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Participer à sa traduction
10 avril 2011Vous pouvez nous aider à améliorer les locutions utilisées dans le logiciel ou à traduire celui-ci dans n’importe qu’elle nouvelle langue permettant sa diffusion à de nouvelles communautés linguistiques.
Pour ce faire, on utilise l’interface de traduction de SPIP où l’ensemble des modules de langue de MediaSPIP sont à disposition. ll vous suffit de vous inscrire sur la liste de discussion des traducteurs pour demander plus d’informations.
Actuellement MediaSPIP n’est disponible qu’en français et (...) -
MediaSPIP 0.1 Beta version
25 avril 2011, parMediaSPIP 0.1 beta is the first version of MediaSPIP proclaimed as "usable".
The zip file provided here only contains the sources of MediaSPIP in its standalone version.
To get a working installation, you must manually install all-software dependencies on the server.
If you want to use this archive for an installation in "farm mode", you will also need to proceed to other manual (...) -
Formulaire personnalisable
21 juin 2013, parCette page présente les champs disponibles dans le formulaire de publication d’un média et il indique les différents champs qu’on peut ajouter. Formulaire de création d’un Media
Dans le cas d’un document de type média, les champs proposés par défaut sont : Texte Activer/Désactiver le forum ( on peut désactiver l’invite au commentaire pour chaque article ) Licence Ajout/suppression d’auteurs Tags
On peut modifier ce formulaire dans la partie :
Administration > Configuration des masques de formulaire. (...)
Sur d’autres sites (13473)
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ffmpeg enciding, Opus sound in the webm container does not work
2 juillet 2017, par MockarutanI’m trying to encode audio and video to a webm file with VP8 and Opus encoding. It almost works. (I use FFmpeg 3.3.2)
I can make a only video webm file and play it in VLC, FFPlay and upload it to YouTube (and all works). If I add Opus sound to the file, it still works in VLC but not in FFPlay or on youtube, on youtube the sound becomes just "ticks".
I have the same problem if I encode only Opus audio to the webm file ; it only works in VLC. But if I encode only Opus audio to a ogg container it works everywhere, and I can even use FFmpeg to combine the ogg file with a video only webm file and produce a fully working webm file with audio and video.
So it seems to me that only when I use my code to encode Opus into a webm container, it just wont work in most players and YouTube. I need it to work in youtube.
Here is the code for the opus to webm only encoding (you can toggle ogg/webm with the define) : https://pastebin.com/jyQ4s3tB
#include <algorithm>
#include <iterator>
extern "C"
{
//#define OGG
#include "libavcodec/avcodec.h"
#include "libavdevice/avdevice.h"
#include "libavfilter/avfilter.h"
#include "libavformat/avformat.h"
#include "libavutil/avutil.h"
#include "libavutil/imgutils.h"
#include "libswscale/swscale.h"
#include "libswresample/swresample.h"
enum InfoCodes
{
ENCODED_VIDEO,
ENCODED_AUDIO,
ENCODED_AUDIO_AND_VIDEO,
NOT_ENOUGH_AUDIO_DATA,
};
enum ErrorCodes
{
RES_NOT_MUL_OF_TWO = -1,
ERROR_FINDING_VID_CODEC = -2,
ERROR_CONTEXT_CREATION = -3,
ERROR_CONTEXT_ALLOCATING = -4,
ERROR_OPENING_VID_CODEC = -5,
ERROR_OPENING_FILE = -6,
ERROR_ALLOCATING_FRAME = -7,
ERROR_ALLOCATING_PIC_BUF = -8,
ERROR_ENCODING_FRAME_SEND = -9,
ERROR_ENCODING_FRAME_RECEIVE = -10,
ERROR_FINDING_AUD_CODEC = -11,
ERROR_OPENING_AUD_CODEC = -12,
ERROR_INIT_RESMPL_CONTEXT = -13,
ERROR_ENCODING_SAMPLES_SEND = -14,
ERROR_ENCODING_SAMPLES_RECEIVE = -15,
ERROR_WRITING_HEADER = -16,
ERROR_INIT_AUDIO_RESPAMLER = -17,
};
AVCodecID aud_codec_comp_id = AV_CODEC_ID_OPUS;
AVSampleFormat sample_fmt_comp = AV_SAMPLE_FMT_FLT;
AVCodecID aud_codec_id;
AVSampleFormat sample_fmt;
#ifndef OGG
char* compressed_cont = "webm";
#endif
#ifdef OGG
char* compressed_cont = "ogg";
#endif
AVCodec *aud_codec = NULL;
AVCodecContext *aud_codec_context = NULL;
AVFormatContext *outctx;
AVStream *audio_st;
AVFrame *aud_frame;
SwrContext *audio_swr_ctx;
int vid_frame_counter, aud_frame_counter;
int vid_width, vid_height;
char* concat(const char *s1, const char *s2)
{
char *result = (char*)malloc(strlen(s1) + strlen(s2) + 1);
strcpy(result, s1);
strcat(result, s2);
return result;
}
int setup_audio_codec()
{
aud_codec_id = aud_codec_comp_id;
sample_fmt = sample_fmt_comp;
// Fixup audio codec
if (aud_codec == NULL)
{
aud_codec = avcodec_find_encoder(aud_codec_id);
avcodec_register(aud_codec);
}
if (!aud_codec)
return ERROR_FINDING_AUD_CODEC;
return 0;
}
int initialize_audio_stream(AVFormatContext *local_outctx, int sample_rate, int per_frame_audio_samples, int audio_bitrate)
{
aud_codec_context = avcodec_alloc_context3(aud_codec);
if (!aud_codec_context)
return ERROR_CONTEXT_CREATION;
aud_codec_context->bit_rate = audio_bitrate;
aud_codec_context->sample_rate = sample_rate;
aud_codec_context->sample_fmt = sample_fmt;
aud_codec_context->channel_layout = AV_CH_LAYOUT_STEREO;
aud_codec_context->channels = av_get_channel_layout_nb_channels(aud_codec_context->channel_layout);
//aud_codec_context->profile = FF_PROFILE_AAC_MAIN;
aud_codec_context->codec = aud_codec;
aud_codec_context->codec_id = aud_codec_id;
AVRational time_base;
time_base.num = per_frame_audio_samples;
time_base.den = aud_codec_context->sample_rate;
aud_codec_context->time_base = time_base;
int ret = avcodec_open2(aud_codec_context, aud_codec, NULL);
if (ret < 0)
return ERROR_OPENING_AUD_CODEC;
local_outctx->audio_codec = aud_codec;
local_outctx->audio_codec_id = aud_codec_id;
audio_st = avformat_new_stream(local_outctx, aud_codec);
audio_st->codecpar->bit_rate = aud_codec_context->bit_rate;
audio_st->codecpar->sample_rate = aud_codec_context->sample_rate;
audio_st->codecpar->channels = aud_codec_context->channels;
audio_st->codecpar->channel_layout = aud_codec_context->channel_layout;
audio_st->codecpar->codec_id = aud_codec_context->codec_id;
audio_st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
audio_st->codecpar->format = aud_codec_context->sample_fmt;
audio_st->codecpar->frame_size = aud_codec_context->frame_size;
audio_st->codecpar->block_align = aud_codec_context->block_align;
audio_st->codecpar->initial_padding = aud_codec_context->initial_padding;
audio_st->codecpar->extradata = aud_codec_context->extradata;
audio_st->codecpar->extradata_size = aud_codec_context->extradata_size;
aud_frame = av_frame_alloc();
aud_frame->nb_samples = aud_codec_context->frame_size;
aud_frame->format = aud_codec_context->sample_fmt;
aud_frame->channel_layout = aud_codec_context->channel_layout;
aud_frame->sample_rate = aud_codec_context->sample_rate;
int buffer_size;
if (aud_codec_context->frame_size == 0)
{
buffer_size = per_frame_audio_samples * 2 * 4;
aud_frame->nb_samples = per_frame_audio_samples;
}
else
{
buffer_size = av_samples_get_buffer_size(NULL, aud_codec_context->channels, aud_codec_context->frame_size,
aud_codec_context->sample_fmt, 0);
}
if (av_sample_fmt_is_planar(sample_fmt))
ret = av_frame_get_buffer(aud_frame, buffer_size / 2);
else
ret = av_frame_get_buffer(aud_frame, buffer_size);
if (!aud_frame || ret < 0)
return ERROR_ALLOCATING_FRAME;
aud_frame_counter = 0;
return 0;
}
int initialize_audio_only_encoding(int sample_rate, int per_frame_audio_samples, int audio_bitrate, const char *filename)
{
int ret;
avcodec_register_all();
av_register_all();
outctx = avformat_alloc_context();
char* with_dot = concat(filename, ".");
char* full_filename = concat(with_dot, compressed_cont);
ret = avformat_alloc_output_context2(&outctx, NULL, compressed_cont, full_filename);
free(with_dot);
if (ret < 0)
{
free(full_filename);
return ERROR_CONTEXT_CREATION;
}
ret = setup_audio_codec();
if (ret < 0)
return ret;
// Setup Audio
ret = initialize_audio_stream(outctx, sample_rate, per_frame_audio_samples, audio_bitrate);
if (ret < 0)
return ret;
av_dump_format(outctx, 0, full_filename, 1);
if (!(outctx->oformat->flags & AVFMT_NOFILE))
{
if (avio_open(&outctx->pb, full_filename, AVIO_FLAG_WRITE) < 0)
{
free(full_filename);
return ERROR_OPENING_FILE;
}
}
free(full_filename);
ret = avformat_write_header(outctx, NULL);
if (ret < 0)
return ERROR_WRITING_HEADER;
return 0;
}
int write_interleaved_audio_frame(float_t *aud_sample)
{
int ret;
aud_frame->data[0] = (uint8_t*)aud_sample;
aud_frame->extended_data[0] = (uint8_t*)aud_sample;
aud_frame->pts = aud_frame_counter++;
ret = avcodec_send_frame(aud_codec_context, aud_frame);
AVPacket pkt;
av_init_packet(&pkt);
pkt.data = NULL;
pkt.size = 0;
while (true)
{
ret = avcodec_receive_packet(aud_codec_context, &pkt);
if (!ret)
{
av_packet_rescale_ts(&pkt, aud_codec_context->time_base, audio_st->time_base);
pkt.stream_index = audio_st->index;
av_interleaved_write_frame(outctx, &pkt);
av_packet_unref(&pkt);
}
if (ret == AVERROR(EAGAIN))
break;
else if (ret < 0)
return ERROR_ENCODING_SAMPLES_RECEIVE;
else
break;
}
return ENCODED_AUDIO;
}
int write_audio_frame(float_t *aud_sample)
{
int ret;
aud_frame->data[0] = (uint8_t*)aud_sample;
aud_frame->extended_data[0] = (uint8_t*)aud_sample;
aud_frame->pts = aud_frame_counter++;
ret = avcodec_send_frame(aud_codec_context, aud_frame);
if (ret < 0)
return ERROR_ENCODING_FRAME_SEND;
AVPacket pkt;
av_init_packet(&pkt);
pkt.data = NULL;
pkt.size = 0;
fflush(stdout);
while (true)
{
ret = avcodec_receive_packet(aud_codec_context, &pkt);
if (!ret)
if (pkt.pts != AV_NOPTS_VALUE)
pkt.pts = av_rescale_q(pkt.pts, aud_codec_context->time_base, audio_st->time_base);
if (pkt.dts != AV_NOPTS_VALUE)
pkt.dts = av_rescale_q(pkt.dts, aud_codec_context->time_base, audio_st->time_base);
{
av_write_frame(outctx, &pkt);
av_packet_unref(&pkt);
}
if (ret == AVERROR(EAGAIN))
break;
else if (ret < 0)
return ERROR_ENCODING_FRAME_RECEIVE;
else
break;
}
return ENCODED_AUDIO;
}
int finish_audio_encoding()
{
AVPacket pkt;
av_init_packet(&pkt);
pkt.data = NULL;
pkt.size = 0;
fflush(stdout);
int ret = avcodec_send_frame(aud_codec_context, NULL);
if (ret < 0)
return ERROR_ENCODING_FRAME_SEND;
while (true)
{
ret = avcodec_receive_packet(aud_codec_context, &pkt);
if (!ret)
{
if (pkt.pts != AV_NOPTS_VALUE)
pkt.pts = av_rescale_q(pkt.pts, aud_codec_context->time_base, audio_st->time_base);
if (pkt.dts != AV_NOPTS_VALUE)
pkt.dts = av_rescale_q(pkt.dts, aud_codec_context->time_base, audio_st->time_base);
av_write_frame(outctx, &pkt);
av_packet_unref(&pkt);
}
if (ret == -AVERROR(AVERROR_EOF))
break;
else if (ret < 0)
return ERROR_ENCODING_FRAME_RECEIVE;
}
av_write_trailer(outctx);
return 0;
}
void cleanup()
{
if (aud_frame)
{
av_frame_free(&aud_frame);
}
if (outctx)
{
for (int i = 0; i < outctx->nb_streams; i++)
av_freep(&outctx->streams[i]);
avio_close(outctx->pb);
av_free(outctx);
}
if (aud_codec_context)
{
avcodec_close(aud_codec_context);
av_free(aud_codec_context);
}
}
void fill_samples(float_t *dst, int nb_samples, int nb_channels, int sample_rate, float_t *t)
{
int i, j;
float_t tincr = 1.0 / sample_rate;
const float_t c = 2 * M_PI * 440.0;
for (i = 0; i < nb_samples; i++) {
*dst = sin(c * *t);
for (j = 1; j < nb_channels; j++)
dst[j] = dst[0];
dst += nb_channels;
*t += tincr;
}
}
int main()
{
int sec = 5;
int frame_rate = 30;
float t = 0, tincr = 0, tincr2 = 0;
int src_samples_linesize;
int src_nb_samples = 960;
int src_channels = 2;
int sample_rate = 48000;
uint8_t **src_data = NULL;
int ret;
initialize_audio_only_encoding(48000, src_nb_samples, 192000, "sound_FLT_960");
ret = av_samples_alloc_array_and_samples(&src_data, &src_samples_linesize, src_channels,
src_nb_samples, AV_SAMPLE_FMT_FLT, 0);
for (size_t i = 0; i < frame_rate * sec; i++)
{
fill_samples((float *)src_data[0], src_nb_samples, src_channels, sample_rate, &t);
write_interleaved_audio_frame((float *)src_data[0]);
}
finish_audio_encoding();
cleanup();
return 0;
}
}
</iterator></algorithm>And some of the files :
The webm audio file that does not work (only in VLC) :
https://drive.google.com/file/d/0B16rIXjPXJCqcU5HVllIYW1iODg/view?usp=sharingThe ogg audio file that works :
https://drive.google.com/file/d/0B16rIXjPXJCqMUZhbW0tTDFjT1E/view?usp=sharingVideo and Audio file that only works in VLC : https://drive.google.com/file/d/0B16rIXjPXJCqX3pEN3B0QVlrekU/view?usp=sharing
If a play the ogg file in FFPlay it says "aq= 30kb", but if I play the webm audio only file i get "aq= 0kb". So that does not seem right either.
Any idea ? Thanks in advance !
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no video above 360P with sound ytdl-core
26 juin 2021, par AshishI am able to download youtube separate videos and audios. but merging not working. 0 KB file is downloading


let info = await ytdl.getInfo(req.body.videoId);
let formatVideo = ytdl.chooseFormat(info.formats, { quality: req.body.format.itag, filter: 'videoonly' });
let formatAudio = ytdl.chooseFormat(info.formats, { quality: req.body.audioTag });
// res.json({video: formatVideo, audio: formatAudio});
const tracker = {
 start: Date.now(),
 audio: { downloaded: 0, total: Infinity },
 video: { downloaded: 0, total: Infinity },
 merged: { frame: 0, speed: '0x', fps: 0 },
};
res.header('Content-Disposition', `attachment; filename=${fileName}`);

cp.exec(`ffmpeg -i ${videoStream} -i ${audioStream} -c copy ${fileName}`, (error, success) => {
 if (!error) {
 console.log(success);
 } else {
 console.log(error);
 }
});
// Prepare the progress bar
let progressbarHandle = null;
const progressbarInterval = 1000;
const showProgress = () => {
 console.log('test');
 readline.cursorTo(process.stdout, 0);
 const toMB = i => (i / 1024 / 1024).toFixed(2);

 process.stdout.write(`Audio | ${(tracker.audio.downloaded / tracker.audio.total * 100).toFixed(2)}% processed `);
 process.stdout.write(`(${toMB(tracker.audio.downloaded)}MB of ${toMB(tracker.audio.total)}MB).${' '.repeat(10)}\n`);

 process.stdout.write(`Video | ${(tracker.video.downloaded / tracker.video.total * 100).toFixed(2)}% processed `);
 process.stdout.write(`(${toMB(tracker.video.downloaded)}MB of ${toMB(tracker.video.total)}MB).${' '.repeat(10)}\n`);

 process.stdout.write(`Merged | processing frame ${tracker.merged.frame} `);
 process.stdout.write(`(at ${tracker.merged.fps} fps => ${tracker.merged.speed}).${' '.repeat(10)}\n`);

 process.stdout.write(`running for: ${((Date.now() - tracker.start) / 1000 / 60).toFixed(2)} Minutes.`);
 readline.moveCursor(process.stdout, 0, -3);
};

// Start the ffmpeg child process
const ffmpegProcess = cp.spawn(ffmpeg, [
 // Remove ffmpeg's console spamming
 '-loglevel', '8', '-hide_banner',
 // Redirect/Enable progress messages
 '-progress', 'pipe:3',
 // Set inputs
 '-i', 'pipe:4',
 '-i', 'pipe:5',
 // Map audio & video from streams
 '-map', '0:a',
 '-map', '1:v',
 // Keep encoding
 '-c:v', 'copy',
 // Define output file
 `${fileName}`,
], {
 windowsHide: true,
 stdio: [
 /* Standard: stdin, stdout, stderr */
 'inherit', 'inherit', 'inherit',
 /* Custom: pipe:3, pipe:4, pipe:5 */
 'pipe', 'pipe', 'pipe', 'pipe'
 ],
});
ffmpegProcess.on('close', () => {
 console.log('done');
 // Cleanup
 process.stdout.write('\n\n\n\n');
 clearInterval(progressbarHandle);
 console.log(tracker, '146');
});

// Link streams
// FFmpeg creates the transformer streams and we just have to insert / read data
ffmpegProcess.stdio[3].on('data', chunk => {
 console.log(chunk, '152');
 // Start the progress bar
 if (!progressbarHandle) progressbarHandle = setInterval(showProgress, progressbarInterval);
 // Parse the param=value list returned by ffmpeg
 const lines = chunk.toString().trim().split('\n');
 const args = {};
 for (const l of lines) {
 const [key, value] = l.split('=');
 args[key.trim()] = value.trim();
 }
 tracker.merged = args;
 // console.log(tracker.merged, '162');
});
// audio.pipe(ffmpegProcess.stdio[4]);
// video.pipe(ffmpegProcess.stdio[5]);
const audioStream = await requestObj.get(formatAudio.url);
const videoStream = await requestObj.get(formatVideo.url);
audioStream.pipe(ffmpegProcess.stdio[4]);
videoStream.pipe(ffmpegProcess.stdio[5]);
ffmpegProcess.stdio[6].pipe(res);



here requestObj is a node js 'request' module


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FFmpeg AVI to MP4 preserves sound, but not images [migrated]
21 février 2013, par user1711384Working with FFmpeg at a conversion (any file to MP4 (H.264/AAC)) :
ffmpeg -y -i testdatei.avi -i logo.jpg -filter_complex overlay=15:15,scale=-1:720 -c:v libx264 -profile:v baseline -preset medium -b:v 880k -g 10 -pass 1 -an -f mp4 -movflags faststart /dev/null
ffmpeg -y -i testdatei.avi -i logo.jpg -filter_complex overlay=15:15,scale=-1:720 -c:v libx264 -profile:v baseline -preset medium -b:v 880k -g 10 -pass 2 -c:a libfdk_aac -b:a 128k -movflags faststart xxx.mp4 2>&1With MPEG and WMV file it's working. With two different AVIs it didn't work. Logfiles from path 1 aren't generated and output 1 is empty, output 2 of course generates an error.
After removing
-profile:v baseline
in both commands, the video file is successfully generated, but it's not possible to play it in JW Player (sound yes, but no image).This is the result of the first command :
ffmpeg version git-2013-02-20-39b0393 Copyright (c) 2000-2013 the FFmpeg developers
built on Feb 20 2013 12:06:36 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5)
configuration: --enable-gpl --enable-libass --enable-libfaac --enable-libfdk-aac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libspeex --enable-librtmp --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-nonfree --enable-version3
libavutil 52. 17.102 / 52. 17.102
libavcodec 54. 92.100 / 54. 92.100
libavformat 54. 63.100 / 54. 63.100
libavdevice 54. 3.103 / 54. 3.103
libavfilter 3. 38.103 / 3. 38.103
libswscale 2. 2.100 / 2. 2.100
libswresample 0. 17.102 / 0. 17.102
libpostproc 52. 2.100 / 52. 2.100
[avi @ 0x23e4d80] non-interleaved AVI
Guessed Channel Layout for Input Stream #0.1 : stereo
Input #0, avi, from 'testdatei.avi':
Metadata:
date : 2013-02-21T14:06:32+01:00
encoder : Adobe Premiere Pro CS6 (Windows)
Duration: 00:00:07.57, start: 0.000000, bitrate: 30330 kb/s
Stream #0:0: Video: dvvideo (dvsd / 0x64737664), yuv411p, 720x480 [SAR 8:9 DAR 4:3], 29.97 tbr, 29.97 tbn, 29.97 tbc
Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, s16, 1536 kb/s
Input #1, image2, from 'logo.jpg':
Duration: 00:00:00.04, start: 0.000000, bitrate: N/A
Stream #1:0: Video: mjpeg, yuvj444p, 170x82, 25 tbr, 25 tbn, 25 tbc
[libx264 @ 0x23e9640] using SAR=8/9
[libx264 @ 0x23e9640] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2 AVX
[libx264 @ 0x23e9640] profile High 4:4:4 Predictive, level 3.1, 4:4:4 8-bit
[libx264 @ 0x23e9640] 264 - core 129 r2 bc13772 - H.264/MPEG-4 AVC codec - Copyleft 2003-2013 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=4 threads=12 lookahead_threads=2 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=2 keyint=10 keyint_min=1 scenecut=40 intra_refresh=0 rc_lookahead=10 rc=2pass mbtree=1 bitrate=880 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 cplxblur=20.0 qblur=0.5 ip_ratio=1.40 aq=1:1.00
Output #0, mp4, to 'xxx.mp4':
Metadata:
date : 2013-02-21T14:06:32+01:00
encoder : Lavf54.63.100
Stream #0:0: Video: h264 ([33][0][0][0] / 0x0021), yuv444p, 1080x720 [SAR 8:9 DAR 4:3], q=-1--1, pass 2, 880 kb/s, 11988 tbn, 29.97 tbc
Stream #0:1: Audio: aac ([64][0][0][0] / 0x0040), 48000 Hz, stereo, s16, 128 kb/s
Stream mapping:
Stream #0:0 (dvvideo) -> overlay:main (graph 0)
Stream #1:0 (mjpeg) -> overlay:overlay (graph 0)
scale (graph 0) -> Stream #0:0 (libx264)
Stream #0:1 -> #0:1 (pcm_s16le -> libfdk_aac)
Press [q] to stop, [?] for help
frame= 79 fps=0.0 q=30.0 size= 291kB time=00:00:02.58 bitrate= 922.4kbits/s
frame= 162 fps=162 q=30.0 size= 620kB time=00:00:05.33 bitrate= 952.9kbits/s
Starting second pass: moving header on top of the file"
frame= 227 fps=154 q=32766.0 Lsize= 958kB time=00:00:07.59 bitrate=1033.5kbits/s
video:829kB audio:120kB subtitle:0 global headers:0kB muxing overhead 0.986027%
[libx264 @ 0x23e9640] frame I:23 Avg QP:19.11 size: 31383
[libx264 @ 0x23e9640] frame P:68 Avg QP:23.91 size: 1240
[libx264 @ 0x23e9640] frame B:136 Avg QP:20.27 size: 310
[libx264 @ 0x23e9640] consecutive B-frames: 19.8% 0.9% 0.0% 79.3%
[libx264 @ 0x23e9640] mb I I16..4: 18.8% 68.4% 12.8%
[libx264 @ 0x23e9640] mb P I16..4: 0.3% 0.3% 0.0% P16..4: 10.7% 2.3% 0.8% 0.0% 0.0% skip:85.6%
[libx264 @ 0x23e9640] mb B I16..4: 0.0% 0.0% 0.0% B16..8: 9.1% 0.1% 0.0% direct: 0.1% skip:90.7% L0:41.2% L1:58.6% BI: 0.2%
[libx264 @ 0x23e9640] 8x8 transform intra:68.3% inter:97.5%
[libx264 @ 0x23e9640] coded y,u,v intra: 53.7% 26.9% 30.8% inter: 0.5% 0.2% 0.3%
[libx264 @ 0x23e9640] i16 v,h,dc,p: 70% 17% 1% 11%
[libx264 @ 0x23e9640] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 28% 25% 18% 4% 3% 4% 4% 6% 8%
[libx264 @ 0x23e9640] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 33% 29% 8% 4% 6% 6% 5% 5% 4%
[libx264 @ 0x23e9640] Weighted P-Frames: Y:0.0% UV:0.0%
[libx264 @ 0x23e9640] ref P L0: 75.9% 5.1% 11.3% 7.7%
[libx264 @ 0x23e9640] ref B L0: 96.0% 3.1% 0.9%
[libx264 @ 0x23e9640] ref B L1: 95.8% 4.2%
[libx264 @ 0x23e9640] kb/s:895.99Output2 :
ffmpeg version git-2013-02-20-39b0393 Copyright (c) 2000-2013 the FFmpeg developers
built on Feb 20 2013 12:06:36 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5)
configuration: --enable-gpl --enable-libass --enable-libfaac --enable-libfdk-aac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libspeex --enable-librtmp --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-nonfree --enable-version3
libavutil 52. 17.102 / 52. 17.102
libavcodec 54. 92.100 / 54. 92.100
libavformat 54. 63.100 / 54. 63.100
libavdevice 54. 3.103 / 54. 3.103
libavfilter 3. 38.103 / 3. 38.103
libswscale 2. 2.100 / 2. 2.100
libswresample 0. 17.102 / 0. 17.102
libpostproc 52. 2.100 / 52. 2.100
[avi @ 0x23e4d80] non-interleaved AVI
Guessed Channel Layout for Input Stream #0.1 : stereo
Input #0, avi, from 'testdatei.avi':
Metadata:
date : 2013-02-21T14:06:32+01:00
encoder : Adobe Premiere Pro CS6 (Windows)
Duration: 00:00:07.57, start: 0.000000, bitrate: 30330 kb/s
Stream #0:0: Video: dvvideo (dvsd / 0x64737664), yuv411p, 720x480 [SAR 8:9 DAR 4:3], 29.97 tbr, 29.97 tbn, 29.97 tbc
Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, s16, 1536 kb/s
Input #1, image2, from 'logo.jpg':
Duration: 00:00:00.04, start: 0.000000, bitrate: N/A
Stream #1:0: Video: mjpeg, yuvj444p, 170x82, 25 tbr, 25 tbn, 25 tbc
[libx264 @ 0x23e9640] using SAR=8/9
[libx264 @ 0x23e9640] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2 AVX
[libx264 @ 0x23e9640] profile High 4:4:4 Predictive, level 3.1, 4:4:4 8-bit
[libx264 @ 0x23e9640] 264 - core 129 r2 bc13772 - H.264/MPEG-4 AVC codec - Copyleft 2003-2013 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=4 threads=12 lookahead_threads=2 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=2 keyint=10 keyint_min=1 scenecut=40 intra_refresh=0 rc_lookahead=10 rc=2pass mbtree=1 bitrate=880 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 cplxblur=20.0 qblur=0.5 ip_ratio=1.40 aq=1:1.00
Output #0, mp4, to 'xxx.mp4':
Metadata:
date : 2013-02-21T14:06:32+01:00
encoder : Lavf54.63.100
Stream #0:0: Video: h264 ([33][0][0][0] / 0x0021), yuv444p, 1080x720 [SAR 8:9 DAR 4:3], q=-1--1, pass 2, 880 kb/s, 11988 tbn, 29.97 tbc
Stream #0:1: Audio: aac ([64][0][0][0] / 0x0040), 48000 Hz, stereo, s16, 128 kb/s
Stream mapping:
Stream #0:0 (dvvideo) -> overlay:main (graph 0)
Stream #1:0 (mjpeg) -> overlay:overlay (graph 0)
scale (graph 0) -> Stream #0:0 (libx264)
Stream #0:1 -> #0:1 (pcm_s16le -> libfdk_aac)
Press [q] to stop, [?] for help
frame= 79 fps=0.0 q=30.0 size= 291kB time=00:00:02.58 bitrate= 922.4kbits/s
frame= 162 fps=162 q=30.0 size= 620kB time=00:00:05.33 bitrate= 952.9kbits/s
Starting second pass: moving header on top of the file"
frame= 227 fps=154 q=32766.0 Lsize= 958kB time=00:00:07.59 bitrate=1033.5kbits/s
video:829kB audio:120kB subtitle:0 global headers:0kB muxing overhead 0.986027%
[libx264 @ 0x23e9640] frame I:23 Avg QP:19.11 size: 31383
[libx264 @ 0x23e9640] frame P:68 Avg QP:23.91 size: 1240
[libx264 @ 0x23e9640] frame B:136 Avg QP:20.27 size: 310
[libx264 @ 0x23e9640] consecutive B-frames: 19.8% 0.9% 0.0% 79.3%
[libx264 @ 0x23e9640] mb I I16..4: 18.8% 68.4% 12.8%
[libx264 @ 0x23e9640] mb P I16..4: 0.3% 0.3% 0.0% P16..4: 10.7% 2.3% 0.8% 0.0% 0.[0% skip:85.6%
[libx264 @ 0x23e9640] mb B I16..4: 0.0% 0.0% 0.0% B16..8: 9.1% 0.1% 0.0% direct: 0.1% skip:90.7% L0:41.2% L1:58.6% BI: 0.2%
[libx264 @ 0x23e9640] 8x8 transform intra:68.3% inter:97.5%
[libx264 @ 0x23e9640] coded y,u,v intra: 53.7% 26.9% 30.8% inter: 0.5% 0.2% 0.3%
[libx264 @ 0x23e9640] i16 v,h,dc,p: 70% 17% 1% 11%
[libx264 @ 0x23e9640] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 28% 25% 18% 4% 3% 4% 4% 6% 8%
[libx264 @ 0x23e9640] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 33% 29% 8% 4% 6% 6% 5% 5% 4%
[libx264 @ 0x23e9640] Weighted P-Frames: Y:0.0% UV:0.0%
[libx264 @ 0x23e9640] ref P L0: 75.9% 5.1% 11.3% 7.7%
[libx264 @ 0x23e9640] ref B L0: 96.0% 3.1% 0.9%
[libx264 @ 0x23e9640] ref B L1: 95.8% 4.2%
[libx264 @ 0x23e9640] kb/s:895.99Do you have a idea why AVI makes problems ? What could be the solution ?