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Bug de détection d’ogg
22 mars 2013, par
Mis à jour : Avril 2013
Langue : français
Type : Video
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Organiser par catégorie
17 mai 2013, parDans MédiaSPIP, une rubrique a 2 noms : catégorie et rubrique.
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Récupération d’informations sur le site maître à l’installation d’une instance
26 novembre 2010, parUtilité
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Il peut donc être tout à fait judicieux de vouloir récupérer certaines de ces informations afin de compléter l’installation d’une instance pour, par exemple : récupérer le (...) -
Support audio et vidéo HTML5
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Sur d’autres sites (3774)
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FFmpeg RTP : Bad packed header lengths
13 septembre 2016, par bot1131357I use the
av_sdp_create()to generate my SDP file but I think something’s wrong. Here is the output :Output #0, rtp, to 'rtp://127.0.0.1:8554':
Stream #0:0: Audio: vorbis (libvorbis), 44100 Hz, stereo, fltp, 64 kb/s
v=0
o=- 0 0 IN IP4 127.0.0.1
s=No Name
c=IN IP4 127.0.0.1
t=0 0
a=tool:libavformat 57.25.101
m=audio 8554 RTP/AVP 96
b=AS:64
a=rtpmap:96 vorbis/44100/2
a=fmtp:96 configuration=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 I try to use the SDP file anyway, I get the following message :
>ffplay -i test2.sdp -protocol_whitelist file,udp,rtp
ffplay version N-78598-g98a0053 Copyright (c) 2003-2016 the FFmpeg developers
built with gcc 5.3.0 (GCC)
configuration: --disable-static --enable-shared --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-decklink --enable-zlib
libavutil 55. 18.100 / 55. 18.100
libavcodec 57. 24.103 / 57. 24.103
libavformat 57. 25.101 / 57. 25.101
libavdevice 57. 0.101 / 57. 0.101
libavfilter 6. 34.100 / 6. 34.100
libswscale 4. 0.100 / 4. 0.100
libswresample 2. 0.101 / 2. 0.101
libpostproc 54. 0.100 / 54. 0.100
[NULL @ 05aea880] Bad packed header lengths (30,0,616,3793) f=0/0
[vorbis @ 05aea880] Extradata missing.
[sdp @ 05ad7ec0] Failed to open codec in av_find_stream_info
nan : 0.000 fd= 0 aq= 0KB vq= 0KB sq= 0B f=0/0Am I overlooking something important ? Am I going about this the wrong way ? Any help would be much appreciated.
Here is my source :
#include "stdafx.h"
#include
extern "C"
{
#include <libavutil></libavutil>opt.h>
#include <libavcodec></libavcodec>avcodec.h>
#include <libavutil></libavutil>channel_layout.h>
#include <libavutil></libavutil>common.h>
#include <libavutil></libavutil>imgutils.h>
#include <libavutil></libavutil>mathematics.h>
#include <libavutil></libavutil>samplefmt.h>
#include <libavformat></libavformat>avformat.h>
}
/* check that a given sample format is supported by the encoder */
static int check_sample_fmt(AVCodec *codec, enum AVSampleFormat sample_fmt)
{
const enum AVSampleFormat *p = codec->sample_fmts;
while (*p != AV_SAMPLE_FMT_NONE) {
if (*p == sample_fmt)
return 1;
p++;
}
return 0;
}
/* just pick the highest supported samplerate */
static int select_sample_rate(AVCodec *codec)
{
const int *p;
int best_samplerate = 0;
if (!codec->supported_samplerates)
return 44100;
p = codec->supported_samplerates;
while (*p) {
best_samplerate = FFMAX(*p, best_samplerate);
p++;
}
return best_samplerate;
}
/* select layout with the highest channel count */
static int select_channel_layout(AVCodec *codec)
{
const uint64_t *p;
uint64_t best_ch_layout = 0;
int best_nb_channels = 0;
if (!codec->channel_layouts)
return AV_CH_LAYOUT_STEREO;
p = codec->channel_layouts;
while (*p) {
int nb_channels = av_get_channel_layout_nb_channels(*p);
if (nb_channels > best_nb_channels) {
best_ch_layout = *p;
best_nb_channels = nb_channels;
}
p++;
}
return best_ch_layout;
}
static int write_frame(AVFormatContext *fmt_ctx, const AVRational *time_base, AVStream *st, AVPacket *pkt)
{
/* rescale output packet timestamp values from codec to stream timebase */
av_packet_rescale_ts(pkt, *time_base, st->time_base);
/* Write the compressed frame to the media file. */
return av_interleaved_write_frame(fmt_ctx, pkt);
}
/*
* Audio encoding example
*/
static void audio_encode_example(const char *filename)
{
AVPacket pkt;
int i, j, k, ret, got_output;
int buffer_size;
uint16_t *samples;
float t, tincr;
AVCodec *outCodec = NULL;
AVCodecContext *outCodecCtx = NULL;
AVFormatContext *outFormatCtx = NULL;
AVStream * outAudioStream = NULL;
AVFrame *outFrame = NULL;
ret = avformat_alloc_output_context2(&outFormatCtx, NULL, "rtp", filename);
if (!outFormatCtx || ret < 0)
{
fprintf(stderr, "Could not allocate output context");
}
outFormatCtx->flags |= AVFMT_FLAG_NOBUFFER | AVFMT_FLAG_FLUSH_PACKETS;
outFormatCtx->max_interleave_delta = 1;
outFormatCtx->oformat->audio_codec = AV_CODEC_ID_VORBIS;
/* find the encoder */
outCodec = avcodec_find_encoder(outFormatCtx->oformat->audio_codec);
if (!outCodec) {
fprintf(stderr, "Codec not found\n");
exit(1);
}
outAudioStream = avformat_new_stream(outFormatCtx, outCodec);
if (!outAudioStream)
{
fprintf(stderr, "Cannot add new audio stream\n");
exit(1);
}
outAudioStream->id = outFormatCtx->nb_streams - 1;
outAudioStream->time_base.den = 44100; // 44.100 kHz
outAudioStream->time_base.num = 1;
outCodecCtx = outAudioStream->codec;
outCodecCtx->bit_rate = 64000;
outCodecCtx->time_base.den = outAudioStream->time_base.den;
outCodecCtx->time_base.num = outAudioStream->time_base.num;
/* check that the encoder supports input */
outCodecCtx->sample_fmt = AV_SAMPLE_FMT_FLTP;
if (!check_sample_fmt(outCodec, outCodecCtx->sample_fmt)) {
fprintf(stderr, "Encoder does not support sample format %s",
av_get_sample_fmt_name(outCodecCtx->sample_fmt));
exit(1);
}
/* select other audio parameters supported by the encoder */
outCodecCtx->sample_rate = select_sample_rate(outCodec);
outCodecCtx->channel_layout = select_channel_layout(outCodec);
outCodecCtx->channels = av_get_channel_layout_nb_channels(outCodecCtx->channel_layout);
/* open it */
if (avcodec_open2(outCodecCtx, outCodec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
exit(1);
}
av_dump_format(outFormatCtx, 0, filename, 1);
char buff[2048] = { 0 };
av_sdp_create(&outFormatCtx, 1, buff, 1024);
printf("%s", buff);
(...SDP printed here...)
ret = avio_open2(&outFormatCtx->pb, filename, AVIO_FLAG_WRITE, NULL, NULL);
ret = avformat_write_header(outFormatCtx, NULL);
printf("ret = %d\n", ret);
if (ret <0)
exit(1);
/* frame containing input audio */
outFrame = av_frame_alloc();
if (!outFrame) {
fprintf(stderr, "Could not allocate audio frame\n");
exit(1);
}
outFrame->nb_samples = outCodecCtx->frame_size;
outFrame->format = outCodecCtx->sample_fmt;
outFrame->channel_layout = outCodecCtx->channel_layout;
/* the codec gives us the frame size, in samples,
* we calculate the size of the samples buffer in bytes */
buffer_size = av_samples_get_buffer_size(NULL, outCodecCtx->channels, outCodecCtx->frame_size,
outCodecCtx->sample_fmt, 0);
if (buffer_size < 0) {
fprintf(stderr, "Could not get sample buffer size\n");
exit(1);
}
samples = (uint16_t*)av_malloc(buffer_size);
if (!samples) {
fprintf(stderr, "Could not allocate %d bytes for samples buffer\n",
buffer_size);
exit(1);
}
/* setup the data pointers in the AVFrame */
ret = avcodec_fill_audio_frame(outFrame, outCodecCtx->channels, outCodecCtx->sample_fmt,
(const uint8_t*)samples, buffer_size, 0);
if (ret < 0) {
fprintf(stderr, "Could not setup audio frame\n");
exit(1);
}
/* encode a single tone sound */
t = 0;
int next_pts = 0;
tincr = 2 * M_PI * 440.0 / outCodecCtx->sample_rate;
for (i = 0; i < 44000; i++) {
av_init_packet(&pkt);
pkt.data = NULL; // packet data will be allocated by the encoder
pkt.size = 0;
for (j = 0; j < outCodecCtx->frame_size; j++) {
samples[2 * j] = (int)(sin(t) * 10000);
for (k = 1; k < outCodecCtx->channels; k++)
samples[2 * j + k] = samples[2 * j];
t += tincr;
}
// Sets time stamp
next_pts += outFrame->nb_samples;
outFrame->pts = next_pts;
/* encode the samples */
ret = avcodec_encode_audio2(outCodecCtx, &pkt, outFrame, &got_output);
if (ret < 0) {
fprintf(stderr, "Error encoding audio frame\n");
exit(1);
}
if (got_output) {
//fwrite(pkt.data, 1, pkt.size, f);
//pkt.stream_index = pktidx++;
write_frame(outFormatCtx, &outCodecCtx->time_base, outAudioStream, &pkt);
av_packet_unref(&pkt);
}
}
/* get the delayed frames */
for (got_output = 1; got_output; i++) {
ret = avcodec_encode_audio2(outCodecCtx, &pkt, NULL, &got_output);
if (ret < 0) {
fprintf(stderr, "Error encoding frame\n");
exit(1);
}
if (got_output) {
//fwrite(pkt.data, 1, pkt.size, f);
pkt.pts = AV_NOPTS_VALUE;
write_frame(outFormatCtx, &outCodecCtx->time_base, outAudioStream, &pkt);
av_packet_unref(&pkt);
}
}
av_freep(&samples);
av_frame_free(&outFrame);
avcodec_close(outCodecCtx);
av_free(outCodecCtx);
}
int main(int argc, char **argv)
{
const char *output;
/* register all the codecs */
//avcodec_register_all();
av_register_all();
avformat_network_init(); // for network streaming
audio_encode_example("rtp://127.0.0.1:8554");
return 0;
} -
ffmpeg command for faster encoding at a decent bitrate with smaller file size
14 septembre 2016, par Blair HolmesI’m currently using an implementation of ffmpeg on my android app. I’m allowing users to take short videos within my app and then when they upload them to the server, I’m crunching them down with ffmpeg to decrease file size so they’re not passing huge amounts of data over the wire.
the problem is, it’s taking forever to encode the videos on the android device. These videos usually aren’t longer than 45 seconds and can take 20 minutes to encode. I’ve done some playing around with different switches / parameters on the ffmpeg command line and I now have it at a more comfortable time, but the file sizes are significantly bigger. I’m just not sure what codec I should use (fastest encoding but with decent quality output), how ffmpeg handles changing size (aspect ratio) in terms of speed of encoding etc.
Here are the two commands I’ve been using. This first one outputs the file size / quality that I want, but it just takes way too long to encode, not to mention it makes my device get really hot while it’s encoding for so long :
ffmpeg -i input.mp4 -b:v 1024k -c:a copy -vf scale=960:540 output.mp4I’ve tweaked the bitrate some on that as well as changed the scale to a smaller size, but I don’t want to have to make the videos really small (in terms of scale) to accomplish a faster encoding. this second command goes a lot faster but makes the file size significantly bigger :
ffmpeg -i input.mp4 -vcodec libx264 -preset fast -c:a copy -s 960x540 output.mp4I’d like to find a happy medium (smaller file size but faster encoding) keeping the video scale size close to the original. ffmpeg just has so many different parameters / switches that it’s difficult to wrap my head around what I should be doing.
EDIT : adding ffmpeg output.
ffmpeg -i input.mp4 -vcodec libx264 -crf 30 -preset veryfast -c:a copy -s 960x540 output.mp4
09-13 11:06:28.330 10881-10881/someapp D/home: ffmpeg version n3.0.1 Copyright (c) 2000-2016 the FFmpeg developers
09-13 11:06:28.330 10881-10881/someapp D/home: built with gcc 4.8 (GCC)
09-13 11:06:28.331 10881-10881/someapp D/home: configuration: --target-os=linux --cross-prefix=/home/vagrant/SourceCode/ffmpeg-android/toolchain-android/bin/arm-linux-androideabi- --arch=arm --cpu=cortex-a8 --enable-runtime-cpudetect --sysroot=/home/vagrant/SourceCode/ffmpeg-android/toolchain-android/sysroot --enable-pic --enable-libx264 --enable-libass --enable-libfreetype --enable-libfribidi --enable-libmp3lame --enable-fontconfig --enable-pthreads --disable-debug --disable-ffserver --enable-version3 --enable-hardcoded-tables --disable-ffplay --disable-ffprobe --enable-gpl --enable-yasm --disable-doc --disable-shared --enable-static --pkg-config=/home/vagrant/SourceCode/ffmpeg-android/ffmpeg-pkg-config --prefix=/home/vagrant/SourceCode/ffmpeg-android/build/armeabi-v7a --extra-cflags='-I/home/vagrant/SourceCode/ffmpeg-android/toolchain-android/include -U_FORTIFY_SOURCE -D_FORTIFY_SOURCE=2 -fno-strict-overflow -fstack-protector-all' --extra-ldflags='-L/home/vagrant/SourceCode/ffmpeg-android/toolchain-android/lib -Wl,-z,relro -Wl,-z,now -pie' --extra-libs='-lpng -lexpat -lm' --extra-cxxflags=
09-13 11:06:28.331 10881-10881/someapp D/home: libavutil 55. 17.103 / 55. 17.103
09-13 11:06:28.331 10881-10881/someapp D/home: libavcodec 57. 24.102 / 57. 24.102
09-13 11:06:28.331 10881-10881/someapp D/home: libavformat 57. 25.100 / 57. 25.100
09-13 11:06:28.331 10881-10881/someapp D/home: libavdevice 57. 0.101 / 57. 0.101
09-13 11:06:28.331 10881-10881/someapp D/home: libavfilter 6. 31.100 / 6. 31.100
09-13 11:06:28.331 10881-10881/someapp D/home: libswscale 4. 0.100 / 4. 0.100
09-13 11:06:28.331 10881-10881/someapp D/home: libswresample 2. 0.101 / 2. 0.101
09-13 11:06:28.331 10881-10881/someapp D/home: libpostproc 54. 0.100 / 54. 0.100
09-13 11:06:28.430 10881-10881/someapp D/home: Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/storage/emulated/0/ExpeditionSpot/Videos/20160913110411.mp4':
09-13 11:06:28.430 10881-10881/someapp D/home: Metadata:
09-13 11:06:28.430 10881-10881/someapp D/home: major_brand : mp42
09-13 11:06:28.430 10881-10881/someapp D/home: minor_version : 0
09-13 11:06:28.430 10881-10881/someapp D/home: compatible_brands: isommp42
09-13 11:06:28.430 10881-10881/someapp D/home: creation_time : 2016-09-13 17:04:33
09-13 11:06:28.430 10881-10881/someapp D/home: com.android.version: 6.0.1
09-13 11:06:28.430 10881-10881/someapp D/home: Duration: 00:00:19.41, start: 0.000000, bitrate: 20222 kb/s
09-13 11:06:28.430 10881-10881/someapp D/home: Stream #0:0(eng): Video: h264 (Baseline) (avc1 / 0x31637661), yuv420p, 1920x1080, 19963 kb/s, SAR 1:1 DAR 16:9, 30.03 fps, 30 tbr, 90k tbn, 180k tbc (default)
09-13 11:06:28.430 10881-10881/someapp D/home: Metadata:
09-13 11:06:28.430 10881-10881/someapp D/home: creation_time : 2016-09-13 17:04:33
09-13 11:06:28.430 10881-10881/someapp D/home: handler_name : VideoHandle
09-13 11:06:28.431 10881-10881/someapp D/home: Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 96 kb/s (default)
09-13 11:06:28.431 10881-10881/someapp D/home: Metadata:
09-13 11:06:28.431 10881-10881/someapp D/home: creation_time : 2016-09-13 17:04:33
09-13 11:06:28.431 10881-10881/someapp D/home: handler_name : SoundHandle
09-13 11:06:28.448 10881-10881/someapp D/home: [libx264 @ 0xf71c4400] using SAR=1/1
09-13 11:06:28.448 10881-10881/someapp D/home: [libx264 @ 0xf71c4400] using cpu capabilities: none!
09-13 11:06:28.516 10881-10881/someapp D/home: [libx264 @ 0xf71c4400] profile High, level 3.1
09-13 11:06:28.516 10881-10881/someapp D/home: [libx264 @ 0xf71c4400] 264 - core 148 - H.264/MPEG-4 AVC codec - Copyleft 2003-2015 - http://www.videolan.org/x264.html - options: cabac=1 ref=1 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=2 psy=1 psy_rd=1.00:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=0 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=0 threads=12 lookahead_threads=4 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=1 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=10 rc=crf mbtree=1 crf=27.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
09-13 11:06:28.532 10881-10881/someapp D/home: Output #0, mp4, to '/storage/emulated/0/ExpeditionSpot/.tmp/small-20160913110411.mp4':
09-13 11:06:28.532 10881-10881/someapp D/home: Metadata:
09-13 11:06:28.532 10881-10881/someapp D/home: major_brand : mp42
09-13 11:06:28.533 10881-10881/someapp D/home: minor_version : 0
09-13 11:06:28.533 10881-10881/someapp D/home: compatible_brands: isommp42
09-13 11:06:28.533 10881-10881/someapp D/home: com.android.version: 6.0.1
09-13 11:06:28.533 10881-10881/someapp D/home: encoder : Lavf57.25.100
09-13 11:06:28.533 10881-10881/someapp D/home: Stream #0:0(eng): Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p, 960x540 [SAR 1:1 DAR 16:9], q=-1--1, 30 fps, 15360 tbn, 30 tbc (default)
09-13 11:06:28.533 10881-10881/someapp D/home: Metadata:
09-13 11:06:28.533 10881-10881/someapp D/home: creation_time : 2016-09-13 17:04:33
09-13 11:06:28.533 10881-10881/someapp D/home: handler_name : VideoHandle
09-13 11:06:28.533 10881-10881/someapp D/home: encoder : Lavc57.24.102 libx264
09-13 11:06:28.533 10881-10881/someapp D/home: Side data:
09-13 11:06:28.533 10881-10881/someapp D/home: unknown side data type 10 (24 bytes)
09-13 11:06:28.533 10881-10881/someapp D/home: Stream #0:1(eng): Audio: aac (LC) ([64][0][0][0] / 0x0040), 48000 Hz, stereo, 96 kb/s (default)
09-13 11:06:28.533 10881-10881/someapp D/home: Metadata:
09-13 11:06:28.533 10881-10881/someapp D/home: creation_time : 2016-09-13 17:04:33
09-13 11:06:28.533 10881-10881/someapp D/home: handler_name : SoundHandle
09-13 11:06:28.533 10881-10881/someapp D/home: Stream mapping:
09-13 11:06:28.533 10881-10881/someapp D/home: Stream #0:0 -> #0:0 (h264 (native) -> h264 (libx264))
09-13 11:06:28.533 10881-10881/someapp D/home: Stream #0:1 -> #0:1 (copy)
09-13 11:06:28.533 10881-10881/someapp D/home: Press [q] to stop, [?] for help
09-13 11:06:29.102 10881-10881/someapp D/home: frame= 7 fps=0.0 q=0.0 size= 0kB time=00:00:01.04 bitrate= 0.4kbits/s speed=2.08x
09-13 11:06:29.699 10881-10881/someapp D/home: frame= 16 fps= 15 q=0.0 size= 0kB time=00:00:01.04 bitrate= 0.4kbits/s speed=0.998x
....
09-13 11:07:12.674 10881-10881/someapp D/home: [libx264 @ 0xf71c4400] frame I:3 Avg QP:26.83 size: 21896
09-13 11:07:12.674 10881-10881/someapp D/home: [libx264 @ 0xf71c4400] frame P:279 Avg QP:28.76 size: 5859
09-13 11:07:12.674 10881-10881/someapp D/home: [libx264 @ 0xf71c4400] frame B:296 Avg QP:29.93 size: 863
09-13 11:07:12.674 10881-10881/someapp D/home: [libx264 @ 0xf71c4400] consecutive B-frames: 18.3% 32.2% 23.9% 25.6%
09-13 11:07:12.674 10881-10881/someapp D/home: [libx264 @ 0xf71c4400] mb I I16..4: 16.9% 54.2% 28.9%
09-13 11:07:12.674 10881-10881/someapp D/home: [libx264 @ 0xf71c4400] mb P I16..4: 7.7% 9.1% 0.4% P16..4: 27.5% 11.1% 4.0% 0.0% 0.0% skip:40.1%
09-13 11:07:12.674 10881-10881/someapp D/home: [libx264 @ 0xf71c4400] mb B I16..4: 0.8% 0.6% 0.0% B16..8: 10.9% 2.4% 0.1% direct: 1.8% skip:83.5% L0:35.8% L1:54.2% BI:10.0%
09-13 11:07:12.675 10881-10881/someapp D/home: [libx264 @ 0xf71c4400] 8x8 transform intra:52.0% inter:41.3%
09-13 11:07:12.675 10881-10881/someapp D/home: [libx264 @ 0xf71c4400] coded y,uvDC,uvAC intra: 28.9% 26.0% 2.1% inter: 6.4% 5.0% 0.0%
09-13 11:07:12.675 10881-10881/someapp D/home: [libx264 @ 0xf71c4400] i16 v,h,dc,p: 57% 20% 17% 6%
09-13 11:07:12.675 10881-10881/someapp D/home: [libx264 @ 0xf71c4400] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 33% 23% 34% 1% 2% 1% 3% 2% 2%
09-13 11:07:12.675 10881-10881/someapp D/home: [libx264 @ 0xf71c4400] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 36% 21% 17% 2% 7% 4% 6% 4% 4%
09-13 11:07:12.675 10881-10881/someapp D/home: [libx264 @ 0xf71c4400] i8c dc,h,v,p: 59% 16% 23% 2%
09-13 11:07:12.675 10881-10881/someapp D/home: [libx264 @ 0xf71c4400] Weighted P-Frames: Y:16.8% UV:4.7%
09-13 11:07:12.675 10881-10881/someapp D/home: [libx264 @ 0xf71c4400] kb/s:812.09 -
FFmpeg RTP payload 96 instead of 97
26 octobre 2016, par bot1131357I am trying to create an rtp audio stream with ffmpeg. The application output and SDP file configuration are as follows :
Output #0, rtp, to 'rtp://127.0.0.1:8554':
Stream #0:0: Audio: pcm_s16be, 8000 Hz, stereo, s16, 256 kb/s
SDP:
v=0
o=- 0 0 IN IP4 127.0.0.1
s=No Name
c=IN IP4 127.0.0.1
t=0 0
a=tool:libavformat 57.25.101
m=audio 8554 RTP/AVP 96
b=AS:256
a=rtpmap:96 L16/8000/2However, when I try to read it with
ffplay -i test.sdp -protocol_whitelist file,udp,rtp, it fails,shows the following :ffplay version N-78598-g98a0053 Copyright (c) 2003-2016 the FFmpeg developers
built with gcc 5.3.0 (GCC)
configuration: --disable-static --enable-shared --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-decklink --enable-zlib
libavutil 55. 18.100 / 55. 18.100
libavcodec 57. 24.103 / 57. 24.103
libavformat 57. 25.101 / 57. 25.101
libavdevice 57. 0.101 / 57. 0.101
libavfilter 6. 34.100 / 6. 34.100
libswscale 4. 0.100 / 4. 0.100
libswresample 2. 0.101 / 2. 0.101
libpostproc 54. 0.100 / 54. 0.100
nan : 0.000 fd= 0 aq= 0KB vq= 0KB sq= 0B f=0/0
(...waits indefinitely.)The only way to make it work again is to modify the payload type in the SDP file from 96 to 97. Can someone tell me why ? Where is this number defined ?
Here is my source. See if you can replicate it.
#include
extern "C"
{
#include <libavutil></libavutil>opt.h>
#include <libavcodec></libavcodec>avcodec.h>
#include <libavutil></libavutil>channel_layout.h>
#include <libavutil></libavutil>common.h>
#include <libavutil></libavutil>imgutils.h>
#include <libavutil></libavutil>mathematics.h>
#include <libavutil></libavutil>samplefmt.h>
#include <libavformat></libavformat>avformat.h>
}
static int write_frame(AVFormatContext *fmt_ctx, const AVRational *time_base, AVStream *st, AVPacket *pkt)
{
/* rescale output packet timestamp values from codec to stream timebase */
av_packet_rescale_ts(pkt, *time_base, st->time_base);
/* Write the compressed frame to the media file. */
return av_interleaved_write_frame(fmt_ctx, pkt);
}
/*
* Audio encoding example
*/
static void audio_encode_example(const char *filename)
{
AVPacket pkt;
int i, j, k, ret, got_output;
int buffer_size;
uint16_t *samples;
float t, tincr;
AVCodec *outCodec = NULL;
AVCodecContext *outCodecCtx = NULL;
AVFormatContext *outFormatCtx = NULL;
AVStream * outAudioStream = NULL;
AVFrame *outFrame = NULL;
ret = avformat_alloc_output_context2(&outFormatCtx, NULL, "rtp", filename);
if (!outFormatCtx || ret < 0)
{
fprintf(stderr, "Could not allocate output context");
}
outFormatCtx->flags |= AVFMT_FLAG_NOBUFFER | AVFMT_FLAG_FLUSH_PACKETS;
outFormatCtx->oformat->audio_codec = AV_CODEC_ID_PCM_S16BE;
/* find the encoder */
outCodec = avcodec_find_encoder(outFormatCtx->oformat->audio_codec);
if (!outCodec) {
fprintf(stderr, "Codec not found\n");
exit(1);
}
outAudioStream = avformat_new_stream(outFormatCtx, outCodec);
if (!outAudioStream)
{
fprintf(stderr, "Cannot add new audio stream\n");
exit(1);
}
outAudioStream->id = outFormatCtx->nb_streams - 1;
outCodecCtx = outAudioStream->codec;
outCodecCtx->sample_fmt = AV_SAMPLE_FMT_S16;
/* select other audio parameters supported by the encoder */
outCodecCtx->sample_rate = 8000;
outCodecCtx->channel_layout = AV_CH_LAYOUT_STEREO;
outCodecCtx->channels = 2;
/* open it */
if (avcodec_open2(outCodecCtx, outCodec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
exit(1);
}
// PCM has no frame, so we have to explicitly specify
outCodecCtx->frame_size = 1152;
av_dump_format(outFormatCtx, 0, filename, 1);
char buff[10000] = { 0 };
ret = av_sdp_create(&outFormatCtx, 1, buff, sizeof(buff));
printf("%s", buff);
ret = avio_open2(&outFormatCtx->pb, filename, AVIO_FLAG_WRITE, NULL, NULL);
ret = avformat_write_header(outFormatCtx, NULL);
printf("ret = %d\n", ret);
if (ret <0) {
exit(1);
}
/* frame containing input audio */
outFrame = av_frame_alloc();
if (!outFrame) {
fprintf(stderr, "Could not allocate audio frame\n");
exit(1);
}
outFrame->nb_samples = outCodecCtx->frame_size;
outFrame->format = outCodecCtx->sample_fmt;
outFrame->channel_layout = outCodecCtx->channel_layout;
/* we calculate the size of the samples buffer in bytes */
buffer_size = av_samples_get_buffer_size(NULL, outCodecCtx->channels, outCodecCtx->frame_size,
outCodecCtx->sample_fmt, 0);
if (buffer_size < 0) {
fprintf(stderr, "Could not get sample buffer size\n");
exit(1);
}
samples = (uint16_t*)av_malloc(buffer_size);
if (!samples) {
fprintf(stderr, "Could not allocate %d bytes for samples buffer\n",
buffer_size);
exit(1);
}
/* setup the data pointers in the AVFrame */
ret = avcodec_fill_audio_frame(outFrame, outCodecCtx->channels, outCodecCtx->sample_fmt,
(const uint8_t*)samples, buffer_size, 0);
if (ret < 0) {
fprintf(stderr, "Could not setup audio frame\n");
exit(1);
}
/* encode a single tone sound */
t = 0;
int next_pts = 0;
tincr = 2 * M_PI * 440.0 / outCodecCtx->sample_rate;
for (i = 0; i < 400000; i++) {
av_init_packet(&pkt);
pkt.data = NULL; // packet data will be allocated by the encoder
pkt.size = 0;
for (j = 0; j < outCodecCtx->frame_size; j++) {
samples[2 * j] = (uint16_t)(sin(t) * 10000);
for (k = 1; k < outCodecCtx->channels; k++)
samples[2 * j + k] = samples[2 * j];
t += tincr;
}
t = (t > 50000) ? 0 : t;
// Sets time stamp
next_pts += outFrame->nb_samples;
outFrame->pts = next_pts;
/* encode the samples */
ret = avcodec_encode_audio2(outCodecCtx, &pkt, outFrame, &got_output);
if (ret < 0) {
fprintf(stderr, "Error encoding audio frame\n");
exit(1);
}
if (got_output) {
write_frame(outFormatCtx, &outCodecCtx->time_base, outAudioStream, &pkt);
av_packet_unref(&pkt);
}
printf("i:%d\n", i); // waste some time to avoid over-filling jitter buffer
printf("Audio: %d\t%d\n", samples[0], samples[1]); // waste some time to avoid over-filling jitter buffer
printf("t: %f\n", t); // waste some time to avoid over-filling jitter buffer
}
/* get the delayed frames */
for (got_output = 1; got_output; i++) {
ret = avcodec_encode_audio2(outCodecCtx, &pkt, NULL, &got_output);
if (ret < 0) {
fprintf(stderr, "Error encoding frame\n");
exit(1);
}
if (got_output) {
pkt.pts = AV_NOPTS_VALUE;
write_frame(outFormatCtx, &outCodecCtx->time_base, outAudioStream, &pkt);
av_packet_unref(&pkt);
}
}
av_freep(&samples);
av_frame_free(&outFrame);
avcodec_close(outCodecCtx);
av_free(outCodecCtx);
}
int main(int argc, char **argv)
{
const char *output;
av_register_all();
avformat_network_init(); // for network streaming
audio_encode_example("rtp://127.0.0.1:8554");
return 0;
}Update
Curiously, running on Linux Ubuntu gives me the following instead :
Output #0, rtp, to 'rtp://127.0.0.1:8554':
Stream #0:0: Unknown: none (pcm_s16be)
v=0
o=- 0 0 IN IP4 127.0.0.1
s=No Name
c=IN IP4 127.0.0.1
t=0 0
a=tool:libavformat 57.48.100
m=application 8554 RTP/AVP 3Does anyone know why the stream has been changed from audio to application ?
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