Initial Commit

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
+