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Merge remote-tracking branch 'qatar/master' 

* qatar/master:
  APIchanges: fill in date and commit for request_sample_fmt
  Add floating-point sample format support to the ac3, eac3, dca, aac, and vorbis decoders.
  Add support for request_sample_format in ffmpeg and ffplay.
  Add APIchanges entry for request_sample_fmt.
  Add request_sample_fmt field to AVCodecContext.
  Add float_interleave() to FmtConvertContext with x86-optimized versions.
  Remove unused make variable SEEK_REFFILE
  fate: remove redundant aref and vref references
  fate: remove do_ffmpeg_nocheck function
  fate: do not collect -benchmark output
  mpegaudiodec: remove decode_end() function
  fate: run aref and vref as regular tests
  mpegaudio: sanitise compute_antialias_* names
  mpeg12: add slice-threading checks to slice-threading initializers.
  h264: copy pixel_shift between slice threading contexts.
  mdec: enable frame-level multithreading.
  mdec.c: fix overread.

Conflicts:
	libavcodec/aacdec.c
	libavcodec/ac3dec.c
	libavcodec/avcodec.h
	libavcodec/dca.c
	libavcodec/h264.c
	libavcodec/mdec.c
	libavcodec/mpeg12.c
	libavcodec/options.c
	libavcodec/version.h
	libavcodec/vorbisdec.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer 2011-05-19 05:12:45 +02:00
parent 8529f9b36b 41e21e4db6
commit 75a37b57a5
26 changed files with 343 additions and 119 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

15
Makefile
View File

@ -186,25 +186,18 @@ check: test
fulltest test: codectest lavftest lavfitest seektest
FFSERVER_REFFILE = $(SRC_PATH)/tests/ffserver.regression.ref
SEEK_REFFILE = $(SRC_PATH)/tests/seek.regression.ref
codectest: fate-codec
lavftest: fate-lavf
lavfitest: fate-lavfi
seektest: fate-seek
AREF = tests/data/acodec.ref.wav
VREF = tests/data/vsynth1.ref.yuv
AREF = fate-acodec-aref
VREF = fate-vsynth1-vref fate-vsynth2-vref
REFS = $(AREF) $(VREF)
$(REFS): TAG = GEN
$(VREF): ffmpeg$(EXESUF) tests/vsynth1/00.pgm tests/vsynth2/00.pgm
$(M)$(SRC_PATH)/tests/codec-regression.sh vref vsynth1 tests/vsynth1 "$(TARGET_EXEC)" "$(TARGET_PATH)"
$(Q)$(SRC_PATH)/tests/codec-regression.sh vref vsynth2 tests/vsynth2 "$(TARGET_EXEC)" "$(TARGET_PATH)"
$(AREF): ffmpeg$(EXESUF) tests/data/asynth1.sw
$(M)$(SRC_PATH)/tests/codec-regression.sh aref acodec tests/acodec "$(TARGET_EXEC)" "$(TARGET_PATH)"
ffservertest: ffserver$(EXESUF) tests/vsynth1/00.pgm tests/data/asynth1.sw
@echo
@ -258,8 +251,8 @@ FATE = $(FATE_ACODEC) \
$(FATE_LAVFI) \
$(FATE_SEEK) \
$(FATE_ACODEC): $(AREF)
$(FATE_VCODEC): $(VREF)
$(filter-out %-aref,$(FATE_ACODEC)): $(AREF)
$(filter-out %-vref,$(FATE_VCODEC)): $(VREF)
$(FATE_LAVF): $(REFS)
$(FATE_LAVFI): $(REFS) tools/lavfi-showfiltfmts$(EXESUF)
$(FATE_SEEK): fate-codec fate-lavf tests/seek_test$(EXESUF)

3
doc/APIchanges
View File

@ -13,6 +13,9 @@ libavutil: 2011-04-18
API changes, most recent first:
2011-05-18 - 64150ff - lavc 53.4.0 - AVCodecContext.request_sample_fmt
Add request_sample_fmt field to AVCodecContext.
2011-05-10 - 188dea1 - lavc 53.3.0 - avcodec.h
Deprecate AVLPCType and the following fields in
AVCodecContext: lpc_coeff_precision, prediction_order_method,

36
libavcodec/aacdec.c
View File

@ -186,7 +186,7 @@ static av_cold int che_configure(AACContext *ac,
if (che_pos[type][id]) {
if (!ac->che[type][id] && !(ac->che[type][id] = av_mallocz(sizeof(ChannelElement))))
return AVERROR(ENOMEM);
ff_aac_sbr_ctx_init(&ac->che[type][id]->sbr);
ff_aac_sbr_ctx_init(ac, &ac->che[type][id]->sbr);
if (type != TYPE_CCE) {
ac->output_data[(*channels)++] = ac->che[type][id]->ch[0].ret;
if (type == TYPE_CPE ||
@ -550,6 +550,7 @@ static void reset_predictor_group(PredictorState *ps, int group_num)
static av_cold int aac_decode_init(AVCodecContext *avctx)
{
AACContext *ac = avctx->priv_data;
float output_scale_factor;
ac->avctx = avctx;
ac->m4ac.sample_rate = avctx->sample_rate;
@ -561,8 +562,13 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
return -1;
}
avctx->sample_fmt = avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT ?
AV_SAMPLE_FMT_FLT : AV_SAMPLE_FMT_S16;
if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
output_scale_factor = 1.0 / 32768.0;
} else {
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
output_scale_factor = 1.0;
}
AAC_INIT_VLC_STATIC( 0, 304);
AAC_INIT_VLC_STATIC( 1, 270);
@ -590,9 +596,9 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]),
352);
ff_mdct_init(&ac->mdct, 11, 1, 1.0/1024.0);
ff_mdct_init(&ac->mdct_small, 8, 1, 1.0/128.0);
ff_mdct_init(&ac->mdct_ltp, 11, 0, -2.0);
ff_mdct_init(&ac->mdct, 11, 1, output_scale_factor/1024.0);
ff_mdct_init(&ac->mdct_small, 8, 1, output_scale_factor/128.0);
ff_mdct_init(&ac->mdct_ltp, 11, 0, -2.0/output_scale_factor);
// window initialization
ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
@ -2174,8 +2180,8 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
avctx->frame_size = samples;
}
data_size_tmp = samples * avctx->channels;
data_size_tmp *= avctx->sample_fmt == AV_SAMPLE_FMT_FLT ? sizeof(float) : sizeof(int16_t);
data_size_tmp = samples * avctx->channels *
(av_get_bits_per_sample_fmt(avctx->sample_fmt) / 8);
if (*data_size < data_size_tmp) {
av_log(avctx, AV_LOG_ERROR,
"Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n",
@ -2185,10 +2191,12 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
*data_size = data_size_tmp;
if (samples) {
if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {
float_interleave(data, (const float **)ac->output_data, samples, avctx->channels);
} else
ac->fmt_conv.float_to_int16_interleave(data, (const float **)ac->output_data, samples, avctx->channels);
if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT)
ac->fmt_conv.float_interleave(data, (const float **)ac->output_data,
samples, avctx->channels);
else
ac->fmt_conv.float_to_int16_interleave(data, (const float **)ac->output_data,
samples, avctx->channels);
}
if (ac->output_configured)
@ -2507,7 +2515,7 @@ AVCodec ff_aac_decoder = {
aac_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
.sample_fmts = (const enum AVSampleFormat[]) {
AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE
AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
},
.channel_layouts = aac_channel_layout,
};
@ -2527,7 +2535,7 @@ AVCodec ff_aac_latm_decoder = {
.decode = latm_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Codec LATM syntax)"),
.sample_fmts = (const enum AVSampleFormat[]) {
AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE
AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
},
.channel_layouts = aac_channel_layout,
};

11
libavcodec/aacsbr.c
View File

@ -127,14 +127,19 @@ av_cold void ff_aac_sbr_init(void)
ff_ps_init();
}
av_cold void ff_aac_sbr_ctx_init(SpectralBandReplication *sbr)
av_cold void ff_aac_sbr_ctx_init(AACContext *ac, SpectralBandReplication *sbr)
{
float mdct_scale;
sbr->kx[0] = sbr->kx[1] = 32; //Typo in spec, kx' inits to 32
sbr->data[0].e_a[1] = sbr->data[1].e_a[1] = -1;
sbr->data[0].synthesis_filterbank_samples_offset = SBR_SYNTHESIS_BUF_SIZE - (1280 - 128);
sbr->data[1].synthesis_filterbank_samples_offset = SBR_SYNTHESIS_BUF_SIZE - (1280 - 128);
ff_mdct_init(&sbr->mdct, 7, 1, 1.0/64);
ff_mdct_init(&sbr->mdct_ana, 7, 1, -2.0);
/* SBR requires samples to be scaled to +/-32768.0 to work correctly.
* mdct scale factors are adjusted to scale up from +/-1.0 at analysis
* and scale back down at synthesis. */
mdct_scale = ac->avctx->sample_fmt == AV_SAMPLE_FMT_FLT ? 32768.0f : 1.0f;
ff_mdct_init(&sbr->mdct, 7, 1, 1.0 / (64 * mdct_scale));
ff_mdct_init(&sbr->mdct_ana, 7, 1, -2.0 * mdct_scale);
ff_ps_ctx_init(&sbr->ps);
}

2
libavcodec/aacsbr.h
View File

@ -36,7 +36,7 @@
/** Initialize SBR. */
av_cold void ff_aac_sbr_init(void);
/** Initialize one SBR context. */
av_cold void ff_aac_sbr_ctx_init(SpectralBandReplication *sbr);
av_cold void ff_aac_sbr_ctx_init(AACContext *ac, SpectralBandReplication *sbr);
/** Close one SBR context. */
av_cold void ff_aac_sbr_ctx_close(SpectralBandReplication *sbr);
/** Decode one SBR element. */

40
libavcodec/ac3dec.c
View File

@ -185,6 +185,15 @@ static av_cold int ac3_decode_init(AVCodecContext *avctx)
ff_fmt_convert_init(&s->fmt_conv, avctx);
av_lfg_init(&s->dith_state, 0);
/* set scale value for float to int16 conversion */
if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
s->mul_bias = 1.0f;
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
} else {
s->mul_bias = 32767.0f;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
}
/* allow downmixing to stereo or mono */
if (avctx->channels > 0 && avctx->request_channels > 0 &&
avctx->request_channels < avctx->channels &&
@ -193,14 +202,6 @@ static av_cold int ac3_decode_init(AVCodecContext *avctx)
}
s->downmixed = 1;
if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
s->mul_bias = 1.0f;
} else {
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
/* set scale value for float to int16 conversion */
s->mul_bias = 32767.0f;
}
return 0;
}
@ -1295,8 +1296,8 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
AC3DecodeContext *s = avctx->priv_data;
float *out_samples_flt = (float *)data;
int16_t *out_samples = (int16_t *)data;
float *out_samples_flt = data;
int16_t *out_samples_s16 = data;
int blk, ch, err;
int data_size_orig, data_size_tmp;
const uint8_t *channel_map;
@ -1400,7 +1401,7 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
for (ch = 0; ch < s->out_channels; ch++)
output[ch] = s->output[channel_map[ch]];
data_size_tmp = s->num_blocks * 256 * avctx->channels;
data_size_tmp *= avctx->sample_fmt == AV_SAMPLE_FMT_FLT ? sizeof(*out_samples_flt) : sizeof(*out_samples);
data_size_tmp *= avctx->sample_fmt == AV_SAMPLE_FMT_FLT ? sizeof(*out_samples_flt) : sizeof(*out_samples_s16);
if (data_size_orig < data_size_tmp)
return -1;
*data_size = data_size_tmp;
@ -1409,14 +1410,19 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
av_log(avctx, AV_LOG_ERROR, "error decoding the audio block\n");
err = 1;
}
if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {
float_interleave_noscale(out_samples_flt, output, 256, s->out_channels);
s->fmt_conv.float_interleave(out_samples_flt, output, 256,
s->out_channels);
out_samples_flt += 256 * s->out_channels;
} else {
s->fmt_conv.float_to_int16_interleave(out_samples, output, 256, s->out_channels);
out_samples += 256 * s->out_channels;
s->fmt_conv.float_to_int16_interleave(out_samples_s16, output, 256,
s->out_channels);
out_samples_s16 += 256 * s->out_channels;
}
}
*data_size = s->num_blocks * 256 * avctx->channels *
(av_get_bits_per_sample_fmt(avctx->sample_fmt) / 8);
return FFMIN(buf_size, s->frame_size);
}
@ -1441,6 +1447,9 @@ AVCodec ff_ac3_decoder = {
.close = ac3_decode_end,
.decode = ac3_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
.sample_fmts = (const enum AVSampleFormat[]) {
AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
},
};
#if CONFIG_EAC3_DECODER
@ -1453,5 +1462,8 @@ AVCodec ff_eac3_decoder = {
.close = ac3_decode_end,
.decode = ac3_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("ATSC A/52B (AC-3, E-AC-3)"),
.sample_fmts = (const enum AVSampleFormat[]) {
AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
},
};
#endif

15
libavcodec/avcodec.h
View File

@ -2880,6 +2880,14 @@ typedef struct AVCodecContext {
*/
enum AVAudioServiceType audio_service_type;
/**
* desired sample format
* - encoding: Not used.
* - decoding: Set by user.
* Decoder will decode to this format if it can.
*/
enum AVSampleFormat request_sample_fmt;
/**
* Current statistics for PTS correction.
* - decoding: maintained and used by libavcodec, not intended to be used by user apps
@ -2890,13 +2898,6 @@ typedef struct AVCodecContext {
int64_t pts_correction_last_pts; /// PTS of the last frame
int64_t pts_correction_last_dts; /// DTS of the last frame
/**
* desired sample format
* - encoding: Not used.
* - decoding: Set by user.
* Decoder will decode to this format if it can.
*/
enum AVSampleFormat request_sample_fmt;
} AVCodecContext;

36
libavcodec/dca.c
View File

@ -1627,8 +1627,9 @@ static int dca_decode_frame(AVCodecContext * avctx,
int lfe_samples;
int num_core_channels = 0;
int i;
float *samples_flt = data;
int16_t *samples = data;
float *samples_flt = data;
int16_t *samples_s16 = data;
int out_size;
DCAContext *s = avctx->priv_data;
int channels;
int core_ss_end;
@ -1818,11 +1819,11 @@ static int dca_decode_frame(AVCodecContext * avctx,
return -1;
}
data_size_tmp = (s->sample_blocks / 8) * 256 * channels;
data_size_tmp *= avctx->sample_fmt == AV_SAMPLE_FMT_FLT ? sizeof(*samples_flt) : sizeof(*samples);
if (*data_size < data_size_tmp)
out_size = 256 / 8 * s->sample_blocks * channels *
(av_get_bits_per_sample_fmt(avctx->sample_fmt) / 8);
if (*data_size < out_size)
return -1;
*data_size = data_size_tmp;
*data_size = out_size;
/* filter to get final output */
for (i = 0; i < (s->sample_blocks / 8); i++) {
@ -1841,13 +1842,15 @@ static int dca_decode_frame(AVCodecContext * avctx,
}
}
/* interleave samples */
if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {
float_interleave(samples_flt, s->samples_chanptr, 256, channels);
s->fmt_conv.float_interleave(samples_flt, s->samples_chanptr, 256,
channels);
samples_flt += 256 * channels;
} else {
s->fmt_conv.float_to_int16_interleave(samples, s->samples_chanptr, 256, channels);
samples += 256 * channels;
s->fmt_conv.float_to_int16_interleave(samples_s16,
s->samples_chanptr, 256,
channels);
samples_s16 += 256 * channels;
}
}
@ -1884,10 +1887,14 @@ static av_cold int dca_decode_init(AVCodecContext * avctx)
for (i = 0; i < DCA_PRIM_CHANNELS_MAX+1; i++)
s->samples_chanptr[i] = s->samples + i * 256;
avctx->sample_fmt = avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT ?
AV_SAMPLE_FMT_FLT : AV_SAMPLE_FMT_S16;
s->scale_bias = 1.0;
if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
s->scale_bias = 1.0 / 32768.0;
} else {
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
s->scale_bias = 1.0;
}
/* allow downmixing to stereo */
if (avctx->channels > 0 && avctx->request_channels < avctx->channels &&
@ -1924,5 +1931,8 @@ AVCodec ff_dca_decoder = {
.close = dca_decode_end,
.long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
.capabilities = CODEC_CAP_CHANNEL_CONF,
.sample_fmts = (const enum AVSampleFormat[]) {
AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
},
.profiles = NULL_IF_CONFIG_SMALL(profiles),
};

20
libavcodec/fmtconvert.c
View File

@ -56,11 +56,31 @@ static void float_to_int16_interleave_c(int16_t *dst, const float **src,
}
}
void ff_float_interleave_c(float *dst, const float **src, unsigned int len,
int channels)
{
int j, c;
unsigned int i;
if (channels == 2) {
for (i = 0; i < len; i++) {
dst[2*i] = src[0][i];
dst[2*i+1] = src[1][i];
}
} else if (channels == 1 && len < INT_MAX / sizeof(float)) {
memcpy(dst, src[0], len * sizeof(float));
} else {
for (c = 0; c < channels; c++)
for (i = 0, j = c; i < len; i++, j += channels)
dst[j] = src[c][i];
}
}
av_cold void ff_fmt_convert_init(FmtConvertContext *c, AVCodecContext *avctx)
{
c->int32_to_float_fmul_scalar = int32_to_float_fmul_scalar_c;
c->float_to_int16 = float_to_int16_c;
c->float_to_int16_interleave = float_to_int16_interleave_c;
c->float_interleave = ff_float_interleave_c;
if (ARCH_ARM) ff_fmt_convert_init_arm(c, avctx);
if (HAVE_ALTIVEC) ff_fmt_convert_init_altivec(c, avctx);

9
libavcodec/fmtconvert.h
View File

@ -68,8 +68,17 @@ typedef struct FmtConvertContext {
*/
void (*float_to_int16_interleave)(int16_t *dst, const float **src,
long len, int channels);
/**
* Convert an array of interleaved float to multiple arrays of float.
*/
void (*float_interleave)(float *dst, const float **src, unsigned int len,
int channels);
} FmtConvertContext;
void ff_float_interleave_c(float *dst, const float **src, unsigned int len,
int channels);
void ff_fmt_convert_init(FmtConvertContext *c, AVCodecContext *avctx);
void ff_fmt_convert_init_arm(FmtConvertContext *c, AVCodecContext *avctx);

1
libavcodec/h264.c
View File

@ -1953,6 +1953,7 @@ static int decode_slice_header(H264Context *h, H264Context *h0){
c->h264dsp = h->h264dsp;
c->sps = h->sps;
c->pps = h->pps;
c->pixel_shift = h->pixel_shift;
init_scan_tables(c);
clone_tables(c, h, i);
}

4
libavcodec/mdec.c
View File

@ -126,7 +126,8 @@ static inline int decode_mb(MDECContext *a, DCTELEM block[6][64]){
a->dsp.clear_blocks(block[0]);
for(i=0; i<6; i++){
if( mdec_decode_block_intra(a, block[ block_index[i] ], block_index[i]) < 0)
if( mdec_decode_block_intra(a, block[ block_index[i] ], block_index[i]) < 0 ||
get_bits_left(&a->gb) < 0)
return -1;
}
return 0;
@ -252,6 +253,7 @@ static av_cold int decode_init_thread_copy(AVCodecContext *avctx){
return 0;
}
static av_cold int decode_end(AVCodecContext *avctx){
MDECContext * const a = avctx->priv_data;

2
libavcodec/mpeg12.c
View File

@ -2342,6 +2342,7 @@ static int decode_chunks(AVCodecContext *avctx,
if(s2->pict_type != AV_PICTURE_TYPE_B || avctx->skip_frame <= AVDISCARD_DEFAULT){
if(HAVE_THREADS && avctx->active_thread_type&FF_THREAD_SLICE){
int i;
assert(avctx->thread_count > 1);
avctx->execute(avctx, slice_decode_thread, &s2->thread_context[0], NULL, s->slice_count, sizeof(void*));
for(i=0; i<s->slice_count; i++)
@ -2510,6 +2511,7 @@ static int decode_chunks(AVCodecContext *avctx,
if(HAVE_THREADS && avctx->active_thread_type&FF_THREAD_SLICE){
int threshold= (s2->mb_height*s->slice_count + avctx->thread_count/2) / avctx->thread_count;
assert(avctx->thread_count > 1);
if(threshold <= mb_y){
MpegEncContext *thread_context= s2->thread_context[s->slice_count];

9
libavcodec/mpegaudiodec.c
View File

@ -41,7 +41,6 @@
#if CONFIG_FLOAT
# define SHR(a,b) ((a)*(1.0f/(1<<(b))))
# define compute_antialias compute_antialias_float
# define FIXR_OLD(a) ((int)((a) * FRAC_ONE + 0.5))
# define FIXR(x) ((float)(x))
# define FIXHR(x) ((float)(x))
@ -51,7 +50,6 @@
# define OUT_FMT AV_SAMPLE_FMT_FLT
#else
# define SHR(a,b) ((a)>>(b))
# define compute_antialias compute_antialias_integer
/* WARNING: only correct for posititive numbers */
# define FIXR_OLD(a) ((int)((a) * FRAC_ONE + 0.5))
# define FIXR(a) ((int)((a) * FRAC_ONE + 0.5))
@ -69,7 +67,7 @@
#include "mpegaudiodata.h"
#include "mpegaudiodectab.h"
static void compute_antialias(MPADecodeContext *s, GranuleDef *g);
static void RENAME(compute_antialias)(MPADecodeContext *s, GranuleDef *g);
static void apply_window_mp3_c(MPA_INT *synth_buf, MPA_INT *window,
int *dither_state, OUT_INT *samples, int incr);
@ -1480,8 +1478,7 @@ static void compute_stereo(MPADecodeContext *s,
}
#if !CONFIG_FLOAT
static void compute_antialias_integer(MPADecodeContext *s,
GranuleDef *g)
static void compute_antialias_fixed(MPADecodeContext *s, GranuleDef *g)
{
int32_t *ptr, *csa;
int n, i;
@ -1848,7 +1845,7 @@ static int mp_decode_layer3(MPADecodeContext *s)
g = &s->granules[ch][gr];
reorder_block(s, g);
compute_antialias(s, g);
RENAME(compute_antialias)(s, g);
compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);
}
} /* gr */

15
libavcodec/mpegaudiodec_float.c
View File

@ -80,13 +80,6 @@ static void compute_antialias_float(MPADecodeContext *s,
}
}
static av_cold int decode_end(AVCodecContext * avctx)
{
MPADecodeContext *s = avctx->priv_data;
ff_dct_end(&s->dct);
return 0;
}
#if CONFIG_MP1FLOAT_DECODER
AVCodec ff_mp1float_decoder =
{
@ -96,7 +89,7 @@ AVCodec ff_mp1float_decoder =
sizeof(MPADecodeContext),
decode_init,
NULL,
decode_end,
.close = NULL,
decode_frame,
CODEC_CAP_PARSE_ONLY,
.flush= flush,
@ -112,7 +105,7 @@ AVCodec ff_mp2float_decoder =
sizeof(MPADecodeContext),
decode_init,
NULL,
decode_end,
.close = NULL,
decode_frame,
CODEC_CAP_PARSE_ONLY,
.flush= flush,
@ -128,7 +121,7 @@ AVCodec ff_mp3float_decoder =
sizeof(MPADecodeContext),
decode_init,
NULL,
decode_end,
.close = NULL,
decode_frame,
CODEC_CAP_PARSE_ONLY,
.flush= flush,
@ -144,7 +137,7 @@ AVCodec ff_mp3adufloat_decoder =
sizeof(MPADecodeContext),
decode_init,
NULL,
decode_end,
.close = NULL,
decode_frame_adu,
CODEC_CAP_PARSE_ONLY,
.flush= flush,

7
libavcodec/options.c
View File

@ -441,7 +441,12 @@ static const AVOption options[]={
{"em", "Emergency", 0, FF_OPT_TYPE_CONST, {.dbl = AV_AUDIO_SERVICE_TYPE_EMERGENCY }, INT_MIN, INT_MAX, A|E, "audio_service_type"},
{"vo", "Voice Over", 0, FF_OPT_TYPE_CONST, {.dbl = AV_AUDIO_SERVICE_TYPE_VOICE_OVER }, INT_MIN, INT_MAX, A|E, "audio_service_type"},
{"ka", "Karaoke", 0, FF_OPT_TYPE_CONST, {.dbl = AV_AUDIO_SERVICE_TYPE_KARAOKE }, INT_MIN, INT_MAX, A|E, "audio_service_type"},
{"request_sample_fmt", "sample format audio decoders should prefer", OFFSET(request_sample_fmt), FF_OPT_TYPE_INT, {.dbl = AV_SAMPLE_FMT_NONE }, AV_SAMPLE_FMT_NONE, AV_SAMPLE_FMT_NB-1, A|D},
{"request_sample_fmt", "sample format audio decoders should prefer", OFFSET(request_sample_fmt), FF_OPT_TYPE_INT, {.dbl = AV_SAMPLE_FMT_NONE }, AV_SAMPLE_FMT_NONE, AV_SAMPLE_FMT_NB-1, A|D, "request_sample_fmt"},
{"u8" , "8-bit unsigned integer", 0, FF_OPT_TYPE_CONST, {.dbl = AV_SAMPLE_FMT_U8 }, INT_MIN, INT_MAX, A|D, "request_sample_fmt"},
{"s16", "16-bit signed integer", 0, FF_OPT_TYPE_CONST, {.dbl = AV_SAMPLE_FMT_S16 }, INT_MIN, INT_MAX, A|D, "request_sample_fmt"},
{"s32", "32-bit signed integer", 0, FF_OPT_TYPE_CONST, {.dbl = AV_SAMPLE_FMT_S32 }, INT_MIN, INT_MAX, A|D, "request_sample_fmt"},
{"flt", "32-bit float", 0, FF_OPT_TYPE_CONST, {.dbl = AV_SAMPLE_FMT_FLT }, INT_MIN, INT_MAX, A|D, "request_sample_fmt"},
{"dbl", "64-bit double", 0, FF_OPT_TYPE_CONST, {.dbl = AV_SAMPLE_FMT_DBL }, INT_MIN, INT_MAX, A|D, "request_sample_fmt"},
{NULL},
};

27
libavcodec/vorbisdec.c
View File

@ -979,7 +979,13 @@ static av_cold int vorbis_decode_init(AVCodecContext *avccontext)
dsputil_init(&vc->dsp, avccontext);
ff_fmt_convert_init(&vc->fmt_conv, avccontext);
vc->scale_bias = 32768.0f;
if (avccontext->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
avccontext->sample_fmt = AV_SAMPLE_FMT_FLT;
vc->scale_bias = 1.0f;
} else {
avccontext->sample_fmt = AV_SAMPLE_FMT_S16;
vc->scale_bias = 32768.0f;
}
if (!headers_len) {
av_log(avccontext, AV_LOG_ERROR, "Extradata missing.\n");
@ -1024,9 +1030,6 @@ static av_cold int vorbis_decode_init(AVCodecContext *avccontext)
avccontext->channels = vc->audio_channels;
avccontext->sample_rate = vc->audio_samplerate;
avccontext->frame_size = FFMIN(vc->blocksize[0], vc->blocksize[1]) >> 2;
avccontext->sample_fmt =
avccontext->request_sample_fmt == AV_SAMPLE_FMT_FLT ?
AV_SAMPLE_FMT_FLT : AV_SAMPLE_FMT_S16;
return 0 ;
}
@ -1636,15 +1639,14 @@ static int vorbis_decode_frame(AVCodecContext *avccontext,
len * ff_vorbis_channel_layout_offsets[vc->audio_channels - 1][i];
}
*data_size = len * vc->audio_channels;
if (avccontext->sample_fmt == AV_SAMPLE_FMT_FLT) {
float_interleave(data, channel_ptrs, len, vc->audio_channels);
*data_size *= sizeof(float);
} else {
if (avccontext->sample_fmt == AV_SAMPLE_FMT_FLT)
vc->fmt_conv.float_interleave(data, channel_ptrs, len, vc->audio_channels);
else
vc->fmt_conv.float_to_int16_interleave(data, channel_ptrs, len,
vc->audio_channels);
*data_size *= 2;
}
*data_size = len * vc->audio_channels *
(av_get_bits_per_sample_fmt(avccontext->sample_fmt) / 8);
return buf_size ;
}
@ -1671,5 +1673,8 @@ AVCodec ff_vorbis_decoder = {
vorbis_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("Vorbis"),
.channel_layouts = ff_vorbis_channel_layouts,
.sample_fmts = (const enum AVSampleFormat[]) {
AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
},
};

141
libavcodec/x86/fmtconvert.asm
View File

@ -20,6 +20,7 @@
;******************************************************************************
%include "x86inc.asm"
%include "x86util.asm"
section .text align=16
@ -89,3 +90,143 @@ FLOAT_TO_INT16_INTERLEAVE6 3dnow
%undef pswapd
FLOAT_TO_INT16_INTERLEAVE6 3dn2
%undef cvtps2pi
;-----------------------------------------------------------------------------
; void ff_float_interleave6(float *dst, const float **src, unsigned int len);
;-----------------------------------------------------------------------------
%macro BUTTERFLYPS 3
movaps m%3, m%1
unpcklps m%1, m%2
unpckhps m%3, m%2
SWAP %2, %3
%endmacro
%macro FLOAT_INTERLEAVE6 2
cglobal float_interleave6_%1, 2,7,%2, dst, src, src1, src2, src3, src4, src5
%ifdef ARCH_X86_64
%define lend r10d
mov lend, r2d
%else
%define lend dword r2m
%endif
mov src1q, [srcq+1*gprsize]
mov src2q, [srcq+2*gprsize]
mov src3q, [srcq+3*gprsize]
mov src4q, [srcq+4*gprsize]
mov src5q, [srcq+5*gprsize]
mov srcq, [srcq]
sub src1q, srcq
sub src2q, srcq
sub src3q, srcq
sub src4q, srcq
sub src5q, srcq
.loop:
%ifidn %1, sse
movaps m0, [srcq]
movaps m1, [srcq+src1q]
movaps m2, [srcq+src2q]
movaps m3, [srcq+src3q]
movaps m4, [srcq+src4q]
movaps m5, [srcq+src5q]
BUTTERFLYPS 0, 1, 6
BUTTERFLYPS 2, 3, 6
BUTTERFLYPS 4, 5, 6
movaps m6, m4
shufps m4, m0, 0xe4
movlhps m0, m2
movhlps m6, m2
movaps [dstq ], m0
movaps [dstq+16], m4
movaps [dstq+32], m6
movaps m6, m5
shufps m5, m1, 0xe4
movlhps m1, m3
movhlps m6, m3
movaps [dstq+48], m1
movaps [dstq+64], m5
movaps [dstq+80], m6
%else ; mmx
movq m0, [srcq]
movq m1, [srcq+src1q]
movq m2, [srcq+src2q]
movq m3, [srcq+src3q]
movq m4, [srcq+src4q]
movq m5, [srcq+src5q]
SBUTTERFLY dq, 0, 1, 6
SBUTTERFLY dq, 2, 3, 6
SBUTTERFLY dq, 4, 5, 6
movq [dstq ], m0
movq [dstq+ 8], m2
movq [dstq+16], m4
movq [dstq+24], m1
movq [dstq+32], m3
movq [dstq+40], m5
%endif
add srcq, mmsize
add dstq, mmsize*6
sub lend, mmsize/4
jg .loop
%ifidn %1, mmx
emms
%endif
REP_RET
%endmacro
INIT_MMX
FLOAT_INTERLEAVE6 mmx, 0
INIT_XMM
FLOAT_INTERLEAVE6 sse, 7
;-----------------------------------------------------------------------------
; void ff_float_interleave2(float *dst, const float **src, unsigned int len);
;-----------------------------------------------------------------------------
%macro FLOAT_INTERLEAVE2 2
cglobal float_interleave2_%1, 3,4,%2, dst, src, len, src1
mov src1q, [srcq+gprsize]
mov srcq, [srcq ]
sub src1q, srcq
.loop
MOVPS m0, [srcq ]
MOVPS m1, [srcq+src1q ]
MOVPS m3, [srcq +mmsize]
MOVPS m4, [srcq+src1q+mmsize]
MOVPS m2, m0
PUNPCKLDQ m0, m1
PUNPCKHDQ m2, m1
MOVPS m1, m3
PUNPCKLDQ m3, m4
PUNPCKHDQ m1, m4
MOVPS [dstq ], m0
MOVPS [dstq+1*mmsize], m2
MOVPS [dstq+2*mmsize], m3
MOVPS [dstq+3*mmsize], m1
add srcq, mmsize*2
add dstq, mmsize*4
sub lend, mmsize/2
jg .loop
%ifidn %1, mmx
emms
%endif
REP_RET
%endmacro
INIT_MMX
%define MOVPS movq
%define PUNPCKLDQ punpckldq
%define PUNPCKHDQ punpckhdq
FLOAT_INTERLEAVE2 mmx, 0
INIT_XMM
%define MOVPS movaps
%define PUNPCKLDQ unpcklps
%define PUNPCKHDQ unpckhps
FLOAT_INTERLEAVE2 sse, 5

30
libavcodec/x86/fmtconvert_mmx.c
View File

@ -235,11 +235,40 @@ static void float_to_int16_interleave_3dn2(int16_t *dst, const float **src, long
float_to_int16_interleave_3dnow(dst, src, len, channels);
}
void ff_float_interleave2_mmx(float *dst, const float **src, unsigned int len);
void ff_float_interleave2_sse(float *dst, const float **src, unsigned int len);
void ff_float_interleave6_mmx(float *dst, const float **src, unsigned int len);
void ff_float_interleave6_sse(float *dst, const float **src, unsigned int len);
static void float_interleave_mmx(float *dst, const float **src,
unsigned int len, int channels)
{
if (channels == 2) {
ff_float_interleave2_mmx(dst, src, len);
} else if (channels == 6)
ff_float_interleave6_mmx(dst, src, len);
else
ff_float_interleave_c(dst, src, len, channels);
}
static void float_interleave_sse(float *dst, const float **src,
unsigned int len, int channels)
{
if (channels == 2) {
ff_float_interleave2_sse(dst, src, len);
} else if (channels == 6)
ff_float_interleave6_sse(dst, src, len);
else
ff_float_interleave_c(dst, src, len, channels);
}
void ff_fmt_convert_init_x86(FmtConvertContext *c, AVCodecContext *avctx)
{
int mm_flags = av_get_cpu_flags();
if (mm_flags & AV_CPU_FLAG_MMX) {
c->float_interleave = float_interleave_mmx;
if(mm_flags & AV_CPU_FLAG_3DNOW){
if(!(avctx->flags & CODEC_FLAG_BITEXACT)){
@ -256,6 +285,7 @@ void ff_fmt_convert_init_x86(FmtConvertContext *c, AVCodecContext *avctx)
c->int32_to_float_fmul_scalar = int32_to_float_fmul_scalar_sse;
c->float_to_int16 = float_to_int16_sse;
c->float_to_int16_interleave = float_to_int16_interleave_sse;
c->float_interleave = float_interleave_sse;
}
if(mm_flags & AV_CPU_FLAG_SSE2){
c->int32_to_float_fmul_scalar = int32_to_float_fmul_scalar_sse2;

5
tests/codec-regression.sh
View File

@ -12,14 +12,13 @@ set -e
eval do_$test=y
rm -f "$logfile"
rm -f "$benchfile"
# generate reference for quality check
if [ -n "$do_vref" ]; then
do_ffmpeg_nocheck $raw_ref -f image2 -vcodec pgmyuv -i $raw_src -an -f rawvideo $target_path/$raw_ref
do_ffmpeg $raw_ref -f image2 -vcodec pgmyuv -i $raw_src -an -f rawvideo
fi
if [ -n "$do_aref" ]; then
do_ffmpeg_nocheck $pcm_ref -ab 128k -ac 2 -ar 44100 -f s16le -i $pcm_src -f wav $target_path/$pcm_ref
do_ffmpeg $pcm_ref -ab 128k -ac 2 -ar 44100 -f s16le -i $pcm_src -f wav
fi
if [ -n "$do_mpeg" ] ; then

5
tests/lavf-regression.sh
View File

@ -44,7 +44,6 @@ do_audio_only()
}
rm -f "$logfile"
rm -f "$benchfile"
if [ -n "$do_avi" ] ; then
do_lavf avi
@ -227,8 +226,8 @@ conversions="yuv420p yuv422p yuv444p yuyv422 yuv410p yuv411p yuvj420p \
monob yuv440p yuvj440p"
for pix_fmt in $conversions ; do
file=${outfile}${pix_fmt}.yuv
do_ffmpeg_nocheck $file $DEC_OPTS -r 1 -t 1 -f image2 -vcodec pgmyuv -i $raw_src \
$ENC_OPTS -f rawvideo -s 352x288 -pix_fmt $pix_fmt $target_path/$raw_dst
run_ffmpeg $DEC_OPTS -r 1 -t 1 -f image2 -vcodec pgmyuv -i $raw_src \
$ENC_OPTS -f rawvideo -s 352x288 -pix_fmt $pix_fmt $target_path/$raw_dst
do_ffmpeg $file $DEC_OPTS -f rawvideo -s 352x288 -pix_fmt $pix_fmt -i $target_path/$raw_dst \
$ENC_OPTS -f rawvideo -s 352x288 -pix_fmt yuv444p
done

1
tests/lavfi-regression.sh
View File

@ -12,7 +12,6 @@ set -e
eval do_$test=y
rm -f "$logfile"
rm -f "$benchfile"
do_video_filter() {
label=$1

2
tests/ref/acodec/aref Normal file
View File

@ -0,0 +1,2 @@
95e54b261530a1bcf6de6fe3b21dc5f6 *./tests/data/acodec.ref.wav
1058444 ./tests/data/acodec.ref.wav

2
tests/ref/vsynth1/vref Normal file
View File

@ -0,0 +1,2 @@
c5ccac874dbf808e9088bc3107860042 *./tests/data/vsynth1.ref.yuv
7603200 ./tests/data/vsynth1.ref.yuv

2
tests/ref/vsynth2/vref Normal file
View File

@ -0,0 +1,2 @@
dde5895817ad9d219f79a52d0bdfb001 *./tests/data/vsynth2.ref.yuv
7603200 ./tests/data/vsynth2.ref.yuv

22
tests/regression-funcs.sh
View File

@ -23,9 +23,6 @@ errfile="$datadir/$this.err"
# various files
ffmpeg="$target_exec ${target_path}/ffmpeg"
tiny_psnr="tests/tiny_psnr"
benchfile="$datadir/$this.bench"
bench="$datadir/$this.bench.tmp"
bench2="$datadir/$this.bench2.tmp"
raw_src="${target_path}/$raw_src_dir/%02d.pgm"
raw_dst="$datadir/$this.out.yuv"
raw_ref="$datadir/$test_ref.ref.yuv"
@ -35,7 +32,7 @@ pcm_ref="$datadir/$test_ref.ref.wav"
crcfile="$datadir/$this.crc"
target_crcfile="$target_datadir/$this.crc"
cleanfiles="$raw_dst $pcm_dst $crcfile $bench $bench2"
cleanfiles="$raw_dst $pcm_dst $crcfile"
trap 'rm -f -- $cleanfiles' EXIT
mkdir -p "$datadir"
@ -69,7 +66,7 @@ do_ffmpeg()
f="$1"
shift
set -- $* ${target_path}/$f
run_ffmpeg -benchmark $* > $bench
run_ffmpeg $*
do_md5sum $f >> $logfile
if [ $f = $raw_dst ] ; then
$tiny_psnr $f $raw_ref >> $logfile
@ -78,8 +75,6 @@ do_ffmpeg()
else
wc -c $f >> $logfile
fi
expr "$(cat $bench)" : '.*utime=\(.*s\)' > $bench2
echo $(cat $bench2) $f >> $benchfile
}
do_ffmpeg_nomd5()
@ -87,7 +82,7 @@ do_ffmpeg_nomd5()
f="$1"
shift
set -- $* ${target_path}/$f
run_ffmpeg -benchmark $* > $bench
run_ffmpeg $*
if [ $f = $raw_dst ] ; then
$tiny_psnr $f $raw_ref >> $logfile
elif [ $f = $pcm_dst ] ; then
@ -95,8 +90,6 @@ do_ffmpeg_nomd5()
else
wc -c $f >> $logfile
fi
expr "$(cat $bench)" : '.*utime=\(.*s\)' > $bench2
echo $(cat $bench2) $f >> $benchfile
}
do_ffmpeg_crc()
@ -107,15 +100,6 @@ do_ffmpeg_crc()
echo "$f $(cat $crcfile)" >> $logfile
}
do_ffmpeg_nocheck()
{
f="$1"
shift
run_ffmpeg -benchmark $* > $bench
expr "$(cat $bench)" : '.*utime=\(.*s\)' > $bench2
echo $(cat $bench2) $f >> $benchfile
}
do_video_decoding()
{
do_ffmpeg $raw_dst $DEC_OPTS $1 -i $target_path/$file -f rawvideo $ENC_OPTS $2