From c668de30852552a275b809a24648aa5a887eac65 Mon Sep 17 00:00:00 2001 From: Thom Johansen Date: Wed, 29 Aug 2007 14:32:52 +0000 Subject: FS #7286. Do correct rounding of final 16 bit samples before sending to DAC, for you golden-eared people. git-svn-id: svn://svn.rockbox.org/rockbox/trunk@14514 a1c6a512-1295-4272-9138-f99709370657 --- apps/dsp.c | 8 +++++--- apps/dsp_cf.S | 40 ++++++++++++++++++++++++++++++---------- apps/plugins/test_codec.c | 13 +++++++------ 3 files changed, 42 insertions(+), 19 deletions(-) diff --git a/apps/dsp.c b/apps/dsp.c index f05c6f14ce..699b0c5c67 100644 --- a/apps/dsp.c +++ b/apps/dsp.c @@ -414,10 +414,11 @@ static void sample_output_mono(int count, struct dsp_data *data, { const int32_t *s0 = src[0]; const int scale = data->output_scale; + const int dc_bias = 1 << (scale - 1); do { - int32_t lr = clip_sample_16(*s0++ >> scale); + int32_t lr = clip_sample_16((*s0++ + dc_bias) >> scale); *dst++ = lr; *dst++ = lr; } @@ -433,11 +434,12 @@ static void sample_output_stereo(int count, struct dsp_data *data, const int32_t *s0 = src[0]; const int32_t *s1 = src[1]; const int scale = data->output_scale; + const int dc_bias = 1 << (scale - 1); do { - *dst++ = clip_sample_16(*s0++ >> scale); - *dst++ = clip_sample_16(*s1++ >> scale); + *dst++ = clip_sample_16((*s0++ + dc_bias) >> scale); + *dst++ = clip_sample_16((*s1++ + dc_bias) >> scale); } while (--count > 0); } diff --git a/apps/dsp_cf.S b/apps/dsp_cf.S index e4869b7c9a..fec00020b5 100644 --- a/apps/dsp_cf.S +++ b/apps/dsp_cf.S @@ -415,11 +415,11 @@ channels_process_sound_chan_karaoke: .align 2 .global sample_output_stereo sample_output_stereo: - lea.l -44(%sp), %sp | save registers + lea.l -48(%sp), %sp | save registers move.l %macsr, %d1 | do it now as at many lines will - movem.l %d1-%d7/%a2-%a5, (%sp) | be the far more common condition + movem.l %d1-%d7/%a2-%a6, (%sp) | be the far more common condition move.l #0x80, %macsr | put emac unit in signed int mode - movem.l 48(%sp), %a0-%a2/%a4 | + movem.l 52(%sp), %a0-%a2/%a4 | lea.l (%a4, %a0.l*4), %a0 | %a0 = end address move.l (%a1), %d1 | %a1 = multiplier: (1 << (16 - scale)) sub.l #16, %d1 | @@ -427,6 +427,7 @@ sample_output_stereo: moveq.l #1, %d0 | asl.l %d1, %d0 | move.l %d0, %a1 | + move.l #0x8000, %a6 | %a6 = rounding term movem.l (%a2), %a2-%a3 | get L/R channel pointers moveq.l #28, %d0 | %d0 = second line bound add.l %a4, %d0 | @@ -438,6 +439,8 @@ sample_output_stereo: bls.b 20f | line loop start | no? start line loop 10: | long loop 0 | move.l (%a2)+, %d1 | read longword from L and R + move.l %a6, %acc0 | + move.l %acc0, %acc1 | mac.l %d1, %a1, (%a3)+, %d2, %acc0 | shift L to high word mac.l %d2, %a1, %acc1 | shift R to high word movclr.l %acc0, %d1 | get possibly saturated results @@ -451,6 +454,10 @@ sample_output_stereo: lea.l -12(%a0), %a5 | %a5 = at or just before last line bound 30: | line loop | move.l (%a3)+, %d4 | get next 4 R samples and scale + move.l %a6, %acc0 | + move.l %acc0, %acc1 | + move.l %acc1, %acc2 | + move.l %acc2, %acc3 | mac.l %d4, %a1, (%a3)+, %d5, %acc0 | with saturation mac.l %d5, %a1, (%a3)+, %d6, %acc1 | mac.l %d6, %a1, (%a3)+, %d7, %acc2 | @@ -460,6 +467,10 @@ sample_output_stereo: movclr.l %acc1, %d5 | movclr.l %acc2, %d6 | movclr.l %acc3, %d7 | + move.l %a6, %acc0 | + move.l %acc0, %acc1 | + move.l %acc1, %acc2 | + move.l %acc2, %acc3 | mac.l %d0, %a1, (%a2)+, %d1, %acc0 | get next 4 L samples and scale mac.l %d1, %a1, (%a2)+, %d2, %acc1 | with saturation mac.l %d2, %a1, (%a2)+, %d3, %acc2 | @@ -484,6 +495,8 @@ sample_output_stereo: bls.b 60f | output end | no? stop 50: | long loop 1 | move.l (%a2)+, %d1 | handle trailing longwords + move.l %a6, %acc0 | + move.l %acc0, %acc1 | mac.l %d1, %a1, (%a3)+, %d2, %acc0 | the same way as leading ones mac.l %d2, %a1, %acc1 | movclr.l %acc0, %d1 | @@ -494,9 +507,9 @@ sample_output_stereo: cmp.l %a4, %a0 | bhi.b 50b | long loop 1 60: | output end | - movem.l (%sp), %d1-%d7/%a2-%a5 | restore registers + movem.l (%sp), %d1-%d7/%a2-%a6 | restore registers move.l %d1, %macsr | - lea.l 44(%sp), %sp | cleanup + lea.l 48(%sp), %sp | cleanup rts | .size sample_output_stereo, .-sample_output_stereo @@ -510,17 +523,18 @@ sample_output_stereo: .align 2 .global sample_output_mono sample_output_mono: - lea.l -28(%sp), %sp | save registers + lea.l -32(%sp), %sp | save registers move.l %macsr, %d1 | do it now as at many lines will - movem.l %d1-%d5/%a2-%a3, (%sp) | be the far more common condition + movem.l %d1-%d5/%a2-%a4, (%sp) | be the far more common condition move.l #0x80, %macsr | put emac unit in signed int mode - movem.l 32(%sp), %a0-%a3 | + movem.l 36(%sp), %a0-%a3 | lea.l (%a3, %a0.l*4), %a0 | %a0 = end address move.l (%a1), %d1 | %d5 = multiplier: (1 << (16 - scale)) sub.l #16, %d1 | neg.l %d1 | moveq.l #1, %d5 | asl.l %d1, %d5 | + move.l #0x8000, %a4 | %a4 = rounding term movem.l (%a2), %a2 | get source channel pointer moveq.l #28, %d0 | %d0 = second line bound add.l %a3, %d0 | @@ -532,6 +546,7 @@ sample_output_mono: bls.b 20f | line loop start | no? start line loop 10: | long loop 0 | move.l (%a2)+, %d1 | read longword from L and R + move.l %a4, %acc0 | mac.l %d1, %d5, %acc0 | shift L to high word movclr.l %acc0, %d1 | get possibly saturated results move.l %d1, %d2 | @@ -544,6 +559,10 @@ sample_output_mono: lea.l -12(%a0), %a1 | %a1 = at or just before last line bound 30: | line loop | move.l (%a2)+, %d0 | get next 4 L samples and scale + move.l %a4, %acc0 | + move.l %acc0, %acc1 | + move.l %acc1, %acc2 | + move.l %acc2, %acc3 | mac.l %d0, %d5, (%a2)+, %d1, %acc0 | with saturation mac.l %d1, %d5, (%a2)+, %d2, %acc1 | mac.l %d2, %d5, (%a2)+, %d3, %acc2 | @@ -573,6 +592,7 @@ sample_output_mono: bls.b 60f | output end | no? stop 50: | loop loop 1 | move.l (%a2)+, %d1 | handle trailing longwords + move.l %a4, %acc0 | mac.l %d1, %d5, %acc0 | the same way as leading ones movclr.l %acc0, %d1 | move.l %d1, %d2 | @@ -582,8 +602,8 @@ sample_output_mono: cmp.l %a3, %a0 | bhi.b 50b | long loop 1 | 60: | output end | - movem.l (%sp), %d1-%d5/%a2-%a3 | restore registers + movem.l (%sp), %d1-%d5/%a2-%a4 | restore registers move.l %d1, %macsr | - lea.l 28(%sp), %sp | cleanup + lea.l 32(%sp), %sp | cleanup rts | .size sample_output_mono, .-sample_output_mono diff --git a/apps/plugins/test_codec.c b/apps/plugins/test_codec.c index c29094b372..4346a23304 100644 --- a/apps/plugins/test_codec.c +++ b/apps/plugins/test_codec.c @@ -221,7 +221,8 @@ static bool pcmbuf_insert_wav(const void *ch1, const void *ch2, int count) const int32_t* data1_32; const int32_t* data2_32; unsigned char* p = wavbuffer; - int scale = wavinfo.sampledepth - 15; + const int scale = wavinfo.sampledepth - 15; + const int dc_bias = 1 << (scale - 1); /* Prevent idle poweroff */ rb->reset_poweroff_timer(); @@ -266,18 +267,18 @@ static bool pcmbuf_insert_wav(const void *ch1, const void *ch2, int count) { case STEREO_INTERLEAVED: while (count--) { - int2le16(p, clip_sample((*data1_32++) >> scale)); + int2le16(p, clip_sample((*data1_32++ + dc_bias) >> scale)); p += 2; - int2le16(p, clip_sample((*data1_32++) >> scale)); + int2le16(p, clip_sample((*data1_32++ + dc_bias) >> scale)); p += 2; } break; case STEREO_NONINTERLEAVED: while (count--) { - int2le16(p, clip_sample((*data1_32++) >> scale)); + int2le16(p, clip_sample((*data1_32++ + dc_bias) >> scale)); p += 2; - int2le16(p, clip_sample((*data2_32++) >> scale)); + int2le16(p, clip_sample((*data2_32++ + dc_bias) >> scale)); p += 2; } @@ -285,7 +286,7 @@ static bool pcmbuf_insert_wav(const void *ch1, const void *ch2, int count) case STEREO_MONO: while (count--) { - int2le16(p, clip_sample((*data1_32++) >> scale)); + int2le16(p, clip_sample((*data1_32++ + dc_bias) >> scale)); p += 2; } break; -- cgit v1.2.3