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Update dr_mp3.

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This commit is contained in:
David Reid
2018-08-25 15:33:12 +10:00
parent 26e9eb620f
commit 726da178d1
1 changed files with 167 additions and 37 deletions
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extras/dr_mp3.h
+167 -37
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@@ -1,5 +1,5 @@
// MP3 audio decoder. Public domain. See "unlicense" statement at the end of this file.
// dr_mp3 - v0.2.11 - 2018-08-08
// dr_mp3 - v0.3.0 - 2018-08-25
//
// David Reid - mackron@gmail.com
//
@@ -113,7 +113,12 @@ typedef struct
void drmp3dec_init(drmp3dec *dec);
// Reads a frame from a low level decoder.
int drmp3dec_decode_frame(drmp3dec *dec, const unsigned char *mp3, int mp3_bytes, short *pcm, drmp3dec_frame_info *info);
int drmp3dec_decode_frame(drmp3dec *dec, const unsigned char *mp3, int mp3_bytes, void *pcm, drmp3dec_frame_info *info);
// Helper for converting between f32 and s16.
void drmp3dec_f32_to_s16(const float *in, drmp3_int16 *out, int num_samples);
// Main API (Pull API)
@@ -213,7 +218,7 @@ typedef struct
drmp3_uint32 frameSampleRate; // The sample rate of the currently loaded MP3 frame. Internal use only.
drmp3_uint32 framesConsumed;
drmp3_uint32 framesRemaining;
drmp3_int16 frames[DRMP3_MAX_SAMPLES_PER_FRAME];
drmp3_uint8 frames[sizeof(float)*DRMP3_MAX_SAMPLES_PER_FRAME]; // <-- Multipled by sizeof(float) to ensure there's enough room for DR_MP3_FLOAT_OUTPUT.
drmp3_src src;
size_t dataSize;
size_t dataCapacity;
@@ -307,6 +312,8 @@ void drmp3_free(void* p);
#define DR_MP3_NO_SIMD
#endif
#define DRMP3_OFFSET_PTR(p, offset) ((void*)((drmp3_uint8*)(p) + (offset)))
#define DRMP3_MAX_FREE_FORMAT_FRAME_SIZE 2304 /* more than ISO spec's */
#define DRMP3_MAX_FRAME_SYNC_MATCHES 10
@@ -1028,7 +1035,7 @@ static void drmp3_L3_huffman(float *dst, drmp3_bs *bs, const drmp3_L3_gr_info *g
{
int tab_num = gr_info->table_select[ireg];
int sfb_cnt = gr_info->region_count[ireg++];
const short *codebook = tabs + tabindex[tab_num];
const drmp3_int16 *codebook = tabs + tabindex[tab_num];
int linbits = g_linbits[tab_num];
do
{
@@ -1634,18 +1641,27 @@ static void drmp3d_DCT_II(float *grbuf, int n)
#endif
}
static short drmp3d_scale_pcm(float sample)
{
if (sample > 32767.0) return (short) 32767;
if (sample < -32768.0) return (short)-32768;
int s = (int)(sample + .5f);
s -= (s < 0); /* away from zero, to be compliant */
if (s > 32767) return (short) 32767;
if (s < -32768) return (short)-32768;
return (short)s;
}
#ifndef DR_MP3_FLOAT_OUTPUT
typedef drmp3_int16 drmp3d_sample_t;
static void drmp3d_synth_pair(short *pcm, int nch, const float *z)
static drmp3_int16 drmp3d_scale_pcm(float sample)
{
if (sample >= 32766.5) return (drmp3_int16) 32767;
if (sample <= -32767.5) return (drmp3_int16)-32768;
drmp3_int16 s = (drmp3_int16)(sample + .5f);
s -= (s < 0); /* away from zero, to be compliant */
return (drmp3_int16)s;
}
#else
typedef float drmp3d_sample_t;
static float drmp3d_scale_pcm(float sample)
{
return sample*(1.f/32768.f);
}
#endif
static void drmp3d_synth_pair(drmp3d_sample_t *pcm, int nch, const float *z)
{
float a;
a = (z[14*64] - z[ 0]) * 29;
@@ -1670,11 +1686,11 @@ static void drmp3d_synth_pair(short *pcm, int nch, const float *z)
pcm[16*nch] = drmp3d_scale_pcm(a);
}
static void drmp3d_synth(float *xl, short *dstl, int nch, float *lins)
static void drmp3d_synth(float *xl, drmp3d_sample_t *dstl, int nch, float *lins)
{
int i;
float *xr = xl + 576*(nch - 1);
short *dstr = dstl + (nch - 1);
drmp3d_sample_t *dstr = dstl + (nch - 1);
static const float g_win[] = {
-1,26,-31,208,218,401,-519,2063,2000,4788,-5517,7134,5959,35640,-39336,74992,
@@ -1731,19 +1747,20 @@ static void drmp3d_synth(float *xl, short *dstl, int nch, float *lins)
DRMP3_V0(0) DRMP3_V2(1) DRMP3_V1(2) DRMP3_V2(3) DRMP3_V1(4) DRMP3_V2(5) DRMP3_V1(6) DRMP3_V2(7)
{
#ifndef DR_MP3_FLOAT_OUTPUT
#if DRMP3_HAVE_SSE
static const drmp3_f4 g_max = { 32767.0f, 32767.0f, 32767.0f, 32767.0f };
static const drmp3_f4 g_min = { -32768.0f, -32768.0f, -32768.0f, -32768.0f };
__m128i pcm8 = _mm_packs_epi32(_mm_cvtps_epi32(_mm_max_ps(_mm_min_ps(a, g_max), g_min)),
_mm_cvtps_epi32(_mm_max_ps(_mm_min_ps(b, g_max), g_min)));
dstr[(15 - i)*nch] = (short)_mm_extract_epi16(pcm8, 1);
dstr[(17 + i)*nch] = (short)_mm_extract_epi16(pcm8, 5);
dstl[(15 - i)*nch] = (short)_mm_extract_epi16(pcm8, 0);
dstl[(17 + i)*nch] = (short)_mm_extract_epi16(pcm8, 4);
dstr[(47 - i)*nch] = (short)_mm_extract_epi16(pcm8, 3);
dstr[(49 + i)*nch] = (short)_mm_extract_epi16(pcm8, 7);
dstl[(47 - i)*nch] = (short)_mm_extract_epi16(pcm8, 2);
dstl[(49 + i)*nch] = (short)_mm_extract_epi16(pcm8, 6);
dstr[(15 - i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 1);
dstr[(17 + i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 5);
dstl[(15 - i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 0);
dstl[(17 + i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 4);
dstr[(47 - i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 3);
dstr[(49 + i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 7);
dstl[(47 - i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 2);
dstl[(49 + i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 6);
#else
int16x4_t pcma, pcmb;
a = DRMP3_VADD(a, DRMP3_VSET(0.5f));
@@ -1759,6 +1776,30 @@ static void drmp3d_synth(float *xl, short *dstl, int nch, float *lins)
vst1_lane_s16(dstl + (47 - i)*nch, pcma, 2);
vst1_lane_s16(dstl + (49 + i)*nch, pcmb, 2);
#endif
#else
static const drmp3_f4 g_scale = { 1.0f/32768.0f, 1.0f/32768.0f, 1.0f/32768.0f, 1.0f/32768.0f };
a = DRMP3_VMUL(a, g_scale);
b = DRMP3_VMUL(b, g_scale);
#if DRMP3_HAVE_SSE
_mm_store_ss(dstr + (15 - i)*nch, _mm_shuffle_ps(a, a, _MM_SHUFFLE(1, 1, 1, 1)));
_mm_store_ss(dstr + (17 + i)*nch, _mm_shuffle_ps(b, b, _MM_SHUFFLE(1, 1, 1, 1)));
_mm_store_ss(dstl + (15 - i)*nch, _mm_shuffle_ps(a, a, _MM_SHUFFLE(0, 0, 0, 0)));
_mm_store_ss(dstl + (17 + i)*nch, _mm_shuffle_ps(b, b, _MM_SHUFFLE(0, 0, 0, 0)));
_mm_store_ss(dstr + (47 - i)*nch, _mm_shuffle_ps(a, a, _MM_SHUFFLE(3, 3, 3, 3)));
_mm_store_ss(dstr + (49 + i)*nch, _mm_shuffle_ps(b, b, _MM_SHUFFLE(3, 3, 3, 3)));
_mm_store_ss(dstl + (47 - i)*nch, _mm_shuffle_ps(a, a, _MM_SHUFFLE(2, 2, 2, 2)));
_mm_store_ss(dstl + (49 + i)*nch, _mm_shuffle_ps(b, b, _MM_SHUFFLE(2, 2, 2, 2)));
#else
vst1q_lane_f32(dstr + (15 - i)*nch, a, 1);
vst1q_lane_f32(dstr + (17 + i)*nch, b, 1);
vst1q_lane_f32(dstl + (15 - i)*nch, a, 0);
vst1q_lane_f32(dstl + (17 + i)*nch, b, 0);
vst1q_lane_f32(dstr + (47 - i)*nch, a, 3);
vst1q_lane_f32(dstr + (49 + i)*nch, b, 3);
vst1q_lane_f32(dstl + (47 - i)*nch, a, 2);
vst1q_lane_f32(dstl + (49 + i)*nch, b, 2);
#endif
#endif /* DR_MP3_FLOAT_OUTPUT */
}
} else
#endif
@@ -1796,7 +1837,7 @@ static void drmp3d_synth(float *xl, short *dstl, int nch, float *lins)
#endif
}
static void drmp3d_synth_granule(float *qmf_state, float *grbuf, int nbands, int nch, short *pcm, float *lins)
static void drmp3d_synth_granule(float *qmf_state, float *grbuf, int nbands, int nch, drmp3d_sample_t *pcm, float *lins)
{
int i;
for (i = 0; i < nch; i++)
@@ -1881,7 +1922,7 @@ void drmp3dec_init(drmp3dec *dec)
dec->header[0] = 0;
}
int drmp3dec_decode_frame(drmp3dec *dec, const unsigned char *mp3, int mp3_bytes, short *pcm, drmp3dec_frame_info *info)
int drmp3dec_decode_frame(drmp3dec *dec, const unsigned char *mp3, int mp3_bytes, void *pcm, drmp3dec_frame_info *info)
{
int i = 0, igr, frame_size = 0, success = 1;
const drmp3_uint8 *hdr;
@@ -1937,7 +1978,7 @@ int drmp3dec_decode_frame(drmp3dec *dec, const unsigned char *mp3, int mp3_bytes
success = drmp3_L3_restore_reservoir(dec, bs_frame, &scratch, main_data_begin);
if (success)
{
for (igr = 0; igr < (DRMP3_HDR_TEST_MPEG1(hdr) ? 2 : 1); igr++, pcm += 576*info->channels)
for (igr = 0; igr < (DRMP3_HDR_TEST_MPEG1(hdr) ? 2 : 1); igr++, pcm = DRMP3_OFFSET_PTR(pcm, sizeof(drmp3d_sample_t)*576*info->channels))
{
memset(scratch.grbuf[0], 0, 576*2*sizeof(float));
drmp3_L3_decode(dec, &scratch, scratch.gr_info + igr*info->channels, info->channels);
@@ -1962,7 +2003,7 @@ int drmp3dec_decode_frame(drmp3dec *dec, const unsigned char *mp3, int mp3_bytes
drmp3_L12_apply_scf_384(sci, sci->scf + igr, scratch.grbuf[0]);
drmp3d_synth_granule(dec->qmf_state, scratch.grbuf[0], 12, info->channels, pcm, scratch.syn[0]);
memset(scratch.grbuf[0], 0, 576*2*sizeof(float));
pcm += 384*info->channels;
pcm = DRMP3_OFFSET_PTR(pcm, sizeof(drmp3d_sample_t)*384*info->channels);
}
if (bs_frame->pos > bs_frame->limit)
{
@@ -1975,6 +2016,64 @@ int drmp3dec_decode_frame(drmp3dec *dec, const unsigned char *mp3, int mp3_bytes
return success*drmp3_hdr_frame_samples(dec->header);
}
void drmp3dec_f32_to_s16(const float *in, drmp3_int16 *out, int num_samples)
{
if(num_samples > 0)
{
int i = 0;
#if DRMP3_HAVE_SIMD
int aligned_count = num_samples & ~7;
for(; i < aligned_count; i+=8)
{
static const drmp3_f4 g_scale = { 32768.0f, 32768.0f, 32768.0f, 32768.0f };
drmp3_f4 a = DRMP3_VMUL(DRMP3_VLD(&in[i ]), g_scale);
drmp3_f4 b = DRMP3_VMUL(DRMP3_VLD(&in[i+4]), g_scale);
#if DRMP3_HAVE_SSE
static const drmp3_f4 g_max = { 32767.0f, 32767.0f, 32767.0f, 32767.0f };
static const drmp3_f4 g_min = { -32768.0f, -32768.0f, -32768.0f, -32768.0f };
__m128i pcm8 = _mm_packs_epi32(_mm_cvtps_epi32(_mm_max_ps(_mm_min_ps(a, g_max), g_min)),
_mm_cvtps_epi32(_mm_max_ps(_mm_min_ps(b, g_max), g_min)));
out[i ] = (drmp3_int16)_mm_extract_epi16(pcm8, 0);
out[i+1] = (drmp3_int16)_mm_extract_epi16(pcm8, 1);
out[i+2] = (drmp3_int16)_mm_extract_epi16(pcm8, 2);
out[i+3] = (drmp3_int16)_mm_extract_epi16(pcm8, 3);
out[i+4] = (drmp3_int16)_mm_extract_epi16(pcm8, 4);
out[i+5] = (drmp3_int16)_mm_extract_epi16(pcm8, 5);
out[i+6] = (drmp3_int16)_mm_extract_epi16(pcm8, 6);
out[i+7] = (drmp3_int16)_mm_extract_epi16(pcm8, 7);
#else
int16x4_t pcma, pcmb;
a = DRMP3_VADD(a, DRMP3_VSET(0.5f));
b = DRMP3_VADD(b, DRMP3_VSET(0.5f));
pcma = vqmovn_s32(vqaddq_s32(vcvtq_s32_f32(a), vreinterpretq_s32_u32(vcltq_f32(a, DRMP3_VSET(0)))));
pcmb = vqmovn_s32(vqaddq_s32(vcvtq_s32_f32(b), vreinterpretq_s32_u32(vcltq_f32(b, DRMP3_VSET(0)))));
vst1_lane_s16(out+i , pcma, 0);
vst1_lane_s16(out+i+1, pcma, 1);
vst1_lane_s16(out+i+2, pcma, 2);
vst1_lane_s16(out+i+3, pcma, 3);
vst1_lane_s16(out+i+4, pcmb, 0);
vst1_lane_s16(out+i+5, pcmb, 1);
vst1_lane_s16(out+i+6, pcmb, 2);
vst1_lane_s16(out+i+7, pcmb, 3);
#endif
}
#endif
for(; i < num_samples; i++)
{
float sample = in[i] * 32768.0f;
if (sample >= 32766.5)
out[i] = (drmp3_int16) 32767;
else if (sample <= -32767.5)
out[i] = (drmp3_int16)-32768;
else
{
short s = (drmp3_int16)(sample + .5f);
s -= (s < 0); /* away from zero, to be compliant */
out[i] = s;
}
}
}
}
@@ -2316,7 +2415,7 @@ static drmp3_bool32 drmp3_decode_next_frame(drmp3* pMP3)
}
drmp3dec_frame_info info;
drmp3_uint32 samplesRead = drmp3dec_decode_frame(&pMP3->decoder, pMP3->pData, (int)pMP3->dataSize, pMP3->frames, &info); // <-- Safe size_t -> int conversion thanks to the check above.
drmp3_uint32 samplesRead = drmp3dec_decode_frame(&pMP3->decoder, pMP3->pData, (int)pMP3->dataSize, (drmp3d_sample_t*)pMP3->frames, &info); // <-- Safe size_t -> int conversion thanks to the check above.
if (samplesRead != 0) {
size_t leftoverDataSize = (pMP3->dataSize - (size_t)info.frame_bytes);
for (size_t i = 0; i < leftoverDataSize; ++i) {
@@ -2369,28 +2468,54 @@ static drmp3_uint64 drmp3_read_src(drmp3_src* pSRC, drmp3_uint64 frameCount, voi
while (frameCount > 0) {
// Read from the in-memory buffer first.
while (pMP3->framesRemaining > 0 && frameCount > 0) {
drmp3d_sample_t* frames = (drmp3d_sample_t*)pMP3->frames;
#ifndef DR_MP3_FLOAT_OUTPUT
if (pMP3->frameChannels == 1) {
if (pMP3->channels == 1) {
// Mono -> Mono.
pFramesOutF[0] = pMP3->frames[pMP3->framesConsumed] / 32768.0f;
pFramesOutF[0] = frames[pMP3->framesConsumed] / 32768.0f;
} else {
// Mono -> Stereo.
pFramesOutF[0] = pMP3->frames[pMP3->framesConsumed] / 32768.0f;
pFramesOutF[1] = pMP3->frames[pMP3->framesConsumed] / 32768.0f;
pFramesOutF[0] = frames[pMP3->framesConsumed] / 32768.0f;
pFramesOutF[1] = frames[pMP3->framesConsumed] / 32768.0f;
}
} else {
if (pMP3->channels == 1) {
// Stereo -> Mono
float sample = 0;
sample += pMP3->frames[(pMP3->framesConsumed*pMP3->frameChannels)+0] / 32768.0f;
sample += pMP3->frames[(pMP3->framesConsumed*pMP3->frameChannels)+1] / 32768.0f;
sample += frames[(pMP3->framesConsumed*pMP3->frameChannels)+0] / 32768.0f;
sample += frames[(pMP3->framesConsumed*pMP3->frameChannels)+1] / 32768.0f;
pFramesOutF[0] = sample * 0.5f;
} else {
// Stereo -> Stereo
pFramesOutF[0] = pMP3->frames[(pMP3->framesConsumed*pMP3->frameChannels)+0] / 32768.0f;
pFramesOutF[1] = pMP3->frames[(pMP3->framesConsumed*pMP3->frameChannels)+1] / 32768.0f;
pFramesOutF[0] = frames[(pMP3->framesConsumed*pMP3->frameChannels)+0] / 32768.0f;
pFramesOutF[1] = frames[(pMP3->framesConsumed*pMP3->frameChannels)+1] / 32768.0f;
}
}
#else
if (pMP3->frameChannels == 1) {
if (pMP3->channels == 1) {
// Mono -> Mono.
pFramesOutF[0] = frames[pMP3->framesConsumed];
} else {
// Mono -> Stereo.
pFramesOutF[0] = frames[pMP3->framesConsumed];
pFramesOutF[1] = frames[pMP3->framesConsumed];
}
} else {
if (pMP3->channels == 1) {
// Stereo -> Mono
float sample = 0;
sample += frames[(pMP3->framesConsumed*pMP3->frameChannels)+0];
sample += frames[(pMP3->framesConsumed*pMP3->frameChannels)+1];
pFramesOutF[0] = sample * 0.5f;
} else {
// Stereo -> Stereo
pFramesOutF[0] = frames[(pMP3->framesConsumed*pMP3->frameChannels)+0];
pFramesOutF[1] = frames[(pMP3->framesConsumed*pMP3->frameChannels)+1];
}
}
#endif
pMP3->framesConsumed += 1;
pMP3->framesRemaining -= 1;
@@ -2763,6 +2888,11 @@ void drmp3_free(void* p)
// REVISION HISTORY
// ===============
//
// v0.3.0 - 2018-08-25
// - Bring up to date with minimp3. This has a minor API change: the "pcm" parameter of drmp3dec_decode_frame() has
// been changed from short* to void* because it can now output both s16 and f32 samples, depending on whether or
// not the DR_MP3_FLOAT_OUTPUT option is set.
//
// v0.2.11 - 2018-08-08
// - Fix a bug where the last part of a file is not read.
//