void DLS_UpdateEnvelope (S_SYNTH_VOICE *pVoice, S_SYNTH_CHANNEL *pChannel,  const S_DLS_ENVELOPE *pEnvParams, EAS_I16 *pValue, EAS_I16 *pIncrement, EAS_U8 *pState)
+{
+    EAS_I32 temp;
+
+    switch (*pState)
+    {
+        /* initial state */
+        case eEnvelopeStateInit:
+            *pState = eEnvelopeStateDelay;
+            *pValue = 0;
+            *pIncrement = pEnvParams->delayTime;
+            if (*pIncrement != 0)
+                return;
+            /*lint -e{825} falls through to next case */
+
+        case eEnvelopeStateDelay:
+            if (*pIncrement)
+            {
+                *pIncrement = *pIncrement - 1;
+                return;
+            }
+
+            /* calculate attack rate */
+            *pState = eEnvelopeStateAttack;
+            if (pEnvParams->attackTime != ZERO_TIME_IN_CENTS)
+            {
+                /*lint -e{702} use shift for performance */
+                temp = pEnvParams->attackTime + ((pEnvParams->velToAttack * pVoice->velocity) >> 7);
+                *pIncrement = ConvertRate(temp);
+                return;
+            }
+
+            *pValue = SYNTH_FULL_SCALE_EG1_GAIN;
+            /*lint -e{825} falls through to next case */
+
+        case eEnvelopeStateAttack:
+            if (*pValue < SYNTH_FULL_SCALE_EG1_GAIN)
+            {
+                temp = *pValue + *pIncrement;
+                *pValue = (EAS_I16) (temp < SYNTH_FULL_SCALE_EG1_GAIN ? temp : SYNTH_FULL_SCALE_EG1_GAIN);
+                return;
+            }
+
+            /* calculate hold time */
+            *pState = eEnvelopeStateHold;
+            if (pEnvParams->holdTime != ZERO_TIME_IN_CENTS)
+            {
+                /*lint -e{702} use shift for performance */
+                temp = pEnvParams->holdTime + ((pEnvParams->keyNumToHold * pVoice->note) >> 7);
+                *pIncrement = ConvertDelay(temp);
+                return;
+            }
+            else
+                *pIncrement = 0;
+            /*lint -e{825} falls through to next case */
+
+        case eEnvelopeStateHold:
+            if (*pIncrement)
+            {
+                *pIncrement = *pIncrement - 1;
+                return;
+            }
+
+            /* calculate decay rate */
+            *pState = eEnvelopeStateDecay;
+            if (pEnvParams->decayTime != ZERO_TIME_IN_CENTS)
+            {
+                /*lint -e{702} use shift for performance */
+                temp = pEnvParams->decayTime + ((pEnvParams->keyNumToDecay * pVoice->note) >> 7);
+                *pIncrement = ConvertRate(temp);
+                return;
+            }
+
+//          *pValue = pEnvParams->sustainLevel;
+            /*lint -e{825} falls through to next case */
+
+        case eEnvelopeStateDecay:
+            if (*pValue > pEnvParams->sustainLevel)
+            {
+                temp = *pValue - *pIncrement;
+                *pValue = (EAS_I16) (temp > pEnvParams->sustainLevel ? temp : pEnvParams->sustainLevel);
+                return;
+            }
+
+            *pState = eEnvelopeStateSustain;
+            *pValue = pEnvParams->sustainLevel;
+            /*lint -e{825} falls through to next case */
+
+        case eEnvelopeStateSustain:
+            return;
+
+        case eEnvelopeStateRelease:
+            temp = *pValue - *pIncrement;
+            if (temp <= 0)
+            {
+                *pState = eEnvelopeStateMuted;
+                *pValue = 0;
+            }
+            else
+                *pValue = (EAS_I16) temp;
+            break;
+
+        case eEnvelopeStateMuted:
+            *pValue = 0;
+            return;
+
+        default:
+            { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Envelope in invalid state %d\n", *pState); */ }
+            break;
+    }
+}
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_dlssynth.h

@@ -0,0 +1,41 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_dlssynth.h
+ *
+ * Contents and purpose:
+ * Implements the Mobile DLS synthesizer.
+ *
+ * Copyright Sonic Network Inc. 2006
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 143 $
+ *   $Date: 2006-07-17 14:09:35 -0700 (Mon, 17 Jul 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_DLSSYNTH_H
+#define _EAS_DLSSYNTH_H
+
+/* prototypes */
+void DLS_MuteVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum);
+void DLS_ReleaseVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum);
+void DLS_SustainPedal (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, S_SYNTH_CHANNEL *pChannel, EAS_I32 voiceNum);
+EAS_RESULT DLS_StartVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum, EAS_U16 regionIndex);
+EAS_BOOL DLS_UpdateVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum, EAS_I32 *pMixBuffer, EAS_I32  numSamples);
+
+#endif
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_effects.h

@@ -0,0 +1,61 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_effects.h
+ *
+ * Contents and purpose:
+ * Defines a generic effects interface.
+ *
+ * Copyright Sonic Network Inc. 2005
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 82 $
+ *   $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_EFFECTS_H
+#define _EAS_EFFECTS_H
+
+#include "eas_types.h"
+
+typedef struct
+{
+    EAS_RESULT  (*pfInit)(EAS_DATA_HANDLE pEASData, EAS_VOID_PTR *pInstData);
+    void        (*pfProcess)(EAS_VOID_PTR pInstData, EAS_PCM *in, EAS_PCM *out, EAS_I32 numSamples);
+    EAS_RESULT  (*pfShutdown)(EAS_DATA_HANDLE pEASData, EAS_VOID_PTR pInstData);
+    EAS_RESULT  (*pFGetParam)(EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 *pValue);
+    EAS_RESULT  (*pFSetParam)(EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 value);
+} S_EFFECTS_INTERFACE;
+
+typedef struct
+{
+    EAS_RESULT  (*pfInit)(EAS_DATA_HANDLE pEASData, EAS_VOID_PTR *pInstData);
+    void        (*pfProcess)(EAS_VOID_PTR pInstData, EAS_I32 *in, EAS_I32 *out, EAS_I32 numSamples);
+    EAS_RESULT  (*pfShutdown)(EAS_DATA_HANDLE pEASData, EAS_VOID_PTR pInstData);
+    EAS_RESULT  (*pFGetParam)(EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 *pValue);
+    EAS_RESULT  (*pFSetParam)(EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 value);
+} S_EFFECTS32_INTERFACE;
+
+/* mixer instance data */
+typedef struct
+{
+    S_EFFECTS_INTERFACE *effect;
+    EAS_VOID_PTR        effectData;
+} S_EFFECTS_MODULE;
+
+#endif /* end _EAS_EFFECTS_H */
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_flog.c

@@ -0,0 +1,96 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_flog2.c
+ *
+ * Contents and purpose:
+ * Fixed point square root
+ *
+ *
+ * Copyright (c) 2006 Sonic Network Inc.
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision$
+ *   $Date$
+ *----------------------------------------------------------------------------
+*/
+
+#include "eas_types.h"
+#include "eas_math.h"
+
+#define MANTISSA_SHIFT          27
+#define MANTISSA_MASK           0x0000000f
+#define MANTISSA_LSB_SHIFT      7
+#define MANTISSA_LSB_MASK       0x000fffff
+#define LOG_EXPONENT_SHIFT      10
+#define INTERPOLATION_SHIFT     20
+#define MAX_NEGATIVE            (-2147483647-1)
+
+/* log lookup table */
+static const EAS_U16 eas_log2_table[] =
+{
+    0, 90, 174, 254, 330, 402, 470, 536,
+    599, 659, 717, 773, 827, 879, 929, 977,
+    1024
+};
+
+/*----------------------------------------------------------------------------
+ * EAS_flog2()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Calculates the log2 of a 32-bit fixed point value
+ *
+ * Inputs:
+ * n = value of interest
+ *
+ * Outputs:
+ * returns the log2 of n
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_I32 EAS_flog2 (EAS_U32 n)
+{
+    EAS_U32 exp;
+    EAS_U32 interp;
+
+    /* check for error condition */
+    if (n == 0)
+        return MAX_NEGATIVE;
+
+    /* find exponent */
+    for (exp = 31; exp > 0; exp--)
+    {
+        /* shift until we get a 1 bit in bit 31 */
+        if ((n & (EAS_U32) MAX_NEGATIVE) != 0)
+            break;
+        n <<= 1;
+    }
+    /*lint -e{701} use shift for performance */
+    exp <<= LOG_EXPONENT_SHIFT;
+
+    /* get the least significant bits for interpolation */
+    interp = (n >> MANTISSA_LSB_SHIFT) & MANTISSA_LSB_MASK;
+
+    /* get the most significant bits for mantissa lookup */
+    n = (n >> MANTISSA_SHIFT) & MANTISSA_MASK;
+
+    /* interpolate mantissa */
+    interp = ((eas_log2_table[n+1] - eas_log2_table[n]) * interp) >> INTERPOLATION_SHIFT;
+    exp += eas_log2_table[n] + interp;
+
+    return (EAS_I32) exp;
+}
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_ima_tables.c

@@ -0,0 +1,54 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_ima_tables.c
+ *
+ * Contents and purpose:
+ * Contains the constant tables for IMA encode/decode
+ *
+ * Copyright (c) 2005 Sonic Network Inc.
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 760 $
+ *   $Date: 2007-07-17 23:09:36 -0700 (Tue, 17 Jul 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#include "eas_types.h"
+
+/*----------------------------------------------------------------------------
+ * ADPCM decode tables
+ *----------------------------------------------------------------------------
+*/
+const EAS_I16 imaIndexTable[16] =
+{
+    -1, -1, -1, -1, 2, 4, 6, 8,
+    -1, -1, -1, -1, 2, 4, 6, 8
+};
+
+const EAS_I16 imaStepSizeTable[89] =
+{
+    7, 8, 9, 10, 11, 12, 13, 14, 16, 17,
+    19, 21, 23, 25, 28, 31, 34, 37, 41, 45,
+    50, 55, 60, 66, 73, 80, 88, 97, 107, 118,
+    130, 143, 157, 173, 190, 209, 230, 253, 279, 307,
+    337, 371, 408, 449, 494, 544, 598, 658, 724, 796,
+    876, 963, 1060, 1166, 1282, 1411, 1552, 1707, 1878, 2066,
+    2272, 2499, 2749, 3024, 3327, 3660, 4026, 4428, 4871, 5358,
+    5894, 6484, 7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899,
+    15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794, 32767
+};
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_imaadpcm.c

@@ -0,0 +1,368 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_imaadpcm.c
+ *
+ * Contents and purpose:
+ * Implements the IMA ADPCM decoder
+ *
+ * Copyright Sonic Network Inc. 2005
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 847 $
+ *   $Date: 2007-08-27 21:30:08 -0700 (Mon, 27 Aug 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#include "eas_data.h"
+#include "eas_host.h"
+#include "eas_pcm.h"
+#include "eas_math.h"
+#include "eas_report.h"
+
+// #define _DEBUG_IMA_ADPCM_LOCATE
+
+/*----------------------------------------------------------------------------
+ * externs
+ *----------------------------------------------------------------------------
+*/
+extern const EAS_I16 imaIndexTable[];
+extern const EAS_I16 imaStepSizeTable[];
+
+/*----------------------------------------------------------------------------
+ * prototypes
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT IMADecoderInit (EAS_DATA_HANDLE pEASData, S_PCM_STATE *pState);
+static EAS_RESULT IMADecoderSample (EAS_DATA_HANDLE pEASData, S_PCM_STATE *pState);
+static void IMADecoderADPCM (S_DECODER_STATE *pState, EAS_U8 nibble);
+static EAS_RESULT IMADecoderLocate (EAS_DATA_HANDLE pEASData, S_PCM_STATE *pState, EAS_I32 time);
+
+/*----------------------------------------------------------------------------
+ * IMA ADPCM Decoder interface
+ *----------------------------------------------------------------------------
+*/
+const S_DECODER_INTERFACE IMADecoder =
+{
+    IMADecoderInit,
+    IMADecoderSample,
+    IMADecoderLocate
+};
+
+/*----------------------------------------------------------------------------
+ * IMADecoderInit()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Initializes the IMA ADPCM decoder
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pEASData) common decoder interface - pEASData not used */
+static EAS_RESULT IMADecoderInit (EAS_DATA_HANDLE pEASData, S_PCM_STATE *pState)
+{
+    pState->decoderL.step = 0;
+    pState->decoderR.step = 0;
+    return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * IMADecoderSample()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Decodes an IMA ADPCM sample
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT IMADecoderSample (EAS_DATA_HANDLE pEASData, S_PCM_STATE *pState)
+{
+    EAS_RESULT result;
+    EAS_I16 sTemp;
+
+    /* if high nibble, decode */
+    if (pState->hiNibble)
+    {
+        IMADecoderADPCM(&pState->decoderL, (EAS_U8)(pState->srcByte >> 4));
+        pState->hiNibble = EAS_FALSE;
+    }
+
+    /* low nibble, need to fetch another byte */
+    else
+    {
+        /* check for loop */
+        if ((pState->bytesLeft == 0) && (pState->loopSamples != 0))
+        {
+            /* seek to start of loop */
+            if ((result = EAS_HWFileSeek(pEASData->hwInstData, pState->fileHandle, (EAS_I32) (pState->startPos + pState->loopLocation))) != EAS_SUCCESS)
+                return result;
+            pState->bytesLeft = pState->byteCount = (EAS_I32) pState->bytesLeftLoop;
+            pState->blockCount = 0;
+            pState->flags &= ~PCM_FLAGS_EMPTY;
+            { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMADecoderSample: Rewind file to %d, bytesLeft = %d\n", pState->startPos, pState->bytesLeft); */ }
+        }
+
+        /* if start of block, fetch new predictor and step index */
+        if ((pState->blockSize != 0) && (pState->blockCount == 0) && (pState->bytesLeft != 0))
+        {
+
+            /* get predicted sample for left channel */
+            if ((result = EAS_HWGetWord(pEASData->hwInstData, pState->fileHandle, &sTemp, EAS_FALSE)) != EAS_SUCCESS)
+                return result;
+#ifdef _DEBUG_IMA_ADPCM
+            { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Predictor: Was %d, now %d\n", pState->decoderL.acc, sTemp); */ }
+#endif
+            pState->decoderL.acc = pState->decoderL.x1 = sTemp;
+
+            /* get step index for left channel - upper 8 bits are reserved */
+            if ((result = EAS_HWGetWord(pEASData->hwInstData, pState->fileHandle, &sTemp, EAS_FALSE)) != EAS_SUCCESS)
+                return result;
+#ifdef _DEBUG_IMA_ADPCM
+            { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Step: Was %d, now %d\n", pState->decoderL.step, sTemp); */ }
+#endif
+            pState->decoderL.step = sTemp & 0xff;
+
+            if (pState->flags & PCM_FLAGS_STEREO)
+            {
+                /* get predicted sample for right channel */
+                if ((result = EAS_HWGetWord(pEASData->hwInstData, pState->fileHandle, &sTemp, EAS_FALSE)) != EAS_SUCCESS)
+                    return result;
+                pState->decoderR.acc = pState->decoderR.x1 = sTemp;
+
+                /* get step index for right channel - upper 8 bits are reserved */
+                if ((result = EAS_HWGetWord(pEASData->hwInstData, pState->fileHandle, &sTemp, EAS_FALSE)) != EAS_SUCCESS)
+                    return result;
+#ifdef _DEBUG_IMA_ADPCM
+                { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Step: Was %d, now %d\n", pState->decoderR.step, sTemp); */ }
+#endif
+                pState->decoderR.step = sTemp & 0xff;
+
+                pState->blockCount = pState->blockSize - 8;
+                pState->bytesLeft -= 8;
+            }
+            else
+            {
+                pState->blockCount = pState->blockSize - 4;
+                pState->bytesLeft -= 4;
+            }
+        }
+        else
+        {
+
+            /* get another ADPCM data pair */
+            if (pState->bytesLeft)
+            {
+
+                if ((result = EAS_HWGetByte(pEASData->hwInstData, pState->fileHandle, &pState->srcByte)) != EAS_SUCCESS)
+                    return result;
+
+                /* decode the low nibble */
+                pState->bytesLeft--;
+                pState->blockCount--;
+                IMADecoderADPCM(&pState->decoderL, (EAS_U8)(pState->srcByte & 0x0f));
+
+                if (pState->flags & PCM_FLAGS_STEREO)
+                    IMADecoderADPCM(&pState->decoderR, (EAS_U8)(pState->srcByte >> 4));
+                else
+                    pState->hiNibble = EAS_TRUE;
+            }
+
+            /* out of ADPCM data, generate enough samples to fill buffer */
+            else
+            {
+                pState->decoderL.x1 = pState->decoderL.x0;
+                pState->decoderR.x1 = pState->decoderR.x0;
+            }
+        }
+    }
+
+    return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * IMADecoderADPCM()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Decodes an IMA ADPCM sample
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static void IMADecoderADPCM (S_DECODER_STATE *pState, EAS_U8 nibble)
+{
+    EAS_INT delta;
+    EAS_INT stepSize;
+
+    /* get stepsize from table */
+    stepSize = imaStepSizeTable[pState->step];
+
+    /* delta = (abs(delta) + 0.5) * step / 4 */
+    delta = 0;
+    if (nibble & 4)
+        delta += stepSize;
+
+    if (nibble & 2)
+        /*lint -e{702} use shift for performance */
+        delta += stepSize >> 1;
+
+    if (nibble & 1)
+        /*lint -e{702} use shift for performance */
+        delta += stepSize >> 2;
+
+    /*lint -e{702} use shift for performance */
+    delta += stepSize >> 3;
+
+    /* integrate the delta */
+    if (nibble & 8)
+      pState->acc -= delta;
+    else
+      pState->acc += delta;
+
+    /* saturate */
+    if (pState->acc > 32767)
+        pState->acc = 32767;
+    if (pState->acc < -32768)
+        pState->acc = -32768;
+    pState->x1 = (EAS_PCM) pState->acc;
+
+    /* compute new step size */
+    pState->step += imaIndexTable[nibble];
+    if (pState->step < 0)
+        pState->step = 0;
+    if (pState->step > 88)
+        pState->step = 88;
+
+#ifdef _DEBUG_IMA_ADPCM
+    { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "In=%u, Pred=%d, Step=%d\n", nibble, pState->acc,  imaStepSizeTable[pState->step]); */ }
+#endif
+}
+
+/*----------------------------------------------------------------------------
+ * IMADecoderLocate()
+ *----------------------------------------------------------------------------
+ * Locate in an IMA ADPCM stream
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT IMADecoderLocate (EAS_DATA_HANDLE pEASData, S_PCM_STATE *pState, EAS_I32 time)
+{
+    EAS_RESULT result;
+    EAS_I32 temp;
+    EAS_I32 samplesPerBlock;
+    EAS_I32 secs, msecs;
+
+    /* no need to calculate if time is zero */
+    if (time == 0)
+        temp = 0;
+
+    /* not zero */
+    else
+    {
+
+        /* can't seek if not a blocked file */
+        if (pState->blockSize == 0)
+            return EAS_ERROR_FEATURE_NOT_AVAILABLE;
+
+        /* calculate number of samples per block */
+        if (pState->flags & PCM_FLAGS_STEREO)
+            samplesPerBlock = pState->blockSize - 7;
+        else
+            samplesPerBlock = (pState->blockSize << 1) - 7;
+
+        /* break down into secs and msecs */
+        secs = time / 1000;
+        msecs = time - (secs * 1000);
+
+        /* calculate sample number fraction from msecs */
+        temp = (msecs * pState->sampleRate);
+        temp = (temp >> 10) + ((temp * 49) >> 21);
+
+        /* add integer sample count */
+        temp += secs * pState->sampleRate;
+
+#ifdef _DEBUG_IMA_ADPCM_LOCATE
+        EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x2380b977, 0x00000006 , time, temp);
+#endif
+
+        /* for looped samples, calculate position in the loop */
+        if ((temp > pState->byteCount) && (pState->loopSamples != 0))
+        {
+            EAS_I32 numBlocks;
+            EAS_I32 samplesPerLoop;
+            EAS_I32 samplesInLastBlock;
+
+            numBlocks = (EAS_I32) (pState->loopStart / pState->blockSize);
+            samplesInLastBlock = (EAS_I32) pState->loopStart - (numBlocks * pState->blockSize);
+            if (samplesInLastBlock)
+            {
+                if (pState->flags & PCM_FLAGS_STEREO)
+                    samplesInLastBlock = samplesInLastBlock - 7;
+                else
+                    /*lint -e{703} use shift for performance */
+                    samplesInLastBlock = (samplesInLastBlock << 1) - 7;
+            }
+            samplesPerLoop = numBlocks * samplesPerBlock + samplesInLastBlock;
+            temp = temp % samplesPerLoop;
+#ifdef _DEBUG_IMA_ADPCM_LOCATE
+            EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x2380b977, 0x00000007 , numBlocks, samplesPerLoop, samplesInLastBlock, temp);
+#endif
+        }
+
+        /* find start of block for requested sample */
+        temp = (temp / samplesPerBlock) * pState->blockSize;
+#ifdef _DEBUG_IMA_ADPCM_LOCATE
+        EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x2380b977, 0x00000008 , temp);
+#endif
+
+    }
+
+    /* seek to new location */
+    if ((result = EAS_PESeek(pEASData, pState, &temp)) != EAS_SUCCESS)
+        return result;
+
+#ifdef _DEBUG_IMA_ADPCM_LOCATE
+    EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x2380b977, 0x00000009 , pState->bytesLeft);
+#endif
+
+    /* reset state */
+    pState->blockCount = 0;
+    pState->hiNibble = EAS_FALSE;
+    if ((pState->state != EAS_STATE_PAUSING) && (pState->state != EAS_STATE_PAUSED))
+        pState->state = EAS_STATE_READY;
+
+    return EAS_SUCCESS;
+}
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_imelodydata.c

@@ -0,0 +1,43 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_imelodydata.c
+ *
+ * Contents and purpose:
+ * SMF File Parser
+ *
+ * This file contains data definitions for the SMF parser.
+ *
+ * Copyright Sonic Network Inc. 2005
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 547 $
+ *   $Date: 2007-01-31 16:30:17 -0800 (Wed, 31 Jan 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#include "eas_types.h"
+#include "eas_imelodydata.h"
+
+/*----------------------------------------------------------------------------
+ *
+ * eas_iMelodyData
+ *
+ * Static memory allocation for iMelody parser
+ *----------------------------------------------------------------------------
+*/
+S_IMELODY_DATA eas_iMelodyData;
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_imelodydata.h

@@ -0,0 +1,74 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_imelodydata.h
+ *
+ * Contents and purpose:
+ * SMF File Parser
+ *
+ * This file contains data declarations for the iMelody parser.
+ *
+ * Copyright Sonic Network Inc. 2005
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 778 $
+ *   $Date: 2007-07-23 16:45:17 -0700 (Mon, 23 Jul 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef EAS_IMELODYDATA_H
+#define EAS_IMELODYDATA_H
+
+#include "eas_data.h"
+
+/* maximum line size as specified in iMelody V1.2 spec */
+#define MAX_LINE_SIZE           75
+
+/*----------------------------------------------------------------------------
+ *
+ * S_IMELODY_DATA
+ *
+ * This structure contains the state data for the iMelody parser
+ *----------------------------------------------------------------------------
+*/
+
+typedef struct
+{
+    EAS_FILE_HANDLE fileHandle;                 /* file handle */
+    S_SYNTH         *pSynth;                    /* pointer to synth */
+    EAS_I32         fileOffset;                 /* offset to start of data */
+    EAS_I32         time;                       /* current time in 256ths of a msec */
+    EAS_I32         tickBase;                   /* basline length of 32nd note in 256th of a msec */
+    EAS_I32         tick;                       /* actual length of 32nd note in 256th of a msec */
+    EAS_I32         restTicks;                  /* ticks to rest after current note */
+    EAS_I32         startLine;                  /* file offset at start of line (for repeats) */
+    EAS_I32         repeatOffset;               /* file offset to start of repeat section */
+    EAS_I32         repeatTime;                 /* time at start of repeat section */
+    S_METADATA_CB   metadata;                   /* metadata callback */
+    EAS_I16         repeatCount;                /* repeat counter */
+    EAS_U8          state;                      /* current state EAS_STATE_XXXX */
+    EAS_U8          style;                      /* from STYLE */
+    EAS_U8          index;                      /* index into buffer */
+    EAS_U8          octave;                     /* octave prefix */
+    EAS_U8          volume;                     /* current volume */
+    EAS_U8          note;                       /* MIDI note number */
+    EAS_I8          noteModifier;               /* sharp or flat */
+    EAS_I8          buffer[MAX_LINE_SIZE+1];    /* buffer for ASCII data */
+} S_IMELODY_DATA;
+
+#endif
+
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_math.c

@@ -0,0 +1,168 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_math.c
+ *
+ * Contents and purpose:
+ * Contains common math routines for the various audio engines.
+ *
+ *
+ * Copyright Sonic Network Inc. 2005
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 586 $
+ *   $Date: 2007-03-08 20:33:04 -0800 (Thu, 08 Mar 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#include "eas.h"
+#include "eas_math.h"
+
+/* anything less than this converts to a fraction too small to represent in 32-bits */
+#define MIN_CENTS   -18000
+
+/*----------------------------------------------------------------------------
+ * EAS_Calculate2toX()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Calculate 2^x
+ *
+ * Inputs:
+ * nCents -     measured in cents
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ * nResult - int.frac result (where frac has NUM_DENTS_FRAC_BITS)
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_I32 EAS_Calculate2toX (EAS_I32 nCents)
+{
+    EAS_I32 nDents;
+    EAS_I32 nExponentInt, nExponentFrac;
+    EAS_I32 nTemp1, nTemp2;
+    EAS_I32 nResult;
+
+    /* check for minimum value */
+    if (nCents < MIN_CENTS)
+        return 0;
+
+    /* for the time being, convert cents to dents */
+    nDents = FMUL_15x15(nCents, CENTS_TO_DENTS);
+
+    nExponentInt = GET_DENTS_INT_PART(nDents);
+    nExponentFrac = GET_DENTS_FRAC_PART(nDents);
+
+    /*
+    implement 2^(fracPart) as a power series
+    */
+    nTemp1 = GN2_TO_X2 + MULT_DENTS_COEF(nExponentFrac, GN2_TO_X3);
+    nTemp2 = GN2_TO_X1 + MULT_DENTS_COEF(nExponentFrac, nTemp1);
+    nTemp1 = GN2_TO_X0 + MULT_DENTS_COEF(nExponentFrac, nTemp2);
+
+    /*
+    implement 2^(intPart) as
+    a left shift for intPart >= 0 or
+    a left shift for intPart <  0
+    */
+    if (nExponentInt >= 0)
+    {
+        /* left shift for positive exponents */
+        /*lint -e{703} <avoid multiply for performance>*/
+        nResult = nTemp1 << nExponentInt;
+    }
+    else
+    {
+        /* right shift for negative exponents */
+        nExponentInt = -nExponentInt;
+        nResult = nTemp1 >> nExponentInt;
+    }
+
+    return nResult;
+}
+
+/*----------------------------------------------------------------------------
+ * EAS_LogToLinear16()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Transform log value to linear gain multiplier using piece-wise linear
+ * approximation
+ *
+ * Inputs:
+ * nGain - log scale value in 20.10 format. Even though gain is normally
+ * stored in 6.10 (16-bit) format we use 32-bit numbers here to eliminate
+ * the need for saturation checking when combining gain values.
+ *
+ * Outputs:
+ * Returns a 16-bit linear value approximately equal to 2^(nGain/1024)
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_U16 EAS_LogToLinear16 (EAS_I32 nGain)
+{
+    EAS_INT nExp;
+    EAS_U16 nTemp;
+
+    /* bias to positive */
+    nGain += 32767;
+
+    /* check for infinite attenuation */
+    if (nGain < 0)
+        return 0;
+
+    /* extract the exponent */
+    nExp = 31 - (nGain >> 10);
+
+    /* check for maximum output */
+    if (nExp < 0)
+        return 0x7fff;
+
+    /* extract mantissa and restore implied 1 bit */
+    nTemp = (EAS_U16)((((nGain & 0x3ff) << 4) | 0x4000) >> nExp);
+
+    /* use shift to approximate power-of-2 operation */
+    return nTemp;
+}
+
+/*----------------------------------------------------------------------------
+ * EAS_VolumeToGain()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Transform volume control in 1dB increments to gain multiplier
+ *
+ * Inputs:
+ * volume - 100 = 0dB, 99 = -1dB, 0 = -inf
+ *
+ * Outputs:
+ * Returns a 16-bit linear value
+ *----------------------------------------------------------------------------
+*/
+EAS_I16 EAS_VolumeToGain (EAS_INT volume)
+{
+    /* check for limits */
+    if (volume <= 0)
+        return 0;
+    if (volume >= 100)
+        return 0x7fff;
+
+    /*lint -e{702} use shift instead of division */
+    return (EAS_I16) EAS_Calculate2toX((((volume - EAS_MAX_VOLUME) * 204099) >> 10) - 1);
+}
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_math.h

@@ -0,0 +1,412 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_math.h
+ *
+ * Contents and purpose:
+ * Contains common math routines for the various audio engines.
+ *
+ *
+ * Copyright Sonic Network Inc. 2005
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 584 $
+ *   $Date: 2007-03-08 09:49:24 -0800 (Thu, 08 Mar 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_MATH_H
+#define _EAS_MATH_H
+
+
+/** coefs for pan, generates sin, cos */
+#define COEFF_PAN_G2    -27146      /* -0.82842712474619 = 2 - 4/sqrt(2) */
+#define COEFF_PAN_G0    23170       /* 0.707106781186547 = 1/sqrt(2) */
+
+/*
+coefficients for approximating
+2^x = gn2toX0 + gn2toX1*x + gn2toX2*x^2 + gn2toX3*x^3
+where x is a int.frac number representing number of octaves.
+Actually, we approximate only the 2^(frac) using the power series
+and implement the 2^(int) as a shift, so that
+2^x == 2^(int.frac) == 2^(int) * 2^(fract)
+    == (gn2toX0 + gn2toX1*x + gn2toX2*x^2 + gn2toX3*x^3) << (int)
+
+The gn2toX.. were generated using a best fit for a 3rd
+order polynomial, instead of taking the coefficients from
+a truncated Taylor (or Maclaurin?) series.
+*/
+
+#define GN2_TO_X0   32768   /*  1                   */
+#define GN2_TO_X1   22833   /*  0.696807861328125   */
+#define GN2_TO_X2   7344    /*  0.22412109375       */
+#define GN2_TO_X3   2588    /*  0.0789794921875     */
+
+/*----------------------------------------------------------------------------
+ * Fixed Point Math
+ *----------------------------------------------------------------------------
+ * These macros are used for fixed point multiplies. If the processor
+ * supports fixed point multiplies, replace these macros with inline
+ * assembly code to improve performance.
+ *----------------------------------------------------------------------------
+*/
+
+/* Fixed point multiply 0.15 x 0.15 = 0.15 returned as 32-bits */
+#define FMUL_15x15(a,b) \
+    /*lint -e(704) <avoid multiply for performance>*/ \
+    (((EAS_I32)(a) * (EAS_I32)(b)) >> 15)
+
+/* Fixed point multiply 0.7 x 0.7 = 0.15 returned as 32-bits */
+#define FMUL_7x7(a,b) \
+    /*lint -e(704) <avoid multiply for performance>*/ \
+    (((EAS_I32)(a) * (EAS_I32)(b) ) << 1)
+
+/* Fixed point multiply 0.8 x 0.8 = 0.15 returned as 32-bits */
+#define FMUL_8x8(a,b) \
+    /*lint -e(704) <avoid multiply for performance>*/ \
+    (((EAS_I32)(a) * (EAS_I32)(b) ) >> 1)
+
+/* Fixed point multiply 0.8 x 1.15 = 0.15 returned as 32-bits */
+#define FMUL_8x15(a,b) \
+    /*lint -e(704) <avoid divide for performance>*/ \
+    (((EAS_I32)((a) << 7) * (EAS_I32)(b)) >> 15)
+
+/* macros for fractional phase accumulator */
+/*
+Note: changed the _U32 to _I32 on 03/14/02. This should not
+affect the phase calculations, and should allow us to reuse these
+macros for other audio sample related math.
+*/
+#define HARDWARE_BIT_WIDTH      32
+
+#define NUM_PHASE_INT_BITS      1
+#define NUM_PHASE_FRAC_BITS     15
+
+#define PHASE_FRAC_MASK         (EAS_U32) ((0x1L << NUM_PHASE_FRAC_BITS) -1)
+
+#define GET_PHASE_INT_PART(x)   (EAS_U32)((EAS_U32)(x) >> NUM_PHASE_FRAC_BITS)
+#define GET_PHASE_FRAC_PART(x)  (EAS_U32)((EAS_U32)(x) & PHASE_FRAC_MASK)
+
+#define DEFAULT_PHASE_FRAC      0
+#define DEFAULT_PHASE_INT       0
+
+/*
+Linear interpolation calculates:
+output = (1-frac) * sample[n] + (frac) * sample[n+1]
+
+where conceptually  0 <= frac < 1
+
+For a fixed point implementation, frac is actually an integer value
+with an implied binary point one position to the left. The value of
+one (unity) is given by PHASE_ONE
+one half and one quarter are useful for 4-point linear interp.
+*/
+#define PHASE_ONE               (EAS_I32) (0x1L << NUM_PHASE_FRAC_BITS)
+
+/*
+ Multiply the signed audio sample by the unsigned fraction.
+-  a is the signed audio sample
+-  b is the unsigned fraction (cast to signed int as long as coef
+    uses (n-1) or less bits, where n == hardware bit width)
+*/
+#define MULT_AUDIO_COEF(audio,coef)         /*lint -e704 <avoid divide for performance>*/ \
+            (EAS_I32)(                                  \
+            (                                           \
+                ((EAS_I32)(audio)) * ((EAS_I32)(coef))  \
+            )                                           \
+            >> NUM_PHASE_FRAC_BITS                      \
+                                        )               \
+                                        /* lint +704 <restore checking>*/
+
+/* wet / dry calculation macros */
+#define NUM_WET_DRY_FRAC_BITS       7   // 15
+#define NUM_WET_DRY_INT_BITS        9   // 1
+
+/* define a 1.0 */
+#define WET_DRY_ONE                 (EAS_I32) ((0x1L << NUM_WET_DRY_FRAC_BITS))
+#define WET_DRY_MINUS_ONE           (EAS_I32) (~WET_DRY_ONE)
+#define WET_DRY_FULL_SCALE          (EAS_I32) (WET_DRY_ONE - 1)
+
+#define MULT_AUDIO_WET_DRY_COEF(audio,coef) /*lint -e(702) <avoid divide for performance>*/ \
+            (EAS_I32)(                                      \
+            (                                               \
+                ((EAS_I32)(audio)) * ((EAS_I32)(coef))      \
+            )                                               \
+            >> NUM_WET_DRY_FRAC_BITS                        \
+                                                     )
+
+/* Envelope 1 (EG1) calculation macros */
+#define NUM_EG1_INT_BITS            1
+#define NUM_EG1_FRAC_BITS           15
+
+/* the max positive gain used in the synth for EG1 */
+/* SYNTH_FULL_SCALE_EG1_GAIN must match the value in the dls2eas
+converter, otherwise, the values we read from the .eas file are bogus. */
+#define SYNTH_FULL_SCALE_EG1_GAIN   (EAS_I32) ((0x1L << NUM_EG1_FRAC_BITS) -1)
+
+/* define a 1.0 */
+#define EG1_ONE                     (EAS_I32) ((0x1L << NUM_EG1_FRAC_BITS))
+#define EG1_MINUS_ONE               (EAS_I32) (~SYNTH_FULL_SCALE_EG1_GAIN)
+
+#define EG1_HALF                    (EAS_I32) (EG1_ONE/2)
+#define EG1_MINUS_HALF              (EAS_I32) (EG1_MINUS_ONE/2)
+
+/*
+We implement the EG1 using a linear gain value, which means that the
+attack segment is handled by incrementing (adding) the linear gain.
+However, EG1 treats the Decay, Sustain, and Release differently than
+the Attack portion. For Decay, Sustain, and Release, the gain is
+linear on dB scale, which is equivalent to exponential damping on
+a linear scale. Because we use a linear gain for EG1, we implement
+the Decay and Release as multiplication (instead of incrementing
+as we did for the attack segment).
+Therefore, we need the following macro to implement the multiplication
+(i.e., exponential damping) during the Decay and Release segments of
+the EG1
+*/
+#define MULT_EG1_EG1(gain,damping)      /*lint -e(704) <avoid divide for performance>*/ \
+            (EAS_I32)(                                      \
+            (                                               \
+                ((EAS_I32)(gain)) * ((EAS_I32)(damping))    \
+            )                                               \
+            >> NUM_EG1_FRAC_BITS                            \
+                                        )
+
+// Use the following macro specifically for the filter, when multiplying
+// the b1 coefficient. The 0 <= |b1| < 2, which therefore might overflow
+// in certain conditions because we store b1 as a 1.15 value.
+// Instead, we could store b1 as b1p (b1' == b1 "prime") where
+// b1p == b1/2, thus ensuring no potential overflow for b1p because
+// 0 <= |b1p| < 1
+// However, during the filter calculation, we must account for the fact
+// that we are using b1p instead of b1, and thereby multiply by
+// an extra factor of 2. Rather than multiply by an extra factor of 2,
+// we can instead shift the result right by one less, hence the
+// modified shift right value of (NUM_EG1_FRAC_BITS -1)
+#define MULT_EG1_EG1_X2(gain,damping)       /*lint -e(702) <avoid divide for performance>*/ \
+            (EAS_I32)(                                      \
+            (                                               \
+                ((EAS_I32)(gain)) * ((EAS_I32)(damping))    \
+            )                                               \
+            >> (NUM_EG1_FRAC_BITS -1)                       \
+                                        )
+
+#define SATURATE_EG1(x)     /*lint -e{734} saturation operation */              \
+    ((EAS_I32)(x) > SYNTH_FULL_SCALE_EG1_GAIN)  ? (SYNTH_FULL_SCALE_EG1_GAIN) : \
+    ((EAS_I32)(x) < EG1_MINUS_ONE)              ? (EG1_MINUS_ONE) : (x);
+
+
+/* use "digital cents" == "dents" instead of cents */
+/* we coudl re-use the phase frac macros, but if we do,
+we must change the phase macros to cast to _I32 instead of _U32,
+because using a _U32 cast causes problems when shifting the exponent
+for the 2^x calculation, because right shift a negative values MUST
+be sign extended, or else the 2^x calculation is wrong */
+
+/* use "digital cents" == "dents" instead of cents */
+#define NUM_DENTS_FRAC_BITS     12
+#define NUM_DENTS_INT_BITS      (HARDWARE_BIT_WIDTH - NUM_DENTS_FRAC_BITS)
+
+#define DENTS_FRAC_MASK             (EAS_I32) ((0x1L << NUM_DENTS_FRAC_BITS) -1)
+
+#define GET_DENTS_INT_PART(x)       /*lint -e(704) <avoid divide for performance>*/ \
+                            (EAS_I32)((EAS_I32)(x) >> NUM_DENTS_FRAC_BITS)
+
+#define GET_DENTS_FRAC_PART(x)  (EAS_I32)((EAS_I32)(x) & DENTS_FRAC_MASK)
+
+#define DENTS_ONE               (EAS_I32) (0x1L << NUM_DENTS_FRAC_BITS)
+
+/* use CENTS_TO_DENTS to convert a value in cents to dents */
+#define CENTS_TO_DENTS (EAS_I32) (DENTS_ONE * (0x1L << NUM_EG1_FRAC_BITS) / 1200L)                          \
+
+
+/*
+For gain, the LFO generates a value that modulates in terms
+of dB. However, we use a linear gain value, so we must convert
+the LFO value in dB to a linear gain. Normally, we would use
+linear gain = 10^x, where x = LFO value in dB / 20.
+Instead, we implement 10^x using our 2^x approximation.
+because
+
+  10^x = 2^(log2(10^x)) = 2^(x * log2(10))
+
+so we need to multiply by log2(10) which is just a constant.
+Ah, but just wait -- our 2^x actually doesn't exactly implement
+2^x, but it actually assumes that the input is in cents, and within
+the 2^x approximation converts its input from cents to octaves
+by dividing its input by 1200.
+
+So, in order to convert the LFO gain value in dB to something
+that our existing 2^x approximation can use, multiply the LFO gain
+by log2(10) * 1200 / 20
+
+The divide by 20 helps convert dB to linear gain, and we might
+as well incorporate that operation into this conversion.
+Of course, we need to keep some fractional bits, so multiply
+the constant by NUM_EG1_FRAC_BITS
+*/
+
+/* use LFO_GAIN_TO_CENTS to convert the LFO gain value to cents */
+#if 0
+#define DOUBLE_LOG2_10  (double) (3.32192809488736) /* log2(10) */
+
+#define DOUBLE_LFO_GAIN_TO_CENTS    (double)                \
+    (                                                       \
+                (DOUBLE_LOG2_10) *                          \
+                1200.0  /                                   \
+                20.0                                        \
+    )
+
+#define LFO_GAIN_TO_CENTS   (EAS_I32)                       \
+    (                                                       \
+                DOUBLE_LFO_GAIN_TO_CENTS *                  \
+                (0x1L << NUM_EG1_FRAC_BITS)                 \
+    )
+#endif
+
+#define LFO_GAIN_TO_CENTS (EAS_I32) (1671981156L >> (23 - NUM_EG1_FRAC_BITS))
+
+
+#define MULT_DENTS_COEF(dents,coef)     /*lint -e704 <avoid divide for performance>*/   \
+            (EAS_I32)(                                  \
+            (                                           \
+                ((EAS_I32)(dents)) * ((EAS_I32)(coef))  \
+            )                                           \
+            >> NUM_DENTS_FRAC_BITS                      \
+                                        )               \
+                                        /* lint +e704 <restore checking>*/
+
+/* we use 16-bits in the PC per audio sample */
+#define BITS_PER_AUDIO_SAMPLE   16
+
+/* we define 1 as 1.0 - 1 LSbit */
+#define DISTORTION_ONE          (EAS_I32)((0x1L << (BITS_PER_AUDIO_SAMPLE-1)) -1)
+#define DISTORTION_MINUS_ONE    (EAS_I32)(~DISTORTION_ONE)
+
+/* drive coef is given as int.frac */
+#define NUM_DRIVE_COEF_INT_BITS     1
+#define NUM_DRIVE_COEF_FRAC_BITS    4
+
+#define MULT_AUDIO_DRIVE(audio,drive)       /*lint -e(702) <avoid divide for performance>*/ \
+            (EAS_I32)   (                               \
+            (                                           \
+                ((EAS_I32)(audio)) * ((EAS_I32)(drive)) \
+            )                                           \
+            >> NUM_DRIVE_COEF_FRAC_BITS                 \
+                                                )
+
+#define MULT_AUDIO_AUDIO(audio1,audio2)         /*lint -e(702) <avoid divide for performance>*/ \
+            (EAS_I32)   (                                   \
+            (                                               \
+                ((EAS_I32)(audio1)) * ((EAS_I32)(audio2))   \
+            )                                               \
+            >> (BITS_PER_AUDIO_SAMPLE-1)                    \
+                                                    )
+
+#define SATURATE(x)                                                         \
+    ((((EAS_I32)(x)) > DISTORTION_ONE)      ? (DISTORTION_ONE) :            \
+    (((EAS_I32)(x)) < DISTORTION_MINUS_ONE) ? (DISTORTION_MINUS_ONE) :  ((EAS_I32)(x)));
+
+
+
+/*----------------------------------------------------------------------------
+ * EAS_Calculate2toX()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Calculate 2^x
+ *
+ * Inputs:
+ * nCents -     measured in cents
+ *
+ * Outputs:
+ * nResult - int.frac result (where frac has NUM_DENTS_FRAC_BITS)
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_I32 EAS_Calculate2toX (EAS_I32 nCents);
+
+/*----------------------------------------------------------------------------
+ * EAS_LogToLinear16()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Transform log value to linear gain multiplier using piece-wise linear
+ * approximation
+ *
+ * Inputs:
+ * nGain - log scale value in 20.10 format. Even though gain is normally
+ * stored in 6.10 (16-bit) format we use 32-bit numbers here to eliminate
+ * the need for saturation checking when combining gain values.
+ *
+ * Outputs:
+ * Returns a 16-bit linear value approximately equal to 2^(nGain/1024)
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_U16 EAS_LogToLinear16 (EAS_I32 nGain);
+
+/*----------------------------------------------------------------------------
+ * EAS_VolumeToGain()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Transform volume control in 1dB increments to gain multiplier
+ *
+ * Inputs:
+ * volume - 100 = 0dB, 99 = -1dB, 0 = -inf
+ *
+ * Outputs:
+ * Returns a 16-bit linear value
+ *----------------------------------------------------------------------------
+*/
+EAS_I16 EAS_VolumeToGain (EAS_INT volume);
+
+/*----------------------------------------------------------------------------
+ * EAS_fsqrt()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Calculates the square root of a 32-bit fixed point value
+ *
+ * Inputs:
+ * n = value of interest
+ *
+ * Outputs:
+ * returns the square root of n
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_U16 EAS_fsqrt (EAS_U32 n);
+
+/*----------------------------------------------------------------------------
+ * EAS_flog2()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Calculates the log2 of a 32-bit fixed point value
+ *
+ * Inputs:
+ * n = value of interest
+ *
+ * Outputs:
+ * returns the log2 of n
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_I32 EAS_flog2 (EAS_U32 n);
+
+#endif
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_mdls.h

@@ -0,0 +1,295 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_mdls.h
+ *
+ * Contents and purpose:
+ * Declarations, interfaces, and prototypes for eas_mdls.c
+ *
+ * Copyright Sonic Network Inc. 2004
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_MDLS_H
+#define _EAS_MDLS_H
+
+/*------------------------------------
+ * includes
+ *------------------------------------
+*/
+#include "eas_data.h"
+
+
+/*------------------------------------
+ * Some defines for dls.h
+ *------------------------------------
+*/
+#ifndef DWORD
+#define DWORD EAS_I32
+#define FAR
+#define SHORT EAS_I16
+#define USHORT EAS_U16
+#define LONG EAS_I32
+#define ULONG EAS_U32
+#endif
+
+
+/* GUID struct (call it DLSID in case GUID is defined elsewhere) */
+typedef struct
+{
+    EAS_U32 Data1;
+    EAS_U16 Data2;
+    EAS_U16 Data3;
+    EAS_U8  Data4[8];
+} DLSID;
+
+#define DEFINE_DLSID(name, l, w1, w2, b1, b2, b3, b4, b5, b6, b7, b8) const DLSID name = { l, w1, w2, { b1, b2, b3, b4, b5, b6, b7, b8 } }
+
+/*------------------------------------
+ * defines
+ *------------------------------------
+*/
+
+/* maximum sample memory for DLS query support */
+#ifndef MAX_DLS_MEMORY
+#define MAX_DLS_MEMORY 65536
+#endif
+
+/* size of conditional chunk stack */
+#ifndef CDL_STACK_SIZE
+#define CDL_STACK_SIZE 8
+#endif
+
+/* size of read buffer for sample conversion */
+#ifndef SAMPLE_CONVERT_CHUNK_SIZE
+#define SAMPLE_CONVERT_CHUNK_SIZE 32
+#endif
+
+
+#define ZERO_TIME_IN_CENTS              -32768
+
+/* Pan calculation macros */
+#define PAN_CONVERSION_FACTOR       4129
+#define MAX_PAN_VALUE               63
+#define MIN_PAN_VALUE               -63
+
+/* multiplier to convert time cents to 10-bit fraction log for EAS_LogToLinear16 */
+#define TIME_CENTS_TO_LOG2          27962
+
+/* conversion factor sustain level from percent to exponent for LogToLinear16 */
+#define SUSTAIN_LOG_CONVERSION_FACTOR   536871
+#define SUSTAIN_LOG_CONVERSION_SHIFT    15
+
+/* conversion factor sustain level from percent to EG full scale */
+#define SUSTAIN_LINEAR_CONVERSION_FACTOR    1073709
+
+/* conversion factor to convert frame period to decay rate */
+#define DECAY_CONVERSION_FACTOR     -16
+
+/*----------------------------------------------------------------------------
+ * These macros define the various characteristics of the defined sample rates
+ *----------------------------------------------------------------------------
+ * DLS_ATTACK_TIME_CONVERT      log offset for conversion from time cents to attack rate
+ * DLS_LFO_FREQUENCY_CONVERT    pitch-cents offset for LFO frequency conversion
+ *----------------------------------------------------------------------------
+*/
+
+#if defined (_SAMPLE_RATE_8000)
+#define DLS_RATE_CONVERT                -9559
+#define DLS_LFO_FREQUENCY_CONVERT       5921
+
+#elif defined (_SAMPLE_RATE_16000)
+#define DLS_RATE_CONVERT                -9559
+#define DLS_LFO_FREQUENCY_CONVERT       5921
+
+#elif defined (_SAMPLE_RATE_20000)
+#define DLS_RATE_CONVERT                -8745
+#define DLS_LFO_FREQUENCY_CONVERT       5108
+
+#elif defined (_SAMPLE_RATE_22050)
+#define DLS_RATE_CONVERT                -8914
+#define DLS_LFO_FREQUENCY_CONVERT       5277
+
+#elif defined (_SAMPLE_RATE_24000)
+#define DLS_RATE_CONVERT                -9061
+#define DLS_LFO_FREQUENCY_CONVERT       5423
+
+#elif defined (_SAMPLE_RATE_32000)
+#define DLS_RATE_CONVERT                -9559
+#define DLS_LFO_FREQUENCY_CONVERT       5921
+
+#elif defined (_SAMPLE_RATE_44100)
+#define DLS_RATE_CONVERT                -8914
+#define DLS_LFO_FREQUENCY_CONVERT       5277
+
+#elif defined (_SAMPLE_RATE_48000)
+#define DLS_RATE_CONVERT                -9061
+#define DLS_LFO_FREQUENCY_CONVERT       5423
+
+#else
+#error "_SAMPLE_RATE_XXXXX must be defined to valid rate"
+#endif
+
+/*
+ * FILTER_Q_CONVERSION_FACTOR convers the 0.1dB steps in the DLS
+ * file to our internal 0.75 dB steps. The value is calculated
+ * as follows:
+ *
+ * 32768 / (10 * <step-size in dB>)
+ *
+ * FILTER_RESONANCE_NUM_ENTRIES is the number of entries in the table
+*/
+#define FILTER_Q_CONVERSION_FACTOR          4369
+#define FILTER_RESONANCE_NUM_ENTRIES        31
+
+/*
+ * Multiplier to convert DLS gain units (10ths of a dB) to a
+ * power-of-two exponent for conversion to linear gain using our
+ * piece-wise linear approximator. Note that we ignore the lower
+ * 16-bits of the DLS gain value. The result is a 10-bit fraction
+ * that works with the EAS_LogToLinear16 function.
+ *
+ * DLS_GAIN_FACTOR = (2^18) / (200 * log10(2))
+ */
+#define DLS_GAIN_FACTOR         4354
+#define DLS_GAIN_SHIFT          8
+
+/*
+ * Reciprocal of 10 for quick divide by 10's
+ *
+ * DLS_GAIN_FACTOR = (2^18) / (200 * log10(2))
+ */
+#define DLS_DIV_10_FACTOR       3277
+#define DLS_DIV_10_SHIFT        16
+
+/*
+ * Multiplier to convert DLS time cents units to a power-of-two
+ * exponent for conversion to absolute time units using our
+ * piece-wise linear approximator.
+ *
+ * DLS_TIME_FACTOR = (2^22) / 1200
+ */
+#define DLS_TIME_FACTOR         3495
+#define DLS_TIME_SHIFT          22
+
+
+/* LFO limits */
+#define MAX_LFO_FREQUENCY_IN_HERTZ          20
+#define MIN_LFO_FREQUENCY_IN_HERTZ          0.1
+#define MAX_LFO_FREQUENCY_IN_PITCHCENTS     1549
+#define MIN_LFO_FREQUENCY_IN_PITCHCENTS     -7624
+#define MAX_LFO_AMPLITUDE_DEPTH             12  /* in dB, DLS2.1 p 31*/
+#define MIN_LFO_AMPLITUDE_DEPTH             -12 /* in dB, DLS2.1 p 31*/
+
+/* add to pitch cents before pow(2.0, n) to convert to frequency */
+#define ABSOLUTE_PITCH_BIAS                 238395828
+
+#define A5_PITCH_OFFSET                     6900
+
+/*
+CHUNK_TYPE is a macro that converts the 4 input args into a 32-bit int
+where
+argument a is placed at the MSB location and
+argument d is placed at the LSB location.
+This is useful for determining the DLS chunk types
+*/
+#define CHUNK_TYPE(a,b,c,d) (   \
+        ( ((EAS_U32)(a) & 0xFF) << 24 ) \
+    +   ( ((EAS_U32)(b) & 0xFF) << 16 ) \
+    +   ( ((EAS_U32)(c) & 0xFF) <<  8 ) \
+    +   ( ((EAS_U32)(d) & 0xFF)       ) )
+
+#define CHUNK_RIFF                  CHUNK_TYPE('R','I','F','F')
+#define CHUNK_DLS                   CHUNK_TYPE('D','L','S',' ')
+#define CHUNK_CDL                   CHUNK_TYPE('c','d','l',' ')
+#define CHUNK_VERS                  CHUNK_TYPE('v','e','r','s')
+#define CHUNK_DLID                  CHUNK_TYPE('d','l','i','d')
+#define CHUNK_LIST                  CHUNK_TYPE('L','I','S','T')
+#define CHUNK_COLH                  CHUNK_TYPE('c','o','l','h')
+#define CHUNK_LINS                  CHUNK_TYPE('l','i','n','s')
+#define CHUNK_PTBL                  CHUNK_TYPE('p','t','b','l')
+#define CHUNK_WVPL                  CHUNK_TYPE('w','v','p','l')
+#define CHUNK_INFO                  CHUNK_TYPE('I','N','F','O')
+#define CHUNK_INAM                  CHUNK_TYPE('I','N','A','M')
+#define CHUNK_INS                   CHUNK_TYPE('i','n','s',' ')
+#define CHUNK_INSH                  CHUNK_TYPE('i','n','s','h')
+#define CHUNK_LRGN                  CHUNK_TYPE('l','r','g','n')
+#define CHUNK_RGN                   CHUNK_TYPE('r','g','n',' ')
+#define CHUNK_RGN2                  CHUNK_TYPE('r','g','n','2')
+#define CHUNK_RGNH                  CHUNK_TYPE('r','g','n','h')
+#define CHUNK_WSMP                  CHUNK_TYPE('w','s','m','p')
+#define CHUNK_WLNK                  CHUNK_TYPE('w','l','n','k')
+#define CHUNK_LART                  CHUNK_TYPE('l','a','r','t')
+#define CHUNK_LAR2                  CHUNK_TYPE('l','a','r','2')
+#define CHUNK_ART1                  CHUNK_TYPE('a','r','t','1')
+#define CHUNK_ART2                  CHUNK_TYPE('a','r','t','2')
+#define CHUNK_WAVE                  CHUNK_TYPE('w','a','v','e')
+#define CHUNK_FMT                   CHUNK_TYPE('f','m','t',' ')
+#define CHUNK_DATA                  CHUNK_TYPE('d','a','t','a')
+#define CHUNK_DMPR                  CHUNK_TYPE('d','m','p','r')
+
+
+#define WAVE_FORMAT_PCM             0x0001 /* Microsoft PCM format, see DLS2.1 p60 */
+#define WAVE_FORMAT_EXTENSIBLE      0xffff
+
+/* defines for wave table structures */
+
+/* initialize each articulation structure to a harmless state */
+/* change art values after we've determined EAS internals */
+#define DEFAULT_DLS_FILTER_CUTOFF_FREQUENCY     0x7FFF      /* DLS2.1, p 31 means leave filter off */
+
+/**********/
+
+/* define the waves that we expect to generate instead of store */
+/* NOTE: our comparison routine converts the input string
+to lowercase, so the following comparison values should all
+be in lowercase.
+*/
+#define STRING_NOISE            "noise"
+
+
+/*------------------------------------
+ * type definitions
+ *------------------------------------
+*/
+#ifdef _STANDALONE_CONVERTER
+typedef struct s_dls_params
+{
+    EAS_INT     sampleRate;
+    EAS_INT     samplesPerFrame;
+    EAS_INT     bitDepth;
+    double      ditherLevel;
+    double      ditherFilterCoeff;
+    EAS_BOOL    compatibility;
+    EAS_BOOL    encodeADPCM;
+} S_DLS_PARAMS;
+#endif
+
+
+/* function prototypes */
+EAS_RESULT DLSParser (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE fileHandle, EAS_I32 offset, S_DLS **pDLS);
+EAS_RESULT DLSCleanup (EAS_HW_DATA_HANDLE hwInstData, S_DLS *pDLS);
+void DLSAddRef (S_DLS *pDLS);
+EAS_I16 ConvertDelay (EAS_I32 timeCents);
+EAS_I16 ConvertRate (EAS_I32 timeCents);
+
+
+#ifdef _STANDALONE_CONVERTER
+void DLSConvParams (S_DLS_PARAMS *pParams, EAS_BOOL set);
+#endif
+
+#endif
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_midi.c

@@ -0,0 +1,569 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_midi.c
+ *
+ * Contents and purpose:
+ * This file implements the MIDI stream parser. It is called by eas_smf.c to parse MIDI messages
+ * that are streamed out of the file. It can also parse live MIDI streams.
+ *
+ * Copyright Sonic Network Inc. 2005
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 794 $
+ *   $Date: 2007-08-01 00:08:48 -0700 (Wed, 01 Aug 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#include "eas_data.h"
+#include "eas_report.h"
+#include "eas_miditypes.h"
+#include "eas_midi.h"
+#include "eas_vm_protos.h"
+#include "eas_parser.h"
+
+#ifdef JET_INTERFACE
+#include "jet_data.h"
+#endif
+
+
+/* state enumerations for ProcessSysExMessage */
+typedef enum
+{
+    eSysEx,
+    eSysExUnivNonRealTime,
+    eSysExUnivNrtTargetID,
+    eSysExGMControl,
+    eSysExUnivRealTime,
+    eSysExUnivRtTargetID,
+    eSysExDeviceControl,
+    eSysExMasterVolume,
+    eSysExMasterVolLSB,
+    eSysExSPMIDI,
+    eSysExSPMIDIchan,
+    eSysExSPMIDIMIP,
+    eSysExMfgID1,
+    eSysExMfgID2,
+    eSysExMfgID3,
+    eSysExEnhancer,
+    eSysExEnhancerSubID,
+    eSysExEnhancerFeedback1,
+    eSysExEnhancerFeedback2,
+    eSysExEnhancerDrive,
+    eSysExEnhancerWet,
+    eSysExEOX,
+    eSysExIgnore
+} E_SYSEX_STATES;
+
+/* local prototypes */
+static EAS_RESULT ProcessMIDIMessage (S_EAS_DATA *pEASData, S_SYNTH *pSynth, S_MIDI_STREAM *pMIDIStream, EAS_INT parserMode);
+static EAS_RESULT ProcessSysExMessage (S_EAS_DATA *pEASData, S_SYNTH *pSynth, S_MIDI_STREAM *pMIDIStream, EAS_U8 c, EAS_INT parserMode);
+
+/*----------------------------------------------------------------------------
+ * EAS_InitMIDIStream()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Initializes the MIDI stream state for parsing.
+ *
+ * Inputs:
+ *
+ * Outputs:
+ * returns EAS_RESULT (EAS_SUCCESS is OK)
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+void EAS_InitMIDIStream (S_MIDI_STREAM *pMIDIStream)
+{
+    pMIDIStream->byte3 = EAS_FALSE;
+    pMIDIStream->pending = EAS_FALSE;
+    pMIDIStream->runningStatus = 0;
+    pMIDIStream->status = 0;
+}
+
+/*----------------------------------------------------------------------------
+ * EAS_ParseMIDIStream()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Parses a MIDI input stream character by character. Characters are pushed (rather than pulled)
+ * so the interface works equally well for both file and stream I/O.
+ *
+ * Inputs:
+ * c            - character from MIDI stream
+ *
+ * Outputs:
+ * returns EAS_RESULT (EAS_SUCCESS is OK)
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_ParseMIDIStream (S_EAS_DATA *pEASData, S_SYNTH *pSynth, S_MIDI_STREAM *pMIDIStream, EAS_U8 c, EAS_INT parserMode)
+{
+
+    /* check for new status byte */
+    if (c & 0x80)
+    {
+        /* save new running status */
+        if (c < 0xf8)
+        {
+            pMIDIStream->runningStatus = c;
+            pMIDIStream->byte3 = EAS_FALSE;
+
+            /* deal with SysEx */
+            if ((c == 0xf7) || (c == 0xf0))
+            {
+                if (parserMode == eParserModeMetaData)
+                    return EAS_SUCCESS;
+                return ProcessSysExMessage(pEASData, pSynth, pMIDIStream, c, parserMode);
+            }
+
+            /* inform the file parser that we're in the middle of a message */
+            if ((c < 0xf4) || (c > 0xf6))
+                pMIDIStream->pending = EAS_TRUE;
+        }
+
+        /* real-time message - ignore it */
+        return EAS_SUCCESS;
+    }
+
+    /* 3rd byte of a 3-byte message? */
+    if (pMIDIStream->byte3)
+    {
+        pMIDIStream->d2 = c;
+        pMIDIStream->byte3 = EAS_FALSE;
+        pMIDIStream->pending = EAS_FALSE;
+        if (parserMode == eParserModeMetaData)
+            return EAS_SUCCESS;
+        return ProcessMIDIMessage(pEASData, pSynth, pMIDIStream, parserMode);
+    }
+
+    /* check for status received */
+    if (pMIDIStream->runningStatus)
+    {
+
+        /* save new status and data byte */
+        pMIDIStream->status = pMIDIStream->runningStatus;
+
+        /* check for 3-byte messages */
+        if (pMIDIStream->status < 0xc0)
+        {
+            pMIDIStream->d1 = c;
+            pMIDIStream->pending = EAS_TRUE;
+            pMIDIStream->byte3 = EAS_TRUE;
+            return EAS_SUCCESS;
+        }
+
+        /* check for 2-byte messages */
+        if (pMIDIStream->status < 0xe0)
+        {
+            pMIDIStream->d1 = c;
+            pMIDIStream->pending = EAS_FALSE;
+            if (parserMode == eParserModeMetaData)
+                return EAS_SUCCESS;
+            return ProcessMIDIMessage(pEASData, pSynth, pMIDIStream, parserMode);
+        }
+
+        /* check for more 3-bytes message */
+        if (pMIDIStream->status < 0xf0)
+        {
+            pMIDIStream->d1 = c;
+            pMIDIStream->pending = EAS_TRUE;
+            pMIDIStream->byte3 = EAS_TRUE;
+            return EAS_SUCCESS;
+        }
+
+        /* SysEx message? */
+        if (pMIDIStream->status == 0xF0)
+        {
+            if (parserMode == eParserModeMetaData)
+                return EAS_SUCCESS;
+            return ProcessSysExMessage(pEASData, pSynth, pMIDIStream, c, parserMode);
+        }
+
+        /* remaining messages all clear running status */
+        pMIDIStream->runningStatus = 0;
+
+        /* F2 is 3-byte message */
+        if (pMIDIStream->status == 0xf2)
+        {
+            pMIDIStream->byte3 = EAS_TRUE;
+            return EAS_SUCCESS;
+        }
+    }
+
+    /* no status byte received, provide a warning, but we should be able to recover */
+    { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Received MIDI data without a valid status byte: %d\n",c); */ }
+    pMIDIStream->pending = EAS_FALSE;
+    return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * ProcessMIDIMessage()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * This function processes a typical MIDI message. All of the data has been received, just need
+ * to take appropriate action.
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT ProcessMIDIMessage (S_EAS_DATA *pEASData, S_SYNTH *pSynth, S_MIDI_STREAM *pMIDIStream, EAS_INT parserMode)
+{
+    EAS_U8 channel;
+
+    channel = pMIDIStream->status & 0x0f;
+    switch (pMIDIStream->status & 0xf0)
+    {
+    case 0x80:
+        { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL,"NoteOff: %02x %02x %02x\n",
+            pMIDIStream->status, pMIDIStream->d1, pMIDIStream->d2); */ }
+        if (parserMode <= eParserModeMute)
+            VMStopNote(pEASData->pVoiceMgr, pSynth, channel, pMIDIStream->d1, pMIDIStream->d2);
+        break;
+
+    case 0x90:
+        if (pMIDIStream->d2)
+        {
+            { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL,"NoteOn: %02x %02x %02x\n",
+                pMIDIStream->status, pMIDIStream->d1, pMIDIStream->d2); */ }
+            pMIDIStream->flags |= MIDI_FLAG_FIRST_NOTE;
+            if (parserMode == eParserModePlay)
+                VMStartNote(pEASData->pVoiceMgr, pSynth, channel, pMIDIStream->d1, pMIDIStream->d2);
+        }
+        else
+        {
+            { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL,"NoteOff: %02x %02x %02x\n",
+                pMIDIStream->status, pMIDIStream->d1, pMIDIStream->d2); */ }
+            if (parserMode <= eParserModeMute)
+                VMStopNote(pEASData->pVoiceMgr, pSynth, channel, pMIDIStream->d1, pMIDIStream->d2);
+        }
+        break;
+
+    case 0xa0:
+        { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL,"PolyPres: %02x %02x %02x\n",
+            pMIDIStream->status, pMIDIStream->d1, pMIDIStream->d2); */ }
+        break;
+
+    case 0xb0:
+        { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL,"Control: %02x %02x %02x\n",
+            pMIDIStream->status, pMIDIStream->d1, pMIDIStream->d2); */ }
+        if (parserMode <= eParserModeMute)
+            VMControlChange(pEASData->pVoiceMgr, pSynth, channel, pMIDIStream->d1, pMIDIStream->d2);
+#ifdef JET_INTERFACE
+        if (pMIDIStream->jetData & MIDI_FLAGS_JET_CB)
+        {
+            JET_Event(pEASData, pMIDIStream->jetData & (JET_EVENT_SEG_MASK | JET_EVENT_TRACK_MASK),
+                channel, pMIDIStream->d1, pMIDIStream->d2);
+        }
+#endif
+        break;
+
+    case 0xc0:
+        { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL,"Program: %02x %02x\n",
+            pMIDIStream->status, pMIDIStream->d1); */ }
+        if (parserMode <= eParserModeMute)
+            VMProgramChange(pEASData->pVoiceMgr, pSynth, channel, pMIDIStream->d1);
+        break;
+
+    case 0xd0:
+        { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL,"ChanPres: %02x %02x\n",
+            pMIDIStream->status, pMIDIStream->d1); */ }
+        if (parserMode <= eParserModeMute)
+            VMChannelPressure(pSynth, channel, pMIDIStream->d1);
+        break;
+
+    case 0xe0:
+        { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL,"PBend: %02x %02x %02x\n",
+            pMIDIStream->status, pMIDIStream->d1, pMIDIStream->d2); */ }
+        if (parserMode <= eParserModeMute)
+            VMPitchBend(pSynth, channel, pMIDIStream->d1, pMIDIStream->d2);
+        break;
+
+    default:
+        { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL,"Unknown: %02x %02x %02x\n",
+            pMIDIStream->status, pMIDIStream->d1, pMIDIStream->d2); */ }
+    }
+    return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * ProcessSysExMessage()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Process a SysEx character byte from the MIDI stream. Since we cannot
+ * simply wait for the next character to arrive, we are forced to save
+ * state after each character. It would be easier to parse at the file
+ * level, but then we lose the nice feature of being able to support
+ * these messages in a real-time MIDI stream.
+ *
+ * Inputs:
+ * pEASData         - pointer to synthesizer instance data
+ * c                - character to be processed
+ * locating         - if true, the sequencer is relocating to a new position
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ * Notes:
+ * These are the SysEx messages we can receive:
+ *
+ * SysEx messages
+ * { f0 7e 7f 09 01 f7 } GM 1 On
+ * { f0 7e 7f 09 02 f7 } GM 1/2 Off
+ * { f0 7e 7f 09 03 f7 } GM 2 On
+ * { f0 7f 7f 04 01 lsb msb } Master Volume
+ * { f0 7f 7f 0b 01 ch mip [ch mip ...] f7 } SP-MIDI
+ * { f0 00 01 3a 04 01 fdbk1 fdbk2 drive wet dry f7 } Enhancer
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT ProcessSysExMessage (S_EAS_DATA *pEASData, S_SYNTH *pSynth, S_MIDI_STREAM *pMIDIStream, EAS_U8 c, EAS_INT parserMode)
+{
+
+    /* check for start byte */
+    if (c == 0xf0)
+    {
+        pMIDIStream->sysExState = eSysEx;
+    }
+    /* check for end byte */
+    else if (c == 0xf7)
+    {
+        /* if this was a MIP message, update the MIP table */
+        if ((pMIDIStream->sysExState == eSysExSPMIDIchan) && (parserMode != eParserModeMetaData))
+            VMUpdateMIPTable(pEASData->pVoiceMgr, pSynth);
+        pMIDIStream->sysExState = eSysExIgnore;
+    }
+
+    /* process SysEx message */
+    else
+    {
+        switch (pMIDIStream->sysExState)
+        {
+        case eSysEx:
+
+            /* first byte, determine message class */
+            switch (c)
+            {
+            case 0x7e:
+                pMIDIStream->sysExState = eSysExUnivNonRealTime;
+                break;
+            case 0x7f:
+                pMIDIStream->sysExState = eSysExUnivRealTime;
+                break;
+            case 0x00:
+                pMIDIStream->sysExState = eSysExMfgID1;
+                break;
+            default:
+                pMIDIStream->sysExState = eSysExIgnore;
+                break;
+            }
+            break;
+
+        /* process GM message */
+        case eSysExUnivNonRealTime:
+            if (c == 0x7f)
+                pMIDIStream->sysExState = eSysExUnivNrtTargetID;
+            else
+                pMIDIStream->sysExState = eSysExIgnore;
+            break;
+
+        case eSysExUnivNrtTargetID:
+            if (c == 0x09)
+                pMIDIStream->sysExState = eSysExGMControl;
+            else
+                pMIDIStream->sysExState = eSysExIgnore;
+            break;
+
+        case eSysExGMControl:
+            if ((c == 1) || (c == 3))
+            {
+                /* GM 1 or GM2 On, reset synth */
+                if (parserMode != eParserModeMetaData)
+                {
+                    pMIDIStream->flags |= MIDI_FLAG_GM_ON;
+                    VMReset(pEASData->pVoiceMgr, pSynth, EAS_FALSE);
+                    VMInitMIPTable(pSynth);
+                }
+                pMIDIStream->sysExState = eSysExEOX;
+            }
+            else
+                pMIDIStream->sysExState = eSysExIgnore;
+            break;
+
+        /* Process Master Volume and SP-MIDI */
+        case eSysExUnivRealTime:
+            if (c == 0x7f)
+                pMIDIStream->sysExState = eSysExUnivRtTargetID;
+            else
+                pMIDIStream->sysExState = eSysExIgnore;
+            break;
+
+        case eSysExUnivRtTargetID:
+            if (c == 0x04)
+                pMIDIStream->sysExState = eSysExDeviceControl;
+            else if (c == 0x0b)
+                pMIDIStream->sysExState = eSysExSPMIDI;
+            else
+                pMIDIStream->sysExState = eSysExIgnore;
+            break;
+
+        /* process master volume */
+        case eSysExDeviceControl:
+            if (c == 0x01)
+                pMIDIStream->sysExState = eSysExMasterVolume;
+            else
+                pMIDIStream->sysExState = eSysExIgnore;
+            break;
+
+        case eSysExMasterVolume:
+            /* save LSB */
+            pMIDIStream->d1 = c;
+            pMIDIStream->sysExState = eSysExMasterVolLSB;
+            break;
+
+        case eSysExMasterVolLSB:
+            if (parserMode != eParserModeMetaData)
+            {
+                EAS_I32 gain = ((EAS_I32) c << 8) | ((EAS_I32) pMIDIStream->d1 << 1);
+                gain = (gain * gain) >> 15;
+                VMSetVolume(pSynth, (EAS_U16) gain);
+            }
+            pMIDIStream->sysExState = eSysExEOX;
+            break;
+
+        /* process SP-MIDI MIP message */
+        case eSysExSPMIDI:
+            if (c == 0x01)
+            {
+                /* assume all channels are muted */
+                if (parserMode != eParserModeMetaData)
+                    VMInitMIPTable(pSynth);
+                pMIDIStream->d1 = 0;
+                pMIDIStream->sysExState = eSysExSPMIDIchan;
+            }
+            else
+                pMIDIStream->sysExState = eSysExIgnore;
+            break;
+
+        case eSysExSPMIDIchan:
+            if (c < NUM_SYNTH_CHANNELS)
+            {
+                pMIDIStream->d2 = c;
+                pMIDIStream->sysExState = eSysExSPMIDIMIP;
+            }
+            else
+            {
+                /* bad MIP message - unmute channels */
+                if (parserMode != eParserModeMetaData)
+                    VMInitMIPTable(pSynth);
+                pMIDIStream->sysExState = eSysExIgnore;
+            }
+            break;
+
+        case eSysExSPMIDIMIP:
+            /* process MIP entry here */
+            if (parserMode != eParserModeMetaData)
+                VMSetMIPEntry(pEASData->pVoiceMgr, pSynth, pMIDIStream->d2, pMIDIStream->d1, c);
+            pMIDIStream->sysExState = eSysExSPMIDIchan;
+
+            /* if 16 channels received, update MIP table */
+            if (++pMIDIStream->d1 == NUM_SYNTH_CHANNELS)
+            {
+                if (parserMode != eParserModeMetaData)
+                    VMUpdateMIPTable(pEASData->pVoiceMgr, pSynth);
+                pMIDIStream->sysExState = eSysExEOX;
+            }
+            break;
+
+        /* process Enhancer */
+        case eSysExMfgID1:
+            if (c == 0x01)
+                pMIDIStream->sysExState = eSysExMfgID1;
+            else
+                pMIDIStream->sysExState = eSysExIgnore;
+            break;
+
+        case eSysExMfgID2:
+            if (c == 0x3a)
+                pMIDIStream->sysExState = eSysExMfgID1;
+            else
+                pMIDIStream->sysExState = eSysExIgnore;
+            break;
+
+        case eSysExMfgID3:
+            if (c == 0x04)
+                pMIDIStream->sysExState = eSysExEnhancer;
+            else
+                pMIDIStream->sysExState = eSysExIgnore;
+            break;
+
+        case eSysExEnhancer:
+            if (c == 0x01)
+                pMIDIStream->sysExState = eSysExEnhancerSubID;
+            else
+                pMIDIStream->sysExState = eSysExIgnore;
+            break;
+
+        case eSysExEnhancerSubID:
+            pMIDIStream->sysExState = eSysExEnhancerFeedback1;
+            break;
+
+        case eSysExEnhancerFeedback1:
+            pMIDIStream->sysExState = eSysExEnhancerFeedback2;
+            break;
+
+        case eSysExEnhancerFeedback2:
+            pMIDIStream->sysExState = eSysExEnhancerDrive;
+            break;
+
+        case eSysExEnhancerDrive:
+            pMIDIStream->sysExState = eSysExEnhancerWet;
+            break;
+
+        case eSysExEnhancerWet:
+            pMIDIStream->sysExState = eSysExEOX;
+            break;
+
+        case eSysExEOX:
+            { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Expected F7, received %02x\n", c); */ }
+            pMIDIStream->sysExState = eSysExIgnore;
+            break;
+
+        case eSysExIgnore:
+            break;
+
+        default:
+            pMIDIStream->sysExState = eSysExIgnore;
+            break;
+        }
+    }
+
+    if (pMIDIStream->sysExState == eSysExIgnore)
+        { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Ignoring SysEx byte %02x\n", c); */ }
+    return EAS_SUCCESS;
+} /* end ProcessSysExMessage */
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_midi.h

@@ -0,0 +1,71 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_midi.h
+ *
+ * Contents and purpose:
+ * Prototypes for MIDI stream parsing functions
+ *
+ *
+ * Copyright Sonic Network Inc. 2005
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 82 $
+ *   $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_MIDI_H
+#define _EAS_MIDI_H
+
+/*----------------------------------------------------------------------------
+ * EAS_InitMIDIStream()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Initializes the MIDI stream state for parsing.
+ *
+ * Inputs:
+ *
+ * Outputs:
+ * returns EAS_RESULT (EAS_SUCCESS is OK)
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+void EAS_InitMIDIStream (S_MIDI_STREAM *pMIDIStream);
+
+/*----------------------------------------------------------------------------
+ * EAS_ParseMIDIStream()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Parses a MIDI input stream character by character. Characters are pushed (rather than pulled)
+ * so the interface works equally well for both file and stream I/O.
+ *
+ * Inputs:
+ * c            - character from MIDI stream
+ *
+ * Outputs:
+ * returns EAS_RESULT (EAS_SUCCESS is OK)
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_ParseMIDIStream (S_EAS_DATA *pEASData, S_SYNTH *pSynth, S_MIDI_STREAM *pMIDIStream, EAS_U8 c, EAS_INT parserMode);
+
+#endif /* #define _EAS_MIDI_H */
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_midictrl.h

@@ -0,0 +1,64 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_midictrl.h
+ *
+ * Contents and purpose:
+ * MIDI controller definitions
+ *
+ * This header only contains declarations that are specific
+ * to this implementation.
+ *
+ * Copyright Sonic Network Inc. 2005
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 82 $
+ *   $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_MIDICTRL_H
+#define _EAS_MIDICTRL_H
+
+/* define controller types */
+/*
+    Note that these controller types are specified in base 10 (decimal)
+    and not in hexadecimal. The above midi messages are specified
+    in hexadecimal.
+*/
+#define MIDI_CONTROLLER_BANK_SELECT         0
+#define MIDI_CONTROLLER_BANK_SELECT_MSB     0
+#define MIDI_CONTROLLER_MOD_WHEEL           1
+#define MIDI_CONTROLLER_ENTER_DATA_MSB      6
+#define MIDI_CONTROLLER_VOLUME              7
+#define MIDI_CONTROLLER_PAN                 10
+#define MIDI_CONTROLLER_EXPRESSION          11
+#define MIDI_CONTROLLER_BANK_SELECT_LSB     32
+#define MIDI_CONTROLLER_ENTER_DATA_LSB      38      /* 0x26 */
+#define MIDI_CONTROLLER_SUSTAIN_PEDAL       64
+#define MIDI_CONTROLLER_SELECT_NRPN_LSB     98
+#define MIDI_CONTROLLER_SELECT_NRPN_MSB     99
+#define MIDI_CONTROLLER_SELECT_RPN_LSB      100     /* 0x64 */
+#define MIDI_CONTROLLER_SELECT_RPN_MSB      101     /* 0x65 */
+#define MIDI_CONTROLLER_ALL_SOUND_OFF       120
+#define MIDI_CONTROLLER_RESET_CONTROLLERS   121
+#define MIDI_CONTROLLER_ALL_NOTES_OFF       123
+#define MIDI_CONTROLLER_OMNI_OFF            124
+#define MIDI_CONTROLLER_OMNI_ON             125
+#define MIDI_CONTROLLER_MONO_ON_POLY_OFF    126
+#define MIDI_CONTROLLER_POLY_ON_MONO_OFF    127
+
+#endif /* #ifndef _EAS_MIDICTRL_H */

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_mididata.c

@@ -0,0 +1,34 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_mididata.c
+ *
+ * Contents and purpose:
+ * Data module for MIDI stream interface
+ *
+ *
+ * Copyright Sonic Network Inc. 2005
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 547 $
+ *   $Date: 2007-01-31 16:30:17 -0800 (Wed, 31 Jan 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#include "eas_miditypes.h"
+
+S_INTERACTIVE_MIDI eas_MIDIData;
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_miditypes.h

@@ -0,0 +1,138 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_miditypes.h
+ *
+ * Contents and purpose:
+ * Contains declarations for the MIDI stream parser.
+ *
+ *
+ * Copyright Sonic Network Inc. 2005
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 778 $
+ *   $Date: 2007-07-23 16:45:17 -0700 (Mon, 23 Jul 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_MIDITYPES_H
+#define _EAS_MIDITYPES_H
+
+#include "eas_data.h"
+#include "eas_parser.h"
+
+/*----------------------------------------------------------------------------
+ * S_MIDI_STREAM
+ *
+ * Maintains parser state for the MIDI stream parser
+ *
+ *----------------------------------------------------------------------------
+*/
+
+typedef struct s_midi_stream_tag
+{
+    EAS_BOOL8           byte3;              /* flag indicates 3rd byte expected */
+    EAS_BOOL8           pending;            /* flag indicates more data expected */
+    EAS_U8              sysExState;         /* maintains the SysEx state */
+    EAS_U8              runningStatus;      /* last running status received */
+    EAS_U8              status;             /* status byte */
+    EAS_U8              d1;                 /* first data byte */
+    EAS_U8              d2;                 /* second data byte */
+    EAS_U8              flags;              /* flags - see below for definition */
+#ifdef JET_INTERFACE
+    EAS_U32             jetData;            /* JET data */
+#endif
+} S_MIDI_STREAM;
+
+/* flags for S_MIDI_STREAM.flags */
+#define MIDI_FLAG_GM_ON         0x01        /* GM System On message received */
+#define MIDI_FLAG_FIRST_NOTE    0x02        /* first note received */
+
+/* flags for S_MIDI_STREAM.jetFlags */
+#define MIDI_FLAGS_JET_MUTE     0x00000001  /* track is muted */
+#define MIDI_FLAGS_JET_CB       0x00000002  /* JET callback enabled */
+
+/*----------------------------------------------------------------------------
+ *
+ * S_SMF_STREAM
+ *
+ * This structure contains data required to parse an SMF stream. For SMF0 files, there
+ * will be a single instance of this per file. For SMF1 files, there will be multiple instance,
+ * one for each separate stream in the file.
+ *
+ *----------------------------------------------------------------------------
+*/
+
+typedef struct s_smf_stream_tag
+{
+    EAS_FILE_HANDLE     fileHandle;         /* host wrapper file handle */
+    EAS_U32             ticks;              /* time of next event in stream */
+    EAS_I32             startFilePos;       /* start location of track within file */
+    S_MIDI_STREAM       midiStream;         /* MIDI stream state */
+} S_SMF_STREAM;
+
+/*----------------------------------------------------------------------------
+ *
+ * S_SMF_DATA
+ *
+ * This structure contains the instance data required to parse an SMF stream.
+ *
+ *----------------------------------------------------------------------------
+*/
+
+typedef struct s_smf_data_tag
+{
+#ifdef _CHECKED_BUILD
+    EAS_U32             handleCheck;        /* signature check for checked build */
+#endif
+    S_SMF_STREAM        *streams;           /* pointer to individual streams in file */
+    S_SMF_STREAM        *nextStream;        /* pointer to next stream with event */
+    S_SYNTH             *pSynth;            /* pointer to synth */
+    EAS_FILE_HANDLE     fileHandle;         /* file handle */
+    S_METADATA_CB       metadata;           /* metadata callback */
+    EAS_I32             fileOffset;         /* for embedded files */
+    EAS_I32             time;               /* current time in milliseconds/256 */
+    EAS_U16             numStreams;         /* actual number of streams */
+    EAS_U16             tickConv;           /* current MIDI tick to msec conversion */
+    EAS_U16             ppqn;               /* ticks per quarter note */
+    EAS_U8              state;              /* current state EAS_STATE_XXXX */
+    EAS_U8              flags;              /* flags - see definitions below */
+} S_SMF_DATA;
+
+#define SMF_FLAGS_CHASE_MODE        0x01    /* chase mode - skip to first note */
+#define SMF_FLAGS_HAS_TIME_SIG      0x02    /* time signature encountered at time 0 */
+#define SMF_FLAGS_HAS_TEMPO         0x04    /* tempo encountered at time 0  */
+#define SMF_FLAGS_HAS_GM_ON         0x08    /* GM System On encountered at time 0 */
+#define SMF_FLAGS_JET_STREAM        0x80    /* JET in use - keep strict timing */
+
+/* combo flags indicate setup bar */
+#define SMF_FLAGS_SETUP_BAR (SMF_FLAGS_HAS_TIME_SIG | SMF_FLAGS_HAS_TEMPO | SMF_FLAGS_HAS_GM_ON)
+
+/*----------------------------------------------------------------------------
+ * Interactive MIDI structure
+ *----------------------------------------------------------------------------
+*/
+typedef struct s_interactive_midi_tag
+{
+#ifdef _CHECKED_BUILD
+    EAS_U32             handleCheck;        /* signature check for checked build */
+#endif
+    S_SYNTH     *pSynth;            /* pointer to synth */
+    S_MIDI_STREAM       stream;             /* stream data */
+} S_INTERACTIVE_MIDI;
+
+#endif /* #ifndef _EAS_MIDITYPES_H */
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_mixbuf.c

@@ -0,0 +1,36 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_mixbuf.c
+ *
+ * Contents and purpose:
+ * Contains a data allocation for synthesizer
+ *
+ * Copyright Sonic Network Inc. 2004
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 82 $
+ *   $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+// includes
+#include "eas_data.h"
+#include "eas_mixer.h"
+
+// globals
+EAS_I32 eas_MixBuffer[BUFFER_SIZE_IN_MONO_SAMPLES * NUM_OUTPUT_CHANNELS];
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_mixer.c

@@ -0,0 +1,464 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_mixer.c
+ *
+ * Contents and purpose:
+ * This file contains the critical components of the mix engine that
+ * must be optimized for best performance.
+ *
+ * Copyright Sonic Network Inc. 2005
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 706 $
+ *   $Date: 2007-05-31 17:22:51 -0700 (Thu, 31 May 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+//3 dls: This module is in the midst of being converted from a synth
+//3 specific module to a general purpose mix engine
+
+/*------------------------------------
+ * includes
+ *------------------------------------
+*/
+#include "eas_data.h"
+#include "eas_host.h"
+#include "eas_math.h"
+#include "eas_mixer.h"
+#include "eas_config.h"
+#include "eas_report.h"
+
+#ifdef _MAXIMIZER_ENABLED
+EAS_I32 MaximizerProcess (EAS_VOID_PTR pInstData, EAS_I32 *pSrc, EAS_I32 *pDst, EAS_I32 numSamples);
+#endif
+
+/*------------------------------------
+ * defines
+ *------------------------------------
+*/
+
+/* need to boost stereo by ~3dB to compensate for the panner */
+#define STEREO_3DB_GAIN_BOOST       512
+
+/*----------------------------------------------------------------------------
+ * EAS_MixEngineInit()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Prepares the mix engine for work, allocates buffers, locates effects modules, etc.
+ *
+ * Inputs:
+ * pEASData         - instance data
+ * pInstData        - pointer to variable to receive instance data handle
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_MixEngineInit (S_EAS_DATA *pEASData)
+{
+
+    /* check Configuration Module for mix buffer allocation */
+    if (pEASData->staticMemoryModel)
+        pEASData->pMixBuffer = EAS_CMEnumData(EAS_CM_MIX_BUFFER);
+    else
+        pEASData->pMixBuffer = EAS_HWMalloc(pEASData->hwInstData, BUFFER_SIZE_IN_MONO_SAMPLES * NUM_OUTPUT_CHANNELS * sizeof(EAS_I32));
+    if (pEASData->pMixBuffer == NULL)
+    {
+        { /* dpp: EAS_ReportEx(_EAS_SEVERITY_FATAL, "Failed to allocate mix buffer memory\n"); */ }
+        return EAS_ERROR_MALLOC_FAILED;
+    }
+    EAS_HWMemSet((void *)(pEASData->pMixBuffer), 0, BUFFER_SIZE_IN_MONO_SAMPLES * NUM_OUTPUT_CHANNELS * sizeof(EAS_I32));
+
+    return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * EAS_MixEnginePrep()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Performs prep before synthesize a buffer of audio, such as clearing
+ * audio buffers, etc.
+ *
+ * Inputs:
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+void EAS_MixEnginePrep (S_EAS_DATA *pEASData, EAS_I32 numSamples)
+{
+
+    /* clear the mix buffer */
+#if (NUM_OUTPUT_CHANNELS == 2)
+    EAS_HWMemSet(pEASData->pMixBuffer, 0, numSamples * (EAS_I32) sizeof(long) * 2);
+#else
+    EAS_HWMemSet(pEASData->pMixBuffer, 0, (EAS_I32) numSamples * (EAS_I32) sizeof(long));
+#endif
+
+    /* need to clear other side-chain effect buffers (chorus & reverb) */
+}
+
+/*----------------------------------------------------------------------------
+ * EAS_MixEnginePost
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * This routine does the post-processing after all voices have been
+ * synthesized. It calls any sweeteners and does the final mixdown to
+ * the output buffer.
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *
+ * Notes:
+ *----------------------------------------------------------------------------
+*/
+void EAS_MixEnginePost (S_EAS_DATA *pEASData, EAS_I32 numSamples)
+{
+    EAS_U16 gain;
+
+//3 dls: Need to restore the mix engine metrics
+
+    /* calculate the gain multiplier */
+#ifdef _MAXIMIZER_ENABLED
+    if (pEASData->effectsModules[EAS_MODULE_MAXIMIZER].effect)
+    {
+        EAS_I32 temp;
+        temp = MaximizerProcess(pEASData->effectsModules[EAS_MODULE_MAXIMIZER].effectData, pEASData->pMixBuffer, pEASData->pMixBuffer, numSamples);
+        temp = (temp * pEASData->masterGain) >> 15;
+        if (temp > 32767)
+            gain = 32767;
+        else
+            gain = (EAS_U16) temp;
+    }
+    else
+        gain = (EAS_U16) pEASData->masterGain;
+#else
+    gain = (EAS_U16) pEASData->masterGain;
+#endif
+
+    /* Not using all the gain bits for now
+     * Reduce the input to the compressor by 6dB to prevent saturation
+     */
+#ifdef _COMPRESSOR_ENABLED
+    if (pEASData->effectsModules[EAS_MODULE_COMPRESSOR].effectData)
+        gain = gain >> 5;
+    else
+        gain = gain >> 4;
+#else
+    gain = gain >> 4;
+#endif
+
+    /* convert 32-bit mix buffer to 16-bit output format */
+#if (NUM_OUTPUT_CHANNELS == 2)
+    SynthMasterGain(pEASData->pMixBuffer, pEASData->pOutputAudioBuffer, gain, (EAS_U16) ((EAS_U16) numSamples * 2));
+#else
+    SynthMasterGain(pEASData->pMixBuffer, pEASData->pOutputAudioBuffer, gain, (EAS_U16) numSamples);
+#endif
+
+#ifdef _ENHANCER_ENABLED
+    /* enhancer effect */
+    if (pEASData->effectsModules[EAS_MODULE_ENHANCER].effectData)
+        (*pEASData->effectsModules[EAS_MODULE_ENHANCER].effect->pfProcess)
+            (pEASData->effectsModules[EAS_MODULE_ENHANCER].effectData,
+            pEASData->pOutputAudioBuffer,
+            pEASData->pOutputAudioBuffer,
+            numSamples);
+#endif
+
+#ifdef _GRAPHIC_EQ_ENABLED
+    /* graphic EQ effect */
+    if (pEASData->effectsModules[EAS_MODULE_GRAPHIC_EQ].effectData)
+        (*pEASData->effectsModules[EAS_MODULE_GRAPHIC_EQ].effect->pfProcess)
+            (pEASData->effectsModules[EAS_MODULE_GRAPHIC_EQ].effectData,
+            pEASData->pOutputAudioBuffer,
+            pEASData->pOutputAudioBuffer,
+            numSamples);
+#endif
+
+#ifdef _COMPRESSOR_ENABLED
+    /* compressor effect */
+    if (pEASData->effectsModules[EAS_MODULE_COMPRESSOR].effectData)
+        (*pEASData->effectsModules[EAS_MODULE_COMPRESSOR].effect->pfProcess)
+            (pEASData->effectsModules[EAS_MODULE_COMPRESSOR].effectData,
+            pEASData->pOutputAudioBuffer,
+            pEASData->pOutputAudioBuffer,
+            numSamples);
+#endif
+
+#ifdef _WOW_ENABLED
+    /* WOW requires a 32-bit buffer, borrow the mix buffer and
+     * pass it as the destination buffer
+     */
+    /*lint -e{740} temporarily passing a parameter through an existing I/F */
+    if (pEASData->effectsModules[EAS_MODULE_WOW].effectData)
+        (*pEASData->effectsModules[EAS_MODULE_WOW].effect->pfProcess)
+            (pEASData->effectsModules[EAS_MODULE_WOW].effectData,
+            pEASData->pOutputAudioBuffer,
+            (EAS_PCM*) pEASData->pMixBuffer,
+            numSamples);
+#endif
+
+#ifdef _TONECONTROLEQ_ENABLED
+    /* ToneControlEQ effect */
+    if (pEASData->effectsModules[EAS_MODULE_TONECONTROLEQ].effectData)
+        (*pEASData->effectsModules[EAS_MODULE_TONECONTROLEQ].effect->pfProcess)
+            (pEASData->effectsModules[EAS_MODULE_TONECONTROLEQ].effectData,
+            pEASData->pOutputAudioBuffer,
+            pEASData->pOutputAudioBuffer,
+            numSamples);
+#endif
+
+#ifdef _REVERB_ENABLED
+    /* Reverb effect */
+    if (pEASData->effectsModules[EAS_MODULE_REVERB].effectData)
+        (*pEASData->effectsModules[EAS_MODULE_REVERB].effect->pfProcess)
+            (pEASData->effectsModules[EAS_MODULE_REVERB].effectData,
+            pEASData->pOutputAudioBuffer,
+            pEASData->pOutputAudioBuffer,
+            numSamples);
+#endif
+
+#ifdef _CHORUS_ENABLED
+    /* Chorus effect */
+    if (pEASData->effectsModules[EAS_MODULE_CHORUS].effectData)
+        (*pEASData->effectsModules[EAS_MODULE_CHORUS].effect->pfProcess)
+            (pEASData->effectsModules[EAS_MODULE_CHORUS].effectData,
+            pEASData->pOutputAudioBuffer,
+            pEASData->pOutputAudioBuffer,
+            numSamples);
+#endif
+
+}
+
+#ifndef NATIVE_EAS_KERNEL
+/*----------------------------------------------------------------------------
+ * SynthMasterGain
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Mixes down audio from 32-bit to 16-bit target buffer
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *
+ *----------------------------------------------------------------------------
+*/
+void SynthMasterGain (long *pInputBuffer, EAS_PCM *pOutputBuffer, EAS_U16 nGain, EAS_U16 numSamples) {
+
+    /* loop through the buffer */
+    while (numSamples--) {
+        long s;
+
+        /* read a sample from the input buffer and add some guard bits */
+        s = *pInputBuffer++;
+
+        /* add some guard bits */
+        /*lint -e{704} <avoid divide for performance>*/
+        s = s >> 7;
+
+        /* apply master gain */
+        s *= (long) nGain;
+
+        /* shift to lower 16-bits */
+        /*lint -e{704} <avoid divide for performance>*/
+        s = s >> 9;
+
+        /* saturate */
+        s = SATURATE(s);
+
+        *pOutputBuffer++ = (EAS_PCM)s;
+    }
+}
+#endif
+
+/*----------------------------------------------------------------------------
+ * EAS_MixEngineShutdown()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Shuts down effects modules and deallocates memory
+ *
+ * Inputs:
+ * pEASData         - instance data
+ * pInstData        - instance data handle
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_MixEngineShutdown (S_EAS_DATA *pEASData)
+{
+
+    /* check Configuration Module for static memory allocation */
+    if (!pEASData->staticMemoryModel && (pEASData->pMixBuffer != NULL))
+        EAS_HWFree(pEASData->hwInstData, pEASData->pMixBuffer);
+
+    return EAS_SUCCESS;
+}
+
+#ifdef UNIFIED_MIXER
+#ifndef NATIVE_MIX_STREAM
+/*----------------------------------------------------------------------------
+ * EAS_MixStream
+ *----------------------------------------------------------------------------
+ * Mix a 16-bit stream into a 32-bit buffer
+ *
+ * pInputBuffer 16-bit input buffer
+ * pMixBuffer   32-bit mix buffer
+ * numSamples   number of samples to mix
+ * gainLeft     initial gain left or mono
+ * gainRight    initial gain right
+ * gainLeft     left gain increment per sample
+ * gainRight    right gain increment per sample
+ * flags        bit 0 = stereo source
+ *              bit 1 = stereo output
+ *----------------------------------------------------------------------------
+*/
+void EAS_MixStream (EAS_PCM *pInputBuffer, EAS_I32 *pMixBuffer, EAS_I32 numSamples, EAS_I32 gainLeft, EAS_I32 gainRight, EAS_I32 gainIncLeft, EAS_I32 gainIncRight, EAS_I32 flags)
+{
+    EAS_I32 temp;
+    EAS_INT src, dest;
+
+    /* NOTE: There are a lot of optimizations that can be done
+     * in the native implementations based on register
+     * availability, etc. For example, it may make sense to
+     * break this down into 8 separate routines:
+     *
+     * 1. Mono source to mono output
+     * 2. Mono source to stereo output
+     * 3. Stereo source to mono output
+     * 4. Stereo source to stereo output
+     * 5. Mono source to mono output - no gain change
+     * 6. Mono source to stereo output - no gain change
+     * 7. Stereo source to mono output - no gain change
+     * 8. Stereo source to stereo output - no gain change
+     *
+     * Other possibilities include loop unrolling, skipping
+     * a gain calculation every 2 or 4 samples, etc.
+     */
+
+    /* no gain change, use fast loops */
+    if ((gainIncLeft == 0) && (gainIncRight == 0))
+    {
+        switch (flags & (MIX_FLAGS_STEREO_SOURCE | MIX_FLAGS_STEREO_OUTPUT))
+        {
+            /* mono to mono */
+            case 0:
+                gainLeft >>= 15;
+                for (src = dest = 0; src < numSamples; src++, dest++)
+                {
+
+                    pMixBuffer[dest] += (pInputBuffer[src] * gainLeft) >> NUM_MIXER_GUARD_BITS;
+                }
+                break;
+
+            /* mono to stereo */
+            case MIX_FLAGS_STEREO_OUTPUT:
+                gainLeft >>= 15;
+                gainRight >>= 15;
+                for (src = dest = 0; src < numSamples; src++, dest+=2)
+                {
+                    pMixBuffer[dest] += (pInputBuffer[src] * gainLeft) >> NUM_MIXER_GUARD_BITS;
+                    pMixBuffer[dest+1] += (pInputBuffer[src] * gainRight) >> NUM_MIXER_GUARD_BITS;
+                }
+                break;
+
+            /* stereo to mono */
+            case MIX_FLAGS_STEREO_SOURCE:
+                gainLeft >>= 15;
+                gainRight >>= 15;
+                for (src = dest = 0; src < numSamples; src+=2, dest++)
+                {
+                    temp = (pInputBuffer[src] * gainLeft) >> NUM_MIXER_GUARD_BITS;
+                    temp += ((pInputBuffer[src+1] * gainRight) >> NUM_MIXER_GUARD_BITS);
+                    pMixBuffer[dest] += temp;
+                }
+                break;
+
+            /* stereo to stereo */
+            case MIX_FLAGS_STEREO_SOURCE | MIX_FLAGS_STEREO_OUTPUT:
+                gainLeft >>= 15;
+                gainRight >>= 15;
+                for (src = dest = 0; src < numSamples; src+=2, dest+=2)
+                {
+                    pMixBuffer[dest] += (pInputBuffer[src] * gainLeft) >> NUM_MIXER_GUARD_BITS;
+                    pMixBuffer[dest+1] += (pInputBuffer[src+1] * gainRight) >> NUM_MIXER_GUARD_BITS;
+                }
+                break;
+        }
+    }
+
+    /* gain change - do gain increment */
+    else
+    {
+        switch (flags & (MIX_FLAGS_STEREO_SOURCE | MIX_FLAGS_STEREO_OUTPUT))
+        {
+            /* mono to mono */
+            case 0:
+                for (src = dest = 0; src < numSamples; src++, dest++)
+                {
+                    gainLeft += gainIncLeft;
+                    pMixBuffer[dest] += (pInputBuffer[src] * (gainLeft >> 15)) >> NUM_MIXER_GUARD_BITS;
+                }
+                break;
+
+            /* mono to stereo */
+            case MIX_FLAGS_STEREO_OUTPUT:
+                for (src = dest = 0; src < numSamples; src++, dest+=2)
+                {
+                    gainLeft += gainIncLeft;
+                    gainRight += gainIncRight;
+                    pMixBuffer[dest] += (pInputBuffer[src] * (gainLeft >> 15)) >> NUM_MIXER_GUARD_BITS;
+                    pMixBuffer[dest+1] += (pInputBuffer[src] * (gainRight >> 15)) >> NUM_MIXER_GUARD_BITS;
+                }
+                break;
+
+            /* stereo to mono */
+            case MIX_FLAGS_STEREO_SOURCE:
+                for (src = dest = 0; src < numSamples; src+=2, dest++)
+                {
+                    gainLeft += gainIncLeft;
+                    gainRight += gainIncRight;
+                    temp = (pInputBuffer[src] * (gainLeft >> 15)) >> NUM_MIXER_GUARD_BITS;
+                    temp += ((pInputBuffer[src+1] * (gainRight >> 15)) >> NUM_MIXER_GUARD_BITS);
+                    pMixBuffer[dest] += temp;
+                }
+                break;
+
+            /* stereo to stereo */
+            case MIX_FLAGS_STEREO_SOURCE | MIX_FLAGS_STEREO_OUTPUT:
+                for (src = dest = 0; src < numSamples; src+=2, dest+=2)
+                {
+                    gainLeft += gainIncLeft;
+                    gainRight += gainIncRight;
+                    pMixBuffer[dest] += (pInputBuffer[src] * (gainLeft >> 15)) >> NUM_MIXER_GUARD_BITS;
+                    pMixBuffer[dest+1] += (pInputBuffer[src+1] * (gainRight >> 15)) >> NUM_MIXER_GUARD_BITS;
+                }
+                break;
+        }
+    }
+}
+#endif
+#endif
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_mixer.h

@@ -0,0 +1,137 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_mixer.h
+ *
+ * Contents and purpose:
+ * This file contains the critical components of the mix engine that
+ * must be optimized for best performance.
+ *
+ * Copyright Sonic Network Inc. 2005
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 706 $
+ *   $Date: 2007-05-31 17:22:51 -0700 (Thu, 31 May 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_MIXER_H
+#define _EAS_MIXER_H
+
+//3 dls: This module is in the midst of being converted from a synth
+//3 specific module to a general purpose mix engine
+
+#define MIX_FLAGS_STEREO_SOURCE     1
+#define MIX_FLAGS_STEREO_OUTPUT     2
+#define NUM_MIXER_GUARD_BITS        4
+
+#include "eas_effects.h"
+
+extern void SynthMasterGain( long *pInputBuffer, EAS_PCM *pOutputBuffer, EAS_U16 nGain, EAS_U16 nNumLoopSamples);
+
+/*----------------------------------------------------------------------------
+ * EAS_MixEngineInit()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Prepares the mix engine for work, allocates buffers, locates effects modules, etc.
+ *
+ * Inputs:
+ * pEASData         - instance data
+ * pInstData        - pointer to variable to receive instance data handle
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_MixEngineInit (EAS_DATA_HANDLE pEASData);
+
+/*----------------------------------------------------------------------------
+ * EAS_MixEnginePrep()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Performs prep before synthesize a buffer of audio, such as clearing
+ * audio buffers, etc.
+ *
+ * Inputs:
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+void EAS_MixEnginePrep (EAS_DATA_HANDLE pEASData, EAS_I32 nNumSamplesToAdd);
+
+/*----------------------------------------------------------------------------
+ * EAS_MixEnginePost
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * This routine does the post-processing after all voices have been
+ * synthesized. It calls any sweeteners and does the final mixdown to
+ * the output buffer.
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *
+ * Notes:
+ *----------------------------------------------------------------------------
+*/
+void EAS_MixEnginePost (EAS_DATA_HANDLE pEASData, EAS_I32 nNumSamplesToAdd);
+
+/*----------------------------------------------------------------------------
+ * EAS_MixEngineShutdown()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Shuts down effects modules and deallocates memory
+ *
+ * Inputs:
+ * pEASData         - instance data
+ * pInstData        - instance data handle
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_MixEngineShutdown (EAS_DATA_HANDLE pEASData);
+
+#ifdef UNIFIED_MIXER
+/*----------------------------------------------------------------------------
+ * EAS_MixStream
+ *----------------------------------------------------------------------------
+ * Mix a 16-bit stream into a 32-bit buffer
+ *
+ * pInputBuffer 16-bit input buffer
+ * pMixBuffer   32-bit mix buffer
+ * numSamples   number of samples to mix
+ * gainLeft     initial gain left or mono
+ * gainRight    initial gain right
+ * gainLeft     left gain increment per sample
+ * gainRight    right gain increment per sample
+ * flags        bit 0 = stereo source
+ *              bit 1 = stereo output
+ *----------------------------------------------------------------------------
+*/
+void EAS_MixStream (EAS_PCM *pInputBuffer, EAS_I32 *pMixBuffer, EAS_I32 numSamples, EAS_I32 gainLeft, EAS_I32 gainRight, EAS_I32 gainIncLeft, EAS_I32 gainIncRight, EAS_I32 flags);
+#endif
+
+#endif /* #ifndef _EAS_MIXER_H */
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_otadata.c

@@ -0,0 +1,41 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_otadata..c
+ *
+ * Contents and purpose:
+ * OTA Stream Parser data module for static memory model
+ *
+ * Copyright Sonic Network Inc. 2005
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 547 $
+ *   $Date: 2007-01-31 16:30:17 -0800 (Wed, 31 Jan 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#include "eas_types.h"
+#include "eas_otadata.h"
+
+/*----------------------------------------------------------------------------
+ *
+ * eas_OTAData
+ *
+ * Static memory allocation for OTA parser
+ *----------------------------------------------------------------------------
+*/
+S_OTA_DATA eas_OTAData;
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_otadata.h

@@ -0,0 +1,81 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_otadata.h
+ *
+ * Contents and purpose:
+ * OTA File Parser
+ *
+ * This file contains data declarations for the OTA parser.
+ *
+ * Copyright Sonic Network Inc. 2005
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 82 $
+ *   $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef EAS_OTADATA_H
+#define EAS_OTADATA_H
+
+#include "eas_data.h"
+
+/* definition for state flags */
+#define OTA_FLAGS_UNICODE           0x01    /* unicode text */
+
+/*----------------------------------------------------------------------------
+ *
+ * S_OTA_DATA
+ *
+ * This structure contains the state data for the OTA parser
+ *----------------------------------------------------------------------------
+*/
+
+typedef struct
+{
+    EAS_I32         fileOffset;             /* offset to location in file */
+    EAS_U8          patternLen;             /* length of current pattern */
+    EAS_U8          dataByte;               /* previous char from file */
+    EAS_U8          bitCount;               /* bit count in char */
+} S_OTA_LOC;
+
+typedef struct
+{
+    EAS_FILE_HANDLE fileHandle;             /* file handle */
+    S_SYNTH *pSynth;                /* synth handle */
+    EAS_I32         fileOffset;             /* offset to start of data */
+    EAS_I32         time;                   /* current time in 256ths of a msec */
+    EAS_U32         tick;                   /* length of 32nd note in 256th of a msec */
+    EAS_U32         restTicks;              /* ticks to rest after current note */
+    S_OTA_LOC       patterns[4];            /* pattern locations */
+    S_OTA_LOC       current;                /* current location */
+    S_OTA_LOC       restore;                /* previous location */
+    S_METADATA_CB   metadata;               /* metadata callback */
+    EAS_U8          flags;                  /* bit flags */
+    EAS_U8          numPatterns;            /* number of patterns left in song */
+    EAS_U8          currentPattern;         /* current pattern for loop */
+    EAS_U8          note;                   /* MIDI note number */
+    EAS_U8          octave;                 /* octave modifier */
+    EAS_U8          style;                  /* from STYLE */
+    EAS_U8          velocity;               /* current volume */
+    EAS_U8          state;                  /* current state EAS_STATE_XXXX */
+    EAS_U8          loopCount;              /* loop count for pattern */
+} S_OTA_DATA;
+
+#endif
+
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_pan.c

@@ -0,0 +1,98 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_pan.c
+ *
+ * Contents and purpose:
+ * Calculates left and right gain multipliers based on a pan value from -63 to +63
+ *
+ * NOTES:
+ * The _CMX_PARSER and _MFI_PARSER preprocessor symbols determine
+ * whether the parser works for those particular file formats.
+ *
+ * Copyright Sonic Network Inc. 2005
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 82 $
+ *   $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+#include "eas_pan.h"
+#include "eas_math.h"
+
+/*----------------------------------------------------------------------------
+ * EAS_CalcPanControl()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Assign the left and right gain values corresponding to the given pan value.
+ *
+ * This routine uses sin/cos approximations for an equal power curve:
+ *
+ * sin(x) = (2-4*c)*x^2 + c + x
+ * cos(x) = (2-4*c)*x^2 + c - x
+ *
+ * where  c = 1/sqrt(2)
+ * using the a0 + x*(a1 + x*a2) approach
+ *
+ * Inputs:
+ * pan          - pan value (-63 to + 63)
+ *
+ * Outputs:
+ * pGainLeft    linear gain multiplier for left channel (15-bit fraction)
+ * pGainRight   linear gain multiplier for left channel (15-bit fraction)
+ *
+ * Side Effects:
+ *----------------------------------------------------------------------------
+*/
+void EAS_CalcPanControl (EAS_INT pan, EAS_I16 *pGainLeft, EAS_I16 *pGainRight)
+{
+    EAS_INT temp;
+    EAS_INT netAngle;
+
+    /* impose hard limit */
+    if (pan < -63)
+        netAngle = -63;
+    else if (pan > 63)
+        netAngle = 63;
+    else
+        netAngle = pan;
+
+    /*lint -e{701} <avoid multiply for performance reasons>*/
+    netAngle = netAngle << 8;
+
+    /* calculate sin */
+    temp = EG1_ONE + FMUL_15x15(COEFF_PAN_G2, netAngle);
+    temp = COEFF_PAN_G0 + FMUL_15x15(temp, netAngle);
+
+    if (temp > SYNTH_FULL_SCALE_EG1_GAIN)
+        temp = SYNTH_FULL_SCALE_EG1_GAIN;
+    else if (temp < 0)
+        temp = 0;
+
+    *pGainRight = (EAS_I16) temp;
+
+    /* calculate cos */
+    temp = -EG1_ONE + FMUL_15x15(COEFF_PAN_G2, netAngle);
+    temp = COEFF_PAN_G0 + FMUL_15x15(temp, netAngle);
+    if (temp > SYNTH_FULL_SCALE_EG1_GAIN)
+        temp = SYNTH_FULL_SCALE_EG1_GAIN;
+    else if (temp < 0)
+        temp = 0;
+
+    *pGainLeft = (EAS_I16) temp;
+}
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_pan.h

@@ -0,0 +1,66 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_pan.h
+ *
+ * Contents and purpose:
+ * Calculates left and right gain multipliers based on a pan value from -63 to +63
+ *
+ * NOTES:
+ * The _CMX_PARSER and _MFI_PARSER preprocessor symbols determine
+ * whether the parser works for those particular file formats.
+ *
+ * Copyright Sonic Network Inc. 2005
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 82 $
+ *   $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef EAS_PAN_H
+#define _EAS_PAN_H
+
+#include "eas_types.h"
+
+/*----------------------------------------------------------------------------
+ * EAS_CalcPanControl()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Assign the left and right gain values corresponding to the given pan value.
+ *
+ * This routine uses sin/cos approximations for an equal power curve:
+ *
+ * sin(x) = (2-4*c)*x^2 + c + x
+ * cos(x) = (2-4*c)*x^2 + c - x
+ *
+ * where  c = 1/sqrt(2)
+ * using the a0 + x*(a1 + x*a2) approach
+ *
+ * Inputs:
+ * pan          - pan value (-63 to + 63)
+ *
+ * Outputs:
+ * pGainLeft    linear gain multiplier for left channel (15-bit fraction)
+ * pGainRight   linear gain multiplier for left channel (15-bit fraction)
+ *
+ * Side Effects:
+ *----------------------------------------------------------------------------
+*/
+void EAS_CalcPanControl (EAS_INT pan, EAS_I16 *pGainLeft, EAS_I16 *pGainRight);
+
+#endif
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_parser.h

@@ -0,0 +1,98 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_parser.h
+ *
+ * Contents and purpose:
+ * Interface declarations for the generic parser interface
+ *
+ * This header only contains declarations that are specific
+ * to this implementation.
+ *
+ * Copyright Sonic Network Inc. 2005
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 767 $
+ *   $Date: 2007-07-19 13:47:31 -0700 (Thu, 19 Jul 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_PARSER_H
+#define _EAS_PARSER_H
+
+#include "eas_types.h"
+
+
+/* metadata callback */
+typedef struct s_metadata_cb_tag
+{
+    EAS_METADATA_CBFUNC     callback;
+    char                    *buffer;
+    EAS_VOID_PTR            pUserData;
+    EAS_I32                 bufferSize;
+} S_METADATA_CB;
+
+/* generic parser interface */
+typedef struct
+{
+    EAS_RESULT (* EAS_CONST pfCheckFileType)(struct s_eas_data_tag *pEASData, EAS_FILE_HANDLE fileHandle, EAS_VOID_PTR *ppHandle, EAS_I32 offset);
+    EAS_RESULT (* EAS_CONST pfPrepare)(struct s_eas_data_tag *pEASData, EAS_VOID_PTR pInstData);
+    EAS_RESULT (* EAS_CONST pfTime)(struct s_eas_data_tag *pEASData, EAS_VOID_PTR pInstData, EAS_U32 *pTime);
+    EAS_RESULT (* EAS_CONST pfEvent)(struct s_eas_data_tag *pEASData, EAS_VOID_PTR pInstData, EAS_INT parseMode);
+    EAS_RESULT (* EAS_CONST pfState)(struct s_eas_data_tag *pEASData, EAS_VOID_PTR pInstData, EAS_STATE *pState);
+    EAS_RESULT (* EAS_CONST pfClose)(struct s_eas_data_tag *pEASData, EAS_VOID_PTR pInstData);
+    EAS_RESULT (* EAS_CONST pfReset)(struct s_eas_data_tag *pEASData, EAS_VOID_PTR pInstData);
+    EAS_RESULT (* EAS_CONST pfPause)(struct s_eas_data_tag *pEASData, EAS_VOID_PTR pInstData);
+    EAS_RESULT (* EAS_CONST pfResume)(struct s_eas_data_tag *pEASData, EAS_VOID_PTR pInstData);
+    EAS_RESULT (* EAS_CONST pfLocate)(struct s_eas_data_tag *pEASData, EAS_VOID_PTR pInstData, EAS_I32 time, EAS_BOOL *pParserLocate);
+    EAS_RESULT (* EAS_CONST pfSetData)(struct s_eas_data_tag *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 value);
+    EAS_RESULT (* EAS_CONST pfGetData)(struct s_eas_data_tag *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 *pValue);
+    EAS_RESULT (* EAS_CONST pfGetMetaData)(struct s_eas_data_tag *pEASData, EAS_VOID_PTR pInstData, EAS_I32 *pMediaLength);
+} S_FILE_PARSER_INTERFACE;
+
+typedef enum
+{
+    eParserModePlay,
+    eParserModeLocate,
+    eParserModeMute,
+    eParserModeMetaData
+} E_PARSE_MODE;
+
+typedef enum
+{
+    PARSER_DATA_FILE_TYPE,
+    PARSER_DATA_PLAYBACK_RATE,
+    PARSER_DATA_TRANSPOSITION,
+    PARSER_DATA_VOLUME,
+    PARSER_DATA_SYNTH_HANDLE,
+    PARSER_DATA_METADATA_CB,
+    PARSER_DATA_DLS_COLLECTION,
+    PARSER_DATA_EAS_LIBRARY,
+    PARSER_DATA_POLYPHONY,
+    PARSER_DATA_PRIORITY,
+    PARSER_DATA_FORMAT,
+    PARSER_DATA_MEDIA_LENGTH,
+    PARSER_DATA_JET_CB,
+    PARSER_DATA_MUTE_FLAGS,
+    PARSER_DATA_SET_MUTE,
+    PARSER_DATA_CLEAR_MUTE,
+    PARSER_DATA_NOTE_COUNT,
+    PARSER_DATA_MAX_PCM_STREAMS,
+    PARSER_DATA_GAIN_OFFSET,
+    PARSER_DATA_PLAY_MODE
+} E_PARSER_DATA;
+
+#endif /* #ifndef _EAS_PARSER_H */

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_pcm.h

@@ -0,0 +1,359 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_pcm.h
+ *
+ * Contents and purpose:
+ * External function prototypes for eas_pcm.c module
+ *
+ *
+ * Copyright Sonic Network Inc. 2005
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 847 $
+ *   $Date: 2007-08-27 21:30:08 -0700 (Mon, 27 Aug 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_PCM_H
+#define _EAS_PCM_H
+
+/* default gain setting - roughly unity gain */
+#define PCM_DEFAULT_GAIN_SETTING    0x6000
+
+typedef struct s_pcm_state_tag *EAS_PCM_HANDLE;
+typedef void (*EAS_PCM_CALLBACK) (EAS_DATA_HANDLE pEASData, EAS_VOID_PTR cbInstData, EAS_PCM_HANDLE pcmHandle, EAS_STATE state);
+
+/* parameters for EAS_PEOpenStream */
+typedef struct s_pcm_open_params_tag
+{
+    EAS_FILE_HANDLE     fileHandle;
+    EAS_I32             decoder;
+    EAS_U32             sampleRate;
+    EAS_I32             size;
+    EAS_U32             loopStart;
+    EAS_U32             loopSamples;
+    EAS_I32             blockSize;
+    EAS_U32             flags;
+    EAS_U32             envData;
+    EAS_I16             volume;
+    EAS_PCM_CALLBACK    pCallbackFunc;
+    EAS_VOID_PTR        cbInstData;
+ } S_PCM_OPEN_PARAMS;
+
+/*----------------------------------------------------------------------------
+ * EAS_PEInit()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Initializes the PCM engine
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_PEInit (EAS_DATA_HANDLE pEASData);
+
+/*----------------------------------------------------------------------------
+ * EAS_PEShutdown()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Shuts down the PCM engine
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_PEShutdown (EAS_DATA_HANDLE pEASData);
+
+/*----------------------------------------------------------------------------
+ * EAS_PEOpenStream()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Starts up a PCM playback
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_PEOpenStream (EAS_DATA_HANDLE pEASData, S_PCM_OPEN_PARAMS *pParams, EAS_PCM_HANDLE *pHandle);
+
+/*----------------------------------------------------------------------------
+ * EAS_PEContinueStream()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Continues a PCM stream
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_PEContinueStream (EAS_DATA_HANDLE pEASData, EAS_PCM_HANDLE handle, EAS_I32 size);
+
+/*----------------------------------------------------------------------------
+ * EAS_PEGetFileHandle()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Returns the file handle of a stream
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_PEGetFileHandle (EAS_DATA_HANDLE pEASData, EAS_PCM_HANDLE handle, EAS_FILE_HANDLE *pFileHandle);
+
+/*----------------------------------------------------------------------------
+ * EAS_PERender()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Render a buffer of PCM audio
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_PERender (EAS_DATA_HANDLE pEASData, EAS_I32 numSamples);
+
+/*----------------------------------------------------------------------------
+ * EAS_PEUpdateParams()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Update the pitch and volume parameters using MIDI controls
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_PEUpdateParams (EAS_DATA_HANDLE pEASData, EAS_PCM_HANDLE pState, EAS_I16 pitch, EAS_I16 gainLeft, EAS_I16 gainRight);
+
+/*----------------------------------------------------------------------------
+ * EAS_PELocate()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * This function seeks to the requested place in the file. Accuracy
+ * is dependent on the sample rate and block size.
+ *
+ * Inputs:
+ * pEASData         - pointer to overall EAS data structure
+ * pState           - stream handle
+ * time             - media time in milliseconds
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_PELocate (EAS_DATA_HANDLE pEASData, EAS_PCM_HANDLE pState, EAS_I32 time);
+
+/*----------------------------------------------------------------------------
+ * EAS_PEUpdateVolume()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Update the volume parameters for a PCM stream
+ *
+ * Inputs:
+ * pEASData         - pointer to EAS library instance data
+ * handle           - pointer to S_PCM_STATE for this stream
+ * gainLeft         - linear gain multipler in 1.15 fraction format
+ * gainRight        - linear gain multipler in 1.15 fraction format
+ * initial          - initial settings, set current gain
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ * Notes
+ * In mono mode, leftGain controls the output gain and rightGain is ignored
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pEASData) reserved for future use */
+EAS_RESULT EAS_PEUpdateVolume (EAS_DATA_HANDLE pEASData, EAS_PCM_HANDLE pState, EAS_I16 volume);
+
+/*----------------------------------------------------------------------------
+ * EAS_PEUpdatePitch()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Update the pitch parameter for a PCM stream
+ *
+ * Inputs:
+ * pEASData         - pointer to EAS library instance data
+ * pState           - pointer to S_PCM_STATE for this stream
+ * pitch            - new pitch value in pitch cents
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pEASData) reserved for future use */
+EAS_RESULT EAS_PEUpdatePitch (EAS_DATA_HANDLE pEASData, EAS_PCM_HANDLE pState, EAS_I16 pitch);
+
+/*----------------------------------------------------------------------------
+ * EAS_PEState()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Returns the current state of the stream
+ *
+ * Inputs:
+ * pEASData         - pointer to overall EAS data structure
+ * handle           - pointer to file handle
+ * pState           - pointer to variable to store state
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ * Notes:
+ * This interface is also exposed in the internal library for use by the other modules.
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_PEState (EAS_DATA_HANDLE pEASData, EAS_PCM_HANDLE handle, EAS_STATE *pState);
+
+/*----------------------------------------------------------------------------
+ * EAS_PEClose()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Close the file and clean up
+ *
+ * Inputs:
+ * pEASData         - pointer to overall EAS data structure
+ * handle           - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_PEClose (EAS_DATA_HANDLE pEASData, EAS_PCM_HANDLE handle);
+
+/*----------------------------------------------------------------------------
+ * EAS_PEReset()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Reset the sequencer. Used for locating backwards in the file.
+ *
+ * Inputs:
+ * pEASData         - pointer to overall EAS data structure
+ * handle           - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_PEReset (EAS_DATA_HANDLE pEASData, EAS_PCM_HANDLE handle);
+
+/*----------------------------------------------------------------------------
+ * EAS_PEPause()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Mute and pause rendering a PCM stream. Sets the gain target to zero and stops the playback
+ * at the end of the next audio frame.
+ *
+ * Inputs:
+ * pEASData         - pointer to EAS library instance data
+ * handle           - pointer to S_PCM_STATE for this stream
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_PEPause (EAS_DATA_HANDLE pEASData, EAS_PCM_HANDLE handle);
+
+/*----------------------------------------------------------------------------
+ * EAS_PEResume()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Resume rendering a PCM stream. Sets the gain target back to its
+ * previous setting and restarts playback at the end of the next audio
+ * frame.
+ *
+ * Inputs:
+ * pEASData         - pointer to EAS library instance data
+ * handle           - pointer to S_PCM_STATE for this stream
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_PEResume (EAS_DATA_HANDLE pEASData, EAS_PCM_HANDLE handle);
+
+/*----------------------------------------------------------------------------
+ * EAS_PERelease()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Put the PCM stream envelope into release.
+ *
+ * Inputs:
+ * pEASData         - pointer to EAS library instance data
+ * handle           - pointer to S_PCM_STATE for this stream
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_PERelease (EAS_DATA_HANDLE pEASData, EAS_PCM_HANDLE handle);
+
+#endif /* end _EAS_PCM_H */
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_pcmdata.c

@@ -0,0 +1,35 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_pcmdata.c
+ *
+ * Contents and purpose:
+ * Contains the static data for the PCM engine.
+ *
+ * Copyright Sonic Network Inc. 2005
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 547 $
+ *   $Date: 2007-01-31 16:30:17 -0800 (Wed, 31 Jan 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#include "eas_data.h"
+
+/* static data allocation */
+S_PCM_STATE eas_PCMData[MAX_PCM_STREAMS];
+
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_pcmdata.h

@@ -0,0 +1,157 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_pcmdata.h
+ *
+ * Contents and purpose:
+ * Data declarations for the PCM engine
+ *
+ *
+ * Copyright Sonic Network Inc. 2005
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 847 $
+ *   $Date: 2007-08-27 21:30:08 -0700 (Mon, 27 Aug 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_PCMDATA_H
+#define _EAS_PCMDATA_H
+
+/* sets the maximum number of simultaneous PCM streams */
+#ifndef MAX_PCM_STREAMS
+#define MAX_PCM_STREAMS             16
+#define PCM_STREAM_THRESHOLD        (MAX_PCM_STREAMS - 4)
+#endif
+
+/* coefficents for high-pass filter in ADPCM */
+#define INTEGRATOR_COEFFICIENT      100     /* coefficient for leaky integrator */
+
+/* additional flags in S_PCM_STATE.flags used internal to module */
+#define PCM_FLAGS_EMPTY             0x01000000  /* unsigned format */
+
+/*----------------------------------------------------------------------------
+ * S_PCM_STATE
+ *
+ * Retains state information for PCM streams.
+ *----------------------------------------------------------------------------
+*/
+typedef struct s_decoder_state_tag
+{
+    EAS_I32             output;             /* last output for DC offset filter */
+    EAS_I32             acc;                /* accumulator for DC offset filter */
+    EAS_I32             step;               /* current ADPCM step size */
+    EAS_PCM             x1;                 /* current generated sample */
+    EAS_PCM             x0;                 /* previous generated sample */
+} S_DECODER_STATE;
+
+typedef enum
+{
+    PCM_ENV_START = 0,
+    PCM_ENV_ATTACK,
+    PCM_ENV_DECAY,
+    PCM_ENV_SUSTAIN,
+    PCM_ENV_RELEASE,
+    PCM_ENV_END
+} E_PCM_ENV_STATE;
+
+typedef struct s_pcm_state_tag
+{
+#ifdef _CHECKED_BUILD
+    EAS_U32             handleCheck;        /* signature check for checked build */
+#endif
+    EAS_FILE_HANDLE     fileHandle;         /* pointer to input file */
+    EAS_PCM_CALLBACK    pCallback;          /* pointer to callback function */
+    EAS_VOID_PTR        cbInstData;         /* instance data for callback function */
+    struct s_decoder_interface_tag EAS_CONST * pDecoder;    /* pointer to decoder interface */
+    EAS_STATE           state;              /* stream state */
+    EAS_I32             time;               /* media time */
+    EAS_I32             startPos;           /* start of PCM stream */
+    EAS_I32             loopLocation;       /* file location where loop starts */
+    EAS_I32             byteCount;          /* size of file */
+    EAS_U32             loopStart;          /* loop start, offset in samples from startPos */
+                                            /* NOTE: For CMF, we use this to store total sample size */
+    EAS_U32             loopSamples;        /* total loop length, in samples, 0 means no loop */
+                                            /* NOTE: For CMF, non-zero means looped */
+    EAS_U32             samplesInLoop;      /* samples left in the loop to play back */
+    EAS_I32             samplesTilLoop;     /* samples left to play until top of loop */
+    EAS_I32             bytesLeft;          /* count of bytes left in stream */
+    EAS_I32             bytesLeftLoop;      /* count of bytes left in stream, value at start of loop */
+    EAS_U32             phase;              /* current phase for interpolator */
+    EAS_U32             basefreq;           /* frequency multiplier */
+    EAS_U32             flags;              /* stream flags */
+    EAS_U32             envData;            /* envelope data (and LFO data) */
+    EAS_U32             envValue;           /* current envelope value */
+    EAS_U32             envScale;           /* current envelope scale */
+    EAS_U32             startOrder;         /* start order index, first is 0, next is 1, etc. */
+    S_DECODER_STATE     decoderL;           /* left (mono) ADPCM state */
+    S_DECODER_STATE     decoderR;           /* right ADPCM state */
+    S_DECODER_STATE     decoderLLoop;       /* left (mono) ADPCM state, value at start of loop */
+    S_DECODER_STATE     decoderRLoop;       /* right ADPCM state, value at start of loop */
+    E_PCM_ENV_STATE     envState;           /* current envelope state */
+    EAS_I16             volume;             /* volume for stream */
+    EAS_I16             pitch;              /* relative pitch in cents - zero is unity playback */
+    EAS_I16             gainLeft;           /* requested gain */
+    EAS_I16             gainRight;          /* requested gain */
+    EAS_I16             currentGainLeft;    /* current gain for anti-zipper filter */
+    EAS_I16             currentGainRight;   /* current gain for anti-zipper filter */
+    EAS_U16             blockSize;          /* block size for ADPCM decoder */
+    EAS_U16             blockCount;         /* block counter for ADPCM decoder */
+    EAS_U16             sampleRate;         /* input sample rate */
+    EAS_U8              srcByte;            /* source byte */
+    EAS_U8              msBitCount;         /* count keeps track of MS bits */
+    EAS_U8              msBitMask;          /* mask keeps track of MS bits */
+    EAS_U8              msBitValue;         /* value keeps track of MS bits */
+    EAS_U8              msBitCountLoop;     /* count keeps track of MS bits, value at loop start */
+    EAS_U8              msBitMaskLoop;      /* mask keeps track of MS bits, value at loop start */
+    EAS_U8              msBitValueLoop;     /* value keeps track of MS bits, value at loop start */
+    EAS_BOOL8           hiNibble;           /* indicates high/low nibble is next */
+    EAS_BOOL8           hiNibbleLoop;       /* indicates high/low nibble is next, value loop start */
+    EAS_U8              rateShift;          /* for playback rate greater than 1.0 */
+} S_PCM_STATE;
+
+/*----------------------------------------------------------------------------
+ * S_DECODER_INTERFACE
+ *
+ * Generic interface for audio decoders
+ *----------------------------------------------------------------------------
+*/
+typedef struct s_decoder_interface_tag
+{
+    EAS_RESULT (* EAS_CONST pfInit)(EAS_DATA_HANDLE pEASData, S_PCM_STATE *pState);
+    EAS_RESULT (* EAS_CONST pfDecodeSample)(EAS_DATA_HANDLE pEASData, S_PCM_STATE *pState);
+    EAS_RESULT (* EAS_CONST pfLocate)(EAS_DATA_HANDLE pEASData, S_PCM_STATE *pState, EAS_I32 time);
+} S_DECODER_INTERFACE;
+
+
+/* header chunk for SMAF ADPCM */
+#define TAG_YAMAHA_ADPCM    0x4d776100
+#define TAG_MASK            0xffffff00
+#define TAG_RIFF_FILE       0x52494646
+#define TAG_WAVE_CHUNK      0x57415645
+#define TAG_FMT_CHUNK       0x666d7420
+
+/*----------------------------------------------------------------------------
+ * EAS_PESeek
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Locate to a particular byte in a PCM stream
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_PESeek (EAS_DATA_HANDLE pEASData, S_PCM_STATE *pState, EAS_I32 *pLocation);
+
+#endif /* _EAS_PCMDATA_H */
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_reverbdata.c

@@ -0,0 +1,34 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_reverbdata.c
+ *
+ * Contents and purpose:
+ * Contains the static data allocation for the Reverb effect
+ *
+ *
+ * Copyright Sonic Network Inc. 2006
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 550 $
+ *   $Date: 2007-02-02 09:37:03 -0800 (Fri, 02 Feb 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#include "eas_reverbdata.h"
+
+S_REVERB_OBJECT eas_ReverbData;
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_reverbdata.h

@@ -0,0 +1,486 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_reverbdata.h
+ *
+ * Contents and purpose:
+ * Contains the prototypes for the Reverb effect.
+ *
+ *
+ * Copyright Sonic Network Inc. 2006
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 499 $
+ *   $Date: 2006-12-11 16:07:20 -0800 (Mon, 11 Dec 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_REVERBDATA_H
+#define _EAS_REVERBDATA_H
+
+#include "eas_types.h"
+#include "eas_audioconst.h"
+
+/*------------------------------------
+ * defines
+ *------------------------------------
+*/
+
+/*
+CIRCULAR() calculates the array index using modulo arithmetic.
+The "trick" is that modulo arithmetic is simplified by masking
+the effective address where the mask is (2^n)-1. This only works
+if the buffer size is a power of two.
+*/
+#define CIRCULAR(base,offset,size) (EAS_U32)(               \
+            (                                               \
+                ((EAS_I32)(base)) + ((EAS_I32)(offset))     \
+            )                                               \
+            & size                                          \
+                                            )
+
+/* reverb parameters are updated every 2^(REVERB_UPDATE_PERIOD_IN_BITS) samples */
+#if defined (_SAMPLE_RATE_8000)
+
+#define REVERB_UPDATE_PERIOD_IN_BITS        5
+#define REVERB_BUFFER_SIZE_IN_SAMPLES       2048
+
+#elif defined (_SAMPLE_RATE_16000)
+
+#define REVERB_UPDATE_PERIOD_IN_BITS        6
+#define REVERB_BUFFER_SIZE_IN_SAMPLES       4096
+
+#elif defined (_SAMPLE_RATE_22050)
+
+#define REVERB_UPDATE_PERIOD_IN_BITS        7
+#define REVERB_BUFFER_SIZE_IN_SAMPLES       4096
+
+#elif defined (_SAMPLE_RATE_32000)
+
+#define REVERB_UPDATE_PERIOD_IN_BITS        7
+#define REVERB_BUFFER_SIZE_IN_SAMPLES       8192
+
+#elif defined (_SAMPLE_RATE_44100)
+
+#define REVERB_UPDATE_PERIOD_IN_BITS        8
+#define REVERB_BUFFER_SIZE_IN_SAMPLES       8192
+
+#elif defined (_SAMPLE_RATE_48000)
+
+#define REVERB_UPDATE_PERIOD_IN_BITS        8
+#define REVERB_BUFFER_SIZE_IN_SAMPLES       8192
+
+#endif
+
+// Define a mask for circular addressing, so that array index
+// can wraparound and stay in array boundary of 0, 1, ..., (buffer size -1)
+// The buffer size MUST be a power of two
+#define REVERB_BUFFER_MASK                  (REVERB_BUFFER_SIZE_IN_SAMPLES -1)
+
+#define REVERB_MAX_ROOM_TYPE            4   // any room numbers larger than this are invalid
+#define REVERB_MAX_NUM_REFLECTIONS      5   // max num reflections per channel
+
+/* synth parameters are updated every SYNTH_UPDATE_PERIOD_IN_SAMPLES */
+#define REVERB_UPDATE_PERIOD_IN_SAMPLES (EAS_I32)(0x1L << REVERB_UPDATE_PERIOD_IN_BITS)
+
+/*
+calculate the update counter by bitwise ANDING with this value to
+generate a 2^n modulo value
+*/
+#define REVERB_MODULO_UPDATE_PERIOD_IN_SAMPLES  (EAS_I32)(REVERB_UPDATE_PERIOD_IN_SAMPLES -1)
+
+/* synth parameters are updated every SYNTH_UPDATE_PERIOD_IN_SECONDS seconds */
+#define REVERB_UPDATE_PERIOD_IN_SECONDS     (REVERB_UPDATE_PERIOD_IN_SAMPLES / _OUTPUT_SAMPLE_RATE)
+
+// xfade parameters
+#define REVERB_XFADE_PERIOD_IN_SECONDS      (100.0 / 1000.0)        // xfade once every this many seconds
+
+#define REVERB_XFADE_PERIOD_IN_SAMPLES      (REVERB_XFADE_PERIOD_IN_SECONDS * _OUTPUT_SAMPLE_RATE)
+
+#define REVERB_XFADE_PHASE_INCREMENT    (EAS_I16)(65536 / ((EAS_I16)REVERB_XFADE_PERIOD_IN_SAMPLES/(EAS_I16)REVERB_UPDATE_PERIOD_IN_SAMPLES))
+
+/**********/
+/* the entire synth uses various flags in a bit field */
+
+/* if flag is set, synth reset has been requested */
+#define REVERB_FLAG_RESET_IS_REQUESTED          0x01    /* bit 0 */
+#define MASK_REVERB_RESET_IS_REQUESTED          0x01
+#define MASK_REVERB_RESET_IS_NOT_REQUESTED      (EAS_U32)(~MASK_REVERB_RESET_IS_REQUESTED)
+
+/*
+by default, we always want to update ALL channel parameters
+when we reset the synth (e.g., during GM ON)
+*/
+#define DEFAULT_REVERB_FLAGS                    0x0
+
+/* coefficients for generating sin, cos */
+#define REVERB_PAN_G2   4294940151          /* -0.82842712474619 = 2 - 4/sqrt(2) */
+/*
+EAS_I32 nPanG1 = +1.0 for sin
+EAS_I32 nPanG1 = -1.0 for cos
+*/
+#define REVERB_PAN_G0   23170               /* 0.707106781186547 = 1/sqrt(2) */
+
+/*************************************************************/
+// define the input injection points
+#define GUARD               5                       // safety guard of this many samples
+
+#define MAX_AP_TIME         (double) (20.0/1000.0)  // delay time in milliseconds
+#define MAX_DELAY_TIME      (double) (65.0/1000.0)  // delay time in milliseconds
+
+#define MAX_AP_SAMPLES      (int)(((double) MAX_AP_TIME)    * ((double) _OUTPUT_SAMPLE_RATE))
+#define MAX_DELAY_SAMPLES   (int)(((double) MAX_DELAY_TIME) * ((double) _OUTPUT_SAMPLE_RATE))
+
+#define AP0_IN              0
+#define AP1_IN              (AP0_IN     + MAX_AP_SAMPLES    + GUARD)
+#define DELAY0_IN           (AP1_IN     + MAX_AP_SAMPLES    + GUARD)
+#define DELAY1_IN           (DELAY0_IN  + MAX_DELAY_SAMPLES + GUARD)
+
+// Define the max offsets for the end points of each section
+// i.e., we don't expect a given section's taps to go beyond
+// the following limits
+#define AP0_OUT             (AP0_IN     + MAX_AP_SAMPLES    -1)
+#define AP1_OUT             (AP1_IN     + MAX_AP_SAMPLES    -1)
+#define DELAY0_OUT          (DELAY0_IN  + MAX_DELAY_SAMPLES -1)
+#define DELAY1_OUT          (DELAY1_IN  + MAX_DELAY_SAMPLES -1)
+
+#define REVERB_DEFAULT_ROOM_NUMBER      1       // default preset number
+#define DEFAULT_AP0_LENGTH              (int)(((double) (17.0/1000.0))  * ((double) _OUTPUT_SAMPLE_RATE))
+#define DEFAULT_AP0_GAIN                19400
+#define DEFAULT_AP1_LENGTH              (int)(((double) (16.5/1000.0))  * ((double) _OUTPUT_SAMPLE_RATE))
+#define DEFAULT_AP1_GAIN                -19400
+
+#define REVERB_DEFAULT_WET              32767
+#define REVERB_DEFAULT_DRY              0
+
+#define EAS_REVERB_WET_MAX              32767
+#define EAS_REVERB_WET_MIN              0
+#define EAS_REVERB_DRY_MAX              32767
+#define EAS_REVERB_DRY_MIN              0
+
+/* parameters for each allpass */
+typedef struct
+{
+    EAS_U16             m_zApOut;       // delay offset for ap out
+
+    EAS_I16             m_nApGain;      // gain for ap
+
+    EAS_U16             m_zApIn;        // delay offset for ap in
+
+} S_ALLPASS_OBJECT;
+
+
+/* parameters for each allpass */
+typedef struct
+{
+    EAS_PCM             m_zLpf;                     // actual state variable, not a length
+
+    EAS_I16             m_nLpfFwd;                  // lpf forward gain
+
+    EAS_I16             m_nLpfFbk;                  // lpf feedback gain
+
+    EAS_U16             m_zDelay[REVERB_MAX_NUM_REFLECTIONS];   // delay offset for ap out
+
+    EAS_I16             m_nGain[REVERB_MAX_NUM_REFLECTIONS];    // gain for ap
+
+} S_EARLY_REFLECTION_OBJECT;
+
+//demo
+typedef struct
+{
+    EAS_I16             m_nLpfFbk;
+    EAS_I16             m_nLpfFwd;
+
+    EAS_I16             m_nEarly;
+    EAS_I16             m_nWet;
+    EAS_I16             m_nDry;
+
+    EAS_I16             m_nEarlyL_LpfFbk;
+    EAS_I16             m_nEarlyL_LpfFwd;
+
+    EAS_I16             m_nEarlyL_Delay0; //8
+    EAS_I16             m_nEarlyL_Gain0;
+    EAS_I16             m_nEarlyL_Delay1;
+    EAS_I16             m_nEarlyL_Gain1;
+    EAS_I16             m_nEarlyL_Delay2;
+    EAS_I16             m_nEarlyL_Gain2;
+    EAS_I16             m_nEarlyL_Delay3;
+    EAS_I16             m_nEarlyL_Gain3;
+    EAS_I16             m_nEarlyL_Delay4;
+    EAS_I16             m_nEarlyL_Gain4;
+
+    EAS_I16             m_nEarlyR_Delay0; //18
+    EAS_I16             m_nEarlyR_Gain0;
+    EAS_I16             m_nEarlyR_Delay1;
+    EAS_I16             m_nEarlyR_Gain1;
+    EAS_I16             m_nEarlyR_Delay2;
+    EAS_I16             m_nEarlyR_Gain2;
+    EAS_I16             m_nEarlyR_Delay3;
+    EAS_I16             m_nEarlyR_Gain3;
+    EAS_I16             m_nEarlyR_Delay4;
+    EAS_I16             m_nEarlyR_Gain4;
+
+    EAS_U16             m_nMaxExcursion; //28
+    EAS_I16             m_nXfadeInterval;
+
+    EAS_I16             m_nAp0_ApGain; //30
+    EAS_I16             m_nAp0_ApOut;
+    EAS_I16             m_nAp1_ApGain;
+    EAS_I16             m_nAp1_ApOut;
+
+    EAS_I16             m_rfu4;
+    EAS_I16             m_rfu5;
+    EAS_I16             m_rfu6;
+    EAS_I16             m_rfu7;
+    EAS_I16             m_rfu8;
+    EAS_I16             m_rfu9;
+    EAS_I16             m_rfu10; //43
+
+} S_REVERB_PRESET;
+
+typedef struct
+{
+    S_REVERB_PRESET     m_sPreset[REVERB_MAX_ROOM_TYPE];    //array of presets
+
+} S_REVERB_PRESET_BANK;
+
+/* parameters for each reverb */
+typedef struct
+{
+    /* controls entire reverb playback volume */
+    /* to conserve memory, use the MSB and ignore the LSB */
+    EAS_U8              m_nMasterVolume;
+
+    /* update counter keeps track of when synth params need updating */
+    /* only needs to be as large as REVERB_UPDATE_PERIOD_IN_SAMPLES */
+    EAS_I16             m_nUpdateCounter;
+
+    EAS_U16             m_nMinSamplesToAdd;         /* ComputeReverb() generates this many samples */
+
+    EAS_U8              m_nFlags;                   /* misc flags/bit fields */
+
+    EAS_PCM             *m_pOutputBuffer;
+    EAS_PCM             *m_pInputBuffer;
+
+    EAS_U16             m_nNumSamplesInOutputBuffer;
+    EAS_U16             m_nNumSamplesInInputBuffer;
+
+    EAS_U16             m_nNumInputSamplesRead;     // if m_nNumInputSamplesRead >= NumSamplesInInputBuffer
+                                                    // then get a new input buffer
+    EAS_PCM             *m_pNextInputSample;
+
+    EAS_U16             m_nBaseIndex;                                   // base index for circular buffer
+
+    // reverb delay line offsets, allpass parameters, etc:
+
+    EAS_PCM             m_nRevOutFbkR;              // combine feedback reverb right out with dry left in
+
+    S_ALLPASS_OBJECT    m_sAp0;                     // allpass 0 (left channel)
+
+    EAS_U16             m_zD0In;                    // delay offset for delay line D0 in
+
+    EAS_PCM             m_nRevOutFbkL;              // combine feedback reverb left out with dry right in
+
+    S_ALLPASS_OBJECT    m_sAp1;                     // allpass 1 (right channel)
+
+    EAS_U16             m_zD1In;                    // delay offset for delay line D1 in
+
+    // delay output taps, notice criss cross order
+    EAS_U16             m_zD0Self;                  // self feeds forward d0 --> d0
+
+    EAS_U16             m_zD1Cross;                 // cross feeds across d1 --> d0
+
+    EAS_PCM             m_zLpf0;                    // actual state variable, not a length
+
+    EAS_U16             m_zD1Self;                  // self feeds forward d1 --> d1
+
+    EAS_U16             m_zD0Cross;                 // cross feeds across d0 --> d1
+
+    EAS_PCM             m_zLpf1;                    // actual state variable, not a length
+
+    EAS_I16             m_nSin;                     // gain for self taps
+
+    EAS_I16             m_nCos;                     // gain for cross taps
+
+    EAS_I16             m_nSinIncrement;            // increment for gain
+
+    EAS_I16             m_nCosIncrement;            // increment for gain
+
+    EAS_I16             m_nLpfFwd;                  // lpf forward gain (includes scaling for mixer)
+
+    EAS_I16             m_nLpfFbk;                  // lpf feedback gain
+
+    EAS_U16             m_nXfadeInterval;           // update/xfade after this many samples
+
+    EAS_U16             m_nXfadeCounter;            // keep track of when to xfade
+
+    EAS_I16             m_nPhase;                   // -1 <= m_nPhase < 1
+                                                    // but during sin,cos calculations
+                                                    // use m_nPhase/2
+
+    EAS_I16             m_nPhaseIncrement;          // add this to m_nPhase each frame
+
+    EAS_I16             m_nNoise;                   // random noise sample
+
+    EAS_U16             m_nMaxExcursion;            // the taps can excurse +/- this amount
+
+    EAS_BOOL            m_bUseNoise;                // if EAS_TRUE, use noise as input signal
+
+    EAS_BOOL            m_bBypass;                  // if EAS_TRUE, then bypass reverb and copy input to output
+
+    EAS_I16             m_nCurrentRoom;             // preset number for current room
+
+    EAS_I16             m_nNextRoom;                // preset number for next room
+
+    EAS_I16             m_nWet;                     // gain for wet (processed) signal
+
+    EAS_I16             m_nDry;                     // gain for dry (unprocessed) signal
+
+    EAS_I16             m_nEarly;                   // gain for early (widen) signal
+
+    S_EARLY_REFLECTION_OBJECT   m_sEarlyL;          // left channel early reflections
+    S_EARLY_REFLECTION_OBJECT   m_sEarlyR;          // right channel early reflections
+
+    EAS_PCM             m_nDelayLine[REVERB_BUFFER_SIZE_IN_SAMPLES];    // one large delay line for all reverb elements
+
+    S_REVERB_PRESET     pPreset;
+
+    S_REVERB_PRESET_BANK    m_sPreset;
+
+    //EAS_I8            preset;
+
+} S_REVERB_OBJECT;
+
+
+/*------------------------------------
+ * prototypes
+ *------------------------------------
+*/
+
+/*----------------------------------------------------------------------------
+ * ReverbUpdateXfade
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Update the xfade parameters as required
+ *
+ * Inputs:
+ * nNumSamplesToAdd - number of samples to write to buffer
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ * - xfade parameters will be changed
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT ReverbUpdateXfade(S_REVERB_OBJECT* pReverbData, EAS_INT nNumSamplesToAdd);
+
+/*----------------------------------------------------------------------------
+ * ReverbCalculateNoise
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Calculate a noise sample and limit its value
+ *
+ * Inputs:
+ * nMaxExcursion - noise value is limited to this value
+ * pnNoise - return new noise sample in this (not limited)
+ *
+ * Outputs:
+ * new limited noise value
+ *
+ * Side Effects:
+ * - *pnNoise noise value is updated
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_U16 ReverbCalculateNoise(EAS_U16 nMaxExcursion, EAS_I16 *pnNoise);
+
+/*----------------------------------------------------------------------------
+ * ReverbCalculateSinCos
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Calculate a new sin and cosine value based on the given phase
+ *
+ * Inputs:
+ * nPhase   - phase angle
+ * pnSin    - input old value, output new value
+ * pnCos    - input old value, output new value
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ * - *pnSin, *pnCos are updated
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT ReverbCalculateSinCos(EAS_I16 nPhase, EAS_I16 *pnSin, EAS_I16 *pnCos);
+
+/*----------------------------------------------------------------------------
+ * Reverb
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * apply reverb to the given signal
+ *
+ * Inputs:
+ * nNu
+ * pnSin    - input old value, output new value
+ * pnCos    - input old value, output new value
+ *
+ * Outputs:
+ * number of samples actually reverberated
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT Reverb(S_REVERB_OBJECT* pReverbData, EAS_INT nNumSamplesToAdd, EAS_PCM *pOutputBuffer, EAS_PCM *pInputBuffer);
+
+/*----------------------------------------------------------------------------
+ * ReverbReadInPresets()
+ *----------------------------------------------------------------------------
+ * Purpose: sets global reverb preset bank to defaults
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT ReverbReadInPresets(S_REVERB_OBJECT* pReverbData);
+
+
+/*----------------------------------------------------------------------------
+ * ReverbUpdateRoom
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Update the room's preset parameters as required
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ * - reverb paramters (fbk, fwd, etc) will be changed
+ * - m_nCurrentRoom := m_nNextRoom
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT ReverbUpdateRoom(S_REVERB_OBJECT* pReverbData);
+
+#endif /* #ifndef _EAS_REVERBDATA_H */
+
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_rtttldata.c

@@ -0,0 +1,41 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_rtttldata.c
+ *
+ * Contents and purpose:
+ * RTTTL File Parser data module for static memory models
+ *
+ * Copyright Sonic Network Inc. 2005
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 547 $
+ *   $Date: 2007-01-31 16:30:17 -0800 (Wed, 31 Jan 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#include "eas_types.h"
+#include "eas_rtttldata.h"
+
+/*----------------------------------------------------------------------------
+ *
+ * eas_RTTTLData
+ *
+ * Static memory allocation for RTTTL parser
+ *----------------------------------------------------------------------------
+*/
+S_RTTTL_DATA eas_RTTTLData;
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_rtttldata.h

@@ -0,0 +1,70 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_rtttldata.h
+ *
+ * Contents and purpose:
+ * SMF File Parser
+ *
+ * This file contains data declarations for the RTTTL parser.
+ *
+ * Copyright Sonic Network Inc. 2005
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 82 $
+ *   $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef EAS_RTTTLDATA_H
+#define EAS_RTTTLDATA_H
+
+#include "eas_data.h"
+
+
+/* maximum line size as specified in iMelody V1.2 spec */
+#define MAX_LINE_SIZE           75
+
+/*----------------------------------------------------------------------------
+ *
+ * S_RTTTL_DATA
+ *
+ * This structure contains the state data for the iMelody parser
+ *----------------------------------------------------------------------------
+*/
+
+typedef struct
+{
+    EAS_FILE_HANDLE fileHandle;             /* file handle */
+    S_SYNTH *pSynth;                    /* synthesizer handle */
+    S_METADATA_CB   metadata;               /* metadata callback */
+    EAS_I32     fileOffset;                 /* offset to start of data */
+    EAS_I32     time;                       /* current time in 256ths of a msec */
+    EAS_I32     tick;                       /* length of 32nd note in 256th of a msec */
+    EAS_I32     restTicks;                  /* ticks to rest after current note */
+    EAS_I32     repeatOffset;               /* file offset to start of repeat section */
+    EAS_U8      repeatCount;                /* repeat counter */
+    EAS_I8      dataByte;                   /* storage for characters that are "put back" */
+    EAS_U8      state;                      /* current state EAS_STATE_XXXX */
+    EAS_I8      style;                      /* from STYLE */
+    EAS_U8      note;                       /* MIDI note number */
+    EAS_U8      octave;                     /* decault octave prefix */
+    EAS_I8      duration;                   /* default note duration */
+} S_RTTTL_DATA;
+
+#endif
+
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_smf.h

@@ -0,0 +1,49 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_smf.h
+ *
+ * Contents and purpose:
+ * SMF Type 0 and 1 File Parser
+ *
+ * Copyright Sonic Network Inc. 2005
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 82 $
+ *   $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_SMF_H
+#define _EAS_SMF_H
+
+/* prototypes for private interface to SMF parser */
+EAS_RESULT SMF_CheckFileType (S_EAS_DATA *pEASData, EAS_FILE_HANDLE fileHandle, EAS_VOID_PTR *ppHandle, EAS_I32 offset);
+EAS_RESULT SMF_Prepare (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData);
+EAS_RESULT SMF_Time (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_U32 *pTime);
+EAS_RESULT SMF_Event (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_INT parserMode);
+EAS_RESULT SMF_State (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_STATE *pState);
+EAS_RESULT SMF_Close (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData);
+EAS_RESULT SMF_Reset (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData);
+EAS_RESULT SMF_Pause (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData);
+EAS_RESULT SMF_Resume (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData);
+EAS_RESULT SMF_SetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 value);
+EAS_RESULT SMF_GetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 *pValue);
+EAS_RESULT SMF_ParseHeader (EAS_HW_DATA_HANDLE hwInstData, S_SMF_DATA *pSMFData);
+
+#endif /* end _EAS_SMF_H */
+
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_smfdata.c

@@ -0,0 +1,66 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_smfdata.c
+ *
+ * Contents and purpose:
+ * SMF File Parser
+ *
+ * This file contains data definitions for the SMF parser.
+ *
+ * Copyright Sonic Network Inc. 2005
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 778 $
+ *   $Date: 2007-07-23 16:45:17 -0700 (Mon, 23 Jul 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#include "eas_miditypes.h"
+#include "eas_smfdata.h"
+
+/*----------------------------------------------------------------------------
+ *
+ * S_SMF_STREAM
+ *
+ * Static memory allocation for SMF parser
+ *----------------------------------------------------------------------------
+*/
+static S_SMF_STREAM eas_SMFStreams[MAX_SMF_STREAMS];
+
+/*----------------------------------------------------------------------------
+ *
+ * eas_SMFData
+ *
+ * Static memory allocation for SMF parser
+ *----------------------------------------------------------------------------
+*/
+S_SMF_DATA eas_SMFData =
+{
+    eas_SMFStreams,     /* pointer to individual streams in file */
+    0,                  /* pointer to next stream with event */
+    0,                  /* pointer to synth */
+    0,                  /* file handle */
+    { 0, 0, 0, 0},      /* metadata callback */
+    0,                  /* file offset */
+    0,                  /* current time in milliseconds/256 */
+    0,                  /* actual number of streams */
+    0,                  /* current MIDI tick to msec conversion */
+    0,                  /* ticks per quarter note */
+    0,                  /* current state EAS_STATE_XXXX */
+    0                   /* flags */
+};
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_smfdata.h

@@ -0,0 +1,66 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_smfdata.h
+ *
+ * Contents and purpose:
+ * SMF File Parser
+ *
+ * This file contains data definitions for the SMF parser.
+ *
+ * Copyright Sonic Network Inc. 2005
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 686 $
+ *   $Date: 2007-05-03 14:10:54 -0700 (Thu, 03 May 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_SMF_DATA_H
+#define _EAS_SMF_DATA_H
+
+#ifndef MAX_SMF_STREAMS
+#define MAX_SMF_STREAMS             128
+#endif
+
+/* offsets in to the SMF file */
+#define SMF_OFS_HEADER_SIZE         4
+#define SMF_OFS_FILE_TYPE           8
+#define SMF_OFS_NUM_TRACKS          10
+
+/* size of chunk info (chunk ID + chunk size) */
+#define SMF_CHUNK_INFO_SIZE         8
+
+/* 'MTrk' track chunk ID */
+#define SMF_CHUNK_TYPE_TRACK        0x4d54726bL
+
+/* some useful meta-events */
+#define SMF_META_TEXT               0x01
+#define SMF_META_COPYRIGHT          0x02
+#define SMF_META_SEQTRK_NAME        0x03
+#define SMF_META_LYRIC              0x05
+#define SMF_META_END_OF_TRACK       0x2f
+#define SMF_META_TEMPO              0x51
+#define SMF_META_TIME_SIGNATURE     0x58
+
+/* default timebase (120BPM) */
+#define SMF_DEFAULT_TIMEBASE        500000L
+
+/* value for pSMFStream->ticks to signify end of track */
+#define SMF_END_OF_TRACK            0xffffffff
+
+#endif
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_sndlib.h

@@ -0,0 +1,406 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_sndlib.h
+ *
+ * Contents and purpose:
+ * Declarations for the sound library
+ *
+ * Copyright Sonic Network Inc. 2005
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 550 $
+ *   $Date: 2007-02-02 09:37:03 -0800 (Fri, 02 Feb 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_SNDLIB_H
+#define _EAS_SNDLIB_H
+
+#include "eas_types.h"
+#include "eas_synthcfg.h"
+
+#ifdef _WT_SYNTH
+#include "eas_wtengine.h"
+#endif
+
+/*----------------------------------------------------------------------------
+ * This is bit of a hack to allow us to keep the same structure
+ * declarations for the DLS parser. Normally, the data is located
+ * in read-only memory, but for DLS, we store the data in RW
+ * memory.
+ *----------------------------------------------------------------------------
+*/
+#ifndef SCNST
+#define SCNST const
+#endif
+
+/*----------------------------------------------------------------------------
+ * sample size
+ *----------------------------------------------------------------------------
+*/
+#ifdef _16_BIT_SAMPLES
+typedef EAS_I16 EAS_SAMPLE;
+#else
+typedef EAS_I8 EAS_SAMPLE;
+#endif
+
+/*----------------------------------------------------------------------------
+ * EAS Library ID - quick check for valid library and version
+ *----------------------------------------------------------------------------
+*/
+#define _EAS_LIBRARY_VERSION        0x01534145
+
+#define NUM_PROGRAMS_IN_BANK        128
+#define INVALID_REGION_INDEX        0xffff
+
+/* this bit in region index indicates that region is for secondary synth */
+#define FLAG_RGN_IDX_FM_SYNTH       0x8000
+#define FLAG_RGN_IDX_DLS_SYNTH      0x4000
+#define REGION_INDEX_MASK           0x3fff
+
+/*----------------------------------------------------------------------------
+ * Generic region data structure
+ *
+ * This must be the first element in each region structure
+ *----------------------------------------------------------------------------
+*/
+typedef struct s_region_tag
+{
+    EAS_U16     keyGroupAndFlags;
+    EAS_U8      rangeLow;
+    EAS_U8      rangeHigh;
+} S_REGION;
+
+/*
+ * Bit fields for m_nKeyGroupAndFlags
+ * Bits 0-2 are mode bits in FM synth
+ * Bits 8-11 are the key group
+ */
+#define REGION_FLAG_IS_LOOPED                   0x01
+#define REGION_FLAG_USE_WAVE_GENERATOR          0x02
+#define REGION_FLAG_USE_ADPCM                   0x04
+#define REGION_FLAG_ONE_SHOT                    0x08
+#define REGION_FLAG_SQUARE_WAVE                 0x10
+#define REGION_FLAG_OFF_CHIP                    0x20
+#define REGION_FLAG_NON_SELF_EXCLUSIVE          0x40
+#define REGION_FLAG_LAST_REGION                 0x8000
+
+/*----------------------------------------------------------------------------
+ * Envelope data structure
+ *----------------------------------------------------------------------------
+*/
+typedef struct s_envelope_tag
+{
+    EAS_I16     attackTime;
+    EAS_I16     decayTime;
+    EAS_I16     sustainLevel;
+    EAS_I16     releaseTime;
+} S_ENVELOPE;
+
+/*----------------------------------------------------------------------------
+ * DLS envelope data structure
+ *----------------------------------------------------------------------------
+*/
+typedef struct s_dls_envelope_tag
+{
+    EAS_I16         delayTime;
+    EAS_I16         attackTime;
+    EAS_I16         holdTime;
+    EAS_I16         decayTime;
+    EAS_I16         sustainLevel;
+    EAS_I16         releaseTime;
+    EAS_I16         velToAttack;
+    EAS_I16         keyNumToDecay;
+    EAS_I16         keyNumToHold;
+} S_DLS_ENVELOPE;
+
+/*----------------------------------------------------------------------------
+ * LFO data structure
+ *----------------------------------------------------------------------------
+*/
+typedef struct s_lfo_params_tag
+{
+    EAS_I16     lfoFreq;
+    EAS_I16     lfoDelay;
+} S_LFO_PARAMS;
+
+/*----------------------------------------------------------------------------
+ * Articulation data structure
+ *----------------------------------------------------------------------------
+*/
+typedef struct s_articulation_tag
+{
+    S_ENVELOPE  eg1;
+    S_ENVELOPE  eg2;
+    EAS_I16     lfoToPitch;
+    EAS_I16     lfoDelay;
+    EAS_I16     lfoFreq;
+    EAS_I16     eg2ToPitch;
+    EAS_I16     eg2ToFc;
+    EAS_I16     filterCutoff;
+    EAS_I8      lfoToGain;
+    EAS_U8      filterQ;
+    EAS_I8      pan;
+} S_ARTICULATION;
+
+/*----------------------------------------------------------------------------
+ * DLS articulation data structure
+ *----------------------------------------------------------------------------
+*/
+
+typedef struct s_dls_articulation_tag
+{
+    S_LFO_PARAMS    modLFO;
+    S_LFO_PARAMS    vibLFO;
+
+    S_DLS_ENVELOPE  eg1;
+    S_DLS_ENVELOPE  eg2;
+
+    EAS_I16         eg1ShutdownTime;
+
+    EAS_I16         filterCutoff;
+    EAS_I16         modLFOToFc;
+    EAS_I16         modLFOCC1ToFc;
+    EAS_I16         modLFOChanPressToFc;
+    EAS_I16         eg2ToFc;
+    EAS_I16         velToFc;
+    EAS_I16         keyNumToFc;
+
+    EAS_I16         modLFOToGain;
+    EAS_I16         modLFOCC1ToGain;
+    EAS_I16         modLFOChanPressToGain;
+
+    EAS_I16         tuning;
+    EAS_I16         keyNumToPitch;
+    EAS_I16         vibLFOToPitch;
+    EAS_I16         vibLFOCC1ToPitch;
+    EAS_I16         vibLFOChanPressToPitch;
+    EAS_I16         modLFOToPitch;
+    EAS_I16         modLFOCC1ToPitch;
+    EAS_I16         modLFOChanPressToPitch;
+    EAS_I16         eg2ToPitch;
+
+    /* pad to 4-byte boundary */
+    EAS_U16         pad;
+
+    EAS_I8          pan;
+    EAS_U8          filterQandFlags;
+
+#ifdef _REVERB
+    EAS_I16         reverbSend;
+    EAS_I16         cc91ToReverbSend;
+#endif
+
+#ifdef _CHORUS
+    EAS_I16         chorusSend;
+    EAS_I16         cc93ToChorusSend;
+#endif
+} S_DLS_ARTICULATION;
+
+/* flags in filterQandFlags
+ * NOTE: Q is stored in bottom 5 bits
+ */
+#define FLAG_DLS_VELOCITY_SENSITIVE     0x80
+#define FILTER_Q_MASK                   0x1f
+
+/*----------------------------------------------------------------------------
+ * Wavetable region data structure
+ *----------------------------------------------------------------------------
+*/
+typedef struct s_wt_region_tag
+{
+    S_REGION    region;
+    EAS_I16     tuning;
+    EAS_I16     gain;
+    EAS_U32     loopStart;
+    EAS_U32     loopEnd;
+    EAS_U16     waveIndex;
+    EAS_U16     artIndex;
+} S_WT_REGION;
+
+/*----------------------------------------------------------------------------
+ * DLS region data structure
+ *----------------------------------------------------------------------------
+*/
+typedef struct s_dls_region_tag
+{
+    S_WT_REGION     wtRegion;
+    EAS_U8          velLow;
+    EAS_U8          velHigh;
+} S_DLS_REGION;
+
+/*----------------------------------------------------------------------------
+ * FM synthesizer data structures
+ *----------------------------------------------------------------------------
+*/
+typedef struct s_fm_oper_tag
+{
+    EAS_I16     tuning;
+    EAS_U8      attackDecay;
+    EAS_U8      velocityRelease;
+    EAS_U8      egKeyScale;
+    EAS_U8      sustain;
+    EAS_U8      gain;
+    EAS_U8      flags;
+} S_FM_OPER;
+
+/* defines for S_FM_OPER.m_nFlags */
+#define FM_OPER_FLAG_MONOTONE           0x01
+#define FM_OPER_FLAG_NO_VIBRATO         0x02
+#define FM_OPER_FLAG_NOISE              0x04
+#define FM_OPER_FLAG_LINEAR_VELOCITY    0x08
+
+/* NOTE: The first two structure elements are common with S_WT_REGION
+ * and we will rely on that in the voice management code and must
+ * remain there unless the voice management code is revisited.
+ */
+typedef struct s_fm_region_tag
+{
+    S_REGION        region;
+    EAS_U8          vibTrem;
+    EAS_U8          lfoFreqDelay;
+    EAS_U8          feedback;
+    EAS_I8          pan;
+    S_FM_OPER       oper[4];
+} S_FM_REGION;
+
+/*----------------------------------------------------------------------------
+ * Common data structures
+ *----------------------------------------------------------------------------
+*/
+
+/*----------------------------------------------------------------------------
+ * Program data structure
+ * Used for individual programs not stored as a complete bank.
+ *----------------------------------------------------------------------------
+*/
+typedef struct s_program_tag
+{
+    EAS_U32 locale;
+    EAS_U16 regionIndex;
+} S_PROGRAM;
+
+/*----------------------------------------------------------------------------
+ * Bank data structure
+ *
+ * A bank always consists of 128 programs. If a bank is less than 128
+ * programs, it should be stored as a spare matrix in the pPrograms
+ * array.
+ *
+ * bankNum:     MSB/LSB of MIDI bank select controller
+ * regionIndex: Index of first region in program
+ *----------------------------------------------------------------------------
+*/
+typedef struct s_bank_tag
+{
+    EAS_U16 locale;
+    EAS_U16 regionIndex[NUM_PROGRAMS_IN_BANK];
+} S_BANK;
+
+
+/* defines for libFormat field
+ * bits 0-17 are the sample rate
+ * bit 18 is true if wavetable is present
+ * bit 19 is true if FM is present
+ * bit 20 is true if filter is enabled
+ * bit 21 is sample depth (0 = 8-bits, 1 = 16-bits)
+ * bits 22-31 are reserved
+ */
+#define LIBFORMAT_SAMPLE_RATE_MASK      0x0003ffff
+#define LIB_FORMAT_TYPE_MASK            0x000c0000
+#define LIB_FORMAT_WAVETABLE            0x00000000
+#define LIB_FORMAT_FM                   0x00040000
+#define LIB_FORMAT_HYBRID               0x00080000
+#define LIB_FORMAT_FILTER_ENABLED       0x00100000
+#define LIB_FORMAT_16_BIT_SAMPLES       0x00200000
+
+#ifdef DLS_SYNTHESIZER
+/*----------------------------------------------------------------------------
+ * DLS data structure
+ *
+ * pDLSPrograms         pointer to array of DLS programs
+ * pDLSRegions          pointer to array of DLS regions
+ * pDLSArticulations    pointer to array of DLS articulations
+ * pSampleLen           pointer to array of sample lengths
+ * ppSamples            pointer to array of sample pointers
+ * numDLSPrograms       number of DLS programs
+ * numDLSRegions        number of DLS regions
+ * numDLSArticulations  number of DLS articulations
+ * numDLSSamples        number of DLS samples
+ *----------------------------------------------------------------------------
+*/
+typedef struct s_eas_dls_tag
+{
+    S_PROGRAM           *pDLSPrograms;
+    S_DLS_REGION        *pDLSRegions;
+    S_DLS_ARTICULATION  *pDLSArticulations;
+    EAS_U32             *pDLSSampleLen;
+    EAS_U32             *pDLSSampleOffsets;
+    EAS_SAMPLE          *pDLSSamples;
+    EAS_U16             numDLSPrograms;
+    EAS_U16             numDLSRegions;
+    EAS_U16             numDLSArticulations;
+    EAS_U16             numDLSSamples;
+    EAS_U8              refCount;
+} S_DLS;
+#endif
+
+/*----------------------------------------------------------------------------
+ * Sound library data structure
+ *
+ * pBanks           pointer to array of banks
+ * pPrograms        pointer to array of programs
+ * pWTRegions       pointer to array of wavetable regions
+ * pFMRegions       pointer to array of FM regions
+ * pArticulations   pointer to array of articulations
+ * pSampleLen       pointer to array of sample lengths
+ * ppSamples        pointer to array of sample pointers
+ * numBanks         number of banks
+ * numPrograms      number of individual program
+ * numRegions       number of regions
+ * numArticulations number of articulations
+ * numSamples       number of samples
+ *----------------------------------------------------------------------------
+*/
+typedef struct s_eas_sndlib_tag
+{
+    SCNST EAS_U32               identifier;
+    SCNST EAS_U32               libAttr;
+
+    SCNST S_BANK                *pBanks;
+    SCNST S_PROGRAM             *pPrograms;
+
+    SCNST S_WT_REGION           *pWTRegions;
+    SCNST S_ARTICULATION        *pArticulations;
+    SCNST EAS_U32               *pSampleLen;
+    SCNST EAS_U32               *pSampleOffsets;
+    SCNST EAS_SAMPLE            *pSamples;
+
+    SCNST S_FM_REGION           *pFMRegions;
+
+    SCNST EAS_U16               numBanks;
+    SCNST EAS_U16               numPrograms;
+
+    SCNST EAS_U16               numWTRegions;
+    SCNST EAS_U16               numArticulations;
+    SCNST EAS_U16               numSamples;
+
+    SCNST EAS_U16               numFMRegions;
+} S_EAS;
+
+#endif
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_synth.h

@@ -0,0 +1,395 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_synth.h
+ *
+ * Contents and purpose:
+ * Declarations, interfaces, and prototypes for synth.
+ *
+ * Copyright Sonic Network Inc. 2004, 2005
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 718 $
+ *   $Date: 2007-06-08 16:43:16 -0700 (Fri, 08 Jun 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_SYNTH_H
+#define _EAS_SYNTH_H
+
+#include "eas_types.h"
+#include "eas_sndlib.h"
+
+#ifdef _WT_SYNTH
+#include "eas_wtsynth.h"
+#endif
+
+#ifdef _FM_SYNTH
+#include "eas_fmsynth.h"
+#endif
+
+#ifndef NUM_OUTPUT_CHANNELS
+#define NUM_OUTPUT_CHANNELS         2
+#endif
+
+#ifndef MAX_SYNTH_VOICES
+#define MAX_SYNTH_VOICES            64
+#endif
+
+#ifndef MAX_VIRTUAL_SYNTHESIZERS
+#define MAX_VIRTUAL_SYNTHESIZERS    4
+#endif
+
+/* defines */
+#ifndef NUM_PRIMARY_VOICES
+#define NUM_PRIMARY_VOICES      MAX_SYNTH_VOICES
+#elif !defined(NUM_SECONDARY_VOICES)
+#define NUM_SECONDARY_VOICES    (MAX_SYNTH_VOICES - NUM_PRIMARY_VOICES)
+#endif
+
+#if defined(EAS_WT_SYNTH)
+#define NUM_WT_VOICES           MAX_SYNTH_VOICES
+
+/* FM on MCU */
+#elif defined(EAS_FM_SYNTH)
+#define NUM_FM_VOICES           MAX_SYNTH_VOICES
+
+/* wavetable drums on MCU, wavetable melodic on DSP */
+#elif defined(EAS_SPLIT_WT_SYNTH)
+#define NUM_WT_VOICES           MAX_SYNTH_VOICES
+
+/* wavetable drums and FM melodic on MCU */
+#elif defined(EAS_HYBRID_SYNTH)
+#define NUM_WT_VOICES           NUM_PRIMARY_VOICES
+#define NUM_FM_VOICES           NUM_SECONDARY_VOICES
+
+/* wavetable drums on MCU, FM melodic on DSP */
+#elif defined(EAS_SPLIT_HYBRID_SYNTH)
+#define NUM_WT_VOICES           NUM_PRIMARY_VOICES
+#define NUM_FM_VOICES           NUM_SECONDARY_VOICES
+
+/* FM synth on DSP */
+#elif defined(EAS_SPLIT_FM_SYNTH)
+#define NUM_FM_VOICES           MAX_SYNTH_VOICES
+
+#else
+#error "Unrecognized architecture option"
+#endif
+
+#define NUM_SYNTH_CHANNELS      16
+
+#define DEFAULT_SYNTH_VOICES    MAX_SYNTH_VOICES
+
+/* use the following values to specify unassigned channels or voices */
+#define UNASSIGNED_SYNTH_CHANNEL    NUM_SYNTH_CHANNELS
+#define UNASSIGNED_SYNTH_VOICE      MAX_SYNTH_VOICES
+
+
+/* synth parameters are updated every SYNTH_UPDATE_PERIOD_IN_SAMPLES */
+#define SYNTH_UPDATE_PERIOD_IN_SAMPLES  (EAS_I32)(0x1L << SYNTH_UPDATE_PERIOD_IN_BITS)
+
+/* stealing weighting factors */
+#define NOTE_AGE_STEAL_WEIGHT           1
+#define NOTE_GAIN_STEAL_WEIGHT          4
+#define CHANNEL_POLY_STEAL_WEIGHT       12
+#define CHANNEL_PRIORITY_STEAL_WEIGHT   2
+#define NOTE_MATCH_PENALTY              128
+#define SYNTH_PRIORITY_WEIGHT           8
+
+/* default synth master volume */
+#define DEFAULT_SYNTH_MASTER_VOLUME     0x7fff
+
+#define DEFAULT_SYNTH_PRIORITY          5
+
+/* default tuning values */
+#define DEFAULT_PITCH_BEND_SENSITIVITY      200     /* 2 semitones */
+#define DEFAULT_FINE_PITCH                  0       /* 0 cents */
+#define DEFAULT_COARSE_PITCH                0       /* 0 semitones */
+
+/* default drum channel is 10, but is internally 9 due to unit offset */
+#define DEFAULT_DRUM_CHANNEL            9
+
+/* drum channel can simultaneously play this many voices at most */
+#define DEFAULT_CHANNEL_POLYPHONY_LIMIT 2
+
+/* default instrument is acoustic piano */
+#define DEFAULT_MELODY_BANK_MSB         0x79
+#define DEFAULT_RHYTHM_BANK_MSB         0x78
+#define DEFAULT_MELODY_BANK_NUMBER      (DEFAULT_MELODY_BANK_MSB << 8)
+#define DEFAULT_RHYTHM_BANK_NUMBER      (DEFAULT_RHYTHM_BANK_MSB << 8)
+#define DEFAULT_SYNTH_PROGRAM_NUMBER    0
+
+#define DEFAULT_PITCH_BEND      0x2000  /* 0x2000 == (0x40 << 7) | 0x00 */
+#define DEFAULT_MOD_WHEEL       0
+#define DEFAULT_CHANNEL_VOLUME  0x64
+#define DEFAULT_PAN             0x40    /* decimal 64, center */
+
+#ifdef _REVERB
+#define DEFAULT_REVERB_SEND     40      /* some reverb */
+#endif
+
+#ifdef _CHORUS
+#define DEFAULT_CHORUS_SEND     0       /* no chorus */
+#endif
+
+#define DEFAULT_EAS_FILTER_CUTOFF_FREQUENCY 0           /* EAS synth uses a different default */
+#define DEFAULT_FILTER_RESONANCE        0
+#define DEFAULT_EXPRESSION              0x7F
+
+#define DEFAULT_CHANNEL_PRESSURE        0
+
+#define DEFAULT_REGISTERED_PARAM        0x3FFF
+
+#define DEFAULT_CHANNEL_STATIC_GAIN     0
+#define DEFAULT_CHANNEL_STATIC_PITCH    0
+
+#define DEFAULT_LFO_MOD_WHEEL_TO_PITCH_CENTS            50
+#define DEFAULT_LFO_CHANNEL_PRESSURE_TO_PITCH_CENTS     50
+
+#define DEFAULT_KEY_NUMBER          0x69
+#define DEFAULT_VELOCITY            0x64
+#define DEFAULT_REGION_INDEX        0
+#define DEFAULT_ARTICULATION_INDEX  0
+#define DEFAULT_VOICE_GAIN          0
+#define DEFAULT_AGE                 0
+#define DEFAULT_SP_MIDI_PRIORITY    16
+
+
+/* filter defines */
+#define DEFAULT_FILTER_ZERO     0
+#define FILTER_CUTOFF_MAX_PITCH_CENTS       1919
+#define FILTER_CUTOFF_MIN_PITCH_CENTS       -4467
+#define A5_PITCH_OFFSET_IN_CENTS            6900
+
+/*------------------------------------
+ * S_SYNTH_CHANNEL data structure
+ *------------------------------------
+*/
+
+/* S_SYNTH_CHANNEL.m_nFlags */
+#define CHANNEL_FLAG_SUSTAIN_PEDAL                      0x01
+#define CHANNEL_FLAG_MUTE                               0x02
+#define CHANNEL_FLAG_UPDATE_CHANNEL_PARAMETERS          0x04
+#define CHANNEL_FLAG_RHYTHM_CHANNEL                     0x08
+#define CHANNEL_FLAG_EXTERNAL_AUDIO                     0x10
+#define DEFAULT_CHANNEL_FLAGS                           0
+
+/* macros for extracting virtual synth and channel numbers */
+#define GET_VSYNTH(a) ((a) >> 4)
+#define GET_CHANNEL(a) ((a) & 15)
+
+typedef struct s_synth_channel_tag
+{
+    /* use static channel parameters to reduce MIPs */
+    /* parameters shared by multiple voices assigned to same channel */
+    EAS_I32     staticPitch;        /* (pitch bend * pitch sens) + fine pitch */
+    EAS_I16     staticGain;         /* (CC7 * CC11 * master vol)^2  */
+
+    EAS_U16     regionIndex;        /* index of first region in program */
+
+    EAS_U16     bankNum;            /* play programs from this bank */
+    EAS_I16     pitchBend;          /* pitch wheel value */
+    EAS_I16     pitchBendSensitivity;
+    EAS_I16     registeredParam;    /* currently selected registered param */
+
+
+#if defined(_FM_SYNTH)
+    EAS_I16     lfoAmt;             /* amount of LFO to apply to voice */
+#endif
+
+    EAS_U8      programNum;         /* play this instrument number */
+    EAS_U8      modWheel;           /* CC1 */
+    EAS_U8      volume;             /* CC7 */
+    EAS_U8      pan;                /* CC10 */
+
+    EAS_U8      expression;         /* CC11 */
+
+    /* the following parameters are controlled by RPNs */
+    EAS_I8      finePitch;
+    EAS_I8      coarsePitch;
+
+    EAS_U8      channelPressure;    /* applied to all voices on a given channel */
+
+    EAS_U8      channelFlags;       /* bit field channelFlags for */
+                                    /* CC64, SP-MIDI channel masking */
+
+    EAS_U8      pool;               /* SPMIDI channel voice pool */
+    EAS_U8      mip;                /* SPMIDI MIP setting */
+
+#ifdef  _REVERB
+    EAS_U8      reverbSend;         /* CC91 */
+#endif
+
+#ifdef  _CHORUS
+    EAS_U8      chorusSend;         /* CC93 */
+#endif
+} S_SYNTH_CHANNEL;
+
+/*------------------------------------
+ * S_SYNTH_VOICE data structure
+ *------------------------------------
+*/
+
+/* S_SYNTH_VOICE.m_nFlags */
+#define VOICE_FLAG_UPDATE_VOICE_PARAMETERS              0x01
+#define VOICE_FLAG_SUSTAIN_PEDAL_DEFER_NOTE_OFF         0x02
+#define VOICE_FLAG_DEFER_MIDI_NOTE_OFF                  0x04
+#define VOICE_FLAG_NO_SAMPLES_SYNTHESIZED_YET           0x08
+#define VOICE_FLAG_DEFER_MUTE                           0x40
+#define DEFAULT_VOICE_FLAGS                             0
+
+/* S_SYNTH_VOICE.m_eState */
+typedef enum {
+
+    eVoiceStateFree = 0,
+    eVoiceStateStart,
+    eVoiceStatePlay,
+    eVoiceStateRelease,
+    eVoiceStateMuting,
+    eVoiceStateStolen,
+    eVoiceStateInvalid          /* should never be in this state! */
+
+} E_VOICE_STATE;
+#define DEFAULT_VOICE_STATE     eVoiceStateFree
+
+typedef struct s_synth_voice_tag
+{
+
+/* These parameters are common to both wavetable and FM
+ * synthesizers. The voice manager should only access this data.
+ * Any other data should be manipulated by the code that is
+ * specific to that synthesizer and reflected back through the
+ * common state data available here.
+ */
+    EAS_U16             regionIndex;        /* index to wave and playback params */
+    EAS_I16             gain;               /* current gain */
+    EAS_U16             age;                /* large value means old note */
+    EAS_U16             nextRegionIndex;    /* index to wave and playback params */
+    EAS_U8              voiceState;         /* current voice state */
+    EAS_U8              voiceFlags;         /* misc flags/bit fields */
+    EAS_U8              channel;            /* this voice plays on this synth channel */
+    EAS_U8              note;               /* 12 <= key number <= 108 */
+    EAS_U8              velocity;           /* 0 <= velocity <= 127 */
+    EAS_U8              nextChannel;        /* play stolen voice on this channel */
+    EAS_U8              nextNote;           /* 12 <= key number <= 108 */
+    EAS_U8              nextVelocity;       /* 0 <= velocity <= 127 */
+} S_SYNTH_VOICE;
+
+/*------------------------------------
+ * S_SYNTH data structure
+ *
+ * One instance for each MIDI stream
+ *------------------------------------
+*/
+
+/* S_SYNTH.m_nFlags */
+#define SYNTH_FLAG_RESET_IS_REQUESTED                   0x01
+#define SYNTH_FLAG_SP_MIDI_ON                           0x02
+#define SYNTH_FLAG_UPDATE_ALL_CHANNEL_PARAMETERS        0x04
+#define SYNTH_FLAG_DEFERRED_MIDI_NOTE_OFF_PENDING       0x08
+#define DEFAULT_SYNTH_FLAGS     SYNTH_FLAG_UPDATE_ALL_CHANNEL_PARAMETERS
+
+typedef struct s_synth_tag
+{
+    struct s_eas_data_tag   *pEASData;
+    const S_EAS             *pEAS;
+
+#ifdef DLS_SYNTHESIZER
+    S_DLS                   *pDLS;
+#endif
+
+#ifdef EXTERNAL_AUDIO
+    EAS_EXT_PRG_CHG_FUNC    cbProgChgFunc;
+    EAS_EXT_EVENT_FUNC      cbEventFunc;
+    EAS_VOID_PTR            *pExtAudioInstData;
+#endif
+
+    S_SYNTH_CHANNEL         channels[NUM_SYNTH_CHANNELS];
+    EAS_I32                 totalNoteCount;
+    EAS_U16                 maxPolyphony;
+    EAS_U16                 numActiveVoices;
+    EAS_U16                 masterVolume;
+    EAS_U8                  channelsByPriority[NUM_SYNTH_CHANNELS];
+    EAS_U8                  poolCount[NUM_SYNTH_CHANNELS];
+    EAS_U8                  poolAlloc[NUM_SYNTH_CHANNELS];
+    EAS_U8                  synthFlags;
+    EAS_I8                  globalTranspose;
+    EAS_U8                  vSynthNum;
+    EAS_U8                  refCount;
+    EAS_U8                  priority;
+} S_SYNTH;
+
+/*------------------------------------
+ * S_VOICE_MGR data structure
+ *
+ * One instance for each EAS library instance
+ *------------------------------------
+*/
+typedef struct s_voice_mgr_tag
+{
+    S_SYNTH                 *pSynth[MAX_VIRTUAL_SYNTHESIZERS];
+    EAS_PCM                 voiceBuffer[SYNTH_UPDATE_PERIOD_IN_SAMPLES];
+
+#ifdef _FM_SYNTH
+    EAS_PCM                 operMixBuffer[SYNTH_UPDATE_PERIOD_IN_SAMPLES];
+    S_FM_VOICE              fmVoices[NUM_FM_VOICES];
+#endif
+
+#ifdef _WT_SYNTH
+    S_WT_VOICE              wtVoices[NUM_WT_VOICES];
+#endif
+
+#ifdef _REVERB
+    EAS_PCM                 reverbSendBuffer[NUM_OUTPUT_CHANNELS * SYNTH_UPDATE_PERIOD_IN_SAMPLES];
+#endif
+
+#ifdef _CHORUS
+    EAS_PCM                 chorusSendBuffer[NUM_OUTPUT_CHANNELS * SYNTH_UPDATE_PERIOD_IN_SAMPLES];
+#endif
+    S_SYNTH_VOICE           voices[MAX_SYNTH_VOICES];
+
+    EAS_SNDLIB_HANDLE       pGlobalEAS;
+
+#ifdef DLS_SYNTHESIZER
+    S_DLS                   *pGlobalDLS;
+#endif
+
+#ifdef _SPLIT_ARCHITECTURE
+    EAS_FRAME_BUFFER_HANDLE pFrameBuffer;
+#endif
+
+#if defined(_SECONDARY_SYNTH) || defined(EAS_SPLIT_WT_SYNTH)
+    EAS_U16                 maxPolyphonyPrimary;
+    EAS_U16                 maxPolyphonySecondary;
+#endif
+
+    EAS_I32                 workload;
+    EAS_I32                 maxWorkLoad;
+
+    EAS_U16                 activeVoices;
+    EAS_U16                 maxPolyphony;
+
+    EAS_U16                 age;
+
+/* limits the number of voice starts in a frame for split architecture */
+#ifdef MAX_VOICE_STARTS
+    EAS_U16                 numVoiceStarts;
+#endif
+} S_VOICE_MGR;
+
+#endif /* #ifdef _EAS_SYNTH_H */
+
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_synth_protos.h

@@ -0,0 +1,60 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_synth_protos.h
+ *
+ * Contents and purpose:
+ * Declarations, interfaces, and prototypes for synth.
+ *
+ * Copyright Sonic Network Inc. 2004
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 82 $
+ *   $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_SYNTH_PROTOS_H
+#define _EAS_SYNTH_PROTOS_H
+
+/* includes */
+#include "eas_data.h"
+#include "eas_sndlib.h"
+
+#ifdef _SPLIT_ARCHITECTURE
+typedef struct s_frame_interface_tag
+{
+    EAS_BOOL (* EAS_CONST pfStartFrame)(EAS_FRAME_BUFFER_HANDLE pFrameBuffer);
+    EAS_BOOL (* EAS_CONST pfEndFrame)(EAS_FRAME_BUFFER_HANDLE pFrameBuffer, EAS_I32 *pMixBuffer, EAS_I16 masterGain);
+} S_FRAME_INTERFACE;
+#endif
+
+/* generic synthesizer interface */
+typedef struct
+{
+    EAS_RESULT (* EAS_CONST pfInitialize)(S_VOICE_MGR *pVoiceMgr);
+    EAS_RESULT (* EAS_CONST pfStartVoice)(S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum, EAS_U16 regionIndex);
+    EAS_BOOL (* EAS_CONST pfUpdateVoice)(S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum, EAS_I32 *pMixBuffer, EAS_I32 numSamples);
+    void (* EAS_CONST pfReleaseVoice)(S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum);
+    void (* EAS_CONST pfMuteVoice)(S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum);
+    void (* EAS_CONST pfSustainPedal)(S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, S_SYNTH_CHANNEL *pChannel, EAS_I32 voiceNum);
+    void (* EAS_CONST pfUpdateChannel)(S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel);
+} S_SYNTH_INTERFACE;
+
+#endif
+
+
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_synthcfg.h

@@ -0,0 +1,70 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_synthcfg.h
+ *
+ * Contents and purpose:
+ * Defines for various synth configurations
+ *
+ * Copyright Sonic Network Inc. 2004, 2005
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 664 $
+ *   $Date: 2007-04-25 13:11:22 -0700 (Wed, 25 Apr 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_SYNTHCFG_H
+#define _EAS_SYNTHCFG_H
+
+#if defined(EAS_WT_SYNTH)
+#define _WT_SYNTH
+
+/* FM on MCU */
+#elif defined(EAS_FM_SYNTH)
+#define _FM_SYNTH
+
+/* wavetable drums and FM melodic on MCU */
+#elif defined(EAS_HYBRID_SYNTH)
+#define _WT_SYNTH
+#define _FM_SYNTH
+#define _SECONDARY_SYNTH
+#define _HYBRID_SYNTH
+
+/* wavetable drums on MCU, wavetable melodic on DSP */
+#elif defined(EAS_SPLIT_WT_SYNTH)
+#define _WT_SYNTH
+#define _SPLIT_ARCHITECTURE
+
+/* wavetable drums on MCU, FM melodic on DSP */
+#elif defined(EAS_SPLIT_HYBRID_SYNTH)
+#define _WT_SYNTH
+#define _FM_SYNTH
+#define _SECONDARY_SYNTH
+#define _SPLIT_ARCHITECTURE
+#define _HYBRID_SYNTH
+
+/* FM synth on DSP */
+#elif defined(EAS_SPLIT_FM_SYNTH)
+#define _FM_SYNTH
+#define _SPLIT_ARCHITECTURE
+
+#else
+#error "Unrecognized architecture option"
+#endif
+
+#endif
+

common/embeddedaudiosynthesis/arm-wt-22k/lib_src/eas_tcdata.c

@@ -0,0 +1,43 @@
+/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_tcdata.c
+ *
+ * Contents and purpose:
+ * ToneControl Parser data
+ *
+ * This file contains static data for the ToneControl parser.
+ *
+ * Copyright Sonic Network Inc. 2005
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ *   $Revision: 547 $
+ *   $Date: 2007-01-31 16:30:17 -0800 (Wed, 31 Jan 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#include "eas_types.h"
+#include "eas_tcdata.h"
+
+/*----------------------------------------------------------------------------
+ *
+ * eas_iMelodyData
+ *
+ * Static memory allocation for iMelody parser
+ *----------------------------------------------------------------------------
+*/
+S_TC_DATA eas_TCData;
+