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ffmpeg.wasm-core
Commit 967facb6 authored by Michael Niedermayer's avatar Michael Niedermayer
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Merge remote-tracking branch 'qatar/master' 

* qatar/master: (26 commits)
  adxenc: use AVCodec.encode2()
  adxenc: Use the AVFrame in ADXContext for coded_frame
  indeo4: fix out-of-bounds function call.
  configure: Restructure help output.
  configure: Internal-only components should not be command-line selectable.
  vorbisenc: use AVCodec.encode2()
  libvorbis: use AVCodec.encode2()
  libopencore-amrnbenc: use AVCodec.encode2()
  ra144enc: use AVCodec.encode2()
  nellymoserenc: use AVCodec.encode2()
  roqaudioenc: use AVCodec.encode2()
  libspeex: use AVCodec.encode2()
  libvo_amrwbenc: use AVCodec.encode2()
  libvo_aacenc: use AVCodec.encode2()
  wmaenc: use AVCodec.encode2()
  mpegaudioenc: use AVCodec.encode2()
  libmp3lame: use AVCodec.encode2()
  libgsmenc: use AVCodec.encode2()
  libfaac: use AVCodec.encode2()
  g726enc: use AVCodec.encode2()
  ...

Conflicts:
	configure
	libavcodec/Makefile
	libavcodec/ac3enc.c
	libavcodec/adxenc.c
	libavcodec/libgsm.c
	libavcodec/libvorbis.c
	libavcodec/vorbisenc.c
	libavcodec/wmaenc.c
	tests/ref/acodec/g722
	tests/ref/lavf/asf
	tests/ref/lavf/ffm
	tests/ref/lavf/mkv
	tests/ref/lavf/mpg
	tests/ref/lavf/rm
	tests/ref/lavf/ts
	tests/ref/seek/lavf_asf
	tests/ref/seek/lavf_ffm
	tests/ref/seek/lavf_mkv
	tests/ref/seek/lavf_mpg
	tests/ref/seek/lavf_rm
	tests/ref/seek/lavf_ts
Merged-by: 's avatarMichael Niedermayer <michaelni@gmx.at>
parents f1fdd208 6aba117f
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configure
View file @ 967facb6
......@@ -59,8 +59,21 @@ cat <<EOF
Usage: configure [options]
Options: [defaults in brackets after descriptions]
Standard options:
Help options:
--help print this message
--list-decoders show all available decoders
--list-encoders show all available encoders
--list-hwaccels show all available hardware accelerators
--list-demuxers show all available demuxers
--list-muxers show all available muxers
--list-parsers show all available parsers
--list-protocols show all available protocols
--list-bsfs show all available bitstream filters
--list-indevs show all available input devices
--list-outdevs show all available output devices
--list-filters show all available filters
Standard options:
--logfile=FILE log tests and output to FILE [config.log]
--disable-logging do not log configure debug information
--prefix=PREFIX install in PREFIX [$prefix]
......@@ -71,14 +84,22 @@ Standard options:
--incdir=DIR install includes in DIR [PREFIX/include]
--mandir=DIR install man page in DIR [PREFIX/share/man]
Configuration options:
--disable-static do not build static libraries [no]
--enable-shared build shared libraries [no]
Licensing options:
--enable-gpl allow use of GPL code, the resulting libs
and binaries will be under GPL [no]
--enable-version3 upgrade (L)GPL to version 3 [no]
--enable-nonfree allow use of nonfree code, the resulting libs
and binaries will be unredistributable [no]
Configuration options:
--disable-static do not build static libraries [no]
--enable-shared build shared libraries [no]
--enable-small optimize for size instead of speed
--enable-runtime-cpudetect detect cpu capabilities at runtime (bigger binary)
--enable-gray enable full grayscale support (slower color)
--disable-swscale-alpha disable alpha channel support in swscale
Component options:
--disable-doc do not build documentation
--disable-ffmpeg disable ffmpeg build
--disable-ffplay disable ffplay build
......@@ -96,29 +117,17 @@ Configuration options:
--disable-os2threads disable OS/2 threads [auto]
--enable-x11grab enable X11 grabbing [no]
--disable-network disable network support [no]
--enable-gray enable full grayscale support (slower color)
--disable-swscale-alpha disable alpha channel support in swscale
--disable-fastdiv disable table-based division
--enable-small optimize for size instead of speed
--disable-aandct disable AAN DCT code
--disable-dct disable DCT code
--disable-fft disable FFT code
--disable-golomb disable Golomb code
--disable-huffman disable Huffman code
--disable-lpc disable LPC code
--disable-mdct disable MDCT code
--disable-rdft disable RDFT code
--disable-fft disable FFT code
--enable-dxva2 enable DXVA2 code
--enable-vaapi enable VAAPI code [autodetect]
--enable-vda enable VDA code [autodetect]
--enable-vda enable VDA code [autodetect]
--enable-vdpau enable VDPAU code [autodetect]
--disable-dxva2 disable DXVA2 code
--disable-vda disable VDA code
--enable-runtime-cpudetect detect cpu capabilities at runtime (bigger binary)
--enable-hardcoded-tables use hardcoded tables instead of runtime generation
--disable-safe-bitstream-reader
disable buffer boundary checking in bitreaders
(faster, but may crash)
--enable-memalign-hack emulate memalign, interferes with memory debuggers
Individual component options:
--disable-everything disable all components listed below
--disable-encoder=NAME disable encoder NAME
--enable-encoder=NAME enable encoder NAME
......@@ -144,25 +153,16 @@ Configuration options:
--enable-protocol=NAME enable protocol NAME
--disable-protocol=NAME disable protocol NAME
--disable-protocols disable all protocols
--enable-indev=NAME enable input device NAME
--disable-indev=NAME disable input device NAME
--disable-outdev=NAME disable output device NAME
--disable-indevs disable input devices
--enable-outdev=NAME enable output device NAME
--disable-outdev=NAME disable output device NAME
--disable-outdevs disable output devices
--disable-devices disable all devices
--enable-filter=NAME enable filter NAME
--disable-filter=NAME disable filter NAME
--disable-filters disable all filters
--list-decoders show all available decoders
--list-encoders show all available encoders
--list-hwaccels show all available hardware accelerators
--list-muxers show all available muxers
--list-demuxers show all available demuxers
--list-parsers show all available parsers
--list-protocols show all available protocols
--list-bsfs show all available bitstream filters
--list-indevs show all available input devices
--list-outdevs show all available output devices
--list-filters show all available filters
External library support:
--enable-avisynth enable reading of AVISynth script files [no]
......@@ -233,11 +233,24 @@ Advanced options (experts only):
--extra-ldflags=ELDFLAGS add ELDFLAGS to LDFLAGS [$LDFLAGS]
--extra-libs=ELIBS add ELIBS [$ELIBS]
--extra-version=STRING version string suffix []
--optflags override optimization-related compiler flags
--build-suffix=SUFFIX library name suffix []
--malloc-prefix=PREFIX prefix malloc and related names with PREFIX
--progs-suffix=SUFFIX program name suffix []
--arch=ARCH select architecture [$arch]
--cpu=CPU select the minimum required CPU (affects
instruction selection, may crash on older CPUs)
--enable-pic build position-independent code
--enable-sram allow use of on-chip SRAM
--disable-symver disable symbol versioning
--disable-fastdiv disable table-based division
--enable-hardcoded-tables use hardcoded tables instead of runtime generation
--disable-safe-bitstream-reader
disable buffer boundary checking in bitreaders
(faster, but may crash)
--enable-memalign-hack emulate memalign, interferes with memory debuggers
Optimization options (experts only):
--disable-asm disable all assembler optimizations
--disable-altivec disable AltiVec optimizations
--disable-amd3dnow disable 3DNow! optimizations
......@@ -255,11 +268,6 @@ Advanced options (experts only):
--disable-neon disable NEON optimizations
--disable-vis disable VIS optimizations
--disable-yasm disable use of yasm assembler
--enable-pic build position-independent code
--malloc-prefix=PFX prefix malloc and related names with PFX
--enable-sram allow use of on-chip SRAM
--disable-symver disable symbol versioning
--optflags override optimization-related compiler flags
--postproc-version=V build libpostproc version V.
Where V can be '$ALT_PP_VER_MAJOR.$ALT_PP_VER_MINOR.$ALT_PP_VER_MICRO' or 'current'. [$postproc_version_default]
......@@ -999,10 +1007,6 @@ PROGRAM_LIST="
CONFIG_LIST="
$COMPONENT_LIST
$PROGRAM_LIST
avplay
avprobe
avserver
aandct
ac3dsp
avcodec
avdevice
......@@ -1019,14 +1023,9 @@ CONFIG_LIST="
fft
frei0r
gnutls
golomb
gpl
gray
h264chroma
h264dsp
h264pred
hardcoded_tables
huffman
libaacplus
libass
libbluray
......@@ -1059,7 +1058,6 @@ CONFIG_LIST="
libx264
libxavs
libxvid
lpc
lsp
mdct
memalign_hack
......@@ -1261,9 +1259,16 @@ HAVE_LIST="
# options emitted with CONFIG_ prefix but not available on command line
CONFIG_EXTRA="
aandct
avutil
golomb
gplv3
h264chroma
h264dsp
h264pred
huffman
lgplv3
lpc
"
CMDLINE_SELECT="
......
libavcodec/Makefile
View file @ 967facb6
......@@ -61,7 +61,8 @@ OBJS-$(CONFIG_AAC_DECODER) += aacdec.o aactab.o aacsbr.o aacps.o \
OBJS-$(CONFIG_AAC_ENCODER) += aacenc.o aaccoder.o \
aacpsy.o aactab.o \
psymodel.o iirfilter.o \
mpeg4audio.o kbdwin.o
mpeg4audio.o kbdwin.o \
audio_frame_queue.o
OBJS-$(CONFIG_AASC_DECODER) += aasc.o msrledec.o
OBJS-$(CONFIG_AC3_DECODER) += ac3dec.o ac3dec_data.o ac3.o kbdwin.o
OBJS-$(CONFIG_AC3_ENCODER) += ac3enc_float.o ac3enc.o ac3tab.o \
......@@ -314,7 +315,8 @@ OBJS-$(CONFIG_MSVIDEO1_ENCODER) += msvideo1enc.o elbg.o
OBJS-$(CONFIG_MSZH_DECODER) += lcldec.o
OBJS-$(CONFIG_MXPEG_DECODER) += mxpegdec.o mjpegdec.o mjpeg.o
OBJS-$(CONFIG_NELLYMOSER_DECODER) += nellymoserdec.o nellymoser.o
OBJS-$(CONFIG_NELLYMOSER_ENCODER) += nellymoserenc.o nellymoser.o
OBJS-$(CONFIG_NELLYMOSER_ENCODER) += nellymoserenc.o nellymoser.o \
audio_frame_queue.o
OBJS-$(CONFIG_NUV_DECODER) += nuv.o rtjpeg.o
OBJS-$(CONFIG_PAM_DECODER) += pnmdec.o pnm.o
OBJS-$(CONFIG_PAM_ENCODER) += pamenc.o pnm.o
......@@ -353,7 +355,8 @@ OBJS-$(CONFIG_R10K_ENCODER) += r210enc.o
OBJS-$(CONFIG_R210_DECODER) += r210dec.o
OBJS-$(CONFIG_R210_ENCODER) += r210enc.o
OBJS-$(CONFIG_RA_144_DECODER) += ra144dec.o ra144.o celp_filters.o
OBJS-$(CONFIG_RA_144_ENCODER) += ra144enc.o ra144.o celp_filters.o
OBJS-$(CONFIG_RA_144_ENCODER) += ra144enc.o ra144.o celp_filters.o \
audio_frame_queue.o
OBJS-$(CONFIG_RA_288_DECODER) += ra288.o celp_math.o celp_filters.o
OBJS-$(CONFIG_RALF_DECODER) += ralf.o
OBJS-$(CONFIG_RAWVIDEO_DECODER) += rawdec.o
......@@ -630,12 +633,13 @@ OBJS-$(CONFIG_WTV_DEMUXER) += mpeg4audio.o mpegaudiodata.o
OBJS-$(CONFIG_LIBAACPLUS_ENCODER) += libaacplus.o
OBJS-$(CONFIG_LIBCELT_DECODER) += libcelt_dec.o
OBJS-$(CONFIG_LIBDIRAC_DECODER) += libdiracdec.o
OBJS-$(CONFIG_LIBFAAC_ENCODER) += libfaac.o
OBJS-$(CONFIG_LIBFAAC_ENCODER) += libfaac.o audio_frame_queue.o
OBJS-$(CONFIG_LIBGSM_DECODER) += libgsm.o
OBJS-$(CONFIG_LIBGSM_ENCODER) += libgsm.o
OBJS-$(CONFIG_LIBGSM_MS_DECODER) += libgsm.o
OBJS-$(CONFIG_LIBGSM_MS_ENCODER) += libgsm.o
OBJS-$(CONFIG_LIBMP3LAME_ENCODER) += libmp3lame.o mpegaudiodecheader.o
OBJS-$(CONFIG_LIBMP3LAME_ENCODER) += libmp3lame.o mpegaudiodecheader.o \
audio_frame_queue.o
OBJS-$(CONFIG_LIBOPENCORE_AMRNB_DECODER) += libopencore-amr.o
OBJS-$(CONFIG_LIBOPENCORE_AMRNB_ENCODER) += libopencore-amr.o
OBJS-$(CONFIG_LIBOPENCORE_AMRWB_DECODER) += libopencore-amr.o
......@@ -648,14 +652,15 @@ OBJS-$(CONFIG_LIBSCHROEDINGER_ENCODER) += libschroedingerenc.o \
libschroedinger.o \
libdirac_libschro.o
OBJS-$(CONFIG_LIBSPEEX_DECODER) += libspeexdec.o
OBJS-$(CONFIG_LIBSPEEX_ENCODER) += libspeexenc.o
OBJS-$(CONFIG_LIBSPEEX_ENCODER) += libspeexenc.o audio_frame_queue.o
OBJS-$(CONFIG_LIBSTAGEFRIGHT_H264_DECODER)+= libstagefright.o
OBJS-$(CONFIG_LIBTHEORA_ENCODER) += libtheoraenc.o
OBJS-$(CONFIG_LIBUTVIDEO_DECODER) += libutvideodec.o
OBJS-$(CONFIG_LIBUTVIDEO_ENCODER) += libutvideoenc.o
OBJS-$(CONFIG_LIBVO_AACENC_ENCODER) += libvo-aacenc.o mpeg4audio.o
OBJS-$(CONFIG_LIBVO_AMRWBENC_ENCODER) += libvo-amrwbenc.o
OBJS-$(CONFIG_LIBVORBIS_ENCODER) += libvorbis.o vorbis_data.o
OBJS-$(CONFIG_LIBVORBIS_ENCODER) += libvorbis.o audio_frame_queue.o \
vorbis_data.o vorbis_parser.o
OBJS-$(CONFIG_LIBVPX_DECODER) += libvpxdec.o
OBJS-$(CONFIG_LIBVPX_ENCODER) += libvpxenc.o
OBJS-$(CONFIG_LIBX264_ENCODER) += libx264.o
......
libavcodec/aacenc.c
View file @ 967facb6
......@@ -34,6 +34,7 @@
#include "avcodec.h"
#include "put_bits.h"
#include "dsputil.h"
#include "internal.h"
#include "mpeg4audio.h"
#include "kbdwin.h"
#include "sinewin.h"
......@@ -476,8 +477,7 @@ static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s,
* Deinterleave input samples.
* Channels are reordered from libavcodec's default order to AAC order.
*/
static void deinterleave_input_samples(AACEncContext *s,
const float *samples, int nb_samples)
static void deinterleave_input_samples(AACEncContext *s, AVFrame *frame)
{
int ch, i;
const int sinc = s->channels;
......@@ -485,35 +485,43 @@ static void deinterleave_input_samples(AACEncContext *s,
/* deinterleave and remap input samples */
for (ch = 0; ch < sinc; ch++) {
const float *sptr = samples + channel_map[ch];
/* copy last 1024 samples of previous frame to the start of the current frame */
memcpy(&s->planar_samples[ch][1024], &s->planar_samples[ch][2048], 1024 * sizeof(s->planar_samples[0][0]));
/* deinterleave */
for (i = 2048; i < 2048 + nb_samples; i++) {
s->planar_samples[ch][i] = *sptr;
sptr += sinc;
i = 2048;
if (frame) {
const float *sptr = ((const float *)frame->data[0]) + channel_map[ch];
for (; i < 2048 + frame->nb_samples; i++) {
s->planar_samples[ch][i] = *sptr;
sptr += sinc;
}
}
memset(&s->planar_samples[ch][i], 0,
(3072 - i) * sizeof(s->planar_samples[0][0]));
}
}
static int aac_encode_frame(AVCodecContext *avctx,
uint8_t *frame, int buf_size, void *data)
static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
AACEncContext *s = avctx->priv_data;
float **samples = s->planar_samples, *samples2, *la, *overlap;
ChannelElement *cpe;
int i, ch, w, g, chans, tag, start_ch;
int i, ch, w, g, chans, tag, start_ch, ret;
int chan_el_counter[4];
FFPsyWindowInfo windows[AAC_MAX_CHANNELS];
if (s->last_frame == 2)
return 0;
deinterleave_input_samples(s, data, data ? avctx->frame_size : 0);
/* add current frame to queue */
if (frame) {
if ((ret = ff_af_queue_add(&s->afq, frame) < 0))
return ret;
}
deinterleave_input_samples(s, frame);
if (s->psypp)
ff_psy_preprocess(s->psypp, s->planar_samples, s->channels);
......@@ -532,7 +540,7 @@ static int aac_encode_frame(AVCodecContext *avctx,
overlap = &samples[cur_channel][0];
samples2 = overlap + 1024;
la = samples2 + (448+64);
if (!data)
if (!frame)
la = NULL;
if (tag == TYPE_LFE) {
wi[ch].window_type[0] = ONLY_LONG_SEQUENCE;
......@@ -565,7 +573,13 @@ static int aac_encode_frame(AVCodecContext *avctx,
}
do {
int frame_bits;
init_put_bits(&s->pb, frame, buf_size*8);
if ((ret = ff_alloc_packet(avpkt, 768 * s->channels))) {
av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
return ret;
}
init_put_bits(&s->pb, avpkt->data, avpkt->size);
if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & CODEC_FLAG_BITEXACT))
put_bitstream_info(avctx, s, LIBAVCODEC_IDENT);
start_ch = 0;
......@@ -645,10 +659,15 @@ static int aac_encode_frame(AVCodecContext *avctx,
s->lambda = FFMIN(s->lambda, 65536.f);
}
if (!data)
if (!frame)
s->last_frame++;
return put_bits_count(&s->pb)>>3;
ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
&avpkt->duration);
avpkt->size = put_bits_count(&s->pb) >> 3;
*got_packet_ptr = 1;
return 0;
}
static av_cold int aac_encode_end(AVCodecContext *avctx)
......@@ -662,6 +681,10 @@ static av_cold int aac_encode_end(AVCodecContext *avctx)
ff_psy_preprocess_end(s->psypp);
av_freep(&s->buffer.samples);
av_freep(&s->cpe);
ff_af_queue_close(&s->afq);
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
return 0;
}
......@@ -695,6 +718,11 @@ static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s)
for(ch = 0; ch < s->channels; ch++)
s->planar_samples[ch] = s->buffer.samples + 3 * 1024 * ch;
#if FF_API_OLD_ENCODE_AUDIO
if (!(avctx->coded_frame = avcodec_alloc_frame()))
goto alloc_fail;
#endif
return 0;
alloc_fail:
return AVERROR(ENOMEM);
......@@ -756,6 +784,9 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
for (i = 0; i < 428; i++)
ff_aac_pow34sf_tab[i] = sqrt(ff_aac_pow2sf_tab[i] * sqrt(ff_aac_pow2sf_tab[i]));
avctx->delay = 1024;
ff_af_queue_init(avctx, &s->afq);
return 0;
fail:
aac_encode_end(avctx);
......@@ -785,7 +816,7 @@ AVCodec ff_aac_encoder = {
.id = CODEC_ID_AAC,
.priv_data_size = sizeof(AACEncContext),
.init = aac_encode_init,
.encode = aac_encode_frame,
.encode2 = aac_encode_frame,
.close = aac_encode_end,
.capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY | CODEC_CAP_EXPERIMENTAL,
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE},
......
libavcodec/aacenc.h
View file @ 967facb6
......@@ -27,7 +27,7 @@
#include "dsputil.h"
#include "aac.h"
#include "audio_frame_queue.h"
#include "psymodel.h"
#define AAC_CODER_NB 4
......@@ -74,6 +74,7 @@ typedef struct AACEncContext {
int cur_channel;
int last_frame;
float lambda;
AudioFrameQueue afq;
DECLARE_ALIGNED(16, int, qcoefs)[96]; ///< quantized coefficients
DECLARE_ALIGNED(32, float, scoefs)[1024]; ///< scaled coefficients
......
libavcodec/ac3enc.c
View file @ 967facb6
......@@ -2050,7 +2050,9 @@ av_cold int ff_ac3_encode_close(AVCodecContext *avctx)
s->mdct_end(s);
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
return 0;
}
......@@ -2434,6 +2436,7 @@ av_cold int ff_ac3_encode_init(AVCodecContext *avctx)
return ret;
avctx->frame_size = AC3_BLOCK_SIZE * s->num_blocks;
avctx->delay = AC3_BLOCK_SIZE;
s->bitstream_mode = avctx->audio_service_type;
if (s->bitstream_mode == AV_AUDIO_SERVICE_TYPE_KARAOKE)
......@@ -2479,9 +2482,13 @@ av_cold int ff_ac3_encode_init(AVCodecContext *avctx)
if (ret)
goto init_fail;
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame= avcodec_alloc_frame();
if (!avctx->coded_frame)
if (!avctx->coded_frame) {
ret = AVERROR(ENOMEM);
goto init_fail;
}
#endif
ff_dsputil_init(&s->dsp, avctx);
ff_ac3dsp_init(&s->ac3dsp, avctx->flags & CODEC_FLAG_BITEXACT);
......
libavcodec/ac3enc.h
View file @ 967facb6
......@@ -297,9 +297,9 @@ int ff_ac3_float_mdct_init(AC3EncodeContext *s);
int ff_ac3_fixed_allocate_sample_buffers(AC3EncodeContext *s);
int ff_ac3_float_allocate_sample_buffers(AC3EncodeContext *s);
int ff_ac3_fixed_encode_frame(AVCodecContext *avctx, unsigned char *frame,
int buf_size, void *data);
int ff_ac3_float_encode_frame(AVCodecContext *avctx, unsigned char *frame,
int buf_size, void *data);
int ff_ac3_fixed_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr);
int ff_ac3_float_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr);
#endif /* AVCODEC_AC3ENC_H */
libavcodec/ac3enc_fixed.c
View file @ 967facb6
......@@ -28,6 +28,7 @@
#define CONFIG_FFT_FLOAT 0
#undef CONFIG_AC3ENC_FLOAT
#include "internal.h"
#include "ac3enc.h"
#include "eac3enc.h"
......@@ -151,7 +152,7 @@ AVCodec ff_ac3_fixed_encoder = {
.id = CODEC_ID_AC3,
.priv_data_size = sizeof(AC3EncodeContext),
.init = ac3_fixed_encode_init,
.encode = ff_ac3_fixed_encode_frame,
.encode2 = ff_ac3_fixed_encode_frame,
.close = ff_ac3_encode_close,
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
......
libavcodec/ac3enc_float.c
View file @ 967facb6
......@@ -27,6 +27,7 @@
*/
#define CONFIG_AC3ENC_FLOAT 1
#include "internal.h"
#include "ac3enc.h"
#include "eac3enc.h"
#include "kbdwin.h"
......@@ -149,7 +150,7 @@ AVCodec ff_ac3_encoder = {
.id = CODEC_ID_AC3,
.priv_data_size = sizeof(AC3EncodeContext),
.init = ff_ac3_encode_init,
.encode = ff_ac3_float_encode_frame,
.encode2 = ff_ac3_float_encode_frame,
.close = ff_ac3_encode_close,
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
......
libavcodec/ac3enc_template.c
View file @ 967facb6
......@@ -386,11 +386,11 @@ static void compute_rematrixing_strategy(AC3EncodeContext *s)
}
int AC3_NAME(encode_frame)(AVCodecContext *avctx, unsigned char *frame,
int buf_size, void *data)
int AC3_NAME(encode_frame)(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
AC3EncodeContext *s = avctx->priv_data;
const SampleType *samples = data;
const SampleType *samples = (const SampleType *)frame->data[0];
int ret;
if (s->options.allow_per_frame_metadata) {
......@@ -437,7 +437,15 @@ int AC3_NAME(encode_frame)(AVCodecContext *avctx, unsigned char *frame,
ff_ac3_quantize_mantissas(s);
ff_ac3_output_frame(s, frame);
if ((ret = ff_alloc_packet(avpkt, s->frame_size))) {
av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
return ret;
}
ff_ac3_output_frame(s, avpkt->data);
return s->frame_size;
if (frame->pts != AV_NOPTS_VALUE)
avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->delay);
*got_packet_ptr = 1;
return 0;
}
libavcodec/adpcmenc.c
View file @ 967facb6
......@@ -24,6 +24,7 @@
#include "bytestream.h"
#include "adpcm.h"
#include "adpcm_data.h"
#include "internal.h"
/**
* @file
......@@ -144,8 +145,10 @@ static av_cold int adpcm_encode_init(AVCodecContext *avctx)
goto error;
}
#if FF_API_OLD_ENCODE_AUDIO
if (!(avctx->coded_frame = avcodec_alloc_frame()))
goto error;
#endif
return 0;
error:
......@@ -156,7 +159,9 @@ error:
static av_cold int adpcm_encode_close(AVCodecContext *avctx)
{
ADPCMEncodeContext *s = avctx->priv_data;
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
av_freep(&s->paths);
av_freep(&s->node_buf);
av_freep(&s->nodep_buf);
......@@ -473,23 +478,31 @@ static void adpcm_compress_trellis(AVCodecContext *avctx,
c->idelta = nodes[0]->step;
}
static int adpcm_encode_frame(AVCodecContext *avctx, uint8_t *frame,
int buf_size, void *data)
static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
int n, i, st;
int16_t *samples;
int n, i, st, pkt_size, ret;
const int16_t *samples;
uint8_t *dst;
ADPCMEncodeContext *c = avctx->priv_data;
uint8_t *buf;
dst = frame;
samples = data;
samples = (const int16_t *)frame->data[0];
st = avctx->channels == 2;
/* n = (BLKSIZE - 4 * avctx->channels) / (2 * 8 * avctx->channels); */
if (avctx->codec_id == CODEC_ID_ADPCM_SWF)
pkt_size = (2 + avctx->channels * (22 + 4 * (frame->nb_samples - 1)) + 7) / 8;
else
pkt_size = avctx->block_align;
if ((ret = ff_alloc_packet(avpkt, pkt_size))) {
av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
return ret;
}
dst = avpkt->data;
switch(avctx->codec->id) {
case CODEC_ID_ADPCM_IMA_WAV:
n = avctx->frame_size / 8;
n = frame->nb_samples / 8;
c->status[0].prev_sample = samples[0];
/* c->status[0].step_index = 0;
XXX: not sure how to init the state machine */
......@@ -557,7 +570,7 @@ static int adpcm_encode_frame(AVCodecContext *avctx, uint8_t *frame,
{
int ch, i;
PutBitContext pb;
init_put_bits(&pb, dst, buf_size * 8);
init_put_bits(&pb, dst, pkt_size * 8);
for (ch = 0; ch < avctx->channels; ch++) {
put_bits(&pb, 9, (c->status[ch].prev_sample + 0x10000) >> 7);
......@@ -581,16 +594,15 @@ static int adpcm_encode_frame(AVCodecContext *avctx, uint8_t *frame,
}
flush_put_bits(&pb);
dst += put_bits_count(&pb) >> 3;
break;
}
case CODEC_ID_ADPCM_SWF:
{
int i;
PutBitContext pb;
init_put_bits(&pb, dst, buf_size * 8);
init_put_bits(&pb, dst, pkt_size * 8);
n = avctx->frame_size - 1;
n = frame->nb_samples - 1;
// store AdpcmCodeSize
put_bits(&pb, 2, 2); // set 4-bit flash adpcm format
......@@ -617,7 +629,7 @@ static int adpcm_encode_frame(AVCodecContext *avctx, uint8_t *frame,
}
av_free(buf);
} else {
for (i = 1; i < avctx->frame_size; i++) {
for (i = 1; i < frame->nb_samples; i++) {
put_bits(&pb, 4, adpcm_ima_compress_sample(&c->status[0],
samples[avctx->channels * i]));
if (avctx->channels == 2)
......@@ -626,7 +638,6 @@ static int adpcm_encode_frame(AVCodecContext *avctx, uint8_t *frame,
}
}
flush_put_bits(&pb);
dst += put_bits_count(&pb) >> 3;
break;
}
case CODEC_ID_ADPCM_MS:
......@@ -674,7 +685,7 @@ static int adpcm_encode_frame(AVCodecContext *avctx, uint8_t *frame,
}
break;
case CODEC_ID_ADPCM_YAMAHA:
n = avctx->frame_size / 2;
n = frame->nb_samples / 2;
if (avctx->trellis > 0) {
FF_ALLOC_OR_GOTO(avctx, buf, 2 * n * 2, error);
n *= 2;
......@@ -700,7 +711,10 @@ static int adpcm_encode_frame(AVCodecContext *avctx, uint8_t *frame,
default:
return AVERROR(EINVAL);
}
return dst - frame;
avpkt->size = pkt_size;
*got_packet_ptr = 1;
return 0;
error:
return AVERROR(ENOMEM);
}
......@@ -713,7 +727,7 @@ AVCodec ff_ ## name_ ## _encoder = { \
.id = id_, \
.priv_data_size = sizeof(ADPCMEncodeContext), \
.init = adpcm_encode_init, \
.encode = adpcm_encode_frame, \
.encode2 = adpcm_encode_frame, \
.close = adpcm_encode_close, \
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16, \
AV_SAMPLE_FMT_NONE}, \
......
libavcodec/adxenc.c
View file @ 967facb6
......@@ -22,6 +22,7 @@
#include "avcodec.h"
#include "adx.h"
#include "bytestream.h"
#include "internal.h"
#include "put_bits.h"
/**
......@@ -87,9 +88,6 @@ static int adx_encode_header(AVCodecContext *avctx, uint8_t *buf, int bufsize)
{
ADXContext *c = avctx->priv_data;
if (bufsize < HEADER_SIZE)
return AVERROR(EINVAL);
bytestream_put_be16(&buf, 0x8000); /* header signature */
bytestream_put_be16(&buf, HEADER_SIZE - 4); /* copyright offset */
bytestream_put_byte(&buf, 3); /* encoding */
......@@ -119,9 +117,11 @@ static av_cold int adx_encode_init(AVCodecContext *avctx)
}
avctx->frame_size = BLOCK_SAMPLES;
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame = avcodec_alloc_frame();
if (!avctx->coded_frame)
return AVERROR(ENOMEM);
#endif
/* the cutoff can be adjusted, but this seems to work pretty well */
c->cutoff = 500;
......@@ -130,40 +130,38 @@ static av_cold int adx_encode_init(AVCodecContext *avctx)
return 0;
}
static av_cold int adx_encode_close(AVCodecContext *avctx)
{
av_freep(&avctx->coded_frame);
return 0;
}
static int adx_encode_frame(AVCodecContext *avctx, uint8_t *frame,
int buf_size, void *data)
static int adx_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
ADXContext *c = avctx->priv_data;
const int16_t *samples = data;
uint8_t *dst = frame;
int ch;
const int16_t *samples = (const int16_t *)frame->data[0];
uint8_t *dst;
int ch, out_size, ret;
out_size = BLOCK_SIZE * avctx->channels + !c->header_parsed * HEADER_SIZE;
if ((ret = ff_alloc_packet(avpkt, out_size)) < 0) {
av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
return ret;
}
dst = avpkt->data;
if (!c->header_parsed) {
int hdrsize;
if ((hdrsize = adx_encode_header(avctx, dst, buf_size)) < 0) {
if ((hdrsize = adx_encode_header(avctx, dst, avpkt->size)) < 0) {
av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
return AVERROR(EINVAL);
}
dst += hdrsize;
buf_size -= hdrsize;
c->header_parsed = 1;
}
if (buf_size < BLOCK_SIZE * avctx->channels) {
av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
return AVERROR(EINVAL);
}
for (ch = 0; ch < avctx->channels; ch++) {
adx_encode(c, dst, samples + ch, &c->prev[ch], avctx->channels);
dst += BLOCK_SIZE;
}
return dst - frame;
*got_packet_ptr = 1;
return 0;
}
AVCodec ff_adpcm_adx_encoder = {
......@@ -172,8 +170,7 @@ AVCodec ff_adpcm_adx_encoder = {
.id = CODEC_ID_ADPCM_ADX,
.priv_data_size = sizeof(ADXContext),
.init = adx_encode_init,
.encode = adx_encode_frame,
.close = adx_encode_close,
.encode2 = adx_encode_frame,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
.long_name = NULL_IF_CONFIG_SMALL("SEGA CRI ADX ADPCM"),
......
libavcodec/audio_frame_queue.c 0 → 100644
View file @ 967facb6
/*
* Audio Frame Queue
* Copyright (c) 2012 Justin Ruggles
*
* This file is part of Libav.
*
* Libav is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* Libav is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/mathematics.h"
#include "internal.h"
#include "audio_frame_queue.h"
void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq)
{
afq->avctx = avctx;
afq->next_pts = AV_NOPTS_VALUE;
afq->remaining_delay = avctx->delay;
afq->remaining_samples = avctx->delay;
afq->frame_queue = NULL;
}
static void delete_next_frame(AudioFrameQueue *afq)
{
AudioFrame *f = afq->frame_queue;
if (f) {
afq->frame_queue = f->next;
f->next = NULL;
av_freep(&f);
}
}
void ff_af_queue_close(AudioFrameQueue *afq)
{
/* remove/free any remaining frames */
while (afq->frame_queue)
delete_next_frame(afq);
memset(afq, 0, sizeof(*afq));
}
int ff_af_queue_add(AudioFrameQueue *afq, const AVFrame *f)
{
AudioFrame *new_frame;
AudioFrame *queue_end = afq->frame_queue;
/* find the end of the queue */
while (queue_end && queue_end->next)
queue_end = queue_end->next;
/* allocate new frame queue entry */
if (!(new_frame = av_malloc(sizeof(*new_frame))))
return AVERROR(ENOMEM);
/* get frame parameters */
new_frame->next = NULL;
new_frame->duration = f->nb_samples;
if (f->pts != AV_NOPTS_VALUE) {
new_frame->pts = av_rescale_q(f->pts,
afq->avctx->time_base,
(AVRational){ 1, afq->avctx->sample_rate });
afq->next_pts = new_frame->pts + new_frame->duration;
} else {
new_frame->pts = AV_NOPTS_VALUE;
afq->next_pts = AV_NOPTS_VALUE;
}
/* add new frame to the end of the queue */
if (!queue_end)
afq->frame_queue = new_frame;
else
queue_end->next = new_frame;
/* add frame sample count */
afq->remaining_samples += f->nb_samples;
#ifdef DEBUG
ff_af_queue_log_state(afq);
#endif
return 0;
}
void ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, int64_t *pts,
int *duration)
{
int64_t out_pts = AV_NOPTS_VALUE;
int removed_samples = 0;
#ifdef DEBUG
ff_af_queue_log_state(afq);
#endif
/* get output pts from the next frame or generated pts */
if (afq->frame_queue) {
if (afq->frame_queue->pts != AV_NOPTS_VALUE)
out_pts = afq->frame_queue->pts - afq->remaining_delay;
} else {
if (afq->next_pts != AV_NOPTS_VALUE)
out_pts = afq->next_pts - afq->remaining_delay;
}
if (pts) {
if (out_pts != AV_NOPTS_VALUE)
*pts = ff_samples_to_time_base(afq->avctx, out_pts);
else
*pts = AV_NOPTS_VALUE;
}
/* if the delay is larger than the packet duration, we use up delay samples
for the output packet and leave all frames in the queue */
if (afq->remaining_delay >= nb_samples) {
removed_samples += nb_samples;
afq->remaining_delay -= nb_samples;
}
/* remove frames from the queue until we have enough to cover the
requested number of samples or until the queue is empty */
while (removed_samples < nb_samples && afq->frame_queue) {
removed_samples += afq->frame_queue->duration;
delete_next_frame(afq);
}
afq->remaining_samples -= removed_samples;
/* if there are no frames left and we have room for more samples, use
any remaining delay samples */
if (removed_samples < nb_samples && afq->remaining_samples > 0) {
int add_samples = FFMIN(afq->remaining_samples,
nb_samples - removed_samples);
removed_samples += add_samples;
afq->remaining_samples -= add_samples;
}
if (removed_samples > nb_samples)
av_log(afq->avctx, AV_LOG_WARNING, "frame_size is too large\n");
if (duration)
*duration = ff_samples_to_time_base(afq->avctx, removed_samples);
}
void ff_af_queue_log_state(AudioFrameQueue *afq)
{
AudioFrame *f;
av_log(afq->avctx, AV_LOG_DEBUG, "remaining delay = %d\n",
afq->remaining_delay);
av_log(afq->avctx, AV_LOG_DEBUG, "remaining samples = %d\n",
afq->remaining_samples);
av_log(afq->avctx, AV_LOG_DEBUG, "frames:\n");
f = afq->frame_queue;
while (f) {
av_log(afq->avctx, AV_LOG_DEBUG, " [ pts=%9"PRId64" duration=%d ]\n",
f->pts, f->duration);
f = f->next;
}
}
libavcodec/audio_frame_queue.h 0 → 100644
View file @ 967facb6
/*
* Audio Frame Queue
* Copyright (c) 2012 Justin Ruggles
*
* This file is part of Libav.
*
* Libav is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* Libav is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVCODEC_AUDIO_FRAME_QUEUE_H
#define AVCODEC_AUDIO_FRAME_QUEUE_H
#include "avcodec.h"
typedef struct AudioFrame {
int64_t pts;
int duration;
struct AudioFrame *next;
} AudioFrame;
typedef struct AudioFrameQueue {
AVCodecContext *avctx;
int64_t next_pts;
int remaining_delay;
int remaining_samples;
AudioFrame *frame_queue;
} AudioFrameQueue;
/**
* Initialize AudioFrameQueue.
*
* @param avctx context to use for time_base and av_log
* @param afq queue context
*/
void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq);
/**
* Close AudioFrameQueue.
*
* Frees memory if needed.
*
* @param afq queue context
*/
void ff_af_queue_close(AudioFrameQueue *afq);
/**
* Add a frame to the queue.
*
* @param afq queue context
* @param f frame to add to the queue
*/
int ff_af_queue_add(AudioFrameQueue *afq, const AVFrame *f);
/**
* Remove frame(s) from the queue.
*
* Retrieves the pts of the next available frame, or a generated pts based on
* the last frame duration if there are no frames left in the queue. The number
* of requested samples should be the full number of samples represented by the
* packet that will be output by the encoder. If fewer samples are available
* in the queue, a smaller value will be used for the output duration.
*
* @param afq queue context
* @param nb_samples number of samples to remove from the queue
* @param[out] pts output packet pts
* @param[out] duration output packet duration
*/
void ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, int64_t *pts,
int *duration);
/**
* Log the current state of the queue.
*
* @param afq queue context
*/
void ff_af_queue_log_state(AudioFrameQueue *afq);
#endif /* AVCODEC_AUDIO_FRAME_QUEUE_H */
libavcodec/eac3enc.c
View file @ 967facb6
......@@ -252,7 +252,7 @@ AVCodec ff_eac3_encoder = {
.id = CODEC_ID_EAC3,
.priv_data_size = sizeof(AC3EncodeContext),
.init = ff_ac3_encode_init,
.encode = ff_ac3_float_encode_frame,
.encode2 = ff_ac3_float_encode_frame,
.close = ff_ac3_encode_close,
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("ATSC A/52 E-AC-3"),
......
libavcodec/flacenc.c
View file @ 967facb6
......@@ -25,6 +25,7 @@
#include "avcodec.h"
#include "get_bits.h"
#include "golomb.h"
#include "internal.h"
#include "lpc.h"
#include "flac.h"
#include "flacdata.h"
......@@ -367,9 +368,11 @@ static av_cold int flac_encode_init(AVCodecContext *avctx)
s->frame_count = 0;
s->min_framesize = s->max_framesize;
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame = avcodec_alloc_frame();
if (!avctx->coded_frame)
return AVERROR(ENOMEM);
#endif
if (channels == 3 &&
avctx->channel_layout != (AV_CH_LAYOUT_STEREO|AV_CH_FRONT_CENTER) ||
......@@ -402,7 +405,7 @@ static av_cold int flac_encode_init(AVCodecContext *avctx)
}
static void init_frame(FlacEncodeContext *s)
static void init_frame(FlacEncodeContext *s, int nb_samples)
{
int i, ch;
FlacFrame *frame;
......@@ -410,7 +413,7 @@ static void init_frame(FlacEncodeContext *s)
frame = &s->frame;
for (i = 0; i < 16; i++) {
if (s->avctx->frame_size == ff_flac_blocksize_table[i]) {
if (nb_samples == ff_flac_blocksize_table[i]) {
frame->blocksize = ff_flac_blocksize_table[i];
frame->bs_code[0] = i;
frame->bs_code[1] = 0;
......@@ -418,7 +421,7 @@ static void init_frame(FlacEncodeContext *s)
}
}
if (i == 16) {
frame->blocksize = s->avctx->frame_size;
frame->blocksize = nb_samples;
if (frame->blocksize <= 256) {
frame->bs_code[0] = 6;
frame->bs_code[1] = frame->blocksize-1;
......@@ -1188,9 +1191,9 @@ static void write_frame_footer(FlacEncodeContext *s)
}
static int write_frame(FlacEncodeContext *s, uint8_t *frame, int buf_size)
static int write_frame(FlacEncodeContext *s, AVPacket *avpkt)
{
init_put_bits(&s->pb, frame, buf_size);
init_put_bits(&s->pb, avpkt->data, avpkt->size);
write_frame_header(s);
write_subframes(s);
write_frame_footer(s);
......@@ -1212,30 +1215,31 @@ static void update_md5_sum(FlacEncodeContext *s, const int16_t *samples)
}
static int flac_encode_frame(AVCodecContext *avctx, uint8_t *frame,
int buf_size, void *data)
static int flac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
FlacEncodeContext *s;
const int16_t *samples = data;
int frame_bytes, out_bytes;
const int16_t *samples;
int frame_bytes, out_bytes, ret;
s = avctx->priv_data;
/* when the last block is reached, update the header in extradata */
if (!data) {
if (!frame) {
s->max_framesize = s->max_encoded_framesize;
av_md5_final(s->md5ctx, s->md5sum);
write_streaminfo(s, avctx->extradata);
return 0;
}
samples = (const int16_t *)frame->data[0];
/* change max_framesize for small final frame */
if (avctx->frame_size < s->frame.blocksize) {
s->max_framesize = ff_flac_get_max_frame_size(avctx->frame_size,
if (frame->nb_samples < s->frame.blocksize) {
s->max_framesize = ff_flac_get_max_frame_size(frame->nb_samples,
s->channels, 16);
}
init_frame(s);
init_frame(s, frame->nb_samples);
copy_samples(s, samples);
......@@ -1250,22 +1254,26 @@ static int flac_encode_frame(AVCodecContext *avctx, uint8_t *frame,
frame_bytes = encode_frame(s);
}
if (buf_size < frame_bytes) {
av_log(avctx, AV_LOG_ERROR, "output buffer too small\n");
return 0;
if ((ret = ff_alloc_packet(avpkt, frame_bytes))) {
av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
return ret;
}
out_bytes = write_frame(s, frame, buf_size);
out_bytes = write_frame(s, avpkt);
s->frame_count++;
avctx->coded_frame->pts = s->sample_count;
s->sample_count += avctx->frame_size;
s->sample_count += frame->nb_samples;
update_md5_sum(s, samples);
if (out_bytes > s->max_encoded_framesize)
s->max_encoded_framesize = out_bytes;
if (out_bytes < s->min_framesize)
s->min_framesize = out_bytes;
return out_bytes;
avpkt->pts = frame->pts;
avpkt->duration = ff_samples_to_time_base(avctx, frame->nb_samples);
avpkt->size = out_bytes;
*got_packet_ptr = 1;
return 0;
}
......@@ -1278,7 +1286,9 @@ static av_cold int flac_encode_close(AVCodecContext *avctx)
}
av_freep(&avctx->extradata);
avctx->extradata_size = 0;
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
return 0;
}
......@@ -1316,7 +1326,7 @@ AVCodec ff_flac_encoder = {
.id = CODEC_ID_FLAC,
.priv_data_size = sizeof(FlacEncodeContext),
.init = flac_encode_init,
.encode = flac_encode_frame,
.encode2 = flac_encode_frame,
.close = flac_encode_close,
.capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY | CODEC_CAP_LOSSLESS,
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
......
libavcodec/g722enc.c
View file @ 967facb6
......@@ -28,6 +28,7 @@
*/
#include "avcodec.h"
#include "internal.h"
#include "g722.h"
#define FREEZE_INTERVAL 128
......@@ -50,6 +51,9 @@ static av_cold int g722_encode_close(AVCodecContext *avctx)
av_freep(&c->node_buf[i]);
av_freep(&c->nodep_buf[i]);
}
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
return 0;
}
......@@ -104,6 +108,7 @@ static av_cold int g722_encode_init(AVCodecContext * avctx)
a common packet size for VoIP applications */
avctx->frame_size = 320;
}
avctx->delay = 22;
if (avctx->trellis) {
/* validate trellis */
......@@ -116,6 +121,14 @@ static av_cold int g722_encode_init(AVCodecContext * avctx)
}
}
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame = avcodec_alloc_frame();
if (!avctx->coded_frame) {
ret = AVERROR(ENOMEM);
goto error;
}
#endif
return 0;
error:
g722_encode_close(avctx);
......@@ -345,27 +358,36 @@ static void g722_encode_no_trellis(G722Context *c,
encode_byte(c, dst++, &samples[i]);
}
static int g722_encode_frame(AVCodecContext *avctx,
uint8_t *dst, int buf_size, void *data)
static int g722_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
G722Context *c = avctx->priv_data;
const int16_t *samples = data;
int nb_samples;
const int16_t *samples = (const int16_t *)frame->data[0];
int nb_samples, out_size, ret;
nb_samples = avctx->frame_size - (avctx->frame_size & 1);
out_size = (frame->nb_samples + 1) / 2;
if ((ret = ff_alloc_packet(avpkt, out_size))) {
av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
return ret;
}
nb_samples = frame->nb_samples - (frame->nb_samples & 1);
if (avctx->trellis)
g722_encode_trellis(c, avctx->trellis, dst, nb_samples, samples);
g722_encode_trellis(c, avctx->trellis, avpkt->data, nb_samples, samples);
else
g722_encode_no_trellis(c, dst, nb_samples, samples);
g722_encode_no_trellis(c, avpkt->data, nb_samples, samples);
/* handle last frame with odd frame_size */
if (nb_samples < avctx->frame_size) {
if (nb_samples < frame->nb_samples) {
int16_t last_samples[2] = { samples[nb_samples], samples[nb_samples] };
encode_byte(c, &dst[nb_samples >> 1], last_samples);
encode_byte(c, &avpkt->data[nb_samples >> 1], last_samples);
}
return (avctx->frame_size + 1) >> 1;
if (frame->pts != AV_NOPTS_VALUE)
avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->delay);
*got_packet_ptr = 1;
return 0;
}
AVCodec ff_adpcm_g722_encoder = {
......@@ -375,7 +397,7 @@ AVCodec ff_adpcm_g722_encoder = {
.priv_data_size = sizeof(G722Context),
.init = g722_encode_init,
.close = g722_encode_close,
.encode = g722_encode_frame,
.encode2 = g722_encode_frame,
.capabilities = CODEC_CAP_SMALL_LAST_FRAME,
.long_name = NULL_IF_CONFIG_SMALL("G.722 ADPCM"),
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
......
libavcodec/g726.c
View file @ 967facb6
......@@ -330,10 +330,12 @@ static av_cold int g726_encode_init(AVCodecContext *avctx)
g726_reset(c);
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame = avcodec_alloc_frame();
if (!avctx->coded_frame)
return AVERROR(ENOMEM);
avctx->coded_frame->key_frame = 1;
#endif
/* select a frame size that will end on a byte boundary and have a size of
approximately 1024 bytes */
......@@ -342,28 +344,37 @@ static av_cold int g726_encode_init(AVCodecContext *avctx)
return 0;
}
#if FF_API_OLD_ENCODE_AUDIO
static av_cold int g726_encode_close(AVCodecContext *avctx)
{
av_freep(&avctx->coded_frame);
return 0;
}
#endif
static int g726_encode_frame(AVCodecContext *avctx,
uint8_t *dst, int buf_size, void *data)
static int g726_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
G726Context *c = avctx->priv_data;
const int16_t *samples = data;
const int16_t *samples = (const int16_t *)frame->data[0];
PutBitContext pb;
int i;
int i, ret, out_size;
init_put_bits(&pb, dst, 1024*1024);
out_size = (frame->nb_samples * c->code_size + 7) / 8;
if ((ret = ff_alloc_packet(avpkt, out_size))) {
av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
return ret;
}
init_put_bits(&pb, avpkt->data, avpkt->size);
for (i = 0; i < avctx->frame_size; i++)
for (i = 0; i < frame->nb_samples; i++)
put_bits(&pb, c->code_size, g726_encode(c, *samples++));
flush_put_bits(&pb);
return put_bits_count(&pb)>>3;
avpkt->size = out_size;
*got_packet_ptr = 1;
return 0;
}
#define OFFSET(x) offsetof(G726Context, x)
......@@ -391,8 +402,10 @@ AVCodec ff_adpcm_g726_encoder = {
.id = CODEC_ID_ADPCM_G726,
.priv_data_size = sizeof(G726Context),
.init = g726_encode_init,
.encode = g726_encode_frame,
.encode2 = g726_encode_frame,
#if FF_API_OLD_ENCODE_AUDIO
.close = g726_encode_close,
#endif
.capabilities = CODEC_CAP_SMALL_LAST_FRAME,
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("G.726 ADPCM"),
......
libavcodec/indeo4.c
View file @ 967facb6
......@@ -372,7 +372,8 @@ static int decode_band_hdr(IVI4DecContext *ctx, IVIBandDesc *band,
if (!get_bits1(&ctx->gb) || ctx->frame_type == FRAMETYPE_INTRA) {
transform_id = get_bits(&ctx->gb, 5);
if (!transforms[transform_id].inv_trans) {
if (transform_id >= FF_ARRAY_ELEMS(transforms) ||
!transforms[transform_id].inv_trans) {
av_log_ask_for_sample(avctx, "Unimplemented transform: %d!\n", transform_id);
return AVERROR_PATCHWELCOME;
}
......
libavcodec/libfaac.c
View file @ 967facb6
......@@ -24,11 +24,19 @@
* Interface to libfaac for aac encoding.
*/
#include "avcodec.h"
#include <faac.h>
#include "avcodec.h"
#include "audio_frame_queue.h"
#include "internal.h"
/* libfaac has an encoder delay of 1024 samples */
#define FAAC_DELAY_SAMPLES 1024
typedef struct FaacAudioContext {
faacEncHandle faac_handle;
AudioFrameQueue afq;
} FaacAudioContext;
static const int channel_maps[][6] = {
......@@ -42,11 +50,15 @@ static av_cold int Faac_encode_close(AVCodecContext *avctx)
{
FaacAudioContext *s = avctx->priv_data;
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
av_freep(&avctx->extradata);
ff_af_queue_close(&s->afq);
if (s->faac_handle)
faacEncClose(s->faac_handle);
return 0;
}
......@@ -118,11 +130,13 @@ static av_cold int Faac_encode_init(AVCodecContext *avctx)
avctx->frame_size = samples_input / avctx->channels;
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame= avcodec_alloc_frame();
if (!avctx->coded_frame) {
ret = AVERROR(ENOMEM);
goto error;
}
#endif
/* Set decoder specific info */
avctx->extradata_size = 0;
......@@ -153,26 +167,52 @@ static av_cold int Faac_encode_init(AVCodecContext *avctx)
goto error;
}
avctx->delay = FAAC_DELAY_SAMPLES;
ff_af_queue_init(avctx, &s->afq);
return 0;
error:
Faac_encode_close(avctx);
return ret;
}
static int Faac_encode_frame(AVCodecContext *avctx,
unsigned char *frame, int buf_size, void *data)
static int Faac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
FaacAudioContext *s = avctx->priv_data;
int bytes_written;
int num_samples = data ? avctx->frame_size : 0;
int bytes_written, ret;
int num_samples = frame ? frame->nb_samples : 0;
void *samples = frame ? frame->data[0] : NULL;
bytes_written = faacEncEncode(s->faac_handle,
data,
if ((ret = ff_alloc_packet(avpkt, (7 + 768) * avctx->channels))) {
av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
return ret;
}
bytes_written = faacEncEncode(s->faac_handle, samples,
num_samples * avctx->channels,
frame,
buf_size);
avpkt->data, avpkt->size);
if (bytes_written < 0) {
av_log(avctx, AV_LOG_ERROR, "faacEncEncode() error\n");
return bytes_written;
}
/* add current frame to the queue */
if (frame) {
if ((ret = ff_af_queue_add(&s->afq, frame) < 0))
return ret;
}
return bytes_written;
if (!bytes_written)
return 0;
/* Get the next frame pts/duration */
ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
&avpkt->duration);
avpkt->size = bytes_written;
*got_packet_ptr = 1;
return 0;
}
static const AVProfile profiles[] = {
......@@ -189,7 +229,7 @@ AVCodec ff_libfaac_encoder = {
.id = CODEC_ID_AAC,
.priv_data_size = sizeof(FaacAudioContext),
.init = Faac_encode_init,
.encode = Faac_encode_frame,
.encode2 = Faac_encode_frame,
.close = Faac_encode_close,
.capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY,
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
......
libavcodec/libgsm.c
View file @ 967facb6
......@@ -30,10 +30,13 @@
#include <gsm/gsm.h>
#include "avcodec.h"
#include "internal.h"
#include "gsm.h"
static av_cold int libgsm_encode_close(AVCodecContext *avctx) {
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
gsm_destroy(avctx->priv_data);
avctx->priv_data = NULL;
return 0;
......@@ -78,9 +81,11 @@ static av_cold int libgsm_encode_init(AVCodecContext *avctx) {
}
}
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame= avcodec_alloc_frame();
if (!avctx->coded_frame)
goto error;
#endif
return 0;
error:
......@@ -88,20 +93,29 @@ error:
return -1;
}
static int libgsm_encode_frame(AVCodecContext *avctx,
unsigned char *frame, int buf_size, void *data) {
// we need a full block
if(buf_size < avctx->block_align) return 0;
static int libgsm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
int ret;
gsm_signal *samples = (gsm_signal *)frame->data[0];
struct gsm_state *state = avctx->priv_data;
if ((ret = ff_alloc_packet(avpkt, avctx->block_align))) {
av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
return ret;
}
switch(avctx->codec_id) {
case CODEC_ID_GSM:
gsm_encode(avctx->priv_data,data,frame);
gsm_encode(state, samples, avpkt->data);
break;
case CODEC_ID_GSM_MS:
gsm_encode(avctx->priv_data,data,frame);
gsm_encode(avctx->priv_data,((short*)data)+GSM_FRAME_SIZE,frame+32);
gsm_encode(state, samples, avpkt->data);
gsm_encode(state, samples + GSM_FRAME_SIZE, avpkt->data + 32);
}
return avctx->block_align;
*got_packet_ptr = 1;
return 0;
}
......@@ -110,7 +124,7 @@ AVCodec ff_libgsm_encoder = {
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_GSM,
.init = libgsm_encode_init,
.encode = libgsm_encode_frame,
.encode2 = libgsm_encode_frame,
.close = libgsm_encode_close,
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("libgsm GSM"),
......@@ -121,7 +135,7 @@ AVCodec ff_libgsm_ms_encoder = {
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_GSM_MS,
.init = libgsm_encode_init,
.encode = libgsm_encode_frame,
.encode2 = libgsm_encode_frame,
.close = libgsm_encode_close,
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("libgsm GSM Microsoft variant"),
......
libavcodec/libmp3lame.c
View file @ 967facb6
......@@ -30,6 +30,7 @@
#include "libavutil/log.h"
#include "libavutil/opt.h"
#include "avcodec.h"
#include "audio_frame_queue.h"
#include "internal.h"
#include "mpegaudio.h"
#include "mpegaudiodecheader.h"
......@@ -44,6 +45,7 @@ typedef struct LAMEContext {
int buffer_index;
int reservoir;
void *planar_samples[2];
AudioFrameQueue afq;
} LAMEContext;
......@@ -51,10 +53,14 @@ static av_cold int mp3lame_encode_close(AVCodecContext *avctx)
{
LAMEContext *s = avctx->priv_data;
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
av_freep(&s->planar_samples[0]);
av_freep(&s->planar_samples[1]);
ff_af_queue_close(&s->afq);
lame_close(s->gfp);
return 0;
}
......@@ -111,12 +117,19 @@ static av_cold int mp3lame_encode_init(AVCodecContext *avctx)
goto error;
}
/* get encoder delay */
avctx->delay = lame_get_encoder_delay(s->gfp) + 528 + 1;
ff_af_queue_init(avctx, &s->afq);
avctx->frame_size = lame_get_framesize(s->gfp);
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame = avcodec_alloc_frame();
if (!avctx->coded_frame) {
ret = AVERROR(ENOMEM);
goto error;
}
#endif
/* sample format */
if (avctx->sample_fmt == AV_SAMPLE_FMT_S32 ||
......@@ -144,67 +157,67 @@ error:
const type *input = samples; \
type *output = s->planar_samples[ch]; \
input += ch; \
for (i = 0; i < s->avctx->frame_size; i++) { \
for (i = 0; i < nb_samples; i++) { \
output[i] = *input * scale; \
input += s->avctx->channels; \
} \
} \
} while (0)
static int encode_frame_int16(LAMEContext *s, void *samples)
static int encode_frame_int16(LAMEContext *s, void *samples, int nb_samples)
{
if (s->avctx->channels > 1) {
return lame_encode_buffer_interleaved(s->gfp, samples,
s->avctx->frame_size,
nb_samples,
s->buffer + s->buffer_index,
BUFFER_SIZE - s->buffer_index);
} else {
return lame_encode_buffer(s->gfp, samples, NULL, s->avctx->frame_size,
return lame_encode_buffer(s->gfp, samples, NULL, nb_samples,
s->buffer + s->buffer_index,
BUFFER_SIZE - s->buffer_index);
}
}
static int encode_frame_int32(LAMEContext *s, void *samples)
static int encode_frame_int32(LAMEContext *s, void *samples, int nb_samples)
{
DEINTERLEAVE(int32_t, 1);
return lame_encode_buffer_int(s->gfp,
s->planar_samples[0], s->planar_samples[1],
s->avctx->frame_size,
nb_samples,
s->buffer + s->buffer_index,
BUFFER_SIZE - s->buffer_index);
}
static int encode_frame_float(LAMEContext *s, void *samples)
static int encode_frame_float(LAMEContext *s, void *samples, int nb_samples)
{
DEINTERLEAVE(float, 32768.0f);
return lame_encode_buffer_float(s->gfp,
s->planar_samples[0], s->planar_samples[1],
s->avctx->frame_size,
nb_samples,
s->buffer + s->buffer_index,
BUFFER_SIZE - s->buffer_index);
}
static int mp3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame,
int buf_size, void *data)
static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
LAMEContext *s = avctx->priv_data;
MPADecodeHeader hdr;
int len;
int len, ret;
int lame_result;
if (data) {
if (frame) {
switch (avctx->sample_fmt) {
case AV_SAMPLE_FMT_S16:
lame_result = encode_frame_int16(s, data);
lame_result = encode_frame_int16(s, frame->data[0], frame->nb_samples);
break;
case AV_SAMPLE_FMT_S32:
lame_result = encode_frame_int32(s, data);
lame_result = encode_frame_int32(s, frame->data[0], frame->nb_samples);
break;
case AV_SAMPLE_FMT_FLT:
lame_result = encode_frame_float(s, data);
lame_result = encode_frame_float(s, frame->data[0], frame->nb_samples);
break;
default:
return AVERROR_BUG;
......@@ -223,6 +236,12 @@ static int mp3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame,
}
s->buffer_index += lame_result;
/* add current frame to the queue */
if (frame) {
if ((ret = ff_af_queue_add(&s->afq, frame) < 0))
return ret;
}
/* Move 1 frame from the LAME buffer to the output packet, if available.
We have to parse the first frame header in the output buffer to
determine the frame size. */
......@@ -236,12 +255,22 @@ static int mp3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame,
av_dlog(avctx, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len,
s->buffer_index);
if (len <= s->buffer_index) {
memcpy(frame, s->buffer, len);
if ((ret = ff_alloc_packet(avpkt, len))) {
av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
return ret;
}
memcpy(avpkt->data, s->buffer, len);
s->buffer_index -= len;
memmove(s->buffer, s->buffer + len, s->buffer_index);
return len;
} else
return 0;
/* Get the next frame pts/duration */
ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
&avpkt->duration);
avpkt->size = len;
*got_packet_ptr = 1;
}
return 0;
}
#define OFFSET(x) offsetof(LAMEContext, x)
......@@ -273,9 +302,9 @@ AVCodec ff_libmp3lame_encoder = {
.id = CODEC_ID_MP3,
.priv_data_size = sizeof(LAMEContext),
.init = mp3lame_encode_init,
.encode = mp3lame_encode_frame,
.encode2 = mp3lame_encode_frame,
.close = mp3lame_encode_close,
.capabilities = CODEC_CAP_DELAY,
.capabilities = CODEC_CAP_DELAY | CODEC_CAP_SMALL_LAST_FRAME,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32,
AV_SAMPLE_FMT_FLT,
AV_SAMPLE_FMT_S16,
......
libavcodec/libopencore-amr.c
View file @ 967facb6
......@@ -22,6 +22,8 @@
#include "avcodec.h"
#include "libavutil/avstring.h"
#include "libavutil/opt.h"
#include "audio_frame_queue.h"
#include "internal.h"
static void amr_decode_fix_avctx(AVCodecContext *avctx)
{
......@@ -85,6 +87,7 @@ typedef struct AMRContext {
int enc_mode;
int enc_dtx;
int enc_last_frame;
AudioFrameQueue afq;
} AMRContext;
static const AVOption options[] = {
......@@ -196,9 +199,12 @@ static av_cold int amr_nb_encode_init(AVCodecContext *avctx)
avctx->frame_size = 160;
avctx->delay = 50;
ff_af_queue_init(avctx, &s->afq);
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame = avcodec_alloc_frame();
if (!avctx->coded_frame)
return AVERROR(ENOMEM);
#endif
s->enc_state = Encoder_Interface_init(s->enc_dtx);
if (!s->enc_state) {
......@@ -218,38 +224,49 @@ static av_cold int amr_nb_encode_close(AVCodecContext *avctx)
AMRContext *s = avctx->priv_data;
Encoder_Interface_exit(s->enc_state);
ff_af_queue_close(&s->afq);
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
return 0;
}
static int amr_nb_encode_frame(AVCodecContext *avctx,
unsigned char *frame/*out*/,
int buf_size, void *data/*in*/)
static int amr_nb_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
AMRContext *s = avctx->priv_data;
int written;
int written, ret;
int16_t *flush_buf = NULL;
const int16_t *samples = data;
const int16_t *samples = frame ? (const int16_t *)frame->data[0] : NULL;
if (s->enc_bitrate != avctx->bit_rate) {
s->enc_mode = get_bitrate_mode(avctx->bit_rate, avctx);
s->enc_bitrate = avctx->bit_rate;
}
if (data) {
if (avctx->frame_size < 160) {
flush_buf = av_mallocz(160 * sizeof(*flush_buf));
if ((ret = ff_alloc_packet(avpkt, 32))) {
av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
return ret;
}
if (frame) {
if (frame->nb_samples < avctx->frame_size) {
flush_buf = av_mallocz(avctx->frame_size * sizeof(*flush_buf));
if (!flush_buf)
return AVERROR(ENOMEM);
memcpy(flush_buf, samples, avctx->frame_size * sizeof(*flush_buf));
memcpy(flush_buf, samples, frame->nb_samples * sizeof(*flush_buf));
samples = flush_buf;
if (avctx->frame_size < 110)
if (frame->nb_samples < avctx->frame_size - avctx->delay)
s->enc_last_frame = -1;
}
if ((ret = ff_af_queue_add(&s->afq, frame) < 0)) {
av_freep(&flush_buf);
return ret;
}
} else {
if (s->enc_last_frame < 0)
return 0;
flush_buf = av_mallocz(160 * sizeof(*flush_buf));
flush_buf = av_mallocz(avctx->frame_size * sizeof(*flush_buf));
if (!flush_buf)
return AVERROR(ENOMEM);
samples = flush_buf;
......@@ -257,12 +274,18 @@ static int amr_nb_encode_frame(AVCodecContext *avctx,
}
written = Encoder_Interface_Encode(s->enc_state, s->enc_mode, samples,
frame, 0);
avpkt->data, 0);
av_dlog(avctx, "amr_nb_encode_frame encoded %u bytes, bitrate %u, first byte was %#02x\n",
written, s->enc_mode, frame[0]);
/* Get the next frame pts/duration */
ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
&avpkt->duration);
avpkt->size = written;
*got_packet_ptr = 1;
av_freep(&flush_buf);
return written;
return 0;
}
AVCodec ff_libopencore_amrnb_encoder = {
......@@ -271,7 +294,7 @@ AVCodec ff_libopencore_amrnb_encoder = {
.id = CODEC_ID_AMR_NB,
.priv_data_size = sizeof(AMRContext),
.init = amr_nb_encode_init,
.encode = amr_nb_encode_frame,
.encode2 = amr_nb_encode_frame,
.close = amr_nb_encode_close,
.capabilities = CODEC_CAP_DELAY | CODEC_CAP_SMALL_LAST_FRAME,
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
......
libavcodec/libspeexenc.c
View file @ 967facb6
......@@ -70,6 +70,7 @@
#include "libavutil/opt.h"
#include "avcodec.h"
#include "internal.h"
#include "audio_frame_queue.h"
typedef struct {
AVClass *class; ///< AVClass for private options
......@@ -81,8 +82,7 @@ typedef struct {
int cbr_quality; ///< CBR quality 0 to 10
int abr; ///< flag to enable ABR
int pkt_frame_count; ///< frame count for the current packet
int64_t next_pts; ///< next pts, in sample_rate time base
int pkt_sample_count; ///< sample count in the current packet
AudioFrameQueue afq; ///< frame queue
} LibSpeexEncContext;
static av_cold void print_enc_params(AVCodecContext *avctx,
......@@ -200,6 +200,7 @@ static av_cold int encode_init(AVCodecContext *avctx)
/* set encoding delay */
speex_encoder_ctl(s->enc_state, SPEEX_GET_LOOKAHEAD, &avctx->delay);
ff_af_queue_init(avctx, &s->afq);
/* create header packet bytes from header struct */
/* note: libspeex allocates the memory for header_data, which is freed
......@@ -208,13 +209,22 @@ static av_cold int encode_init(AVCodecContext *avctx)
/* allocate extradata and coded_frame */
avctx->extradata = av_malloc(header_size + FF_INPUT_BUFFER_PADDING_SIZE);
if (!avctx->extradata) {
speex_header_free(header_data);
speex_encoder_destroy(s->enc_state);
av_log(avctx, AV_LOG_ERROR, "memory allocation error\n");
return AVERROR(ENOMEM);
}
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame = avcodec_alloc_frame();
if (!avctx->extradata || !avctx->coded_frame) {
if (!avctx->coded_frame) {
av_freep(&avctx->extradata);
speex_header_free(header_data);
speex_encoder_destroy(s->enc_state);
av_log(avctx, AV_LOG_ERROR, "memory allocation error\n");
return AVERROR(ENOMEM);
}
#endif
/* copy header packet to extradata */
memcpy(avctx->extradata, header_data, header_size);
......@@ -228,19 +238,21 @@ static av_cold int encode_init(AVCodecContext *avctx)
return 0;
}
static int encode_frame(AVCodecContext *avctx, uint8_t *frame, int buf_size,
void *data)
static int encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
LibSpeexEncContext *s = avctx->priv_data;
int16_t *samples = data;
int16_t *samples = frame ? (int16_t *)frame->data[0] : NULL;
int ret;
if (data) {
if (samples) {
/* encode Speex frame */
if (avctx->channels == 2)
speex_encode_stereo_int(samples, s->header.frame_size, &s->bits);
speex_encode_int(s->enc_state, samples, &s->bits);
s->pkt_frame_count++;
s->pkt_sample_count += avctx->frame_size;
if ((ret = ff_af_queue_add(&s->afq, frame) < 0))
return ret;
} else {
/* handle end-of-stream */
if (!s->pkt_frame_count)
......@@ -255,18 +267,20 @@ static int encode_frame(AVCodecContext *avctx, uint8_t *frame, int buf_size,
/* write output if all frames for the packet have been encoded */
if (s->pkt_frame_count == s->frames_per_packet) {
s->pkt_frame_count = 0;
avctx->coded_frame->pts = ff_samples_to_time_base(avctx, s->next_pts -
avctx->delay);
s->next_pts += s->pkt_sample_count;
s->pkt_sample_count = 0;
if (buf_size > speex_bits_nbytes(&s->bits)) {
int ret = speex_bits_write(&s->bits, frame, buf_size);
speex_bits_reset(&s->bits);
if ((ret = ff_alloc_packet(avpkt, speex_bits_nbytes(&s->bits)))) {
av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
return ret;
} else {
av_log(avctx, AV_LOG_ERROR, "output buffer too small");
return AVERROR(EINVAL);
}
ret = speex_bits_write(&s->bits, avpkt->data, avpkt->size);
speex_bits_reset(&s->bits);
/* Get the next frame pts/duration */
ff_af_queue_remove(&s->afq, s->frames_per_packet * avctx->frame_size,
&avpkt->pts, &avpkt->duration);
avpkt->size = ret;
*got_packet_ptr = 1;
return 0;
}
return 0;
}
......@@ -278,7 +292,10 @@ static av_cold int encode_close(AVCodecContext *avctx)
speex_bits_destroy(&s->bits);
speex_encoder_destroy(s->enc_state);
ff_af_queue_close(&s->afq);
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
av_freep(&avctx->extradata);
return 0;
......@@ -312,7 +329,7 @@ AVCodec ff_libspeex_encoder = {
.id = CODEC_ID_SPEEX,
.priv_data_size = sizeof(LibSpeexEncContext),
.init = encode_init,
.encode = encode_frame,
.encode2 = encode_frame,
.close = encode_close,
.capabilities = CODEC_CAP_DELAY,
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE },
......
libavcodec/libvo-aacenc.c
View file @ 967facb6
......@@ -23,25 +23,61 @@
#include <vo-aacenc/cmnMemory.h>
#include "avcodec.h"
#include "audio_frame_queue.h"
#include "internal.h"
#include "mpeg4audio.h"
#define FRAME_SIZE 1024
#define ENC_DELAY 1600
typedef struct AACContext {
VO_AUDIO_CODECAPI codec_api;
VO_HANDLE handle;
VO_MEM_OPERATOR mem_operator;
VO_CODEC_INIT_USERDATA user_data;
VO_PBYTE end_buffer;
AudioFrameQueue afq;
int last_frame;
int last_samples;
} AACContext;
static int aac_encode_close(AVCodecContext *avctx)
{
AACContext *s = avctx->priv_data;
s->codec_api.Uninit(s->handle);
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
av_freep(&avctx->extradata);
ff_af_queue_close(&s->afq);
av_freep(&s->end_buffer);
return 0;
}
static av_cold int aac_encode_init(AVCodecContext *avctx)
{
AACContext *s = avctx->priv_data;
AACENC_PARAM params = { 0 };
int index;
int index, ret;
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame = avcodec_alloc_frame();
if (!avctx->coded_frame)
return AVERROR(ENOMEM);
avctx->frame_size = 1024;
#endif
avctx->frame_size = FRAME_SIZE;
avctx->delay = ENC_DELAY;
s->last_frame = 2;
ff_af_queue_init(avctx, &s->afq);
s->end_buffer = av_mallocz(avctx->frame_size * avctx->channels * 2);
if (!s->end_buffer) {
ret = AVERROR(ENOMEM);
goto error;
}
voGetAACEncAPI(&s->codec_api);
......@@ -61,7 +97,8 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
if (s->codec_api.SetParam(s->handle, VO_PID_AAC_ENCPARAM, &params)
!= VO_ERR_NONE) {
av_log(avctx, AV_LOG_ERROR, "Unable to set encoding parameters\n");
return AVERROR(EINVAL);
ret = AVERROR(EINVAL);
goto error;
}
for (index = 0; index < 16; index++)
......@@ -70,43 +107,69 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
if (index == 16) {
av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate %d\n",
avctx->sample_rate);
return AVERROR(ENOSYS);
ret = AVERROR(ENOSYS);
goto error;
}
if (avctx->flags & CODEC_FLAG_GLOBAL_HEADER) {
avctx->extradata_size = 2;
avctx->extradata = av_mallocz(avctx->extradata_size +
FF_INPUT_BUFFER_PADDING_SIZE);
if (!avctx->extradata)
return AVERROR(ENOMEM);
if (!avctx->extradata) {
ret = AVERROR(ENOMEM);
goto error;
}
avctx->extradata[0] = 0x02 << 3 | index >> 1;
avctx->extradata[1] = (index & 0x01) << 7 | avctx->channels << 3;
}
return 0;
error:
aac_encode_close(avctx);
return ret;
}
static int aac_encode_close(AVCodecContext *avctx)
{
AACContext *s = avctx->priv_data;
s->codec_api.Uninit(s->handle);
av_freep(&avctx->coded_frame);
return 0;
}
static int aac_encode_frame(AVCodecContext *avctx,
unsigned char *frame/*out*/,
int buf_size, void *data/*in*/)
static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
AACContext *s = avctx->priv_data;
VO_CODECBUFFER input = { 0 }, output = { 0 };
VO_AUDIO_OUTPUTINFO output_info = { { 0 } };
VO_PBYTE samples;
int ret;
/* handle end-of-stream small frame and flushing */
if (!frame) {
if (s->last_frame <= 0)
return 0;
if (s->last_samples > 0 && s->last_samples < ENC_DELAY - FRAME_SIZE) {
s->last_samples = 0;
s->last_frame--;
}
s->last_frame--;
memset(s->end_buffer, 0, 2 * avctx->channels * avctx->frame_size);
samples = s->end_buffer;
} else {
if (frame->nb_samples < avctx->frame_size) {
s->last_samples = frame->nb_samples;
memcpy(s->end_buffer, frame->data[0], 2 * avctx->channels * frame->nb_samples);
samples = s->end_buffer;
} else {
samples = (VO_PBYTE)frame->data[0];
}
/* add current frame to the queue */
if ((ret = ff_af_queue_add(&s->afq, frame) < 0))
return ret;
}
if ((ret = ff_alloc_packet(avpkt, FFMAX(8192, 768 * avctx->channels)))) {
av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
return ret;
}
input.Buffer = data;
input.Length = 2 * avctx->channels * avctx->frame_size;
output.Buffer = frame;
output.Length = buf_size;
input.Buffer = samples;
input.Length = 2 * avctx->channels * avctx->frame_size;
output.Buffer = avpkt->data;
output.Length = avpkt->size;
s->codec_api.SetInputData(s->handle, &input);
if (s->codec_api.GetOutputData(s->handle, &output, &output_info)
......@@ -114,7 +177,14 @@ static int aac_encode_frame(AVCodecContext *avctx,
av_log(avctx, AV_LOG_ERROR, "Unable to encode frame\n");
return AVERROR(EINVAL);
}
return output.Length;
/* Get the next frame pts/duration */
ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
&avpkt->duration);
avpkt->size = output.Length;
*got_packet_ptr = 1;
return 0;
}
AVCodec ff_libvo_aacenc_encoder = {
......@@ -123,8 +193,9 @@ AVCodec ff_libvo_aacenc_encoder = {
.id = CODEC_ID_AAC,
.priv_data_size = sizeof(AACContext),
.init = aac_encode_init,
.encode = aac_encode_frame,
.encode2 = aac_encode_frame,
.close = aac_encode_close,
.capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY,
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("Android VisualOn AAC"),
};
libavcodec/libvo-amrwbenc.c
View file @ 967facb6
......@@ -21,9 +21,12 @@
#include <vo-amrwbenc/enc_if.h>
#include "avcodec.h"
#include "libavutil/avstring.h"
#include "libavutil/opt.h"
#include "avcodec.h"
#include "internal.h"
#define MAX_PACKET_SIZE (1 + (477 + 7) / 8)
typedef struct AMRWBContext {
AVClass *av_class;
......@@ -86,9 +89,12 @@ static av_cold int amr_wb_encode_init(AVCodecContext *avctx)
s->last_bitrate = avctx->bit_rate;
avctx->frame_size = 320;
avctx->delay = 80;
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame = avcodec_alloc_frame();
if (!avctx->coded_frame)
return AVERROR(ENOMEM);
#endif
s->state = E_IF_init();
......@@ -104,19 +110,34 @@ static int amr_wb_encode_close(AVCodecContext *avctx)
return 0;
}
static int amr_wb_encode_frame(AVCodecContext *avctx,
unsigned char *frame/*out*/,
int buf_size, void *data/*in*/)
static int amr_wb_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
AMRWBContext *s = avctx->priv_data;
int size;
const int16_t *samples = (const int16_t *)frame->data[0];
int size, ret;
if ((ret = ff_alloc_packet(avpkt, MAX_PACKET_SIZE))) {
av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
return ret;
}
if (s->last_bitrate != avctx->bit_rate) {
s->mode = get_wb_bitrate_mode(avctx->bit_rate, avctx);
s->last_bitrate = avctx->bit_rate;
}
size = E_IF_encode(s->state, s->mode, data, frame, s->allow_dtx);
return size;
size = E_IF_encode(s->state, s->mode, samples, avpkt->data, s->allow_dtx);
if (size <= 0 || size > MAX_PACKET_SIZE) {
av_log(avctx, AV_LOG_ERROR, "Error encoding frame\n");
return AVERROR(EINVAL);
}
if (frame->pts != AV_NOPTS_VALUE)
avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->delay);
avpkt->size = size;
*got_packet_ptr = 1;
return 0;
}
AVCodec ff_libvo_amrwbenc_encoder = {
......@@ -125,7 +146,7 @@ AVCodec ff_libvo_amrwbenc_encoder = {
.id = CODEC_ID_AMR_WB,
.priv_data_size = sizeof(AMRWBContext),
.init = amr_wb_encode_init,
.encode = amr_wb_encode_frame,
.encode2 = amr_wb_encode_frame,
.close = amr_wb_encode_close,
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("Android VisualOn Adaptive Multi-Rate "
......
libavcodec/libvorbis.c
View file @ 967facb6
......@@ -29,9 +29,11 @@
#include "libavutil/fifo.h"
#include "libavutil/opt.h"
#include "avcodec.h"
#include "audio_frame_queue.h"
#include "bytestream.h"
#include "internal.h"
#include "vorbis.h"
#include "vorbis_parser.h"
#undef NDEBUG
#include <assert.h>
......@@ -56,6 +58,8 @@ typedef struct OggVorbisContext {
vorbis_comment vc; /**< VorbisComment info */
ogg_packet op; /**< ogg packet */
double iblock; /**< impulse block bias option */
VorbisParseContext vp; /**< parse context to get durations */
AudioFrameQueue afq; /**< frame queue for timestamps */
} OggVorbisContext;
static const AVOption options[] = {
......@@ -186,7 +190,10 @@ static av_cold int oggvorbis_encode_close(AVCodecContext *avctx)
vorbis_info_clear(&s->vi);
av_fifo_free(s->pkt_fifo);
ff_af_queue_close(&s->afq);
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
av_freep(&avctx->extradata);
return 0;
......@@ -247,9 +254,15 @@ static av_cold int oggvorbis_encode_init(AVCodecContext *avctx)
offset += header_code.bytes;
assert(offset == avctx->extradata_size);
if ((ret = avpriv_vorbis_parse_extradata(avctx, &s->vp)) < 0) {
av_log(avctx, AV_LOG_ERROR, "invalid extradata\n");
return ret;
}
vorbis_comment_clear(&s->vc);
avctx->frame_size = OGGVORBIS_FRAME_SIZE;
ff_af_queue_init(avctx, &s->afq);
s->pkt_fifo = av_fifo_alloc(BUFFER_SIZE);
if (!s->pkt_fifo) {
......@@ -257,11 +270,13 @@ static av_cold int oggvorbis_encode_init(AVCodecContext *avctx)
goto error;
}
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame = avcodec_alloc_frame();
if (!avctx->coded_frame) {
ret = AVERROR(ENOMEM);
goto error;
}
#endif
return 0;
error:
......@@ -269,17 +284,17 @@ error:
return ret;
}
static int oggvorbis_encode_frame(AVCodecContext *avctx, unsigned char *packets,
int buf_size, void *data)
static int oggvorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
OggVorbisContext *s = avctx->priv_data;
ogg_packet op;
float *audio = data;
int pkt_size, ret;
int ret, duration;
/* send samples to libvorbis */
if (data) {
const int samples = avctx->frame_size;
if (frame) {
const float *audio = (const float *)frame->data[0];
const int samples = frame->nb_samples;
float **buffer;
int c, channels = s->vi.channels;
......@@ -295,6 +310,8 @@ static int oggvorbis_encode_frame(AVCodecContext *avctx, unsigned char *packets,
av_log(avctx, AV_LOG_ERROR, "error in vorbis_analysis_wrote()\n");
return vorbis_error_to_averror(ret);
}
if ((ret = ff_af_queue_add(&s->afq, frame) < 0))
return ret;
} else {
if (!s->eof)
if ((ret = vorbis_analysis_wrote(&s->vd, 0)) < 0) {
......@@ -330,22 +347,34 @@ static int oggvorbis_encode_frame(AVCodecContext *avctx, unsigned char *packets,
return vorbis_error_to_averror(ret);
}
/* output then next packet from the output buffer, if available */
pkt_size = 0;
if (av_fifo_size(s->pkt_fifo) >= sizeof(ogg_packet)) {
av_fifo_generic_read(s->pkt_fifo, &op, sizeof(ogg_packet), NULL);
pkt_size = op.bytes;
// FIXME: we should use the user-supplied pts and duration
avctx->coded_frame->pts = ff_samples_to_time_base(avctx,
op.granulepos);
if (pkt_size > buf_size) {
av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
return AVERROR(EINVAL);
/* check for available packets */
if (av_fifo_size(s->pkt_fifo) < sizeof(ogg_packet))
return 0;
av_fifo_generic_read(s->pkt_fifo, &op, sizeof(ogg_packet), NULL);
if ((ret = ff_alloc_packet(avpkt, op.bytes))) {
av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
return ret;
}
av_fifo_generic_read(s->pkt_fifo, avpkt->data, op.bytes, NULL);
avpkt->pts = ff_samples_to_time_base(avctx, op.granulepos);
duration = avpriv_vorbis_parse_frame(&s->vp, avpkt->data, avpkt->size);
if (duration > 0) {
/* we do not know encoder delay until we get the first packet from
* libvorbis, so we have to update the AudioFrameQueue counts */
if (!avctx->delay) {
avctx->delay = duration;
s->afq.remaining_delay += duration;
s->afq.remaining_samples += duration;
}
av_fifo_generic_read(s->pkt_fifo, packets, pkt_size, NULL);
ff_af_queue_remove(&s->afq, duration, &avpkt->pts, &avpkt->duration);
}
return pkt_size;
*got_packet_ptr = 1;
return 0;
}
AVCodec ff_libvorbis_encoder = {
......@@ -354,7 +383,7 @@ AVCodec ff_libvorbis_encoder = {
.id = CODEC_ID_VORBIS,
.priv_data_size = sizeof(OggVorbisContext),
.init = oggvorbis_encode_init,
.encode = oggvorbis_encode_frame,
.encode2 = oggvorbis_encode_frame,
.close = oggvorbis_encode_close,
.capabilities = CODEC_CAP_DELAY,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT,
......
libavcodec/mpegaudioenc.c
View file @ 967facb6
......@@ -80,6 +80,7 @@ static av_cold int MPA_encode_init(AVCodecContext *avctx)
bitrate = bitrate / 1000;
s->nb_channels = channels;
avctx->frame_size = MPA_FRAME_SIZE;
avctx->delay = 512 - 32 + 1;
/* encoding freq */
s->lsf = 0;
......@@ -180,9 +181,11 @@ static av_cold int MPA_encode_init(AVCodecContext *avctx)
total_quant_bits[i] = 12 * v;
}
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame= avcodec_alloc_frame();
if (!avctx->coded_frame)
return AVERROR(ENOMEM);
#endif
return 0;
}
......@@ -726,14 +729,14 @@ static void encode_frame(MpegAudioContext *s,
flush_put_bits(p);
}
static int MPA_encode_frame(AVCodecContext *avctx,
unsigned char *frame, int buf_size, void *data)
static int MPA_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
MpegAudioContext *s = avctx->priv_data;
const short *samples = data;
const int16_t *samples = (const int16_t *)frame->data[0];
short smr[MPA_MAX_CHANNELS][SBLIMIT];
unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
int padding, i;
int padding, i, ret;
for(i=0;i<s->nb_channels;i++) {
filter(s, i, samples + i, s->nb_channels);
......@@ -748,16 +751,28 @@ static int MPA_encode_frame(AVCodecContext *avctx,
}
compute_bit_allocation(s, smr, bit_alloc, &padding);
init_put_bits(&s->pb, frame, MPA_MAX_CODED_FRAME_SIZE);
if ((ret = ff_alloc_packet(avpkt, MPA_MAX_CODED_FRAME_SIZE))) {
av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
return ret;
}
init_put_bits(&s->pb, avpkt->data, avpkt->size);
encode_frame(s, bit_alloc, padding);
return put_bits_ptr(&s->pb) - s->pb.buf;
if (frame->pts != AV_NOPTS_VALUE)
avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->delay);
avpkt->size = put_bits_count(&s->pb) / 8;
*got_packet_ptr = 1;
return 0;
}
static av_cold int MPA_encode_close(AVCodecContext *avctx)
{
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
return 0;
}
......@@ -772,7 +787,7 @@ AVCodec ff_mp2_encoder = {
.id = CODEC_ID_MP2,
.priv_data_size = sizeof(MpegAudioContext),
.init = MPA_encode_init,
.encode = MPA_encode_frame,
.encode2 = MPA_encode_frame,
.close = MPA_encode_close,
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
.supported_samplerates= (const int[]){44100, 48000, 32000, 22050, 24000, 16000, 0},
......
libavcodec/nellymoserenc.c
View file @ 967facb6
......@@ -38,8 +38,10 @@
#include "libavutil/mathematics.h"
#include "nellymoser.h"
#include "avcodec.h"
#include "audio_frame_queue.h"
#include "dsputil.h"
#include "fft.h"
#include "internal.h"
#include "sinewin.h"
#define BITSTREAM_WRITER_LE
......@@ -54,6 +56,7 @@ typedef struct NellyMoserEncodeContext {
int last_frame;
DSPContext dsp;
FFTContext mdct_ctx;
AudioFrameQueue afq;
DECLARE_ALIGNED(32, float, mdct_out)[NELLY_SAMPLES];
DECLARE_ALIGNED(32, float, in_buff)[NELLY_SAMPLES];
DECLARE_ALIGNED(32, float, buf)[3 * NELLY_BUF_LEN]; ///< sample buffer
......@@ -136,7 +139,10 @@ static av_cold int encode_end(AVCodecContext *avctx)
av_free(s->opt);
av_free(s->path);
}
ff_af_queue_close(&s->afq);
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
return 0;
}
......@@ -161,6 +167,7 @@ static av_cold int encode_init(AVCodecContext *avctx)
avctx->frame_size = NELLY_SAMPLES;
avctx->delay = NELLY_BUF_LEN;
ff_af_queue_init(avctx, &s->afq);
s->avctx = avctx;
if ((ret = ff_mdct_init(&s->mdct_ctx, 8, 0, 32768.0)) < 0)
goto error;
......@@ -180,11 +187,13 @@ static av_cold int encode_init(AVCodecContext *avctx)
}
}
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame = avcodec_alloc_frame();
if (!avctx->coded_frame) {
ret = AVERROR(ENOMEM);
goto error;
}
#endif
return 0;
error:
......@@ -366,30 +375,44 @@ static void encode_block(NellyMoserEncodeContext *s, unsigned char *output, int
memset(put_bits_ptr(&pb), 0, output + output_size - put_bits_ptr(&pb));
}
static int encode_frame(AVCodecContext *avctx, uint8_t *frame, int buf_size, void *data)
static int encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
NellyMoserEncodeContext *s = avctx->priv_data;
const float *samples = data;
int ret;
if (s->last_frame)
return 0;
memcpy(s->buf, s->buf + NELLY_SAMPLES, NELLY_BUF_LEN * sizeof(*s->buf));
if (data) {
memcpy(s->buf + NELLY_BUF_LEN, samples, avctx->frame_size * sizeof(*s->buf));
if (avctx->frame_size < NELLY_SAMPLES) {
if (frame) {
memcpy(s->buf + NELLY_BUF_LEN, frame->data[0],
frame->nb_samples * sizeof(*s->buf));
if (frame->nb_samples < NELLY_SAMPLES) {
memset(s->buf + NELLY_BUF_LEN + avctx->frame_size, 0,
(NELLY_SAMPLES - avctx->frame_size) * sizeof(*s->buf));
if (avctx->frame_size >= NELLY_BUF_LEN)
(NELLY_SAMPLES - frame->nb_samples) * sizeof(*s->buf));
if (frame->nb_samples >= NELLY_BUF_LEN)
s->last_frame = 1;
}
if ((ret = ff_af_queue_add(&s->afq, frame) < 0))
return ret;
} else {
memset(s->buf + NELLY_BUF_LEN, 0, NELLY_SAMPLES * sizeof(*s->buf));
s->last_frame = 1;
}
encode_block(s, frame, buf_size);
return NELLY_BLOCK_LEN;
if ((ret = ff_alloc_packet(avpkt, NELLY_BLOCK_LEN))) {
av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
return ret;
}
encode_block(s, avpkt->data, avpkt->size);
/* Get the next frame pts/duration */
ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
&avpkt->duration);
*got_packet_ptr = 1;
return 0;
}
AVCodec ff_nellymoser_encoder = {
......@@ -398,7 +421,7 @@ AVCodec ff_nellymoser_encoder = {
.id = CODEC_ID_NELLYMOSER,
.priv_data_size = sizeof(NellyMoserEncodeContext),
.init = encode_init,
.encode = encode_frame,
.encode2 = encode_frame,
.close = encode_end,
.capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY,
.long_name = NULL_IF_CONFIG_SMALL("Nellymoser Asao"),
......
libavcodec/ra144.h
View file @ 967facb6
......@@ -24,6 +24,7 @@
#include <stdint.h>
#include "lpc.h"
#include "audio_frame_queue.h"
#define NBLOCKS 4 ///< number of subblocks within a block
#define BLOCKSIZE 40 ///< subblock size in 16-bit words
......@@ -36,6 +37,7 @@ typedef struct {
AVCodecContext *avctx;
AVFrame frame;
LPCContext lpc_ctx;
AudioFrameQueue afq;
int last_frame;
unsigned int old_energy; ///< previous frame energy
......
libavcodec/ra144enc.c
View file @ 967facb6
......@@ -28,6 +28,8 @@
#include <float.h>
#include "avcodec.h"
#include "audio_frame_queue.h"
#include "internal.h"
#include "put_bits.h"
#include "celp_filters.h"
#include "ra144.h"
......@@ -37,7 +39,10 @@ static av_cold int ra144_encode_close(AVCodecContext *avctx)
{
RA144Context *ractx = avctx->priv_data;
ff_lpc_end(&ractx->lpc_ctx);
ff_af_queue_close(&ractx->afq);
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
return 0;
}
......@@ -64,11 +69,15 @@ static av_cold int ra144_encode_init(AVCodecContext * avctx)
if (ret < 0)
goto error;
ff_af_queue_init(avctx, &ractx->afq);
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame = avcodec_alloc_frame();
if (!avctx->coded_frame) {
ret = AVERROR(ENOMEM);
goto error;
}
#endif
return 0;
error:
......@@ -429,8 +438,8 @@ static void ra144_encode_subblock(RA144Context *ractx,
}
static int ra144_encode_frame(AVCodecContext *avctx, uint8_t *frame,
int buf_size, void *data)
static int ra144_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
static const uint8_t sizes[LPC_ORDER] = {64, 32, 32, 16, 16, 8, 8, 8, 8, 4};
static const uint8_t bit_sizes[LPC_ORDER] = {6, 5, 5, 4, 4, 3, 3, 3, 3, 2};
......@@ -442,16 +451,16 @@ static int ra144_encode_frame(AVCodecContext *avctx, uint8_t *frame,
int16_t block_coefs[NBLOCKS][LPC_ORDER];
int lpc_refl[LPC_ORDER]; /**< reflection coefficients of the frame */
unsigned int refl_rms[NBLOCKS]; /**< RMS of the reflection coefficients */
const int16_t *samples = data;
const int16_t *samples = frame ? (const int16_t *)frame->data[0] : NULL;
int energy = 0;
int i, idx;
int i, idx, ret;
if (ractx->last_frame)
return 0;
if (buf_size < FRAMESIZE) {
av_log(avctx, AV_LOG_ERROR, "output buffer too small\n");
return 0;
if ((ret = ff_alloc_packet(avpkt, FRAMESIZE))) {
av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
return ret;
}
/**
......@@ -465,9 +474,9 @@ static int ra144_encode_frame(AVCodecContext *avctx, uint8_t *frame,
lpc_data[i] = ractx->curr_block[BLOCKSIZE + BLOCKSIZE / 2 + i];
energy += (lpc_data[i] * lpc_data[i]) >> 4;
}
if (data) {
if (frame) {
int j;
for (j = 0; j < avctx->frame_size && i < NBLOCKS * BLOCKSIZE; i++, j++) {
for (j = 0; j < frame->nb_samples && i < NBLOCKS * BLOCKSIZE; i++, j++) {
lpc_data[i] = samples[j] >> 2;
energy += (lpc_data[i] * lpc_data[i]) >> 4;
}
......@@ -499,7 +508,7 @@ static int ra144_encode_frame(AVCodecContext *avctx, uint8_t *frame,
memset(lpc_refl, 0, sizeof(lpc_refl));
}
}
init_put_bits(&pb, frame, buf_size);
init_put_bits(&pb, avpkt->data, avpkt->size);
for (i = 0; i < LPC_ORDER; i++) {
idx = quantize(lpc_refl[i], ff_lpc_refl_cb[i], sizes[i]);
put_bits(&pb, bit_sizes[i], idx);
......@@ -525,15 +534,24 @@ static int ra144_encode_frame(AVCodecContext *avctx, uint8_t *frame,
/* copy input samples to current block for processing in next call */
i = 0;
if (data) {
for (; i < avctx->frame_size; i++)
if (frame) {
for (; i < frame->nb_samples; i++)
ractx->curr_block[i] = samples[i] >> 2;
if ((ret = ff_af_queue_add(&ractx->afq, frame) < 0))
return ret;
} else
ractx->last_frame = 1;
memset(&ractx->curr_block[i], 0,
(NBLOCKS * BLOCKSIZE - i) * sizeof(*ractx->curr_block));
return FRAMESIZE;
/* Get the next frame pts/duration */
ff_af_queue_remove(&ractx->afq, avctx->frame_size, &avpkt->pts,
&avpkt->duration);
avpkt->size = FRAMESIZE;
*got_packet_ptr = 1;
return 0;
}
......@@ -543,7 +561,7 @@ AVCodec ff_ra_144_encoder = {
.id = CODEC_ID_RA_144,
.priv_data_size = sizeof(RA144Context),
.init = ra144_encode_init,
.encode = ra144_encode_frame,
.encode2 = ra144_encode_frame,
.close = ra144_encode_close,
.capabilities = CODEC_CAP_DELAY | CODEC_CAP_SMALL_LAST_FRAME,
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
......
libavcodec/roqaudioenc.c
View file @ 967facb6
......@@ -24,6 +24,7 @@
#include "libavutil/intmath.h"
#include "avcodec.h"
#include "bytestream.h"
#include "internal.h"
#define ROQ_FRAME_SIZE 735
#define ROQ_HEADER_SIZE 8
......@@ -37,6 +38,7 @@ typedef struct
int input_frames;
int buffered_samples;
int16_t *frame_buffer;
int64_t first_pts;
} ROQDPCMContext;
......@@ -44,7 +46,9 @@ static av_cold int roq_dpcm_encode_close(AVCodecContext *avctx)
{
ROQDPCMContext *context = avctx->priv_data;
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
av_freep(&context->frame_buffer);
return 0;
......@@ -77,11 +81,13 @@ static av_cold int roq_dpcm_encode_init(AVCodecContext *avctx)
context->lastSample[0] = context->lastSample[1] = 0;
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame= avcodec_alloc_frame();
if (!avctx->coded_frame) {
ret = AVERROR(ENOMEM);
goto error;
}
#endif
return 0;
error:
......@@ -129,23 +135,25 @@ static unsigned char dpcm_predict(short *previous, short current)
return result;
}
static int roq_dpcm_encode_frame(AVCodecContext *avctx,
unsigned char *frame, int buf_size, void *data)
static int roq_dpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
int i, stereo, data_size;
const int16_t *in = data;
uint8_t *out = frame;
int i, stereo, data_size, ret;
const int16_t *in = frame ? (const int16_t *)frame->data[0] : NULL;
uint8_t *out;
ROQDPCMContext *context = avctx->priv_data;
stereo = (avctx->channels == 2);
if (!data && context->input_frames >= 8)
if (!in && context->input_frames >= 8)
return 0;
if (data && context->input_frames < 8) {
if (in && context->input_frames < 8) {
memcpy(&context->frame_buffer[context->buffered_samples * avctx->channels],
in, avctx->frame_size * avctx->channels * sizeof(*in));
context->buffered_samples += avctx->frame_size;
if (context->input_frames == 0)
context->first_pts = frame->pts;
if (context->input_frames < 7) {
context->input_frames++;
return 0;
......@@ -158,15 +166,16 @@ static int roq_dpcm_encode_frame(AVCodecContext *avctx,
context->lastSample[1] &= 0xFF00;
}
if (context->input_frames == 7 || !data)
if (context->input_frames == 7 || !in)
data_size = avctx->channels * context->buffered_samples;
else
data_size = avctx->channels * avctx->frame_size;
if (buf_size < ROQ_HEADER_SIZE + data_size) {
av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
return AVERROR(EINVAL);
if ((ret = ff_alloc_packet(avpkt, ROQ_HEADER_SIZE + data_size))) {
av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
return ret;
}
out = avpkt->data;
bytestream_put_byte(&out, stereo ? 0x21 : 0x20);
bytestream_put_byte(&out, 0x10);
......@@ -182,12 +191,15 @@ static int roq_dpcm_encode_frame(AVCodecContext *avctx,
for (i = 0; i < data_size; i++)
*out++ = dpcm_predict(&context->lastSample[i & 1], *in++);
avpkt->pts = context->input_frames <= 7 ? context->first_pts : frame->pts;
avpkt->duration = data_size / avctx->channels;
context->input_frames++;
if (!data)
if (!in)
context->input_frames = FFMAX(context->input_frames, 8);
/* Return the result size */
return ROQ_HEADER_SIZE + data_size;
*got_packet_ptr = 1;
return 0;
}
AVCodec ff_roq_dpcm_encoder = {
......@@ -196,7 +208,7 @@ AVCodec ff_roq_dpcm_encoder = {
.id = CODEC_ID_ROQ_DPCM,
.priv_data_size = sizeof(ROQDPCMContext),
.init = roq_dpcm_encode_init,
.encode = roq_dpcm_encode_frame,
.encode2 = roq_dpcm_encode_frame,
.close = roq_dpcm_encode_close,
.capabilities = CODEC_CAP_DELAY,
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
......
libavcodec/vorbisenc.c
View file @ 967facb6
......@@ -27,6 +27,7 @@
#include <float.h>
#include "avcodec.h"
#include "dsputil.h"
#include "internal.h"
#include "fft.h"
#include "vorbis.h"
#include "vorbis_enc_data.h"
......@@ -123,7 +124,7 @@ typedef struct {
int nmodes;
vorbis_enc_mode *modes;
int64_t sample_count;
int64_t next_pts;
} vorbis_enc_context;
#define MAX_CHANNELS 2
......@@ -1014,23 +1015,27 @@ static int apply_window_and_mdct(vorbis_enc_context *venc, const signed short *a
return 1;
}
static int vorbis_encode_frame(AVCodecContext *avccontext,
unsigned char *packets,
int buf_size, void *data)
static int vorbis_encode_frame(AVCodecContext *avccontext, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
vorbis_enc_context *venc = avccontext->priv_data;
const signed short *audio = data;
int samples = data ? avccontext->frame_size : 0;
const int16_t *audio = frame ? (const int16_t *)frame->data[0] : NULL;
int samples = frame ? frame->nb_samples : 0;
vorbis_enc_mode *mode;
vorbis_enc_mapping *mapping;
PutBitContext pb;
int i;
int i, ret;
if (!apply_window_and_mdct(venc, audio, samples))
return 0;
samples = 1 << (venc->log2_blocksize[0] - 1);
init_put_bits(&pb, packets, buf_size);
if ((ret = ff_alloc_packet(avpkt, 8192))) {
av_log(avccontext, AV_LOG_ERROR, "Error getting output packet\n");
return ret;
}
init_put_bits(&pb, avpkt->data, avpkt->size);
if (pb.size_in_bits - put_bits_count(&pb) < 1 + ilog(venc->nmodes - 1)) {
av_log(avccontext, AV_LOG_ERROR, "output buffer is too small\n");
......@@ -1081,10 +1086,20 @@ static int vorbis_encode_frame(AVCodecContext *avccontext,
return AVERROR(EINVAL);
}
avccontext->coded_frame->pts = venc->sample_count;
venc->sample_count += avccontext->frame_size;
flush_put_bits(&pb);
return put_bits_count(&pb) >> 3;
avpkt->size = put_bits_count(&pb) >> 3;
avpkt->duration = ff_samples_to_time_base(avccontext, avccontext->frame_size);
if (frame)
if (frame->pts != AV_NOPTS_VALUE)
avpkt->pts = ff_samples_to_time_base(avccontext, frame->pts);
else
avpkt->pts = venc->next_pts;
if (avpkt->pts != AV_NOPTS_VALUE)
venc->next_pts = avpkt->pts + avpkt->duration;
*got_packet_ptr = 1;
return 0;
}
......@@ -1142,7 +1157,9 @@ static av_cold int vorbis_encode_close(AVCodecContext *avccontext)
ff_mdct_end(&venc->mdct[0]);
ff_mdct_end(&venc->mdct[1]);
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avccontext->coded_frame);
#endif
av_freep(&avccontext->extradata);
return 0 ;
......@@ -1173,11 +1190,13 @@ static av_cold int vorbis_encode_init(AVCodecContext *avccontext)
avccontext->frame_size = 1 << (venc->log2_blocksize[0] - 1);
#if FF_API_OLD_ENCODE_AUDIO
avccontext->coded_frame = avcodec_alloc_frame();
if (!avccontext->coded_frame) {
ret = AVERROR(ENOMEM);
goto error;
}
#endif
return 0;
error:
......@@ -1191,7 +1210,7 @@ AVCodec ff_vorbis_encoder = {
.id = CODEC_ID_VORBIS,
.priv_data_size = sizeof(vorbis_enc_context),
.init = vorbis_encode_init,
.encode = vorbis_encode_frame,
.encode2 = vorbis_encode_frame,
.close = vorbis_encode_close,
.capabilities= CODEC_CAP_DELAY | CODEC_CAP_EXPERIMENTAL,
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
......
libavcodec/wmaenc.c
View file @ 967facb6
......@@ -20,6 +20,7 @@
*/
#include "avcodec.h"
#include "internal.h"
#include "wma.h"
#include "libavutil/avassert.h"
......@@ -87,7 +88,12 @@ static int encode_init(AVCodecContext * avctx){
avctx->bit_rate = avctx->block_align * 8LL * avctx->sample_rate /
s->frame_len;
//av_log(NULL, AV_LOG_ERROR, "%d %d %d %d\n", s->block_align, avctx->bit_rate, s->frame_len, avctx->sample_rate);
avctx->frame_size= s->frame_len;
avctx->frame_size = avctx->delay = s->frame_len;
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame = &s->frame;
avcodec_get_frame_defaults(avctx->coded_frame);
#endif
return 0;
}
......@@ -341,16 +347,17 @@ static int encode_frame(WMACodecContext *s, float (*src_coefs)[BLOCK_MAX_SIZE],
return put_bits_count(&s->pb)/8 - s->block_align;
}
static int encode_superframe(AVCodecContext *avctx,
unsigned char *buf, int buf_size, void *data){
static int encode_superframe(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
WMACodecContext *s = avctx->priv_data;
const short *samples = data;
int i, total_gain;
const int16_t *samples = (const int16_t *)frame->data[0];
int i, total_gain, ret;
s->block_len_bits= s->frame_len_bits; //required by non variable block len
s->block_len = 1 << s->block_len_bits;
apply_window_and_mdct(avctx, samples, avctx->frame_size);
apply_window_and_mdct(avctx, samples, frame->nb_samples);
if (s->ms_stereo) {
float a, b;
......@@ -364,24 +371,25 @@ static int encode_superframe(AVCodecContext *avctx,
}
}
if (buf_size < 2 * MAX_CODED_SUPERFRAME_SIZE) {
av_log(avctx, AV_LOG_ERROR, "output buffer size is too small\n");
return AVERROR(EINVAL);
if ((ret = ff_alloc_packet(avpkt, 2 * MAX_CODED_SUPERFRAME_SIZE))) {
av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
return ret;
}
#if 1
total_gain= 128;
for(i=64; i; i>>=1){
int error= encode_frame(s, s->coefs, buf, buf_size, total_gain-i);
int error = encode_frame(s, s->coefs, avpkt->data, avpkt->size,
total_gain - i);
if(error<0)
total_gain-= i;
}
#else
total_gain= 90;
best= encode_frame(s, s->coefs, buf, buf_size, total_gain);
best = encode_frame(s, s->coefs, avpkt->data, avpkt->size, total_gain);
for(i=32; i; i>>=1){
int scoreL= encode_frame(s, s->coefs, buf, buf_size, total_gain-i);
int scoreR= encode_frame(s, s->coefs, buf, buf_size, total_gain+i);
int scoreL = encode_frame(s, s->coefs, avpkt->data, avpkt->size, total_gain - i);
int scoreR = encode_frame(s, s->coefs, avpkt->data, avpkt->size, total_gain + i);
av_log(NULL, AV_LOG_ERROR, "%d %d %d (%d)\n", scoreL, best, scoreR, total_gain);
if(scoreL < FFMIN(best, scoreR)){
best = scoreL;
......@@ -393,7 +401,7 @@ static int encode_superframe(AVCodecContext *avctx,
}
#endif
encode_frame(s, s->coefs, buf, buf_size, total_gain);
encode_frame(s, s->coefs, avpkt->data, avpkt->size, total_gain);
av_assert0((put_bits_count(&s->pb) & 7) == 0);
i= s->block_align - (put_bits_count(&s->pb)+7)/8;
av_assert0(i>=0);
......@@ -402,7 +410,13 @@ static int encode_superframe(AVCodecContext *avctx,
flush_put_bits(&s->pb);
av_assert0(put_bits_ptr(&s->pb) - s->pb.buf == s->block_align);
return s->block_align;
if (frame->pts != AV_NOPTS_VALUE)
avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->delay);
avpkt->size = s->block_align;
*got_packet_ptr = 1;
return 0;
}
AVCodec ff_wmav1_encoder = {
......@@ -411,7 +425,7 @@ AVCodec ff_wmav1_encoder = {
.id = CODEC_ID_WMAV1,
.priv_data_size = sizeof(WMACodecContext),
.init = encode_init,
.encode = encode_superframe,
.encode2 = encode_superframe,
.close = ff_wma_end,
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 1"),
......@@ -423,7 +437,7 @@ AVCodec ff_wmav2_encoder = {
.id = CODEC_ID_WMAV2,
.priv_data_size = sizeof(WMACodecContext),
.init = encode_init,
.encode = encode_superframe,
.encode2 = encode_superframe,
.close = ff_wma_end,
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 2"),
......
tests/ref/acodec/g722
View file @ 967facb6
d1a10c4d35f752f60798114a156be3a8 *./tests/data/acodec/g722.wav
e4d5ae038f29659c03fcf68818f7be6c *./tests/data/acodec/g722.wav
48053 ./tests/data/acodec/g722.wav
8dafe5b74ccd5f08fed2fb2a69c5475f *./tests/data/g722.acodec.out.wav
stddev: 8939.47 PSNR: 17.30 MAXDIFF:40370 bytes: 191980/ 1058400
tests/ref/lavf/asf
View file @ 967facb6
3d410176ebf9ffdf99d2738922cef260 *./tests/data/lavf/lavf.asf
e60b298a4db9bcedaabaaee9f90d2a42 *./tests/data/lavf/lavf.asf
333489 ./tests/data/lavf/lavf.asf
./tests/data/lavf/lavf.asf CRC=0x9f5ab3e6
tests/ref/lavf/ffm
View file @ 967facb6
8ce2ea9a73a1187647df7bf3c8e1b8fd *./tests/data/lavf/lavf.ffm
793e977bc8b7f0d86f785a9062c4d978 *./tests/data/lavf/lavf.ffm
376832 ./tests/data/lavf/lavf.ffm
./tests/data/lavf/lavf.ffm CRC=0xf361ed74
tests/ref/lavf/mkv
View file @ 967facb6
f99ae18e1212ee184188243107a4b824 *./tests/data/lavf/lavf.mkv
19c989b2a18dc352ede9754af5fcb5f2 *./tests/data/lavf/lavf.mkv
320521 ./tests/data/lavf/lavf.mkv
./tests/data/lavf/lavf.mkv CRC=0x2a83e6b0
./tests/data/lavf/lavf.mkv CRC=0x5b4ae6b0
tests/ref/lavf/mpg
View file @ 967facb6
6103dbae73aec6c9bf05bbbc6ea35f89 *./tests/data/lavf/lavf.mpg
855384c0cd3d0e3843d48698441c1384 *./tests/data/lavf/lavf.mpg
372736 ./tests/data/lavf/lavf.mpg
./tests/data/lavf/lavf.mpg CRC=0xf361ed74
91b42dd3352e21dd0dee57f6a7241ca2 *./tests/data/lavf/lavf.mpg
612b686e2c035b18175ccefdacf9532c *./tests/data/lavf/lavf.mpg
387072 ./tests/data/lavf/lavf.mpg
./tests/data/lavf/lavf.mpg CRC=0x3d6ddf56
dd60652c2193670abffb8c2a123a820e *./tests/data/lavf/lavf.mpg
fcf2c242b41373186d43de3d5c518e5a *./tests/data/lavf/lavf.mpg
372736 ./tests/data/lavf/lavf.mpg
./tests/data/lavf/lavf.mpg CRC=0xf361ed74
tests/ref/lavf/rm
View file @ 967facb6
2b0eebb5814825c9c4b385cbf8e5b0da *./tests/data/lavf/lavf.rm
be73bce6e371fd543f93f668406f3430 *./tests/data/lavf/lavf.rm
346714 ./tests/data/lavf/lavf.rm
tests/ref/lavf/ts
View file @ 967facb6
34f95a300355d474767b436430eba15b *./tests/data/lavf/lavf.ts
258a64dbc1724438e90560294be4be5c *./tests/data/lavf/lavf.ts
406644 ./tests/data/lavf/lavf.ts
./tests/data/lavf/lavf.ts CRC=0x133216c1
tests/ref/lavf/wtv
View file @ 967facb6
7bd312f32538a14f248c2dff85394118 *./tests/data/lavf/lavf.wtv
de9c3be54bafeba1b7f9618609bd0f62 *./tests/data/lavf/lavf.wtv
413696 ./tests/data/lavf/lavf.wtv
./tests/data/lavf/lavf.wtv CRC=0x133216c1
tests/ref/seek/lavf_asf
View file @ 967facb6
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 689 size: 28487
ret: 0 st: 1 flags:1 dts:-0.011000 pts:-0.011000 pos: 689 size: 208
ret: 0 st:-1 flags:0 ts:-1.000000
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 689 size: 28487
ret: 0 st: 1 flags:1 dts:-0.011000 pts:-0.011000 pos: 689 size: 208
ret: 0 st:-1 flags:1 ts: 1.894167
ret: 0 st: 1 flags:1 dts: 0.940000 pts: 0.940000 pos: 301489 size: 209
ret: 0 st: 1 flags:1 dts: 0.930000 pts: 0.930000 pos: 301489 size: 209
ret: 0 st: 0 flags:0 ts: 0.788000
ret: 0 st: 1 flags:1 dts: 0.940000 pts: 0.940000 pos: 301489 size: 209
ret: 0 st: 1 flags:1 dts: 0.930000 pts: 0.930000 pos: 301489 size: 209
ret: 0 st: 0 flags:1 ts:-0.317000
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 689 size: 28487
ret: 0 st: 1 flags:1 dts:-0.011000 pts:-0.011000 pos: 689 size: 208
ret: 0 st: 1 flags:0 ts: 2.577000
ret: 0 st: 1 flags:1 dts: 0.967000 pts: 0.967000 pos: 330289 size: 209
ret: 0 st: 1 flags:1 dts: 0.982000 pts: 0.982000 pos: 330289 size: 209
ret: 0 st: 1 flags:1 ts: 1.471000
ret: 0 st: 1 flags:1 dts: 0.967000 pts: 0.967000 pos: 330289 size: 209
ret: 0 st: 1 flags:1 dts: 0.982000 pts: 0.982000 pos: 330289 size: 209
ret: 0 st:-1 flags:0 ts: 0.365002
ret: 0 st: 1 flags:1 dts: 0.444000 pts: 0.444000 pos: 147889 size: 209
ret: 0 st: 1 flags:1 dts: 0.459000 pts: 0.459000 pos: 147889 size: 209
ret: 0 st:-1 flags:1 ts:-0.740831
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 689 size: 28487
ret: 0 st: 1 flags:1 dts:-0.011000 pts:-0.011000 pos: 689 size: 208
ret: 0 st: 0 flags:0 ts: 2.153000
ret: 0 st: 1 flags:1 dts: 0.940000 pts: 0.940000 pos: 301489 size: 209
ret: 0 st: 1 flags:1 dts: 0.930000 pts: 0.930000 pos: 301489 size: 209
ret: 0 st: 0 flags:1 ts: 1.048000
ret: 0 st: 1 flags:1 dts: 0.940000 pts: 0.940000 pos: 301489 size: 209
ret: 0 st: 1 flags:1 dts: 0.930000 pts: 0.930000 pos: 301489 size: 209
ret: 0 st: 1 flags:0 ts:-0.058000
ret: 0 st: 1 flags:1 dts: 0.000000 pts: 0.000000 pos: 29489 size: 208
ret: 0 st: 1 flags:1 dts:-0.011000 pts:-0.011000 pos: 689 size: 208
ret: 0 st: 1 flags:1 ts: 2.836000
ret: 0 st: 1 flags:1 dts: 0.967000 pts: 0.967000 pos: 330289 size: 209
ret: 0 st: 1 flags:1 dts: 0.982000 pts: 0.982000 pos: 330289 size: 209
ret: 0 st:-1 flags:0 ts: 1.730004
ret: 0 st: 1 flags:1 dts: 0.940000 pts: 0.940000 pos: 301489 size: 209
ret: 0 st: 1 flags:1 dts: 0.930000 pts: 0.930000 pos: 301489 size: 209
ret: 0 st:-1 flags:1 ts: 0.624171
ret: 0 st: 1 flags:1 dts: 0.444000 pts: 0.444000 pos: 147889 size: 209
ret: 0 st: 1 flags:1 dts: 0.459000 pts: 0.459000 pos: 147889 size: 209
ret: 0 st: 0 flags:0 ts:-0.482000
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 689 size: 28487
ret: 0 st: 1 flags:1 dts:-0.011000 pts:-0.011000 pos: 689 size: 208
ret: 0 st: 0 flags:1 ts: 2.413000
ret: 0 st: 1 flags:1 dts: 0.940000 pts: 0.940000 pos: 301489 size: 209
ret: 0 st: 1 flags:1 dts: 0.930000 pts: 0.930000 pos: 301489 size: 209
ret: 0 st: 1 flags:0 ts: 1.307000
ret: 0 st: 1 flags:1 dts: 0.967000 pts: 0.967000 pos: 330289 size: 209
ret: 0 st: 1 flags:1 dts: 0.982000 pts: 0.982000 pos: 330289 size: 209
ret: 0 st: 1 flags:1 ts: 0.201000
ret: 0 st: 1 flags:1 dts: 0.183000 pts: 0.183000 pos: 71089 size: 209
ret: 0 st: 1 flags:1 dts: 0.198000 pts: 0.198000 pos: 74289 size: 209
ret: 0 st:-1 flags:0 ts:-0.904994
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 689 size: 28487
ret: 0 st: 1 flags:1 dts:-0.011000 pts:-0.011000 pos: 689 size: 208
ret: 0 st:-1 flags:1 ts: 1.989173
ret: 0 st: 1 flags:1 dts: 0.940000 pts: 0.940000 pos: 301489 size: 209
ret: 0 st: 1 flags:1 dts: 0.930000 pts: 0.930000 pos: 301489 size: 209
ret: 0 st: 0 flags:0 ts: 0.883000
ret: 0 st: 1 flags:1 dts: 0.940000 pts: 0.940000 pos: 301489 size: 209
ret: 0 st: 1 flags:1 dts: 0.930000 pts: 0.930000 pos: 301489 size: 209
ret: 0 st: 0 flags:1 ts:-0.222000
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 689 size: 28487
ret: 0 st: 1 flags:1 dts:-0.011000 pts:-0.011000 pos: 689 size: 208
ret: 0 st: 1 flags:0 ts: 2.672000
ret: 0 st: 1 flags:1 dts: 0.967000 pts: 0.967000 pos: 330289 size: 209
ret: 0 st: 1 flags:1 dts: 0.982000 pts: 0.982000 pos: 330289 size: 209
ret: 0 st: 1 flags:1 ts: 1.566000
ret: 0 st: 1 flags:1 dts: 0.967000 pts: 0.967000 pos: 330289 size: 209
ret: 0 st: 1 flags:1 dts: 0.982000 pts: 0.982000 pos: 330289 size: 209
ret: 0 st:-1 flags:0 ts: 0.460008
ret: 0 st: 1 flags:1 dts: 0.444000 pts: 0.444000 pos: 147889 size: 209
ret: 0 st: 1 flags:1 dts: 0.459000 pts: 0.459000 pos: 147889 size: 209
ret: 0 st:-1 flags:1 ts:-0.645825
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 689 size: 28487
ret: 0 st: 1 flags:1 dts:-0.011000 pts:-0.011000 pos: 689 size: 208
tests/ref/seek/lavf_ffm
View file @ 967facb6
......@@ -2,52 +2,52 @@ ret: 0 st: 0 flags:1 dts:-0.040000 pts: 0.000000 pos: 8192 size: 24664
ret: 0 st:-1 flags:0 ts:-1.000000
ret: 0 st: 0 flags:1 dts:-0.040000 pts: 0.000000 pos: 8192 size: 24664
ret: 0 st:-1 flags:1 ts: 1.894167
ret: 0 st: 1 flags:1 dts: 0.940408 pts: 0.940408 pos: 376832 size: 209
ret: 0 st: 1 flags:1 dts: 0.929501 pts: 0.929501 pos: 376832 size: 209
ret: 0 st: 0 flags:0 ts: 0.788334
ret: 0 st: 1 flags:1 dts: 0.809796 pts: 0.809796 pos: 327680 size: 209
ret: 0 st: 1 flags:1 dts: 0.825011 pts: 0.825011 pos: 327680 size: 209
ret: 0 st: 0 flags:1 ts:-0.317499
ret: 0 st: 0 flags:1 dts:-0.040000 pts: 0.000000 pos: 8192 size: 24664
ret: 0 st: 1 flags:0 ts: 2.576668
ret: 0 st: 1 flags:1 dts: 0.940408 pts: 0.940408 pos: 376832 size: 209
ret: 0 st: 1 flags:1 dts: 0.929501 pts: 0.929501 pos: 376832 size: 209
ret: 0 st: 1 flags:1 ts: 1.470835
ret: 0 st: 1 flags:1 dts: 0.940408 pts: 0.940408 pos: 376832 size: 209
ret: 0 st: 1 flags:1 dts: 0.929501 pts: 0.929501 pos: 376832 size: 209
ret: 0 st:-1 flags:0 ts: 0.365002
ret: 0 st: 1 flags:1 dts: 0.365714 pts: 0.365714 pos: 163840 size: 209
ret: 0 st: 1 flags:1 dts: 0.380930 pts: 0.380930 pos: 167936 size: 209
ret: 0 st:-1 flags:1 ts:-0.740831
ret: 0 st: 0 flags:1 dts:-0.040000 pts: 0.000000 pos: 8192 size: 24664
ret: 0 st: 0 flags:0 ts: 2.153336
ret: 0 st: 1 flags:1 dts: 0.940408 pts: 0.940408 pos: 376832 size: 209
ret: 0 st: 1 flags:1 dts: 0.929501 pts: 0.929501 pos: 376832 size: 209
ret: 0 st: 0 flags:1 ts: 1.047503
ret: 0 st: 1 flags:1 dts: 0.940408 pts: 0.940408 pos: 376832 size: 209
ret: 0 st: 1 flags:1 dts: 0.929501 pts: 0.929501 pos: 376832 size: 209
ret: 0 st: 1 flags:0 ts:-0.058330
ret: 0 st: 0 flags:1 dts:-0.040000 pts: 0.000000 pos: 8192 size: 24664
ret: 0 st: 1 flags:1 ts: 2.835837
ret: 0 st: 1 flags:1 dts: 0.940408 pts: 0.940408 pos: 376832 size: 209
ret: 0 st: 1 flags:1 dts: 0.929501 pts: 0.929501 pos: 376832 size: 209
ret: 0 st:-1 flags:0 ts: 1.730004
ret: 0 st: 1 flags:1 dts: 0.940408 pts: 0.940408 pos: 376832 size: 209
ret: 0 st: 1 flags:1 dts: 0.929501 pts: 0.929501 pos: 376832 size: 209
ret: 0 st:-1 flags:1 ts: 0.624171
ret: 0 st: 1 flags:1 dts: 0.653061 pts: 0.653061 pos: 274432 size: 209
ret: 0 st: 1 flags:1 dts: 0.642154 pts: 0.642154 pos: 274432 size: 209
ret: 0 st: 0 flags:0 ts:-0.481662
ret: 0 st: 0 flags:1 dts:-0.040000 pts: 0.000000 pos: 8192 size: 24664
ret: 0 st: 0 flags:1 ts: 2.412505
ret: 0 st: 1 flags:1 dts: 0.940408 pts: 0.940408 pos: 376832 size: 209
ret: 0 st: 1 flags:1 dts: 0.929501 pts: 0.929501 pos: 376832 size: 209
ret: 0 st: 1 flags:0 ts: 1.306672
ret: 0 st: 1 flags:1 dts: 0.940408 pts: 0.940408 pos: 376832 size: 209
ret: 0 st: 1 flags:1 dts: 0.929501 pts: 0.929501 pos: 376832 size: 209
ret: 0 st: 1 flags:1 ts: 0.200839
ret: 0 st: 1 flags:1 dts: 0.208980 pts: 0.208980 pos: 114688 size: 209
ret: 0 st: 1 flags:1 dts: 0.224195 pts: 0.224195 pos: 114688 size: 209
ret: 0 st:-1 flags:0 ts:-0.904994
ret: 0 st: 0 flags:1 dts:-0.040000 pts: 0.000000 pos: 8192 size: 24664
ret: 0 st:-1 flags:1 ts: 1.989173
ret: 0 st: 1 flags:1 dts: 0.940408 pts: 0.940408 pos: 376832 size: 209
ret: 0 st: 1 flags:1 dts: 0.929501 pts: 0.929501 pos: 376832 size: 209
ret: 0 st: 0 flags:0 ts: 0.883340
ret: 0 st: 1 flags:1 dts: 0.888163 pts: 0.888163 pos: 352256 size: 209
ret: 0 st: 1 flags:1 dts: 0.877256 pts: 0.877256 pos: 339968 size: 209
ret: 0 st: 0 flags:1 ts:-0.222493
ret: 0 st: 0 flags:1 dts:-0.040000 pts: 0.000000 pos: 8192 size: 24664
ret: 0 st: 1 flags:0 ts: 2.671674
ret: 0 st: 1 flags:1 dts: 0.940408 pts: 0.940408 pos: 376832 size: 209
ret: 0 st: 1 flags:1 dts: 0.929501 pts: 0.929501 pos: 376832 size: 209
ret: 0 st: 1 flags:1 ts: 1.565841
ret: 0 st: 1 flags:1 dts: 0.940408 pts: 0.940408 pos: 376832 size: 209
ret: 0 st: 1 flags:1 dts: 0.929501 pts: 0.929501 pos: 376832 size: 209
ret: 0 st:-1 flags:0 ts: 0.460008
ret: 0 st: 1 flags:1 dts: 0.496327 pts: 0.496327 pos: 221184 size: 209
ret: 0 st: 1 flags:1 dts: 0.485420 pts: 0.485420 pos: 221184 size: 209
ret: 0 st:-1 flags:1 ts:-0.645825
ret: 0 st: 0 flags:1 dts:-0.040000 pts: 0.000000 pos: 8192 size: 24664
tests/ref/seek/lavf_mkv
View file @ 967facb6
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 556 size: 27837
ret: 0 st: 1 flags:1 dts:-0.011000 pts:-0.011000 pos: 555 size: 208
ret: 0 st:-1 flags:0 ts:-1.000000
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 556 size: 27837
ret: 0 st: 1 flags:1 dts: 0.000000 pts: 0.000000 pos: 555 size: 208
ret: 0 st:-1 flags:1 ts: 1.894167
ret: 0 st: 0 flags:1 dts: 0.960000 pts: 0.960000 pos: 291977 size: 27834
ret: 0 st: 0 flags:1 dts: 0.960000 pts: 0.960000 pos: 292193 size: 27834
ret: 0 st: 0 flags:0 ts: 0.788000
ret: 0 st: 0 flags:1 dts: 0.960000 pts: 0.960000 pos: 291977 size: 27834
ret: 0 st: 0 flags:1 dts: 0.960000 pts: 0.960000 pos: 292193 size: 27834
ret: 0 st: 0 flags:1 ts:-0.317000
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 556 size: 27837
ret: 0 st: 1 flags:1 dts: 0.000000 pts: 0.000000 pos: 555 size: 208
ret:-1 st: 1 flags:0 ts: 2.577000
ret: 0 st: 1 flags:1 ts: 1.471000
ret: 0 st: 1 flags:1 dts: 1.019000 pts: 1.019000 pos: 320250 size: 209
ret: 0 st: 1 flags:1 dts: 1.008000 pts: 1.008000 pos: 320250 size: 209
ret: 0 st:-1 flags:0 ts: 0.365002
ret: 0 st: 0 flags:1 dts: 0.480000 pts: 0.480000 pos: 146746 size: 27925
ret: 0 st:-1 flags:1 ts:-0.740831
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 556 size: 27837
ret: 0 st: 1 flags:1 dts: 0.000000 pts: 0.000000 pos: 555 size: 208
ret:-1 st: 0 flags:0 ts: 2.153000
ret: 0 st: 0 flags:1 ts: 1.048000
ret: 0 st: 0 flags:1 dts: 0.960000 pts: 0.960000 pos: 291977 size: 27834
ret: 0 st: 0 flags:1 dts: 0.960000 pts: 0.960000 pos: 292193 size: 27834
ret: 0 st: 1 flags:0 ts:-0.058000
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 556 size: 27837
ret: 0 st: 1 flags:1 dts: 0.015000 pts: 0.015000 pos: 555 size: 208
ret: 0 st: 1 flags:1 ts: 2.836000
ret: 0 st: 1 flags:1 dts: 1.019000 pts: 1.019000 pos: 320250 size: 209
ret: 0 st: 1 flags:1 dts: 1.008000 pts: 1.008000 pos: 320250 size: 209
ret:-1 st:-1 flags:0 ts: 1.730004
ret: 0 st:-1 flags:1 ts: 0.624171
ret: 0 st: 0 flags:1 dts: 0.480000 pts: 0.480000 pos: 146746 size: 27925
ret: 0 st: 0 flags:0 ts:-0.482000
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 556 size: 27837
ret: 0 st: 1 flags:1 dts: 0.000000 pts: 0.000000 pos: 555 size: 208
ret: 0 st: 0 flags:1 ts: 2.413000
ret: 0 st: 0 flags:1 dts: 0.960000 pts: 0.960000 pos: 291977 size: 27834
ret: 0 st: 0 flags:1 dts: 0.960000 pts: 0.960000 pos: 292193 size: 27834
ret:-1 st: 1 flags:0 ts: 1.307000
ret: 0 st: 1 flags:1 ts: 0.201000
ret: 0 st: 1 flags:1 dts: 0.183000 pts: 0.183000 pos: 72126 size: 209
ret: 0 st: 1 flags:1 dts: 0.198000 pts: 0.198000 pos: 555 size: 208
ret: 0 st:-1 flags:0 ts:-0.904994
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 556 size: 27837
ret: 0 st: 1 flags:1 dts: 0.000000 pts: 0.000000 pos: 555 size: 208
ret: 0 st:-1 flags:1 ts: 1.989173
ret: 0 st: 0 flags:1 dts: 0.960000 pts: 0.960000 pos: 291977 size: 27834
ret: 0 st: 0 flags:1 dts: 0.960000 pts: 0.960000 pos: 292193 size: 27834
ret: 0 st: 0 flags:0 ts: 0.883000
ret: 0 st: 0 flags:1 dts: 0.960000 pts: 0.960000 pos: 291977 size: 27834
ret: 0 st: 0 flags:1 dts: 0.960000 pts: 0.960000 pos: 292193 size: 27834
ret: 0 st: 0 flags:1 ts:-0.222000
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 556 size: 27837
ret: 0 st: 1 flags:1 dts: 0.000000 pts: 0.000000 pos: 555 size: 208
ret:-1 st: 1 flags:0 ts: 2.672000
ret: 0 st: 1 flags:1 ts: 1.566000
ret: 0 st: 1 flags:1 dts: 1.019000 pts: 1.019000 pos: 320250 size: 209
ret: 0 st: 1 flags:1 dts: 1.008000 pts: 1.008000 pos: 320250 size: 209
ret: 0 st:-1 flags:0 ts: 0.460008
ret: 0 st: 0 flags:1 dts: 0.480000 pts: 0.480000 pos: 146746 size: 27925
ret: 0 st:-1 flags:1 ts:-0.645825
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 556 size: 27837
ret: 0 st: 1 flags:1 dts: 0.000000 pts: 0.000000 pos: 555 size: 208
tests/ref/seek/lavf_mpg
View file @ 967facb6
ret: 0 st: 1 flags:1 dts: 1.000000 pts: 1.000000 pos: 2048 size: 208
ret: 0 st: 1 flags:1 dts: 0.989089 pts: 0.989089 pos: 2048 size: 208
ret: 0 st:-1 flags:0 ts:-1.000000
ret: 0 st: 1 flags:1 dts: 1.000000 pts: 1.000000 pos: 2048 size: 208
ret: 0 st: 1 flags:1 dts: 0.989089 pts: 0.989089 pos: 2048 size: 208
ret: 0 st:-1 flags:1 ts: 1.894167
ret: 0 st: 0 flags:0 dts: 1.880000 pts: 1.920000 pos: 327680 size: 12894
ret: 0 st: 0 flags:0 ts: 0.788333
ret: 0 st: 1 flags:1 dts: 1.000000 pts: 1.000000 pos: 2048 size: 208
ret: 0 st: 1 flags:1 dts: 0.989089 pts: 0.989089 pos: 2048 size: 208
ret: 0 st: 0 flags:1 ts:-0.317500
ret: 0 st: 1 flags:1 dts: 1.000000 pts: 1.000000 pos: 2048 size: 208
ret: 0 st: 1 flags:1 dts: 0.989089 pts: 0.989089 pos: 2048 size: 208
ret: 0 st: 1 flags:0 ts: 2.576667
ret: 0 st: 1 flags:1 dts: 2.018778 pts: 2.018778 pos: 370700 size: 235
ret: 0 st: 1 flags:1 dts: 2.007867 pts: 2.007867 pos: 370700 size: 235
ret: 0 st: 1 flags:1 ts: 1.470833
ret: 0 st: 1 flags:1 dts: 1.261222 pts: 1.261222 pos: 145408 size: 261
ret: 0 st: 1 flags:1 dts: 1.250322 pts: 1.250322 pos: 145408 size: 261
ret: 0 st:-1 flags:0 ts: 0.365002
ret: 0 st: 1 flags:1 dts: 1.000000 pts: 1.000000 pos: 2048 size: 208
ret: 0 st: 1 flags:1 dts: 0.989089 pts: 0.989089 pos: 2048 size: 208
ret: 0 st:-1 flags:1 ts:-0.740831
ret: 0 st: 1 flags:1 dts: 1.000000 pts: 1.000000 pos: 2048 size: 208
ret: 0 st: 1 flags:1 dts: 0.989089 pts: 0.989089 pos: 2048 size: 208
ret: 0 st: 0 flags:0 ts: 2.153333
ret: 0 st: 0 flags:0 dts: 1.920000 pts: 1.960000 pos: 339968 size: 681
ret: 0 st: 0 flags:1 ts: 1.047500
ret: 0 st: 0 flags:0 dts: 1.040000 pts: 1.080000 pos: 40960 size: 16073
ret: 0 st: 1 flags:0 ts:-0.058333
ret: 0 st: 1 flags:1 dts: 1.000000 pts: 1.000000 pos: 2048 size: 208
ret: 0 st: 1 flags:1 dts: 0.989089 pts: 0.989089 pos: 2048 size: 208
ret: 0 st: 1 flags:1 ts: 2.835833
ret: 0 st: 1 flags:1 dts: 2.018778 pts: 2.018778 pos: 370700 size: 235
ret: 0 st: 1 flags:1 dts: 2.007867 pts: 2.007867 pos: 370700 size: 235
ret: 0 st:-1 flags:0 ts: 1.730004
ret: 0 st: 0 flags:0 dts: 1.760000 pts: 1.800000 pos: 292864 size: 13170
ret: 0 st:-1 flags:1 ts: 0.624171
ret: 0 st: 1 flags:1 dts: 1.000000 pts: 1.000000 pos: 2048 size: 208
ret: 0 st: 1 flags:1 dts: 0.989089 pts: 0.989089 pos: 2048 size: 208
ret: 0 st: 0 flags:0 ts:-0.481667
ret: 0 st: 1 flags:1 dts: 1.000000 pts: 1.000000 pos: 2048 size: 208
ret: 0 st: 1 flags:1 dts: 0.989089 pts: 0.989089 pos: 2048 size: 208
ret: 0 st: 0 flags:1 ts: 2.412500
ret: 0 st: 0 flags:0 dts: 1.920000 pts: 1.960000 pos: 339968 size: 681
ret: 0 st: 1 flags:0 ts: 1.306667
ret: 0 st: 1 flags:1 dts: 1.522444 pts: 1.522444 pos: 342028 size: 314
ret: 0 st: 1 flags:1 dts: 1.511544 pts: 1.511544 pos: 342028 size: 314
ret: 0 st: 1 flags:1 ts: 0.200844
ret: 0 st: 1 flags:1 dts: 1.000000 pts: 1.000000 pos: 2048 size: 208
ret: 0 st: 1 flags:1 dts: 0.989089 pts: 0.989089 pos: 2048 size: 208
ret: 0 st:-1 flags:0 ts:-0.904994
ret: 0 st: 1 flags:1 dts: 1.000000 pts: 1.000000 pos: 2048 size: 208
ret: 0 st: 1 flags:1 dts: 0.989089 pts: 0.989089 pos: 2048 size: 208
ret: 0 st:-1 flags:1 ts: 1.989173
ret: 0 st: 0 flags:0 dts: 1.920000 pts: 1.960000 pos: 339968 size: 681
ret: 0 st: 0 flags:0 ts: 0.883344
ret: 0 st: 1 flags:1 dts: 1.000000 pts: 1.000000 pos: 2048 size: 208
ret: 0 st: 1 flags:1 dts: 0.989089 pts: 0.989089 pos: 2048 size: 208
ret: 0 st: 0 flags:1 ts:-0.222489
ret: 0 st: 1 flags:1 dts: 1.000000 pts: 1.000000 pos: 2048 size: 208
ret: 0 st: 1 flags:1 dts: 0.989089 pts: 0.989089 pos: 2048 size: 208
ret: 0 st: 1 flags:0 ts: 2.671678
ret: 0 st: 1 flags:1 dts: 2.018778 pts: 2.018778 pos: 370700 size: 235
ret: 0 st: 1 flags:1 dts: 2.007867 pts: 2.007867 pos: 370700 size: 235
ret: 0 st: 1 flags:1 ts: 1.565844
ret: 0 st: 1 flags:1 dts: 1.522444 pts: 1.522444 pos: 342028 size: 314
ret: 0 st: 1 flags:1 dts: 1.511544 pts: 1.511544 pos: 342028 size: 314
ret: 0 st:-1 flags:0 ts: 0.460008
ret: 0 st: 1 flags:1 dts: 1.000000 pts: 1.000000 pos: 2048 size: 208
ret: 0 st: 1 flags:1 dts: 0.989089 pts: 0.989089 pos: 2048 size: 208
ret: 0 st:-1 flags:1 ts:-0.645825
ret: 0 st: 1 flags:1 dts: 1.000000 pts: 1.000000 pos: 2048 size: 208
ret: 0 st: 1 flags:1 dts: 0.989089 pts: 0.989089 pos: 2048 size: 208
tests/ref/seek/lavf_rm
View file @ 967facb6
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 406 size: 31082
ret: 0 st: 1 flags:1 dts: 0.000000 pts: 0.000000 pos: 395 size: 278
ret: 0 st:-1 flags:0 ts:-1.000000
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 406 size: 31082
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 696 size: 31082
ret: 0 st:-1 flags:1 ts: 1.894167
ret: 0 st: 0 flags:1 dts: 0.960000 pts: 0.960000 pos: 314990 size: 31143
ret: 0 st: 0 flags:0 ts: 0.788000
ret: 0 st: 0 flags:1 dts: 0.960000 pts: 0.960000 pos: 314990 size: 31143
ret: 0 st: 0 flags:1 ts:-0.317000
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 406 size: 31082
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 696 size: 31082
ret: 0 st: 1 flags:0 ts: 2.577000
ret: 0 st: 1 flags:1 dts: 0.975000 pts: 0.975000 pos: 346136 size: 278
ret: 0 st: 1 flags:1 ts: 1.471000
......@@ -14,13 +14,13 @@ ret: 0 st: 1 flags:1 dts: 0.975000 pts: 0.975000 pos: 346136 size: 278
ret: 0 st:-1 flags:0 ts: 0.365002
ret: 0 st: 0 flags:1 dts: 0.480000 pts: 0.480000 pos: 158523 size: 31134
ret: 0 st:-1 flags:1 ts:-0.740831
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 406 size: 31082
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 696 size: 31082
ret: 0 st: 0 flags:0 ts: 2.153000
ret: 0 st: 0 flags:1 dts: 0.960000 pts: 0.960000 pos: 314990 size: 31143
ret: 0 st: 0 flags:1 ts: 1.048000
ret: 0 st: 0 flags:1 dts: 0.960000 pts: 0.960000 pos: 314990 size: 31143
ret: 0 st: 1 flags:0 ts:-0.058000
ret: 0 st: 1 flags:1 dts: 0.000000 pts: 0.000000 pos: 31491 size: 278
ret: 0 st: 1 flags:1 dts: 0.000000 pts: 0.000000 pos: 395 size: 278
ret: 0 st: 1 flags:1 ts: 2.836000
ret: 0 st: 1 flags:1 dts: 0.975000 pts: 0.975000 pos: 346136 size: 278
ret: 0 st:-1 flags:0 ts: 1.730004
......@@ -28,7 +28,7 @@ ret: 0 st: 0 flags:1 dts: 0.960000 pts: 0.960000 pos: 314990 size: 31143
ret: 0 st:-1 flags:1 ts: 0.624171
ret: 0 st: 0 flags:1 dts: 0.480000 pts: 0.480000 pos: 158523 size: 31134
ret: 0 st: 0 flags:0 ts:-0.482000
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 406 size: 31082
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 696 size: 31082
ret: 0 st: 0 flags:1 ts: 2.413000
ret: 0 st: 0 flags:1 dts: 0.960000 pts: 0.960000 pos: 314990 size: 31143
ret: 0 st: 1 flags:0 ts: 1.307000
......@@ -36,13 +36,13 @@ ret: 0 st: 1 flags:1 dts: 0.975000 pts: 0.975000 pos: 346136 size: 278
ret: 0 st: 1 flags:1 ts: 0.201000
ret: 0 st: 1 flags:1 dts: 0.174000 pts: 0.174000 pos: 78977 size: 278
ret: 0 st:-1 flags:0 ts:-0.904994
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 406 size: 31082
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 696 size: 31082
ret: 0 st:-1 flags:1 ts: 1.989173
ret: 0 st: 0 flags:1 dts: 0.960000 pts: 0.960000 pos: 314990 size: 31143
ret: 0 st: 0 flags:0 ts: 0.883000
ret: 0 st: 0 flags:1 dts: 0.960000 pts: 0.960000 pos: 314990 size: 31143
ret: 0 st: 0 flags:1 ts:-0.222000
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 406 size: 31082
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 696 size: 31082
ret: 0 st: 1 flags:0 ts: 2.672000
ret: 0 st: 1 flags:1 dts: 0.975000 pts: 0.975000 pos: 346136 size: 278
ret: 0 st: 1 flags:1 ts: 1.566000
......@@ -50,4 +50,4 @@ ret: 0 st: 1 flags:1 dts: 0.975000 pts: 0.975000 pos: 346136 size: 278
ret: 0 st:-1 flags:0 ts: 0.460008
ret: 0 st: 0 flags:1 dts: 0.480000 pts: 0.480000 pos: 158523 size: 31134
ret: 0 st:-1 flags:1 ts:-0.645825
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 406 size: 31082
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 696 size: 31082
tests/ref/seek/lavf_ts
View file @ 967facb6
......@@ -8,9 +8,9 @@ ret: 0 st: 0 flags:1 dts: 1.360000 pts: 1.400000 pos: 564 size: 24801
ret: 0 st: 0 flags:1 ts:-0.317500
ret: 0 st: 0 flags:1 dts: 1.360000 pts: 1.400000 pos: 564 size: 24801
ret: 0 st: 1 flags:0 ts: 2.576667
ret: 0 st: 1 flags:1 dts: 2.131433 pts: 2.131433 pos: 404012 size: 209
ret: 0 st: 1 flags:1 dts: 2.120522 pts: 2.120522 pos: 404012 size: 209
ret: 0 st: 1 flags:1 ts: 1.470833
ret: 0 st: 1 flags:1 dts: 1.400000 pts: 1.400000 pos: 172584 size: 208
ret: 0 st: 1 flags:1 dts: 1.389089 pts: 1.389089 pos: 159800 size: 208
ret: 0 st:-1 flags:0 ts: 0.365002
ret: 0 st: 0 flags:1 dts: 1.360000 pts: 1.400000 pos: 564 size: 24801
ret: 0 st:-1 flags:1 ts:-0.740831
......@@ -20,21 +20,21 @@ ret: 0 st: 0 flags:0 dts: 2.160000 pts: 2.200000 pos: 325616 size: 12679
ret: 0 st: 0 flags:1 ts: 1.047500
ret: 0 st: 0 flags:1 dts: 1.360000 pts: 1.400000 pos: 564 size: 24801
ret: 0 st: 1 flags:0 ts:-0.058333
ret: 0 st: 1 flags:1 dts: 1.400000 pts: 1.400000 pos: 172584 size: 208
ret: 0 st: 1 flags:1 dts: 1.389089 pts: 1.389089 pos: 159800 size: 208
ret: 0 st: 1 flags:1 ts: 2.835833
ret: 0 st: 1 flags:1 dts: 2.131433 pts: 2.131433 pos: 404012 size: 209
ret: 0 st: 1 flags:1 dts: 2.120522 pts: 2.120522 pos: 404012 size: 209
ret: 0 st:-1 flags:0 ts: 1.730004
ret: 0 st: 1 flags:1 dts: 1.400000 pts: 1.400000 pos: 172584 size: 208
ret: 0 st: 0 flags:0 dts: 1.760000 pts: 1.800000 pos: 162808 size: 12075
ret: 0 st:-1 flags:1 ts: 0.624171
ret: 0 st: 0 flags:1 dts: 1.360000 pts: 1.400000 pos: 564 size: 24801
ret: 0 st: 0 flags:0 ts:-0.481667
ret: 0 st: 0 flags:1 dts: 1.360000 pts: 1.400000 pos: 564 size: 24801
ret: 0 st: 0 flags:1 ts: 2.412500
ret: 0 st: 1 flags:1 dts: 2.131433 pts: 2.131433 pos: 404012 size: 209
ret: 0 st: 1 flags:1 dts: 2.120522 pts: 2.120522 pos: 404012 size: 209
ret: 0 st: 1 flags:0 ts: 1.306667
ret: 0 st: 1 flags:1 dts: 1.400000 pts: 1.400000 pos: 172584 size: 208
ret: 0 st: 1 flags:1 dts: 1.389089 pts: 1.389089 pos: 159800 size: 208
ret: 0 st: 1 flags:1 ts: 0.200844
ret: 0 st: 1 flags:1 dts: 1.400000 pts: 1.400000 pos: 172584 size: 208
ret: 0 st: 1 flags:1 dts: 1.389089 pts: 1.389089 pos: 159800 size: 208
ret: 0 st:-1 flags:0 ts:-0.904994
ret: 0 st: 0 flags:1 dts: 1.360000 pts: 1.400000 pos: 564 size: 24801
ret: 0 st:-1 flags:1 ts: 1.989173
......@@ -44,9 +44,9 @@ ret: 0 st: 0 flags:1 dts: 1.360000 pts: 1.400000 pos: 564 size: 24801
ret: 0 st: 0 flags:1 ts:-0.222489
ret: 0 st: 0 flags:1 dts: 1.360000 pts: 1.400000 pos: 564 size: 24801
ret: 0 st: 1 flags:0 ts: 2.671678
ret: 0 st: 1 flags:1 dts: 2.131433 pts: 2.131433 pos: 404012 size: 209
ret: 0 st: 1 flags:1 dts: 2.120522 pts: 2.120522 pos: 404012 size: 209
ret: 0 st: 1 flags:1 ts: 1.565844
ret: 0 st: 1 flags:1 dts: 1.400000 pts: 1.400000 pos: 172584 size: 208
ret: 0 st: 1 flags:1 dts: 1.389089 pts: 1.389089 pos: 159800 size: 208
ret: 0 st:-1 flags:0 ts: 0.460008
ret: 0 st: 0 flags:1 dts: 1.360000 pts: 1.400000 pos: 564 size: 24801
ret: 0 st:-1 flags:1 ts:-0.645825
......
tests/ref/seek/lavf_wtv
View file @ 967facb6
ret: 0 st: 0 flags:1 dts:-0.040000 pts: 0.000000 pos: 2144 size: 24801
ret: 0 st: 1 flags:1 dts:-0.010907 pts:-0.010907 pos: 27072 size: 208
ret: 0 st:-1 flags:0 ts:-1.000000
ret: 0 st: 0 flags:1 dts: NOPTS pts: 0.000000 pos: 2144 size: 24801
ret: 0 st: 1 flags:1 dts:-0.010907 pts:-0.010907 pos: 27072 size: 208
ret:-1 st:-1 flags:1 ts: 1.894167
ret: 0 st: 0 flags:0 ts: 0.788334
ret: 0 st: 1 flags:1 dts: 0.783674 pts: 0.783674 pos: 321176 size: 209
ret: 0 st: 1 flags:1 dts: 0.772766 pts: 0.772766 pos: 321176 size: 209
ret: 0 st: 0 flags:1 ts:-0.317499
ret: 0 st: 0 flags:1 dts: NOPTS pts: 0.000000 pos: 2144 size: 24801
ret: 0 st: 1 flags:1 dts:-0.010907 pts:-0.010907 pos: 27072 size: 208
ret:-1 st: 1 flags:0 ts: 2.576668
ret:-1 st: 1 flags:1 ts: 1.470835
ret: 0 st:-1 flags:0 ts: 0.365002
ret: 0 st: 1 flags:1 dts: 0.365714 pts: 0.365714 pos: 167160 size: 209
ret: 0 st: 1 flags:1 dts: 0.380930 pts: 0.380930 pos: 167496 size: 209
ret: 0 st:-1 flags:1 ts:-0.740831
ret: 0 st: 0 flags:1 dts: NOPTS pts: 0.000000 pos: 2144 size: 24801
ret: 0 st: 1 flags:1 dts:-0.010907 pts:-0.010907 pos: 27072 size: 208
ret:-1 st: 0 flags:0 ts: 2.153336
ret:-1 st: 0 flags:1 ts: 1.047503
ret: 0 st: 1 flags:0 ts:-0.058330
ret: 0 st: 0 flags:1 dts: NOPTS pts: 0.000000 pos: 2144 size: 24801
ret: 0 st: 1 flags:1 dts:-0.010907 pts:-0.010907 pos: 27072 size: 208
ret:-1 st: 1 flags:1 ts: 2.835837
ret:-1 st:-1 flags:0 ts: 1.730004
ret: 0 st:-1 flags:1 ts: 0.624171
ret: 0 st: 1 flags:1 dts: 0.600816 pts: 0.600816 pos: 266240 size: 209
ret: 0 st: 1 flags:1 dts: 0.616032 pts: 0.616032 pos: 266576 size: 209
ret: 0 st: 0 flags:0 ts:-0.481662
ret: 0 st: 0 flags:1 dts: NOPTS pts: 0.000000 pos: 2144 size: 24801
ret: 0 st: 1 flags:1 dts:-0.010907 pts:-0.010907 pos: 27072 size: 208
ret:-1 st: 0 flags:1 ts: 2.412505
ret:-1 st: 1 flags:0 ts: 1.306672
ret: 0 st: 1 flags:1 ts: 0.200839
ret: 0 st: 1 flags:1 dts: 0.208980 pts: 0.208980 pos: 113304 size: 209
ret: 0 st: 1 flags:1 dts: 0.224195 pts: 0.224195 pos: 113640 size: 209
ret: 0 st:-1 flags:0 ts:-0.904994
ret: 0 st: 0 flags:1 dts: NOPTS pts: 0.000000 pos: 2144 size: 24801
ret: 0 st: 1 flags:1 dts:-0.010907 pts:-0.010907 pos: 27072 size: 208
ret:-1 st:-1 flags:1 ts: 1.989173
ret: 0 st: 0 flags:0 ts: 0.883340
ret: 0 st: 1 flags:1 dts: 0.888163 pts: 0.888163 pos: 357608 size: 209
ret: 0 st: 1 flags:1 dts: 0.903379 pts: 0.903379 pos: 357944 size: 209
ret: 0 st: 0 flags:1 ts:-0.222493
ret: 0 st: 0 flags:1 dts: NOPTS pts: 0.000000 pos: 2144 size: 24801
ret: 0 st: 1 flags:1 dts:-0.010907 pts:-0.010907 pos: 27072 size: 208
ret:-1 st: 1 flags:0 ts: 2.671674
ret:-1 st: 1 flags:1 ts: 1.565841
ret: 0 st:-1 flags:0 ts: 0.460008
ret: 0 st: 1 flags:1 dts: 0.444082 pts: 0.444082 pos: 205440 size: 209
ret: 0 st: 1 flags:1 dts: 0.459297 pts: 0.459297 pos: 205776 size: 209
ret: 0 st:-1 flags:1 ts:-0.645825
ret: 0 st: 0 flags:1 dts: NOPTS pts: 0.000000 pos: 2144 size: 24801
ret: 0 st: 1 flags:1 dts:-0.010907 pts:-0.010907 pos: 27072 size: 208
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