mirror of
https://github.com/love2d/megasource.git
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Added OpenAL-Soft 1.16.0.
This commit is contained in:
rude
@@ -0,0 +1,707 @@
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/**
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* OpenAL cross platform audio library
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* Copyright (C) 1999-2007 by authors.
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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* Or go to http://www.gnu.org/copyleft/lgpl.html
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*/
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#include "config.h"
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#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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#include <alloca.h>
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#include "alMain.h"
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#include "alu.h"
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#include <CoreServices/CoreServices.h>
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#include <unistd.h>
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#include <AudioUnit/AudioUnit.h>
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#include <AudioToolbox/AudioToolbox.h>
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typedef struct {
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AudioUnit audioUnit;
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ALuint frameSize;
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ALdouble sampleRateRatio; // Ratio of hardware sample rate / requested sample rate
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AudioStreamBasicDescription format; // This is the OpenAL format as a CoreAudio ASBD
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AudioConverterRef audioConverter; // Sample rate converter if needed
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AudioBufferList *bufferList; // Buffer for data coming from the input device
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ALCvoid *resampleBuffer; // Buffer for returned RingBuffer data when resampling
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RingBuffer *ring;
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} ca_data;
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static const ALCchar ca_device[] = "CoreAudio Default";
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static void destroy_buffer_list(AudioBufferList* list)
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{
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if(list)
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{
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UInt32 i;
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for(i = 0;i < list->mNumberBuffers;i++)
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free(list->mBuffers[i].mData);
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free(list);
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}
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}
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static AudioBufferList* allocate_buffer_list(UInt32 channelCount, UInt32 byteSize)
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{
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AudioBufferList *list;
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list = calloc(1, sizeof(AudioBufferList) + sizeof(AudioBuffer));
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if(list)
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{
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list->mNumberBuffers = 1;
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list->mBuffers[0].mNumberChannels = channelCount;
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list->mBuffers[0].mDataByteSize = byteSize;
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list->mBuffers[0].mData = malloc(byteSize);
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if(list->mBuffers[0].mData == NULL)
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{
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free(list);
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list = NULL;
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}
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}
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return list;
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}
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static OSStatus ca_callback(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags, const AudioTimeStamp *inTimeStamp,
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UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList *ioData)
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{
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ALCdevice *device = (ALCdevice*)inRefCon;
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ca_data *data = (ca_data*)device->ExtraData;
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aluMixData(device, ioData->mBuffers[0].mData,
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ioData->mBuffers[0].mDataByteSize / data->frameSize);
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return noErr;
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}
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static OSStatus ca_capture_conversion_callback(AudioConverterRef inAudioConverter, UInt32 *ioNumberDataPackets,
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AudioBufferList *ioData, AudioStreamPacketDescription **outDataPacketDescription, void* inUserData)
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{
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ALCdevice *device = (ALCdevice*)inUserData;
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ca_data *data = (ca_data*)device->ExtraData;
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// Read from the ring buffer and store temporarily in a large buffer
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ReadRingBuffer(data->ring, data->resampleBuffer, (ALsizei)(*ioNumberDataPackets));
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// Set the input data
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ioData->mNumberBuffers = 1;
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ioData->mBuffers[0].mNumberChannels = data->format.mChannelsPerFrame;
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ioData->mBuffers[0].mData = data->resampleBuffer;
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ioData->mBuffers[0].mDataByteSize = (*ioNumberDataPackets) * data->format.mBytesPerFrame;
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return noErr;
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}
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static OSStatus ca_capture_callback(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags,
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const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber,
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UInt32 inNumberFrames, AudioBufferList *ioData)
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{
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ALCdevice *device = (ALCdevice*)inRefCon;
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ca_data *data = (ca_data*)device->ExtraData;
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AudioUnitRenderActionFlags flags = 0;
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OSStatus err;
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// fill the bufferList with data from the input device
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err = AudioUnitRender(data->audioUnit, &flags, inTimeStamp, 1, inNumberFrames, data->bufferList);
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if(err != noErr)
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{
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ERR("AudioUnitRender error: %d\n", err);
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return err;
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}
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WriteRingBuffer(data->ring, data->bufferList->mBuffers[0].mData, inNumberFrames);
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return noErr;
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}
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static ALCenum ca_open_playback(ALCdevice *device, const ALCchar *deviceName)
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{
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ComponentDescription desc;
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Component comp;
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ca_data *data;
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OSStatus err;
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if(!deviceName)
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deviceName = ca_device;
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else if(strcmp(deviceName, ca_device) != 0)
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return ALC_INVALID_VALUE;
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/* open the default output unit */
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desc.componentType = kAudioUnitType_Output;
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desc.componentSubType = kAudioUnitSubType_DefaultOutput;
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desc.componentManufacturer = kAudioUnitManufacturer_Apple;
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desc.componentFlags = 0;
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desc.componentFlagsMask = 0;
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comp = FindNextComponent(NULL, &desc);
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if(comp == NULL)
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{
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ERR("FindNextComponent failed\n");
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return ALC_INVALID_VALUE;
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}
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data = calloc(1, sizeof(*data));
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err = OpenAComponent(comp, &data->audioUnit);
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if(err != noErr)
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{
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ERR("OpenAComponent failed\n");
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free(data);
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return ALC_INVALID_VALUE;
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}
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/* init and start the default audio unit... */
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err = AudioUnitInitialize(data->audioUnit);
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if(err != noErr)
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{
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ERR("AudioUnitInitialize failed\n");
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CloseComponent(data->audioUnit);
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free(data);
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return ALC_INVALID_VALUE;
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}
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al_string_copy_cstr(&device->DeviceName, deviceName);
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device->ExtraData = data;
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return ALC_NO_ERROR;
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}
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static void ca_close_playback(ALCdevice *device)
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{
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ca_data *data = (ca_data*)device->ExtraData;
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AudioUnitUninitialize(data->audioUnit);
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CloseComponent(data->audioUnit);
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free(data);
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device->ExtraData = NULL;
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}
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static ALCboolean ca_reset_playback(ALCdevice *device)
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{
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ca_data *data = (ca_data*)device->ExtraData;
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AudioStreamBasicDescription streamFormat;
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AURenderCallbackStruct input;
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OSStatus err;
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UInt32 size;
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err = AudioUnitUninitialize(data->audioUnit);
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if(err != noErr)
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ERR("-- AudioUnitUninitialize failed.\n");
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/* retrieve default output unit's properties (output side) */
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size = sizeof(AudioStreamBasicDescription);
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err = AudioUnitGetProperty(data->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 0, &streamFormat, &size);
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if(err != noErr || size != sizeof(AudioStreamBasicDescription))
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{
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ERR("AudioUnitGetProperty failed\n");
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return ALC_FALSE;
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}
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#if 0
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TRACE("Output streamFormat of default output unit -\n");
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TRACE(" streamFormat.mFramesPerPacket = %d\n", streamFormat.mFramesPerPacket);
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TRACE(" streamFormat.mChannelsPerFrame = %d\n", streamFormat.mChannelsPerFrame);
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TRACE(" streamFormat.mBitsPerChannel = %d\n", streamFormat.mBitsPerChannel);
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TRACE(" streamFormat.mBytesPerPacket = %d\n", streamFormat.mBytesPerPacket);
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TRACE(" streamFormat.mBytesPerFrame = %d\n", streamFormat.mBytesPerFrame);
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TRACE(" streamFormat.mSampleRate = %5.0f\n", streamFormat.mSampleRate);
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#endif
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/* set default output unit's input side to match output side */
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err = AudioUnitSetProperty(data->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamFormat, size);
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if(err != noErr)
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{
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ERR("AudioUnitSetProperty failed\n");
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return ALC_FALSE;
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}
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if(device->Frequency != streamFormat.mSampleRate)
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{
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device->UpdateSize = (ALuint)((ALuint64)device->UpdateSize *
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streamFormat.mSampleRate /
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device->Frequency);
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device->Frequency = streamFormat.mSampleRate;
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}
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/* FIXME: How to tell what channels are what in the output device, and how
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* to specify what we're giving? eg, 6.0 vs 5.1 */
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switch(streamFormat.mChannelsPerFrame)
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{
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case 1:
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device->FmtChans = DevFmtMono;
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break;
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case 2:
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device->FmtChans = DevFmtStereo;
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break;
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case 4:
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device->FmtChans = DevFmtQuad;
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break;
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case 6:
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device->FmtChans = DevFmtX51;
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break;
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case 7:
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device->FmtChans = DevFmtX61;
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break;
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case 8:
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device->FmtChans = DevFmtX71;
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break;
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default:
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ERR("Unhandled channel count (%d), using Stereo\n", streamFormat.mChannelsPerFrame);
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device->FmtChans = DevFmtStereo;
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streamFormat.mChannelsPerFrame = 2;
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break;
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}
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SetDefaultWFXChannelOrder(device);
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/* use channel count and sample rate from the default output unit's current
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* parameters, but reset everything else */
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streamFormat.mFramesPerPacket = 1;
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streamFormat.mFormatFlags = 0;
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switch(device->FmtType)
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{
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case DevFmtUByte:
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device->FmtType = DevFmtByte;
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/* fall-through */
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case DevFmtByte:
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streamFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger;
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streamFormat.mBitsPerChannel = 8;
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break;
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case DevFmtUShort:
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device->FmtType = DevFmtShort;
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/* fall-through */
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case DevFmtShort:
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streamFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger;
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streamFormat.mBitsPerChannel = 16;
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break;
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case DevFmtUInt:
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device->FmtType = DevFmtInt;
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/* fall-through */
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case DevFmtInt:
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streamFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger;
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streamFormat.mBitsPerChannel = 32;
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break;
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case DevFmtFloat:
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streamFormat.mFormatFlags = kLinearPCMFormatFlagIsFloat;
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streamFormat.mBitsPerChannel = 32;
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break;
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}
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streamFormat.mBytesPerFrame = streamFormat.mChannelsPerFrame *
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streamFormat.mBitsPerChannel / 8;
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streamFormat.mBytesPerPacket = streamFormat.mBytesPerFrame;
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streamFormat.mFormatID = kAudioFormatLinearPCM;
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streamFormat.mFormatFlags |= kAudioFormatFlagsNativeEndian |
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kLinearPCMFormatFlagIsPacked;
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err = AudioUnitSetProperty(data->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamFormat, sizeof(AudioStreamBasicDescription));
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if(err != noErr)
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{
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ERR("AudioUnitSetProperty failed\n");
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return ALC_FALSE;
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}
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/* setup callback */
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data->frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType);
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input.inputProc = ca_callback;
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input.inputProcRefCon = device;
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err = AudioUnitSetProperty(data->audioUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, 0, &input, sizeof(AURenderCallbackStruct));
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if(err != noErr)
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{
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ERR("AudioUnitSetProperty failed\n");
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return ALC_FALSE;
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}
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/* init the default audio unit... */
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err = AudioUnitInitialize(data->audioUnit);
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if(err != noErr)
|
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{
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ERR("AudioUnitInitialize failed\n");
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return ALC_FALSE;
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}
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return ALC_TRUE;
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}
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static ALCboolean ca_start_playback(ALCdevice *device)
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{
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ca_data *data = (ca_data*)device->ExtraData;
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OSStatus err;
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||||
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err = AudioOutputUnitStart(data->audioUnit);
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if(err != noErr)
|
||||
{
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ERR("AudioOutputUnitStart failed\n");
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||||
return ALC_FALSE;
|
||||
}
|
||||
|
||||
return ALC_TRUE;
|
||||
}
|
||||
|
||||
static void ca_stop_playback(ALCdevice *device)
|
||||
{
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||||
ca_data *data = (ca_data*)device->ExtraData;
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||||
OSStatus err;
|
||||
|
||||
err = AudioOutputUnitStop(data->audioUnit);
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||||
if(err != noErr)
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ERR("AudioOutputUnitStop failed\n");
|
||||
}
|
||||
|
||||
static ALCenum ca_open_capture(ALCdevice *device, const ALCchar *deviceName)
|
||||
{
|
||||
AudioStreamBasicDescription requestedFormat; // The application requested format
|
||||
AudioStreamBasicDescription hardwareFormat; // The hardware format
|
||||
AudioStreamBasicDescription outputFormat; // The AudioUnit output format
|
||||
AURenderCallbackStruct input;
|
||||
ComponentDescription desc;
|
||||
AudioDeviceID inputDevice;
|
||||
UInt32 outputFrameCount;
|
||||
UInt32 propertySize;
|
||||
UInt32 enableIO;
|
||||
Component comp;
|
||||
ca_data *data;
|
||||
OSStatus err;
|
||||
|
||||
desc.componentType = kAudioUnitType_Output;
|
||||
desc.componentSubType = kAudioUnitSubType_HALOutput;
|
||||
desc.componentManufacturer = kAudioUnitManufacturer_Apple;
|
||||
desc.componentFlags = 0;
|
||||
desc.componentFlagsMask = 0;
|
||||
|
||||
// Search for component with given description
|
||||
comp = FindNextComponent(NULL, &desc);
|
||||
if(comp == NULL)
|
||||
{
|
||||
ERR("FindNextComponent failed\n");
|
||||
return ALC_INVALID_VALUE;
|
||||
}
|
||||
|
||||
data = calloc(1, sizeof(*data));
|
||||
device->ExtraData = data;
|
||||
|
||||
// Open the component
|
||||
err = OpenAComponent(comp, &data->audioUnit);
|
||||
if(err != noErr)
|
||||
{
|
||||
ERR("OpenAComponent failed\n");
|
||||
goto error;
|
||||
}
|
||||
|
||||
// Turn off AudioUnit output
|
||||
enableIO = 0;
|
||||
err = AudioUnitSetProperty(data->audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, 0, &enableIO, sizeof(ALuint));
|
||||
if(err != noErr)
|
||||
{
|
||||
ERR("AudioUnitSetProperty failed\n");
|
||||
goto error;
|
||||
}
|
||||
|
||||
// Turn on AudioUnit input
|
||||
enableIO = 1;
|
||||
err = AudioUnitSetProperty(data->audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, 1, &enableIO, sizeof(ALuint));
|
||||
if(err != noErr)
|
||||
{
|
||||
ERR("AudioUnitSetProperty failed\n");
|
||||
goto error;
|
||||
}
|
||||
|
||||
// Get the default input device
|
||||
propertySize = sizeof(AudioDeviceID);
|
||||
err = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultInputDevice, &propertySize, &inputDevice);
|
||||
if(err != noErr)
|
||||
{
|
||||
ERR("AudioHardwareGetProperty failed\n");
|
||||
goto error;
|
||||
}
|
||||
|
||||
if(inputDevice == kAudioDeviceUnknown)
|
||||
{
|
||||
ERR("No input device found\n");
|
||||
goto error;
|
||||
}
|
||||
|
||||
// Track the input device
|
||||
err = AudioUnitSetProperty(data->audioUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0, &inputDevice, sizeof(AudioDeviceID));
|
||||
if(err != noErr)
|
||||
{
|
||||
ERR("AudioUnitSetProperty failed\n");
|
||||
goto error;
|
||||
}
|
||||
|
||||
// set capture callback
|
||||
input.inputProc = ca_capture_callback;
|
||||
input.inputProcRefCon = device;
|
||||
|
||||
err = AudioUnitSetProperty(data->audioUnit, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, 0, &input, sizeof(AURenderCallbackStruct));
|
||||
if(err != noErr)
|
||||
{
|
||||
ERR("AudioUnitSetProperty failed\n");
|
||||
goto error;
|
||||
}
|
||||
|
||||
// Initialize the device
|
||||
err = AudioUnitInitialize(data->audioUnit);
|
||||
if(err != noErr)
|
||||
{
|
||||
ERR("AudioUnitInitialize failed\n");
|
||||
goto error;
|
||||
}
|
||||
|
||||
// Get the hardware format
|
||||
propertySize = sizeof(AudioStreamBasicDescription);
|
||||
err = AudioUnitGetProperty(data->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 1, &hardwareFormat, &propertySize);
|
||||
if(err != noErr || propertySize != sizeof(AudioStreamBasicDescription))
|
||||
{
|
||||
ERR("AudioUnitGetProperty failed\n");
|
||||
goto error;
|
||||
}
|
||||
|
||||
// Set up the requested format description
|
||||
switch(device->FmtType)
|
||||
{
|
||||
case DevFmtUByte:
|
||||
requestedFormat.mBitsPerChannel = 8;
|
||||
requestedFormat.mFormatFlags = kAudioFormatFlagIsPacked;
|
||||
break;
|
||||
case DevFmtShort:
|
||||
requestedFormat.mBitsPerChannel = 16;
|
||||
requestedFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagsNativeEndian | kAudioFormatFlagIsPacked;
|
||||
break;
|
||||
case DevFmtInt:
|
||||
requestedFormat.mBitsPerChannel = 32;
|
||||
requestedFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagsNativeEndian | kAudioFormatFlagIsPacked;
|
||||
break;
|
||||
case DevFmtFloat:
|
||||
requestedFormat.mBitsPerChannel = 32;
|
||||
requestedFormat.mFormatFlags = kAudioFormatFlagIsPacked;
|
||||
break;
|
||||
case DevFmtByte:
|
||||
case DevFmtUShort:
|
||||
case DevFmtUInt:
|
||||
ERR("%s samples not supported\n", DevFmtTypeString(device->FmtType));
|
||||
goto error;
|
||||
}
|
||||
|
||||
switch(device->FmtChans)
|
||||
{
|
||||
case DevFmtMono:
|
||||
requestedFormat.mChannelsPerFrame = 1;
|
||||
break;
|
||||
case DevFmtStereo:
|
||||
requestedFormat.mChannelsPerFrame = 2;
|
||||
break;
|
||||
|
||||
case DevFmtQuad:
|
||||
case DevFmtX51:
|
||||
case DevFmtX51Side:
|
||||
case DevFmtX61:
|
||||
case DevFmtX71:
|
||||
ERR("%s not supported\n", DevFmtChannelsString(device->FmtChans));
|
||||
goto error;
|
||||
}
|
||||
|
||||
requestedFormat.mBytesPerFrame = requestedFormat.mChannelsPerFrame * requestedFormat.mBitsPerChannel / 8;
|
||||
requestedFormat.mBytesPerPacket = requestedFormat.mBytesPerFrame;
|
||||
requestedFormat.mSampleRate = device->Frequency;
|
||||
requestedFormat.mFormatID = kAudioFormatLinearPCM;
|
||||
requestedFormat.mReserved = 0;
|
||||
requestedFormat.mFramesPerPacket = 1;
|
||||
|
||||
// save requested format description for later use
|
||||
data->format = requestedFormat;
|
||||
data->frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType);
|
||||
|
||||
// Use intermediate format for sample rate conversion (outputFormat)
|
||||
// Set sample rate to the same as hardware for resampling later
|
||||
outputFormat = requestedFormat;
|
||||
outputFormat.mSampleRate = hardwareFormat.mSampleRate;
|
||||
|
||||
// Determine sample rate ratio for resampling
|
||||
data->sampleRateRatio = outputFormat.mSampleRate / device->Frequency;
|
||||
|
||||
// The output format should be the requested format, but using the hardware sample rate
|
||||
// This is because the AudioUnit will automatically scale other properties, except for sample rate
|
||||
err = AudioUnitSetProperty(data->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 1, (void *)&outputFormat, sizeof(outputFormat));
|
||||
if(err != noErr)
|
||||
{
|
||||
ERR("AudioUnitSetProperty failed\n");
|
||||
goto error;
|
||||
}
|
||||
|
||||
// Set the AudioUnit output format frame count
|
||||
outputFrameCount = device->UpdateSize * data->sampleRateRatio;
|
||||
err = AudioUnitSetProperty(data->audioUnit, kAudioUnitProperty_MaximumFramesPerSlice, kAudioUnitScope_Output, 0, &outputFrameCount, sizeof(outputFrameCount));
|
||||
if(err != noErr)
|
||||
{
|
||||
ERR("AudioUnitSetProperty failed: %d\n", err);
|
||||
goto error;
|
||||
}
|
||||
|
||||
// Set up sample converter
|
||||
err = AudioConverterNew(&outputFormat, &requestedFormat, &data->audioConverter);
|
||||
if(err != noErr)
|
||||
{
|
||||
ERR("AudioConverterNew failed: %d\n", err);
|
||||
goto error;
|
||||
}
|
||||
|
||||
// Create a buffer for use in the resample callback
|
||||
data->resampleBuffer = malloc(device->UpdateSize * data->frameSize * data->sampleRateRatio);
|
||||
|
||||
// Allocate buffer for the AudioUnit output
|
||||
data->bufferList = allocate_buffer_list(outputFormat.mChannelsPerFrame, device->UpdateSize * data->frameSize * data->sampleRateRatio);
|
||||
if(data->bufferList == NULL)
|
||||
goto error;
|
||||
|
||||
data->ring = CreateRingBuffer(data->frameSize, (device->UpdateSize * data->sampleRateRatio) * device->NumUpdates);
|
||||
if(data->ring == NULL)
|
||||
goto error;
|
||||
|
||||
al_string_copy_cstr(&device->DeviceName, deviceName);
|
||||
|
||||
return ALC_NO_ERROR;
|
||||
|
||||
error:
|
||||
DestroyRingBuffer(data->ring);
|
||||
free(data->resampleBuffer);
|
||||
destroy_buffer_list(data->bufferList);
|
||||
|
||||
if(data->audioConverter)
|
||||
AudioConverterDispose(data->audioConverter);
|
||||
if(data->audioUnit)
|
||||
CloseComponent(data->audioUnit);
|
||||
|
||||
free(data);
|
||||
device->ExtraData = NULL;
|
||||
|
||||
return ALC_INVALID_VALUE;
|
||||
}
|
||||
|
||||
static void ca_close_capture(ALCdevice *device)
|
||||
{
|
||||
ca_data *data = (ca_data*)device->ExtraData;
|
||||
|
||||
DestroyRingBuffer(data->ring);
|
||||
free(data->resampleBuffer);
|
||||
destroy_buffer_list(data->bufferList);
|
||||
|
||||
AudioConverterDispose(data->audioConverter);
|
||||
CloseComponent(data->audioUnit);
|
||||
|
||||
free(data);
|
||||
device->ExtraData = NULL;
|
||||
}
|
||||
|
||||
static void ca_start_capture(ALCdevice *device)
|
||||
{
|
||||
ca_data *data = (ca_data*)device->ExtraData;
|
||||
OSStatus err = AudioOutputUnitStart(data->audioUnit);
|
||||
if(err != noErr)
|
||||
ERR("AudioOutputUnitStart failed\n");
|
||||
}
|
||||
|
||||
static void ca_stop_capture(ALCdevice *device)
|
||||
{
|
||||
ca_data *data = (ca_data*)device->ExtraData;
|
||||
OSStatus err = AudioOutputUnitStop(data->audioUnit);
|
||||
if(err != noErr)
|
||||
ERR("AudioOutputUnitStop failed\n");
|
||||
}
|
||||
|
||||
static ALCenum ca_capture_samples(ALCdevice *device, ALCvoid *buffer, ALCuint samples)
|
||||
{
|
||||
ca_data *data = (ca_data*)device->ExtraData;
|
||||
AudioBufferList *list;
|
||||
UInt32 frameCount;
|
||||
OSStatus err;
|
||||
|
||||
// If no samples are requested, just return
|
||||
if(samples == 0)
|
||||
return ALC_NO_ERROR;
|
||||
|
||||
// Allocate a temporary AudioBufferList to use as the return resamples data
|
||||
list = alloca(sizeof(AudioBufferList) + sizeof(AudioBuffer));
|
||||
|
||||
// Point the resampling buffer to the capture buffer
|
||||
list->mNumberBuffers = 1;
|
||||
list->mBuffers[0].mNumberChannels = data->format.mChannelsPerFrame;
|
||||
list->mBuffers[0].mDataByteSize = samples * data->frameSize;
|
||||
list->mBuffers[0].mData = buffer;
|
||||
|
||||
// Resample into another AudioBufferList
|
||||
frameCount = samples;
|
||||
err = AudioConverterFillComplexBuffer(data->audioConverter, ca_capture_conversion_callback,
|
||||
device, &frameCount, list, NULL);
|
||||
if(err != noErr)
|
||||
{
|
||||
ERR("AudioConverterFillComplexBuffer error: %d\n", err);
|
||||
return ALC_INVALID_VALUE;
|
||||
}
|
||||
return ALC_NO_ERROR;
|
||||
}
|
||||
|
||||
static ALCuint ca_available_samples(ALCdevice *device)
|
||||
{
|
||||
ca_data *data = device->ExtraData;
|
||||
return RingBufferSize(data->ring) / data->sampleRateRatio;
|
||||
}
|
||||
|
||||
|
||||
static const BackendFuncs ca_funcs = {
|
||||
ca_open_playback,
|
||||
ca_close_playback,
|
||||
ca_reset_playback,
|
||||
ca_start_playback,
|
||||
ca_stop_playback,
|
||||
ca_open_capture,
|
||||
ca_close_capture,
|
||||
ca_start_capture,
|
||||
ca_stop_capture,
|
||||
ca_capture_samples,
|
||||
ca_available_samples,
|
||||
ALCdevice_GetLatencyDefault
|
||||
};
|
||||
|
||||
ALCboolean alc_ca_init(BackendFuncs *func_list)
|
||||
{
|
||||
*func_list = ca_funcs;
|
||||
return ALC_TRUE;
|
||||
}
|
||||
|
||||
void alc_ca_deinit(void)
|
||||
{
|
||||
}
|
||||
|
||||
void alc_ca_probe(enum DevProbe type)
|
||||
{
|
||||
switch(type)
|
||||
{
|
||||
case ALL_DEVICE_PROBE:
|
||||
AppendAllDevicesList(ca_device);
|
||||
break;
|
||||
case CAPTURE_DEVICE_PROBE:
|
||||
AppendCaptureDeviceList(ca_device);
|
||||
break;
|
||||
}
|
||||
}
|
||||
Reference in New Issue
Block a user