sonivox/arm-wt-22k/lib_src/ARM-E_interpolate_noloop_gnu.s

@***********************************************************
@ 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