mirror of
https://github.com/librempeg/librempeg
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Merge remote-tracking branch 'qatar/master'
* qatar/master: avcodec: add a cook parser to get subpacket duration FATE: allow lavf tests to alter input parameters FATE: replace the acodec-pcm_s24daud test with an enc_dec_pcm checksum test FATE: replace the acodec-g726 test with 4 new encode/decode tests FATE: replace current g722 encoding tests with an encode/decode test FATE: add a pattern rule for generating asynth wav files FATE: optionally write a WAVE header in audiogen avutil: add audio fifo buffer Conflicts: doc/APIchanges libavcodec/version.h libavutil/avutil.h tests/Makefile tests/codec-regression.sh tests/fate/voice.mak tests/lavf-regression.sh tests/ref/acodec/g722 tests/ref/acodec/g726 tests/ref/acodec/pcm_s24daud tests/ref/lavf/dv_fmt tests/ref/lavf/gxf tests/ref/lavf/mxf tests/ref/lavf/mxf_d10 tests/ref/seek/lavf_dv Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer
commit
3194ab78a6
@ -22,6 +22,18 @@ API changes, most recent first:
|
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2012-03-26 - a67d9cf - lavfi 2.66.100
|
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Add avfilter_fill_frame_from_{audio_,}buffer_ref() functions.
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|
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2012-xx-xx - xxxxxxx - lavu 51.28.0 - audio_fifo.h
|
||||
Add audio FIFO functions:
|
||||
av_audio_fifo_free()
|
||||
av_audio_fifo_alloc()
|
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av_audio_fifo_realloc()
|
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av_audio_fifo_write()
|
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av_audio_fifo_read()
|
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av_audio_fifo_drain()
|
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av_audio_fifo_reset()
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av_audio_fifo_size()
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av_audio_fifo_space()
|
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|
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2012-xx-xx - xxxxxxx - lavfi 2.16.0 - avfiltergraph.h
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Add avfilter_graph_parse2()
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@ -679,6 +679,7 @@ OBJS-$(CONFIG_AC3_PARSER) += ac3_parser.o ac3tab.o \
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aac_ac3_parser.o
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OBJS-$(CONFIG_ADX_PARSER) += adx_parser.o adx.o
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OBJS-$(CONFIG_CAVSVIDEO_PARSER) += cavs_parser.o
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OBJS-$(CONFIG_COOK_PARSER) += cook_parser.o
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OBJS-$(CONFIG_DCA_PARSER) += dca_parser.o
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OBJS-$(CONFIG_DIRAC_PARSER) += dirac_parser.o
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OBJS-$(CONFIG_DNXHD_PARSER) += dnxhd_parser.o
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@ -437,6 +437,7 @@ void avcodec_register_all(void)
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REGISTER_PARSER (AC3, ac3);
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REGISTER_PARSER (ADX, adx);
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REGISTER_PARSER (CAVSVIDEO, cavsvideo);
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REGISTER_PARSER (COOK, cook);
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REGISTER_PARSER (DCA, dca);
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REGISTER_PARSER (DIRAC, dirac);
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REGISTER_PARSER (DNXHD, dnxhd);
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59
libavcodec/cook_parser.c
Normal file
59
libavcodec/cook_parser.c
Normal file
@ -0,0 +1,59 @@
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/*
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* Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
|
||||
*
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* This file is part of Libav.
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*
|
||||
* 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
|
||||
*/
|
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|
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/**
|
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* @file
|
||||
* Cook audio parser
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*
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* Determines subpacket duration from extradata.
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*/
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#include <stdint.h>
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#include "libavutil/intreadwrite.h"
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#include "parser.h"
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typedef struct CookParseContext {
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int duration;
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} CookParseContext;
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static int cook_parse(AVCodecParserContext *s1, AVCodecContext *avctx,
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const uint8_t **poutbuf, int *poutbuf_size,
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const uint8_t *buf, int buf_size)
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{
|
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CookParseContext *s = s1->priv_data;
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||||
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if (s->duration)
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s1->duration = s->duration;
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else if (avctx->extradata && avctx->extradata_size >= 8 && avctx->channels)
|
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s->duration = AV_RB16(avctx->extradata + 4) / avctx->channels;
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||||
|
||||
/* always return the full packet. this parser isn't doing any splitting or
|
||||
combining, only setting packet duration */
|
||||
*poutbuf = buf;
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||||
*poutbuf_size = buf_size;
|
||||
return buf_size;
|
||||
}
|
||||
|
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AVCodecParser ff_cook_parser = {
|
||||
.codec_ids = { CODEC_ID_COOK },
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.priv_data_size = sizeof(CookParseContext),
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.parser_parse = cook_parse,
|
||||
};
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@ -27,8 +27,8 @@
|
||||
*/
|
||||
|
||||
#define LIBAVCODEC_VERSION_MAJOR 54
|
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#define LIBAVCODEC_VERSION_MINOR 14
|
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#define LIBAVCODEC_VERSION_MICRO 101
|
||||
#define LIBAVCODEC_VERSION_MINOR 15
|
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#define LIBAVCODEC_VERSION_MICRO 100
|
||||
|
||||
#define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
|
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LIBAVCODEC_VERSION_MINOR, \
|
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|
||||
@ -75,7 +75,7 @@ AVInputFormat ff_daud_demuxer = {
|
||||
.long_name = NULL_IF_CONFIG_SMALL("D-Cinema audio format"),
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.read_header = daud_header,
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||||
.read_packet = daud_packet,
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||||
.extensions = "302",
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||||
.extensions = "302,daud",
|
||||
};
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||||
#endif
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||||
|
||||
|
||||
@ -205,6 +205,7 @@ static int rm_read_audio_stream_info(AVFormatContext *s, AVIOContext *pb,
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st->codec->block_align = coded_framesize;
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break;
|
||||
case CODEC_ID_COOK:
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st->need_parsing = AVSTREAM_PARSE_HEADERS;
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case CODEC_ID_ATRAC3:
|
||||
case CODEC_ID_SIPR:
|
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avio_rb16(pb); avio_r8(pb);
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|
||||
@ -5,6 +5,7 @@ NAME = avutil
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HEADERS = adler32.h \
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aes.h \
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||||
attributes.h \
|
||||
audio_fifo.h \
|
||||
audioconvert.h \
|
||||
avassert.h \
|
||||
avstring.h \
|
||||
@ -45,6 +46,7 @@ BUILT_HEADERS = avconfig.h
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||||
OBJS = adler32.o \
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||||
aes.o \
|
||||
audio_fifo.o \
|
||||
audioconvert.o \
|
||||
avstring.o \
|
||||
base64.o \
|
||||
|
||||
193
libavutil/audio_fifo.c
Normal file
193
libavutil/audio_fifo.c
Normal file
@ -0,0 +1,193 @@
|
||||
/*
|
||||
* Audio FIFO
|
||||
* Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
|
||||
*
|
||||
* 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
|
||||
*/
|
||||
|
||||
/**
|
||||
* @file
|
||||
* Audio FIFO
|
||||
*/
|
||||
|
||||
#include "avutil.h"
|
||||
#include "audio_fifo.h"
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#include "fifo.h"
|
||||
#include "mem.h"
|
||||
#include "samplefmt.h"
|
||||
|
||||
struct AVAudioFifo {
|
||||
AVFifoBuffer **buf; /**< single buffer for interleaved, per-channel buffers for planar */
|
||||
int nb_buffers; /**< number of buffers */
|
||||
int nb_samples; /**< number of samples currently in the FIFO */
|
||||
int allocated_samples; /**< current allocated size, in samples */
|
||||
|
||||
int channels; /**< number of channels */
|
||||
enum AVSampleFormat sample_fmt; /**< sample format */
|
||||
int sample_size; /**< size, in bytes, of one sample in a buffer */
|
||||
};
|
||||
|
||||
void av_audio_fifo_free(AVAudioFifo *af)
|
||||
{
|
||||
if (af) {
|
||||
if (af->buf) {
|
||||
int i;
|
||||
for (i = 0; i < af->nb_buffers; i++) {
|
||||
if (af->buf[i])
|
||||
av_fifo_free(af->buf[i]);
|
||||
}
|
||||
av_free(af->buf);
|
||||
}
|
||||
av_free(af);
|
||||
}
|
||||
}
|
||||
|
||||
AVAudioFifo *av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels,
|
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int nb_samples)
|
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{
|
||||
AVAudioFifo *af;
|
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int buf_size, i;
|
||||
|
||||
/* get channel buffer size (also validates parameters) */
|
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if (av_samples_get_buffer_size(&buf_size, channels, nb_samples, sample_fmt, 1) < 0)
|
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return NULL;
|
||||
|
||||
af = av_mallocz(sizeof(*af));
|
||||
if (!af)
|
||||
return NULL;
|
||||
|
||||
af->channels = channels;
|
||||
af->sample_fmt = sample_fmt;
|
||||
af->sample_size = buf_size / nb_samples;
|
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af->nb_buffers = av_sample_fmt_is_planar(sample_fmt) ? channels : 1;
|
||||
|
||||
af->buf = av_mallocz(af->nb_buffers * sizeof(*af->buf));
|
||||
if (!af->buf)
|
||||
goto error;
|
||||
|
||||
for (i = 0; i < af->nb_buffers; i++) {
|
||||
af->buf[i] = av_fifo_alloc(buf_size);
|
||||
if (!af->buf[i])
|
||||
goto error;
|
||||
}
|
||||
af->allocated_samples = nb_samples;
|
||||
|
||||
return af;
|
||||
|
||||
error:
|
||||
av_audio_fifo_free(af);
|
||||
return NULL;
|
||||
}
|
||||
|
||||
int av_audio_fifo_realloc(AVAudioFifo *af, int nb_samples)
|
||||
{
|
||||
int i, ret, buf_size;
|
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|
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if ((ret = av_samples_get_buffer_size(&buf_size, af->channels, nb_samples,
|
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af->sample_fmt, 1)) < 0)
|
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return ret;
|
||||
|
||||
for (i = 0; i < af->nb_buffers; i++) {
|
||||
if ((ret = av_fifo_realloc2(af->buf[i], buf_size)) < 0)
|
||||
return ret;
|
||||
}
|
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af->allocated_samples = nb_samples;
|
||||
return 0;
|
||||
}
|
||||
|
||||
int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples)
|
||||
{
|
||||
int i, ret, size;
|
||||
|
||||
/* automatically reallocate buffers if needed */
|
||||
if (av_audio_fifo_space(af) < nb_samples) {
|
||||
int current_size = av_audio_fifo_size(af);
|
||||
/* check for integer overflow in new size calculation */
|
||||
if (INT_MAX / 2 - current_size < nb_samples)
|
||||
return AVERROR(EINVAL);
|
||||
/* reallocate buffers */
|
||||
if ((ret = av_audio_fifo_realloc(af, 2 * (current_size + nb_samples))) < 0)
|
||||
return ret;
|
||||
}
|
||||
|
||||
size = nb_samples * af->sample_size;
|
||||
for (i = 0; i < af->nb_buffers; i++) {
|
||||
ret = av_fifo_generic_write(af->buf[i], data[i], size, NULL);
|
||||
if (ret != size)
|
||||
return AVERROR_BUG;
|
||||
}
|
||||
af->nb_samples += nb_samples;
|
||||
|
||||
return nb_samples;
|
||||
}
|
||||
|
||||
int av_audio_fifo_read(AVAudioFifo *af, void **data, int nb_samples)
|
||||
{
|
||||
int i, ret, size;
|
||||
|
||||
if (nb_samples < 0)
|
||||
return AVERROR(EINVAL);
|
||||
nb_samples = FFMIN(nb_samples, af->nb_samples);
|
||||
if (!nb_samples)
|
||||
return 0;
|
||||
|
||||
size = nb_samples * af->sample_size;
|
||||
for (i = 0; i < af->nb_buffers; i++) {
|
||||
if ((ret = av_fifo_generic_read(af->buf[i], data[i], size, NULL)) < 0)
|
||||
return AVERROR_BUG;
|
||||
}
|
||||
af->nb_samples -= nb_samples;
|
||||
|
||||
return nb_samples;
|
||||
}
|
||||
|
||||
int av_audio_fifo_drain(AVAudioFifo *af, int nb_samples)
|
||||
{
|
||||
int i, size;
|
||||
|
||||
if (nb_samples < 0)
|
||||
return AVERROR(EINVAL);
|
||||
nb_samples = FFMIN(nb_samples, af->nb_samples);
|
||||
|
||||
if (nb_samples) {
|
||||
size = nb_samples * af->sample_size;
|
||||
for (i = 0; i < af->nb_buffers; i++)
|
||||
av_fifo_drain(af->buf[i], size);
|
||||
af->nb_samples -= nb_samples;
|
||||
}
|
||||
return 0;
|
||||
}
|
||||
|
||||
void av_audio_fifo_reset(AVAudioFifo *af)
|
||||
{
|
||||
int i;
|
||||
|
||||
for (i = 0; i < af->nb_buffers; i++)
|
||||
av_fifo_reset(af->buf[i]);
|
||||
|
||||
af->nb_samples = 0;
|
||||
}
|
||||
|
||||
int av_audio_fifo_size(AVAudioFifo *af)
|
||||
{
|
||||
return af->nb_samples;
|
||||
}
|
||||
|
||||
int av_audio_fifo_space(AVAudioFifo *af)
|
||||
{
|
||||
return af->allocated_samples - af->nb_samples;
|
||||
}
|
||||
146
libavutil/audio_fifo.h
Normal file
146
libavutil/audio_fifo.h
Normal file
@ -0,0 +1,146 @@
|
||||
/*
|
||||
* Audio FIFO
|
||||
* Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
|
||||
*
|
||||
* 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
|
||||
*/
|
||||
|
||||
/**
|
||||
* @file
|
||||
* Audio FIFO Buffer
|
||||
*/
|
||||
|
||||
#ifndef AVUTIL_AUDIO_FIFO_H
|
||||
#define AVUTIL_AUDIO_FIFO_H
|
||||
|
||||
#include "avutil.h"
|
||||
#include "fifo.h"
|
||||
#include "samplefmt.h"
|
||||
|
||||
/**
|
||||
* @addtogroup lavu_audio
|
||||
* @{
|
||||
*/
|
||||
|
||||
/**
|
||||
* Context for an Audio FIFO Buffer.
|
||||
*
|
||||
* - Operates at the sample level rather than the byte level.
|
||||
* - Supports multiple channels with either planar or packed sample format.
|
||||
* - Automatic reallocation when writing to a full buffer.
|
||||
*/
|
||||
typedef struct AVAudioFifo AVAudioFifo;
|
||||
|
||||
/**
|
||||
* Free an AVAudioFifo.
|
||||
*
|
||||
* @param af AVAudioFifo to free
|
||||
*/
|
||||
void av_audio_fifo_free(AVAudioFifo *af);
|
||||
|
||||
/**
|
||||
* Allocate an AVAudioFifo.
|
||||
*
|
||||
* @param sample_fmt sample format
|
||||
* @param channels number of channels
|
||||
* @param nb_samples initial allocation size, in samples
|
||||
* @return newly allocated AVAudioFifo, or NULL on error
|
||||
*/
|
||||
AVAudioFifo *av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels,
|
||||
int nb_samples);
|
||||
|
||||
/**
|
||||
* Reallocate an AVAudioFifo.
|
||||
*
|
||||
* @param af AVAudioFifo to reallocate
|
||||
* @param nb_samples new allocation size, in samples
|
||||
* @return 0 if OK, or negative AVERROR code on failure
|
||||
*/
|
||||
int av_audio_fifo_realloc(AVAudioFifo *af, int nb_samples);
|
||||
|
||||
/**
|
||||
* Write data to an AVAudioFifo.
|
||||
*
|
||||
* The AVAudioFifo will be reallocated automatically if the available space
|
||||
* is less than nb_samples.
|
||||
*
|
||||
* @see enum AVSampleFormat
|
||||
* The documentation for AVSampleFormat describes the data layout.
|
||||
*
|
||||
* @param af AVAudioFifo to write to
|
||||
* @param data audio data plane pointers
|
||||
* @param nb_samples number of samples to write
|
||||
* @return number of samples actually written, or negative AVERROR
|
||||
* code on failure.
|
||||
*/
|
||||
int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples);
|
||||
|
||||
/**
|
||||
* Read data from an AVAudioFifo.
|
||||
*
|
||||
* @see enum AVSampleFormat
|
||||
* The documentation for AVSampleFormat describes the data layout.
|
||||
*
|
||||
* @param af AVAudioFifo to read from
|
||||
* @param data audio data plane pointers
|
||||
* @param nb_samples number of samples to read
|
||||
* @return number of samples actually read, or negative AVERROR code
|
||||
* on failure.
|
||||
*/
|
||||
int av_audio_fifo_read(AVAudioFifo *af, void **data, int nb_samples);
|
||||
|
||||
/**
|
||||
* Drain data from an AVAudioFifo.
|
||||
*
|
||||
* Removes the data without reading it.
|
||||
*
|
||||
* @param af AVAudioFifo to drain
|
||||
* @param nb_samples number of samples to drain
|
||||
* @return 0 if OK, or negative AVERROR code on failure
|
||||
*/
|
||||
int av_audio_fifo_drain(AVAudioFifo *af, int nb_samples);
|
||||
|
||||
/**
|
||||
* Reset the AVAudioFifo buffer.
|
||||
*
|
||||
* This empties all data in the buffer.
|
||||
*
|
||||
* @param af AVAudioFifo to reset
|
||||
*/
|
||||
void av_audio_fifo_reset(AVAudioFifo *af);
|
||||
|
||||
/**
|
||||
* Get the current number of samples in the AVAudioFifo available for reading.
|
||||
*
|
||||
* @param af the AVAudioFifo to query
|
||||
* @return number of samples available for reading
|
||||
*/
|
||||
int av_audio_fifo_size(AVAudioFifo *af);
|
||||
|
||||
/**
|
||||
* Get the current number of samples in the AVAudioFifo available for writing.
|
||||
*
|
||||
* @param af the AVAudioFifo to query
|
||||
* @return number of samples available for writing
|
||||
*/
|
||||
int av_audio_fifo_space(AVAudioFifo *af);
|
||||
|
||||
/**
|
||||
* @}
|
||||
*/
|
||||
|
||||
#endif /* AVUTIL_AUDIO_FIFO_H */
|
||||
@ -153,7 +153,7 @@
|
||||
*/
|
||||
|
||||
#define LIBAVUTIL_VERSION_MAJOR 51
|
||||
#define LIBAVUTIL_VERSION_MINOR 46
|
||||
#define LIBAVUTIL_VERSION_MINOR 47
|
||||
#define LIBAVUTIL_VERSION_MICRO 100
|
||||
|
||||
#define LIBAVUTIL_VERSION_INT AV_VERSION_INT(LIBAVUTIL_VERSION_MAJOR, \
|
||||
|
||||
@ -29,6 +29,9 @@ tests/data/asynth1.sw: tests/audiogen$(HOSTEXESUF) | tests/data
|
||||
tests/data/asynth-16000-1.sw: tests/audiogen$(HOSTEXESUF) | tests/data
|
||||
$(M)./$< $@ 16000 1
|
||||
|
||||
tests/data/asynth-%.wav: tests/audiogen$(HOSTEXESUF) | tests/data
|
||||
$(M)./$< $@ $(subst -, ,$*)
|
||||
|
||||
tests/data/mapchan-6ch.sw: tests/audiogen$(HOSTEXESUF)
|
||||
@mkdir -p tests/data
|
||||
$(M)./$< $@ 22050 6
|
||||
@ -37,7 +40,7 @@ tests/data/mapchan-mono.sw: tests/audiogen$(HOSTEXESUF)
|
||||
@mkdir -p tests/data
|
||||
$(M)./$< $@ 22050 1
|
||||
|
||||
tests/data/%.sw tests/vsynth%/00.pgm: TAG = GEN
|
||||
tests/data/%.sw tests/data/asynth% tests/vsynth%/00.pgm: TAG = GEN
|
||||
|
||||
include $(SRC_PATH)/tests/fate/aac.mak
|
||||
include $(SRC_PATH)/tests/fate/ac3.mak
|
||||
|
||||
@ -22,7 +22,9 @@
|
||||
*/
|
||||
|
||||
#include <stdlib.h>
|
||||
#include <stdint.h>
|
||||
#include <stdio.h>
|
||||
#include <string.h>
|
||||
|
||||
#define MAX_CHANNELS 8
|
||||
|
||||
@ -93,12 +95,45 @@ static int int_cos(int a)
|
||||
|
||||
FILE *outfile;
|
||||
|
||||
static void put_sample(int v)
|
||||
static void put16(int16_t v)
|
||||
{
|
||||
fputc(v & 0xff, outfile);
|
||||
fputc( v & 0xff, outfile);
|
||||
fputc((v >> 8) & 0xff, outfile);
|
||||
}
|
||||
|
||||
static void put32(uint32_t v)
|
||||
{
|
||||
fputc( v & 0xff, outfile);
|
||||
fputc((v >> 8) & 0xff, outfile);
|
||||
fputc((v >> 16) & 0xff, outfile);
|
||||
fputc((v >> 24) & 0xff, outfile);
|
||||
}
|
||||
|
||||
#define HEADER_SIZE 46
|
||||
#define FMT_SIZE 18
|
||||
#define SAMPLE_SIZE 2
|
||||
#define WFORMAT_PCM 0x0001
|
||||
|
||||
static void put_wav_header(int sample_rate, int channels, int nb_samples)
|
||||
{
|
||||
int block_align = SAMPLE_SIZE * channels;
|
||||
int data_size = block_align * nb_samples;
|
||||
|
||||
fputs("RIFF", outfile);
|
||||
put32(HEADER_SIZE + data_size);
|
||||
fputs("WAVEfmt ", outfile);
|
||||
put32(FMT_SIZE);
|
||||
put16(WFORMAT_PCM);
|
||||
put16(channels);
|
||||
put32(sample_rate);
|
||||
put32(block_align * sample_rate);
|
||||
put16(block_align);
|
||||
put16(SAMPLE_SIZE * 8);
|
||||
put16(0);
|
||||
fputs("data", outfile);
|
||||
put32(data_size);
|
||||
}
|
||||
|
||||
int main(int argc, char **argv)
|
||||
{
|
||||
int i, a, v, j, f, amp, ampa;
|
||||
@ -107,10 +142,12 @@ int main(int argc, char **argv)
|
||||
int taba[MAX_CHANNELS];
|
||||
int sample_rate = 44100;
|
||||
int nb_channels = 2;
|
||||
char *ext;
|
||||
|
||||
if (argc < 2 || argc > 4) {
|
||||
printf("usage: %s file [<sample rate> [<channels>]]\n"
|
||||
"generate a test raw 16 bit audio stream\n"
|
||||
"If the file extension is .wav a WAVE header will be added.\n"
|
||||
"default: 44100 Hz stereo\n", argv[0]);
|
||||
exit(1);
|
||||
}
|
||||
@ -137,12 +174,15 @@ int main(int argc, char **argv)
|
||||
return 1;
|
||||
}
|
||||
|
||||
if ((ext = strrchr(argv[1], '.')) != NULL && !strcmp(ext, ".wav"))
|
||||
put_wav_header(sample_rate, nb_channels, 6 * sample_rate);
|
||||
|
||||
/* 1 second of single freq sinus at 1000 Hz */
|
||||
a = 0;
|
||||
for (i = 0; i < 1 * sample_rate; i++) {
|
||||
v = (int_cos(a) * 10000) >> FRAC_BITS;
|
||||
for (j = 0; j < nb_channels; j++)
|
||||
put_sample(v);
|
||||
put16(v);
|
||||
a += (1000 * FRAC_ONE) / sample_rate;
|
||||
}
|
||||
|
||||
@ -151,7 +191,7 @@ int main(int argc, char **argv)
|
||||
for (i = 0; i < 1 * sample_rate; i++) {
|
||||
v = (int_cos(a) * 10000) >> FRAC_BITS;
|
||||
for (j = 0; j < nb_channels; j++)
|
||||
put_sample(v);
|
||||
put16(v);
|
||||
f = 100 + (((10000 - 100) * i) / sample_rate);
|
||||
a += (f * FRAC_ONE) / sample_rate;
|
||||
}
|
||||
@ -160,14 +200,14 @@ int main(int argc, char **argv)
|
||||
for (i = 0; i < sample_rate / 2; i++) {
|
||||
v = myrnd(&seed, 20000) - 10000;
|
||||
for (j = 0; j < nb_channels; j++)
|
||||
put_sample(v);
|
||||
put16(v);
|
||||
}
|
||||
|
||||
/* 0.5 second of high amplitude white noise */
|
||||
for (i = 0; i < sample_rate / 2; i++) {
|
||||
v = myrnd(&seed, 65535) - 32768;
|
||||
for (j = 0; j < nb_channels; j++)
|
||||
put_sample(v);
|
||||
put16(v);
|
||||
}
|
||||
|
||||
/* 1 second of unrelated ramps for each channel */
|
||||
@ -179,7 +219,7 @@ int main(int argc, char **argv)
|
||||
for (i = 0; i < 1 * sample_rate; i++) {
|
||||
for (j = 0; j < nb_channels; j++) {
|
||||
v = (int_cos(taba[j]) * 10000) >> FRAC_BITS;
|
||||
put_sample(v);
|
||||
put16(v);
|
||||
f = tabf1[j] + (((tabf2[j] - tabf1[j]) * i) / sample_rate);
|
||||
taba[j] += (f * FRAC_ONE) / sample_rate;
|
||||
}
|
||||
@ -194,7 +234,7 @@ int main(int argc, char **argv)
|
||||
if (j & 1)
|
||||
amp = 10000 - amp;
|
||||
v = (int_cos(a) * amp) >> FRAC_BITS;
|
||||
put_sample(v);
|
||||
put16(v);
|
||||
a += (500 * FRAC_ONE) / sample_rate;
|
||||
ampa += (2 * FRAC_ONE) / sample_rate;
|
||||
}
|
||||
|
||||
@ -395,16 +395,6 @@ do_audio_encoding g723_1.tco "-b:a 6.3k -ac 1 -ar 8000 -acodec g723_1"
|
||||
do_audio_decoding
|
||||
fi
|
||||
|
||||
if [ -n "$do_g722" ] ; then
|
||||
do_audio_encoding g722.wav "-b 64k -ac 1 -ar 16000 -acodec g722"
|
||||
do_audio_decoding
|
||||
fi
|
||||
|
||||
if [ -n "$do_g726" ] ; then
|
||||
do_audio_encoding g726.wav "-b:a 32k -ac 1 -ar 8000 -acodec g726"
|
||||
do_audio_decoding
|
||||
fi
|
||||
|
||||
if [ -n "$do_adpcm_adx" ] ; then
|
||||
do_audio_encoding adpcm_adx.adx "-acodec adpcm_adx"
|
||||
do_audio_decoding
|
||||
@ -534,6 +524,3 @@ fi
|
||||
if [ -n "$do_pcm_f64le" ] ; then
|
||||
do_audio_enc_dec wav dbl pcm_f64le
|
||||
fi
|
||||
if [ -n "$do_pcm_s24daud" ] ; then
|
||||
do_audio_enc_dec 302 s16 pcm_s24daud "-ac 6 -ar 96000"
|
||||
fi
|
||||
|
||||
@ -25,5 +25,10 @@ fate-pcm_u8-stereo: CMD = md5 -i $(SAMPLES)/qt-surge-suite/surge-2-8-raw.mov -f
|
||||
FATE_PCM += fate-w64
|
||||
fate-w64: CMD = crc -i $(SAMPLES)/w64/w64-pcm16.w64
|
||||
|
||||
FATE_PCM += fate-dcinema-encode
|
||||
fate-dcinema-encode: tests/data/asynth-96000-6.wav
|
||||
fate-dcinema-encode: SRC = tests/data/asynth-96000-6.wav
|
||||
fate-dcinema-encode: CMD = enc_dec_pcm daud md5 s16le $(SRC) -c:a pcm_s24daud
|
||||
|
||||
FATE_TESTS += $(FATE_PCM)
|
||||
fate-pcm: $(FATE_PCM)
|
||||
|
||||
@ -1,9 +1,36 @@
|
||||
FATE_VOICE += fate-g722dec-1
|
||||
FATE_G722 += fate-g722dec-1
|
||||
fate-g722dec-1: CMD = framecrc -i $(SAMPLES)/g722/conf-adminmenu-162.g722
|
||||
|
||||
FATE_VOICE += fate-g722enc
|
||||
fate-g722enc: tests/data/asynth-16000-1.sw
|
||||
fate-g722enc: CMD = md5 -ar 16000 -ac 1 -f s16le -i $(TARGET_PATH)/tests/data/asynth-16000-1.sw -acodec g722 -ac 1 -f g722
|
||||
FATE_G722 += fate-g722-encode
|
||||
fate-g722-encode: tests/data/asynth-16000-1.wav
|
||||
fate-g722-encode: SRC = tests/data/asynth-16000-1.wav
|
||||
fate-g722-encode: CMD = enc_dec_pcm wav md5 s16le $(SRC) -c:a g722
|
||||
|
||||
FATE_VOICE += $(FATE_G722)
|
||||
fate-g722: $(FATE_G722)
|
||||
|
||||
FATE_G726 += fate-g726-encode-2bit
|
||||
fate-g726-encode-2bit: tests/data/asynth-8000-1.wav
|
||||
fate-g726-encode-2bit: SRC = tests/data/asynth-8000-1.wav
|
||||
fate-g726-encode-2bit: CMD = enc_dec_pcm wav md5 s16le $(SRC) -c:a g726 -b:a 16k
|
||||
|
||||
FATE_G726 += fate-g726-encode-3bit
|
||||
fate-g726-encode-3bit: tests/data/asynth-8000-1.wav
|
||||
fate-g726-encode-3bit: SRC = tests/data/asynth-8000-1.wav
|
||||
fate-g726-encode-3bit: CMD = enc_dec_pcm wav md5 s16le $(SRC) -c:a g726 -b:a 24k
|
||||
|
||||
FATE_G726 += fate-g726-encode-4bit
|
||||
fate-g726-encode-4bit: tests/data/asynth-8000-1.wav
|
||||
fate-g726-encode-4bit: SRC = tests/data/asynth-8000-1.wav
|
||||
fate-g726-encode-4bit: CMD = enc_dec_pcm wav md5 s16le $(SRC) -c:a g726 -b:a 32k
|
||||
|
||||
FATE_G726 += fate-g726-encode-5bit
|
||||
fate-g726-encode-5bit: tests/data/asynth-8000-1.wav
|
||||
fate-g726-encode-5bit: SRC = tests/data/asynth-8000-1.wav
|
||||
fate-g726-encode-5bit: CMD = enc_dec_pcm wav md5 s16le $(SRC) -c:a g726 -b:a 40k
|
||||
|
||||
FATE_VOICE += $(FATE_G726)
|
||||
fate-g726: $(FATE_G726)
|
||||
|
||||
FATE_GSM += fate-gsm-ms
|
||||
fate-gsm-ms: CMD = framecrc -i $(SAMPLES)/gsm/ciao.wav
|
||||
|
||||
@ -24,18 +24,18 @@ do_lavf_fate()
|
||||
do_lavf()
|
||||
{
|
||||
file=${outfile}lavf.$1
|
||||
do_avconv $file $DEC_OPTS -f image2 -vcodec pgmyuv -i $raw_src $DEC_OPTS -ar 44100 -f s16le -i $pcm_src $ENC_OPTS -b:a 64k -t 1 -qscale:v 10 $2
|
||||
do_avconv_crc $file $DEC_OPTS -i $target_path/$file $3
|
||||
do_avconv $file $DEC_OPTS -f image2 -vcodec pgmyuv -i $raw_src $DEC_OPTS -ar 44100 -f s16le $2 -i $pcm_src $ENC_OPTS -b:a 64k -t 1 -qscale:v 10 $3
|
||||
do_avconv_crc $file $DEC_OPTS -i $target_path/$file $4
|
||||
}
|
||||
|
||||
do_lavf_timecode_nodrop() { do_lavf $1 "$2 -timecode 02:56:14:13"; }
|
||||
do_lavf_timecode_drop() { do_lavf $1 "$2 -timecode 02:56:14.13 -r 30000/1001"; }
|
||||
do_lavf_timecode_nodrop() { do_lavf $1 "" "$2 -timecode 02:56:14:13"; }
|
||||
do_lavf_timecode_drop() { do_lavf $1 "" "$2 -timecode 02:56:14.13 -r 30000/1001"; }
|
||||
|
||||
do_lavf_timecode()
|
||||
{
|
||||
do_lavf_timecode_nodrop "$@"
|
||||
do_lavf_timecode_drop "$@"
|
||||
do_lavf "$@"
|
||||
do_lavf $1 "" "$2"
|
||||
}
|
||||
|
||||
do_streamed_images()
|
||||
@ -64,11 +64,11 @@ do_audio_only()
|
||||
}
|
||||
|
||||
if [ -n "$do_avi" ] ; then
|
||||
do_lavf avi "-acodec mp2 -ab 64k"
|
||||
do_lavf avi "" "-acodec mp2 -ab 64k"
|
||||
fi
|
||||
|
||||
if [ -n "$do_asf" ] ; then
|
||||
do_lavf asf "-acodec mp2 -ab 64k" "-r 25"
|
||||
do_lavf asf "" "-acodec mp2 -ab 64k" "-r 25"
|
||||
fi
|
||||
|
||||
if [ -n "$do_rm" ] ; then
|
||||
@ -87,15 +87,15 @@ do_lavf_timecode mxf "-ar 48000 -bf 2"
|
||||
fi
|
||||
|
||||
if [ -n "$do_mxf_d10" ]; then
|
||||
do_lavf mxf_d10 "-ar 48000 -ac 2 -r 25 -s 720x576 -vf pad=720:608:0:32 -vcodec mpeg2video -g 0 -flags +ildct+low_delay -dc 10 -non_linear_quant 1 -intra_vlc 1 -qscale 1 -ps 1 -qmin 1 -rc_max_vbv_use 1 -rc_min_vbv_use 1 -pix_fmt yuv422p -minrate 30000k -maxrate 30000k -b 30000k -bufsize 1200000 -top 1 -rc_init_occupancy 1200000 -qmax 12 -f mxf_d10"
|
||||
do_lavf mxf_d10 "-ar 48000 -ac 2" "-r 25 -s 720x576 -vf pad=720:608:0:32 -vcodec mpeg2video -g 0 -flags +ildct+low_delay -dc 10 -non_linear_quant 1 -intra_vlc 1 -qscale 1 -ps 1 -qmin 1 -rc_max_vbv_use 1 -rc_min_vbv_use 1 -pix_fmt yuv422p -minrate 30000k -maxrate 30000k -b 30000k -bufsize 1200000 -top 1 -rc_init_occupancy 1200000 -qmax 12 -f mxf_d10"
|
||||
fi
|
||||
|
||||
if [ -n "$do_ts" ] ; then
|
||||
do_lavf ts "-ab 64k -mpegts_transport_stream_id 42"
|
||||
do_lavf ts "" "-ab 64k -mpegts_transport_stream_id 42"
|
||||
fi
|
||||
|
||||
if [ -n "$do_swf" ] ; then
|
||||
do_lavf swf -an
|
||||
do_lavf swf "" "-an"
|
||||
fi
|
||||
|
||||
if [ -n "$do_ffm" ] ; then
|
||||
@ -103,11 +103,11 @@ do_lavf ffm "-ab 64k"
|
||||
fi
|
||||
|
||||
if [ -n "$do_flv_fmt" ] ; then
|
||||
do_lavf flv -an
|
||||
do_lavf flv "" "-an"
|
||||
fi
|
||||
|
||||
if [ -n "$do_mov" ] ; then
|
||||
do_lavf mov "-movflags +rtphint -acodec pcm_alaw -vcodec mpeg4"
|
||||
do_lavf mov "" "-movflags +rtphint -acodec pcm_alaw -vcodec mpeg4"
|
||||
do_lavf_timecode mov "-acodec pcm_alaw -vcodec mpeg4"
|
||||
fi
|
||||
|
||||
@ -118,21 +118,21 @@ fi
|
||||
if [ -n "$do_dv_fmt" ] ; then
|
||||
do_lavf_timecode_nodrop dv "-ar 48000 -r 25 -s pal -ac 2"
|
||||
do_lavf_timecode_drop dv "-ar 48000 -pix_fmt yuv411p -s ntsc -ac 2"
|
||||
do_lavf dv "-ar 48000 -r 25 -s pal -ac 2"
|
||||
do_lavf dv "-ar 48000" "-r 25 -s pal -ac 2"
|
||||
fi
|
||||
|
||||
if [ -n "$do_gxf" ] ; then
|
||||
do_lavf_timecode_nodrop gxf "-ar 48000 -r 25 -s pal -ac 1"
|
||||
do_lavf_timecode_drop gxf "-ar 48000 -s ntsc -ac 1"
|
||||
do_lavf gxf "-ar 48000 -r 25 -s pal -ac 1"
|
||||
do_lavf gxf "-ar 48000" "-r 25 -s pal -ac 1"
|
||||
fi
|
||||
|
||||
if [ -n "$do_nut" ] ; then
|
||||
do_lavf nut "-acodec mp2 -ab 64k"
|
||||
do_lavf nut "" "-acodec mp2 -ab 64k"
|
||||
fi
|
||||
|
||||
if [ -n "$do_mkv" ] ; then
|
||||
do_lavf mkv "-acodec mp2 -ab 64k -vcodec mpeg4"
|
||||
do_lavf mkv "" "-acodec mp2 -ab 64k -vcodec mpeg4"
|
||||
fi
|
||||
|
||||
if [ -n "$do_mp3" ] ; then
|
||||
@ -150,7 +150,7 @@ do_lavf_fate ogg "vp3/coeff_level64.mkv"
|
||||
fi
|
||||
|
||||
if [ -n "$do_wtv" ] ; then
|
||||
do_lavf wtv "-acodec mp2"
|
||||
do_lavf wtv "" "-acodec mp2"
|
||||
fi
|
||||
|
||||
|
||||
|
||||
@ -1,4 +0,0 @@
|
||||
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
|
||||
@ -1,4 +0,0 @@
|
||||
331fcf91f4483b508059d0933af97987 *./tests/data/acodec/g726.wav
|
||||
24054 ./tests/data/acodec/g726.wav
|
||||
fac563ba7947d8fc42b4af048707c145 *./tests/data/g726.acodec.out.wav
|
||||
stddev: 8553.69 PSNR: 17.69 MAXDIFF:29353 bytes: 95984/ 1058400
|
||||
@ -1,4 +0,0 @@
|
||||
1b75d5198ae789ab3c48f7024e08f4a9 *./tests/data/acodec/pcm_s24daud.302
|
||||
10368730 ./tests/data/acodec/pcm_s24daud.302
|
||||
70ec0ba6bc151ddc7509c09804d95d3b *./tests/data/pcm_s24daud.acodec.out.wav
|
||||
stddev: 8967.92 PSNR: 17.28 MAXDIFF:42548 bytes: 6911796/ 1058400
|
||||
1
tests/ref/fate/dcinema-encode
Normal file
1
tests/ref/fate/dcinema-encode
Normal file
@ -0,0 +1 @@
|
||||
MD5=2d7c6897c315493647db159f4bfd6edc
|
||||
1
tests/ref/fate/g722-encode
Normal file
1
tests/ref/fate/g722-encode
Normal file
@ -0,0 +1 @@
|
||||
MD5=7106189574186051c0497b287e2e5f19
|
||||
@ -1 +0,0 @@
|
||||
94e2f200d6e05b47cec4aa3e94571cf3
|
||||
1
tests/ref/fate/g726-encode-2bit
Normal file
1
tests/ref/fate/g726-encode-2bit
Normal file
@ -0,0 +1 @@
|
||||
MD5=215eaef5778a16e2bf4f3725a557f355
|
||||
1
tests/ref/fate/g726-encode-3bit
Normal file
1
tests/ref/fate/g726-encode-3bit
Normal file
@ -0,0 +1 @@
|
||||
MD5=0bebd949dfd5ac0ae3f2c3ceb2e3fac1
|
||||
1
tests/ref/fate/g726-encode-4bit
Normal file
1
tests/ref/fate/g726-encode-4bit
Normal file
@ -0,0 +1 @@
|
||||
MD5=a21cfea116ab2179eabe5d84b6bfc09a
|
||||
1
tests/ref/fate/g726-encode-5bit
Normal file
1
tests/ref/fate/g726-encode-5bit
Normal file
@ -0,0 +1 @@
|
||||
MD5=9cad98cf5205bf76d6e9d1241e56141a
|
||||
@ -4,6 +4,6 @@
|
||||
cc33ae4f9e6828914dea0f09d1241b7e *./tests/data/lavf/lavf.dv
|
||||
3480000 ./tests/data/lavf/lavf.dv
|
||||
./tests/data/lavf/lavf.dv CRC=0x8d5e9e8f
|
||||
3a6a9163a67b729b4a6b5d972ccceb97 *./tests/data/lavf/lavf.dv
|
||||
b36c83cd0ba0ebe719f09f885c4bbcd3 *./tests/data/lavf/lavf.dv
|
||||
3600000 ./tests/data/lavf/lavf.dv
|
||||
./tests/data/lavf/lavf.dv CRC=0x5ce4e5e4
|
||||
./tests/data/lavf/lavf.dv CRC=0x2bc2ae3a
|
||||
|
||||
@ -4,6 +4,6 @@ befc1a39c37a4ecd9264942a3e34b3f6 *./tests/data/lavf/lavf.gxf
|
||||
267d2b2b6e357209d76c366302cf35c3 *./tests/data/lavf/lavf.gxf
|
||||
794572 ./tests/data/lavf/lavf.gxf
|
||||
./tests/data/lavf/lavf.gxf CRC=0xab47d02d
|
||||
1c1693cf2358025f1e37ac76e1da925a *./tests/data/lavf/lavf.gxf
|
||||
0a1a37fa79b62435545271b4e8e882f5 *./tests/data/lavf/lavf.gxf
|
||||
796392 ./tests/data/lavf/lavf.gxf
|
||||
./tests/data/lavf/lavf.gxf CRC=0x102918fd
|
||||
./tests/data/lavf/lavf.gxf CRC=0x3b1a8e91
|
||||
|
||||
@ -1,3 +1,3 @@
|
||||
23177c8a72f34e243e9ffc4f6c70d3c7 *./tests/data/lavf/lavf.mxf_d10
|
||||
0d72247067569901a2e351586ddc0b82 *./tests/data/lavf/lavf.mxf_d10
|
||||
5330989 ./tests/data/lavf/lavf.mxf_d10
|
||||
./tests/data/lavf/lavf.mxf_d10 CRC=0x81602ff1
|
||||
./tests/data/lavf/lavf.mxf_d10 CRC=0x4474d480
|
||||
|
||||
@ -1,53 +0,0 @@
|
||||
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 58 size: 4096
|
||||
ret: 0 st:-1 flags:0 ts:-1.000000
|
||||
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 58 size: 4096
|
||||
ret: 0 st:-1 flags:1 ts: 1.894167
|
||||
ret: 0 st: 0 flags:1 dts: 1.894000 pts: 1.894000 pos: 7634 size: 4096
|
||||
ret: 0 st: 0 flags:0 ts: 0.788375
|
||||
ret: 0 st: 0 flags:1 dts: 0.788500 pts: 0.788500 pos: 3212 size: 4096
|
||||
ret: 0 st: 0 flags:1 ts:-0.317500
|
||||
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 58 size: 4096
|
||||
ret: 0 st:-1 flags:0 ts: 2.576668
|
||||
ret: 0 st: 0 flags:1 dts: 2.576750 pts: 2.576750 pos: 10365 size: 4096
|
||||
ret: 0 st:-1 flags:1 ts: 1.470835
|
||||
ret: 0 st: 0 flags:1 dts: 1.470750 pts: 1.470750 pos: 5941 size: 4096
|
||||
ret: 0 st: 0 flags:0 ts: 0.365000
|
||||
ret: 0 st: 0 flags:1 dts: 0.365000 pts: 0.365000 pos: 1518 size: 4096
|
||||
ret: 0 st: 0 flags:1 ts:-0.740875
|
||||
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 58 size: 4096
|
||||
ret: 0 st:-1 flags:0 ts: 2.153336
|
||||
ret: 0 st: 0 flags:1 dts: 2.153500 pts: 2.153500 pos: 8672 size: 4096
|
||||
ret: 0 st:-1 flags:1 ts: 1.047503
|
||||
ret: 0 st: 0 flags:1 dts: 1.047500 pts: 1.047500 pos: 4248 size: 4096
|
||||
ret: 0 st: 0 flags:0 ts:-0.058375
|
||||
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 58 size: 4096
|
||||
ret: 0 st: 0 flags:1 ts: 2.835875
|
||||
ret: 0 st: 0 flags:1 dts: 2.835750 pts: 2.835750 pos: 11401 size: 4096
|
||||
ret: 0 st:-1 flags:0 ts: 1.730004
|
||||
ret: 0 st: 0 flags:1 dts: 1.730000 pts: 1.730000 pos: 6978 size: 4096
|
||||
ret: 0 st:-1 flags:1 ts: 0.624171
|
||||
ret: 0 st: 0 flags:1 dts: 0.624000 pts: 0.624000 pos: 2554 size: 4096
|
||||
ret: 0 st: 0 flags:0 ts:-0.481625
|
||||
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 58 size: 4096
|
||||
ret: 0 st: 0 flags:1 ts: 2.412500
|
||||
ret: 0 st: 0 flags:1 dts: 2.412500 pts: 2.412500 pos: 9708 size: 4096
|
||||
ret: 0 st:-1 flags:0 ts: 1.306672
|
||||
ret: 0 st: 0 flags:1 dts: 1.306750 pts: 1.306750 pos: 5285 size: 4096
|
||||
ret: 0 st:-1 flags:1 ts: 0.200839
|
||||
ret: 0 st: 0 flags:1 dts: 0.200750 pts: 0.200750 pos: 861 size: 4096
|
||||
ret: 0 st: 0 flags:0 ts:-0.905000
|
||||
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 58 size: 4096
|
||||
ret: 0 st: 0 flags:1 ts: 1.989125
|
||||
ret: 0 st: 0 flags:1 dts: 1.989000 pts: 1.989000 pos: 8014 size: 4096
|
||||
ret: 0 st:-1 flags:0 ts: 0.883340
|
||||
ret: 0 st: 0 flags:1 dts: 0.883500 pts: 0.883500 pos: 3592 size: 4096
|
||||
ret: 0 st:-1 flags:1 ts:-0.222493
|
||||
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 58 size: 4096
|
||||
ret: 0 st: 0 flags:0 ts: 2.671625
|
||||
ret: 0 st: 0 flags:1 dts: 2.671750 pts: 2.671750 pos: 10745 size: 4096
|
||||
ret: 0 st: 0 flags:1 ts: 1.565875
|
||||
ret: 0 st: 0 flags:1 dts: 1.565750 pts: 1.565750 pos: 6321 size: 4096
|
||||
ret: 0 st:-1 flags:0 ts: 0.460008
|
||||
ret: 0 st: 0 flags:1 dts: 0.460000 pts: 0.460000 pos: 1898 size: 4096
|
||||
ret: 0 st:-1 flags:1 ts:-0.645825
|
||||
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 58 size: 4096
|
||||
@ -1,27 +0,0 @@
|
||||
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 4 size: 39816
|
||||
ret:-1 st:-1 flags:0 ts:-1.000000
|
||||
ret:-1 st:-1 flags:1 ts: 1.894167
|
||||
ret:-1 st: 0 flags:0 ts: 0.788333
|
||||
ret:-1 st: 0 flags:1 ts:-0.317500
|
||||
ret:-1 st:-1 flags:0 ts: 2.576668
|
||||
ret:-1 st:-1 flags:1 ts: 1.470835
|
||||
ret:-1 st: 0 flags:0 ts: 0.365000
|
||||
ret:-1 st: 0 flags:1 ts:-0.740833
|
||||
ret:-1 st:-1 flags:0 ts: 2.153336
|
||||
ret:-1 st:-1 flags:1 ts: 1.047503
|
||||
ret:-1 st: 0 flags:0 ts:-0.058333
|
||||
ret:-1 st: 0 flags:1 ts: 2.835833
|
||||
ret:-1 st:-1 flags:0 ts: 1.730004
|
||||
ret:-1 st:-1 flags:1 ts: 0.624171
|
||||
ret:-1 st: 0 flags:0 ts:-0.481667
|
||||
ret:-1 st: 0 flags:1 ts: 2.412500
|
||||
ret:-1 st:-1 flags:0 ts: 1.306672
|
||||
ret:-1 st:-1 flags:1 ts: 0.200839
|
||||
ret:-1 st: 0 flags:0 ts:-0.904989
|
||||
ret:-1 st: 0 flags:1 ts: 1.989178
|
||||
ret:-1 st:-1 flags:0 ts: 0.883340
|
||||
ret:-1 st:-1 flags:1 ts:-0.222493
|
||||
ret:-1 st: 0 flags:0 ts: 2.671678
|
||||
ret:-1 st: 0 flags:1 ts: 1.565844
|
||||
ret:-1 st:-1 flags:0 ts: 0.460008
|
||||
ret:-1 st:-1 flags:1 ts:-0.645825
|
||||
Loading…
Reference in New Issue
Block a user