Merge pull request #625 from digitall/rate-hack

AUDIO: Add support for sample rates >65kHz.

3 files changed, 54 insertions, 33 deletions

diff --git a/audio/rate.cpp b/audio/rate.cpp
index 19d9c8c61e..6264465e19 100644
--- a/audio/rate.cpp
+++ b/audio/rate.cpp
@@ -46,6 +46,16 @@ namespace Audio {
  */
 #define INTERMEDIATE_BUFFER_SIZE 512
 
+/**
+ * The default fractional type in frac.h (with 16 fractional bits) limits
+ * the rate conversion code to 65536Hz audio: we need to able to handle
+ * 96kHz audio, so we use fewer fractional bits in this code.
+ */
+enum {
+	FRAC_BITS_LOW = 15,
+	FRAC_ONE_LOW = (1L << FRAC_BITS_LOW),
+	FRAC_HALF_LOW = (1L << (FRAC_BITS_LOW-1))
+};
 
 /**
  * Audio rate converter based on simple resampling. Used when no
@@ -187,18 +197,18 @@ public:
  */
 template<bool stereo, bool reverseStereo>
 LinearRateConverter<stereo, reverseStereo>::LinearRateConverter(st_rate_t inrate, st_rate_t outrate) {
-	if (inrate >= 65536 || outrate >= 65536) {
-		error("rate effect can only handle rates < 65536");
+	if (inrate >= 131072 || outrate >= 131072) {
+		error("rate effect can only handle rates < 131072");
 	}
 
-	opos = FRAC_ONE;
+	opos = FRAC_ONE_LOW;
 
 	// Compute the linear interpolation increment.
-	// This will overflow if inrate >= 2^16, and underflow if outrate >= 2^16.
+	// This will overflow if inrate >= 2^17, and underflow if outrate >= 2^17.
 	// Also, if the quotient of the two rate becomes too small / too big, that
 	// would cause problems, but since we rarely scale from 1 to 65536 Hz or vice
 	// versa, I think we can live with that limitation ;-).
-	opos_inc = (inrate << FRAC_BITS) / outrate;
+	opos_inc = (inrate << FRAC_BITS_LOW) / outrate;
 
 	ilast0 = ilast1 = 0;
 	icur0 = icur1 = 0;
@@ -220,7 +230,7 @@ int LinearRateConverter<stereo, reverseStereo>::flow(AudioStream &input, st_samp
 	while (obuf < oend) {
 
 		// read enough input samples so that opos < 0
-		while ((frac_t)FRAC_ONE <= opos) {
+		while ((frac_t)FRAC_ONE_LOW <= opos) {
 			// Check if we have to refill the buffer
 			if (inLen == 0) {
 				inPtr = inBuf;
@@ -235,17 +245,17 @@ int LinearRateConverter<stereo, reverseStereo>::flow(AudioStream &input, st_samp
 				ilast1 = icur1;
 				icur1 = *inPtr++;
 			}
-			opos -= FRAC_ONE;
+			opos -= FRAC_ONE_LOW;
 		}
 
 		// Loop as long as the outpos trails behind, and as long as there is
 		// still space in the output buffer.
-		while (opos < (frac_t)FRAC_ONE && obuf < oend) {
+		while (opos < (frac_t)FRAC_ONE_LOW && obuf < oend) {
 			// interpolate
 			st_sample_t out0, out1;
-			out0 = (st_sample_t)(ilast0 + (((icur0 - ilast0) * opos + FRAC_HALF) >> FRAC_BITS));
+			out0 = (st_sample_t)(ilast0 + (((icur0 - ilast0) * opos + FRAC_HALF_LOW) >> FRAC_BITS_LOW));
 			out1 = (stereo ?
-						  (st_sample_t)(ilast1 + (((icur1 - ilast1) * opos + FRAC_HALF) >> FRAC_BITS)) :
+						  (st_sample_t)(ilast1 + (((icur1 - ilast1) * opos + FRAC_HALF_LOW) >> FRAC_BITS_LOW)) :
 						  out0);
 
 			// output left channel
@@ -333,7 +343,7 @@ public:
 template<bool stereo, bool reverseStereo>
 RateConverter *makeRateConverter(st_rate_t inrate, st_rate_t outrate) {
 	if (inrate != outrate) {
-		if ((inrate % outrate) == 0) {
+		if ((inrate % outrate) == 0 && (inrate < 65536)) {
 			return new SimpleRateConverter<stereo, reverseStereo>(inrate, outrate);
 		} else {
 			return new LinearRateConverter<stereo, reverseStereo>(inrate, outrate);
diff --git a/audio/rate_arm.cpp b/audio/rate_arm.cpp
index 4ad8d71a34..7765266673 100644
--- a/audio/rate_arm.cpp
+++ b/audio/rate_arm.cpp
@@ -68,6 +68,16 @@ namespace Audio {
  */
 #define INTERMEDIATE_BUFFER_SIZE 512
 
+/**
+ * The default fractional type in frac.h (with 16 fractional bits) limits
+ * the rate conversion code to 65536Hz audio: we need to able to handle
+ * 96kHz audio, so we use fewer fractional bits in this code.
+ */
+enum {
+	FRAC_BITS_LOW = 15,
+	FRAC_ONE_LOW = (1L << FRAC_BITS_LOW),
+	FRAC_HALF_LOW = (1L << (FRAC_BITS_LOW-1))
+};
 
 /**
  * Audio rate converter based on simple resampling. Used when no
@@ -287,17 +297,18 @@ LinearRateConverter<stereo, reverseStereo>::LinearRateConverter(st_rate_t inrate
 		error("Input and Output rates must be different to use rate effect");
 	}
 
-	if (inrate >= 65536 || outrate >= 65536) {
-		error("rate effect can only handle rates < 65536");
+	if (inrate >= 131072 || outrate >= 131072) {
+		error("rate effect can only handle rates < 131072");
 	}
 
-	lr.opos = FRAC_ONE;
+	lr.opos = FRAC_ONE_LOW;
 
 	/* increment */
-	incr = (inrate << FRAC_BITS) / outrate;
+	incr = (inrate << FRAC_BITS_LOW) / outrate;
 
 	lr.opos_inc = incr;
 
+	// FIXME: Does 32768 here need changing to 65536 or 0? Compare to rate.cpp code...
 	lr.ilast[0] = lr.ilast[1] = 32768;
 	lr.icur[0] = lr.icur[1] = 0;
 
@@ -438,7 +449,7 @@ public:
  */
 RateConverter *makeRateConverter(st_rate_t inrate, st_rate_t outrate, bool stereo, bool reverseStereo) {
 	if (inrate != outrate) {
-		if ((inrate % outrate) == 0) {
+		if ((inrate % outrate) == 0 && (inrate < 65536)) {
 			if (stereo) {
 				if (reverseStereo)
 					return new SimpleRateConverter<true, true>(inrate, outrate);
diff --git a/audio/rate_arm_asm.s b/audio/rate_arm_asm.s
index a727209d39..bb01c614c2 100644
--- a/audio/rate_arm_asm.s
+++ b/audio/rate_arm_asm.s
@@ -441,17 +441,17 @@ LinearRate_M_part2:
 
         LDRSH   r4, [r3]                @ r4 = obuf[0]
         LDRSH   r5, [r3,#2]             @ r5 = obuf[1]
-        MOV     r6, r6, ASR #16         @ r6 = tmp0 = tmp1 >>= 16
+        MOV     r6, r6, ASR #15         @ r6 = tmp0 = tmp1 >>= 15
         MUL     r7, r12,r6              @ r7 = tmp0*vol_l
         MUL     r6, r14,r6              @ r6 = tmp1*vol_r
 
-        ADDS    r7, r7, r4, LSL #16     @ r7 = obuf[0]<<16 + tmp0*vol_l
+        ADDS    r7, r7, r4, LSL #15     @ r7 = obuf[0]<<15 + tmp0*vol_l
         RSCVS   r7, r10, #0x80000000    @ Clamp r7
-        ADDS    r6, r6, r5, LSL #16     @ r6 = obuf[1]<<16 + tmp1*vol_r
+        ADDS    r6, r6, r5, LSL #15     @ r6 = obuf[1]<<15 + tmp1*vol_r
         RSCVS   r6, r10, #0x80000000    @ Clamp r6
 
-        MOV     r7, r7, LSR #16         @ Shift back to halfword
-        MOV     r6, r6, LSR #16         @ Shift back to halfword
+        MOV     r7, r7, LSR #15         @ Shift back to halfword
+        MOV     r6, r6, LSR #15         @ Shift back to halfword
 
         LDR     r5, [r2,#12]            @ r5 = opos_inc
         STRH    r7, [r3],#2             @ Store output value
@@ -538,23 +538,23 @@ LinearRate_S_part2:
         LDR     r7, [r2,#24]            @ r7 = ilast[1]<<16 + 32768
         LDRSH   r5, [r2,#18]            @ r5 = icur[1]
         LDRSH   r10,[r3]                @ r10= obuf[0]
-        MOV     r6, r6, ASR #16         @ r6 = tmp1 >>= 16
+        MOV     r6, r6, ASR #15         @ r6 = tmp1 >>= 15
         SUB     r5, r5, r7, ASR #16     @ r5 = icur[1] - ilast[1]
         MLA     r7, r4, r5, r7  @ r7 = (icur[1]-ilast[1])*opos_frac+ilast[1]
 
         LDRSH   r5, [r3,#2]             @ r5 = obuf[1]
-        MOV     r7, r7, ASR #16         @ r7 = tmp0 >>= 16
+        MOV     r7, r7, ASR #15         @ r7 = tmp0 >>= 15
         MUL     r7, r12,r7              @ r7 = tmp0*vol_l
         MUL     r6, r14,r6              @ r6 = tmp1*vol_r
 
-        ADDS    r7, r7, r10, LSL #16    @ r7 = obuf[0]<<16 + tmp0*vol_l
+        ADDS    r7, r7, r10, LSL #15    @ r7 = obuf[0]<<15 + tmp0*vol_l
         MOV     r4, #0
         RSCVS   r7, r4, #0x80000000     @ Clamp r7
-        ADDS    r6, r6, r5, LSL #16     @ r6 = obuf[1]<<16 + tmp1*vol_r
+        ADDS    r6, r6, r5, LSL #15     @ r6 = obuf[1]<<15 + tmp1*vol_r
         RSCVS   r6, r4, #0x80000000     @ Clamp r6
 
-        MOV     r7, r7, LSR #16         @ Shift back to halfword
-        MOV     r6, r6, LSR #16         @ Shift back to halfword
+        MOV     r7, r7, LSR #15         @ Shift back to halfword
+        MOV     r6, r6, LSR #15         @ Shift back to halfword
 
         LDR     r5, [r2,#12]            @ r5 = opos_inc
         STRH    r7, [r3],#2             @ Store output value
@@ -641,23 +641,23 @@ LinearRate_R_part2:
         LDR     r7, [r2,#24]            @ r7 = ilast[1]<<16 + 32768
         LDRSH   r5, [r2,#18]            @ r5 = icur[1]
         LDRSH   r10,[r3,#2]             @ r10= obuf[1]
-        MOV     r6, r6, ASR #16         @ r6 = tmp1 >>= 16
+        MOV     r6, r6, ASR #15         @ r6 = tmp1 >>= 15
         SUB     r5, r5, r7, ASR #16     @ r5 = icur[1] - ilast[1]
         MLA     r7, r4, r5, r7  @ r7 = (icur[1]-ilast[1])*opos_frac+ilast[1]
 
         LDRSH   r5, [r3]                @ r5 = obuf[0]
-        MOV     r7, r7, ASR #16         @ r7 = tmp0 >>= 16
+        MOV     r7, r7, ASR #15         @ r7 = tmp0 >>= 15
         MUL     r7, r12,r7              @ r7 = tmp0*vol_l
         MUL     r6, r14,r6              @ r6 = tmp1*vol_r
 
-        ADDS    r7, r7, r10, LSL #16    @ r7 = obuf[1]<<16 + tmp0*vol_l
+        ADDS    r7, r7, r10, LSL #15    @ r7 = obuf[1]<<15 + tmp0*vol_l
         MOV     r4, #0
         RSCVS   r7, r4, #0x80000000     @ Clamp r7
-        ADDS    r6, r6, r5, LSL #16     @ r6 = obuf[0]<<16 + tmp1*vol_r
+        ADDS    r6, r6, r5, LSL #15     @ r6 = obuf[0]<<15 + tmp1*vol_r
         RSCVS   r6, r4, #0x80000000     @ Clamp r6
 
-        MOV     r7, r7, LSR #16         @ Shift back to halfword
-        MOV     r6, r6, LSR #16         @ Shift back to halfword
+        MOV     r7, r7, LSR #15         @ Shift back to halfword
+        MOV     r6, r6, LSR #15         @ Shift back to halfword
 
         LDR     r5, [r2,#12]            @ r5 = opos_inc
         STRH    r6, [r3],#2             @ Store output value