From 613e76debb43c42457429d4a7e81fde8ce5831b2 Mon Sep 17 00:00:00 2001
From: Ozkan Sezer <[EMAIL REDACTED]>
Date: Tue, 13 Dec 2022 08:55:40 +0300
Subject: [PATCH] music_wavpack.c: add 4x decimator to allow noise-free DSD
playback
... or any other file over 256,000 Hz sampling rate. The code isn't
particularly easy on the CPU, but it doesn't use any more resources
unless it's actually used.
Also, remove the unnecessary OPEN_WVC flag from WavpackOpenFileInputEx
calls.
Patch from David Bryant (@dbry at github.)
---
src/codecs/music_wavpack.c | 150 +++++++++++++++++++++++++++++++++++--
1 file changed, 144 insertions(+), 6 deletions(-)
diff --git a/src/codecs/music_wavpack.c b/src/codecs/music_wavpack.c
index 998d1dd0..3b92684d 100644
--- a/src/codecs/music_wavpack.c
+++ b/src/codecs/music_wavpack.c
@@ -45,6 +45,11 @@
#define WAVPACK4_OR_OLDER
#endif
+static void *decimation_init (int num_channels, int ratio);
+static int decimation_run (void *context, int32_t *samples, int num_samples);
+static void decimation_reset (void *context);
+static void *decimation_destroy (void *context);
+
#ifdef WAVPACK4_OR_OLDER
typedef struct {
int32_t (*read_bytes)(void *id, void *data, int32_t bcount);
@@ -185,7 +190,8 @@ typedef struct {
WavpackContext *ctx;
int64_t numsamples;
uint32_t samplerate;
- int bps, channels, mode;
+ int bps, channels, mode, decimation;
+ void *decimation_cnxt;
SDL_AudioStream *stream;
int32_t *buffer;
@@ -349,8 +355,8 @@ static void *WAVPACK_CreateFromRW_internal(SDL_RWops *src1, SDL_RWops *src2, int
music->volume = MIX_MAX_VOLUME;
music->ctx = (wvpk.WavpackOpenFileInputEx64 != NULL) ?
- wvpk.WavpackOpenFileInputEx64(&sdl_reader64, src1, src2, err, OPEN_WVC|OPEN_NORMALIZE|OPEN_TAGS, 0) :
- wvpk.WavpackOpenFileInputEx(&sdl_reader32, src1, src2, err, OPEN_WVC|OPEN_NORMALIZE|OPEN_TAGS, 0);
+ wvpk.WavpackOpenFileInputEx64(&sdl_reader64, src1, src2, err, OPEN_NORMALIZE|OPEN_TAGS|OPEN_DSD_AS_PCM, 0) :
+ wvpk.WavpackOpenFileInputEx(&sdl_reader32, src1, src2, err, OPEN_NORMALIZE|OPEN_TAGS, 0);
if (!music->ctx) {
Mix_SetError("%s", err);
SDL_free(music);
@@ -367,18 +373,32 @@ static void *WAVPACK_CreateFromRW_internal(SDL_RWops *src1, SDL_RWops *src2, int
music->bps = wvpk.WavpackGetBytesPerSample(music->ctx) << 3;
music->channels = wvpk.WavpackGetNumChannels(music->ctx);
music->mode = wvpk.WavpackGetMode(music->ctx);
+ music->decimation = 1;
if (freesrc2) {
*freesrc2 = 0; /* WAVPACK_Delete() will free it. */
}
+ /* for very high sample rates (including DSD, which will normally be 352,800 Hz)
+ * decimate 4x here before sending on */
+ if (music->samplerate >= 256000) {
+ music->decimation = 4;
+ music->decimation_cnxt = decimation_init (music->channels, music->decimation);
+
+ if (!music->decimation_cnxt) {
+ SDL_OutOfMemory();
+ WAVPACK_Delete(music);
+ return NULL;
+ }
+ }
+
#if WAVPACK_DBG
SDL_Log("WavPack loader:\n numsamples: %" SDL_PRIs64 "\n samplerate: %d\n bitspersample: %d\n channels: %d\n mode: 0x%x\n lossy: %d\n duration: %f\n",
(Sint64)music->numsamples, music->samplerate, music->bps, music->channels, music->mode, !(music->mode & MODE_LOSSLESS), music->numsamples/(double)music->samplerate);
#endif
format = (music->mode & MODE_FLOAT) ? AUDIO_F32SYS : AUDIO_S32SYS;
- music->stream = SDL_NewAudioStream(format, (Uint8)music->channels, (int)music->samplerate,
+ music->stream = SDL_NewAudioStream(format, (Uint8)music->channels, (int)music->samplerate / music->decimation,
music_spec.format, music_spec.channels, music_spec.freq);
if (!music->stream) {
WAVPACK_Delete(music);
@@ -386,7 +406,7 @@ static void *WAVPACK_CreateFromRW_internal(SDL_RWops *src1, SDL_RWops *src2, int
}
music->frames = music_spec.samples;
- music->buffer = (int32_t *)SDL_malloc(music->frames * music->channels * sizeof(int32_t));
+ music->buffer = (int32_t *)SDL_malloc(music->frames * music->channels * sizeof(int32_t) * music->decimation);
if (!music->buffer) {
SDL_OutOfMemory();
WAVPACK_Delete(music);
@@ -475,7 +495,12 @@ static int WAVPACK_GetSome(void *context, void *data, int bytes, SDL_bool *done)
return 0;
}
- amount = (int) wvpk.WavpackUnpackSamples(music->ctx, music->buffer, music->frames);
+ amount = (int) wvpk.WavpackUnpackSamples(music->ctx, music->buffer, music->frames * music->decimation);
+
+ if (amount && music->decimation_cnxt) {
+ amount = decimation_run (music->decimation_cnxt, music->buffer, amount);
+ }
+
if (amount) {
amount *= music->channels;
if (music->bps < 32) {
@@ -522,6 +547,9 @@ static int WAVPACK_Seek(void *context, double time)
if (!success) {
return Mix_SetError("%s", wvpk.WavpackGetErrorMessage(music->ctx));
}
+ if (music->decimation_cnxt) {
+ decimation_reset (music->decimation_cnxt);
+ }
return 0;
}
@@ -547,6 +575,9 @@ static void WAVPACK_Delete(void *context)
WAVPACK_music *music = (WAVPACK_music *)context;
meta_tags_clear(&music->tags);
wvpk.WavpackCloseFile(music->ctx);
+ if (music->decimation_cnxt) {
+ decimation_destroy (music->decimation_cnxt);
+ }
if (music->stream) {
SDL_FreeAudioStream(music->stream);
}
@@ -562,6 +593,113 @@ static void WAVPACK_Delete(void *context)
SDL_free(music);
}
+/* Decimation code for playing DSD (which comes from the library already decimated 8x) */
+
+/* sinc low-pass filter, cutoff = fs/12, 80 terms */
+
+static const int32_t filter[] = {
+ 50, 464, 968, 711, -1203, -5028, -9818, -13376,
+ -12870, -6021, 7526, 25238, 41688, 49778, 43050, 18447,
+ -21428, -67553, -105876, -120890, -100640, -41752, 47201, 145510,
+ 224022, 252377, 208224, 86014, -97312, -301919, -470919, -541796,
+ -461126, -199113, 239795, 813326, 1446343, 2043793, 2509064, 2763659,
+ 2763659, 2509064, 2043793, 1446343, 813326, 239795, -199113, -461126,
+ -541796, -470919, -301919, -97312, 86014, 208224, 252377, 224022,
+ 145510, 47201, -41752, -100640, -120890, -105876, -67553, -21428,
+ 18447, 43050, 49778, 41688, 25238, 7526, -6021, -12870,
+ -13376, -9818, -5028, -1203, 711, 968, 464, 50
+};
+
+#define NUM_TERMS ((int)(sizeof(filter) / sizeof(filter[0])))
+
+typedef struct chan_state {
+ int delay[NUM_TERMS], index, num_channels, ratio;
+} ChanState;
+
+static void *decimation_init (int num_channels, int ratio)
+{
+ ChanState *sp = (ChanState *)SDL_calloc(num_channels, sizeof(ChanState));
+ int i;
+
+ if (sp) {
+ for (i = 0; i < num_channels; ++i) {
+ sp[i].num_channels = num_channels;
+ sp[i].index = NUM_TERMS - ratio;
+ sp[i].ratio = ratio;
+ }
+ }
+
+ return sp;
+}
+
+static int decimation_run (void *context, int32_t *samples, int num_samples)
+{
+ int32_t *in_samples = samples, *out_samples = samples;
+ ChanState *sp = (ChanState *) context;
+ int num_channels, ratio, chan;
+
+ if (!sp) {
+ return 0;
+ }
+
+ num_channels = sp->num_channels;
+ ratio = sp->ratio;
+ chan = 0;
+
+ while (num_samples) {
+ sp = ((ChanState *) context) + chan;
+
+ sp->delay[sp->index++] = *in_samples++;
+
+ if (sp->index == NUM_TERMS) {
+ int64_t sum = 0;
+ int i;
+
+ for (i = 0; i < NUM_TERMS; ++i) {
+ sum += (int64_t) filter[i] * sp->delay[i];
+ }
+
+ *out_samples++ = (int32_t)(sum >> 24);
+ SDL_memmove(sp->delay, sp->delay + ratio, sizeof(sp->delay[0]) * (NUM_TERMS - ratio));
+ sp->index = NUM_TERMS - ratio;
+ }
+
+ if (++chan == num_channels) {
+ num_samples--;
+ chan = 0;
+ }
+ }
+
+ return (int)(out_samples - samples) / num_channels;
+}
+
+static void decimation_reset (void *context)
+{
+ ChanState *sp = (ChanState *) context;
+ int num_channels, ratio, i;
+
+ if (sp) {
+ num_channels = sp->num_channels;
+ ratio = sp->ratio;
+
+ SDL_memset(sp, 0, sizeof(ChanState) * num_channels);
+
+ for (i = 0; i < num_channels; ++i) {
+ sp[i].num_channels = num_channels;
+ sp[i].index = NUM_TERMS - ratio;
+ sp[i].ratio = ratio;
+ }
+ }
+}
+
+static void *decimation_destroy (void *context)
+{
+ if (context) {
+ SDL_free (context);
+ }
+ return NULL;
+}
+
Mix_MusicInterface Mix_MusicInterface_WAVPACK =
{
"WAVPACK",