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Merge remote-tracking branch 'qatar/master'

Browse Source 
* 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
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