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  • ffmpeg enciding, Opus sound in the webm container does not work

    2 juillet 2017, par Mockarutan

    I’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 &lt; 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 &lt; 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(&amp;outctx, NULL, compressed_cont, full_filename);

           free(with_dot);

           if (ret &lt; 0)
           {
               free(full_filename);
               return ERROR_CONTEXT_CREATION;
           }

           ret = setup_audio_codec();
           if (ret &lt; 0)
               return ret;

           // Setup Audio
           ret = initialize_audio_stream(outctx, sample_rate, per_frame_audio_samples, audio_bitrate);
           if (ret &lt; 0)
               return ret;

           av_dump_format(outctx, 0, full_filename, 1);

           if (!(outctx->oformat->flags &amp; AVFMT_NOFILE))
           {
               if (avio_open(&amp;outctx->pb, full_filename, AVIO_FLAG_WRITE) &lt; 0)
               {
                   free(full_filename);
                   return ERROR_OPENING_FILE;
               }
           }

           free(full_filename);

           ret = avformat_write_header(outctx, NULL);
           if (ret &lt; 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(&amp;pkt);
           pkt.data = NULL;
           pkt.size = 0;

           while (true)
           {
               ret = avcodec_receive_packet(aud_codec_context, &amp;pkt);
               if (!ret)
               {
                   av_packet_rescale_ts(&amp;pkt, aud_codec_context->time_base, audio_st->time_base);

                   pkt.stream_index = audio_st->index;

                   av_interleaved_write_frame(outctx, &amp;pkt);

                   av_packet_unref(&amp;pkt);
               }
               if (ret == AVERROR(EAGAIN))
                   break;
               else if (ret &lt; 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 &lt; 0)
               return ERROR_ENCODING_FRAME_SEND;

           AVPacket pkt;
           av_init_packet(&amp;pkt);
           pkt.data = NULL;
           pkt.size = 0;

           fflush(stdout);

           while (true)
           {
               ret = avcodec_receive_packet(aud_codec_context, &amp;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, &amp;pkt);
                   av_packet_unref(&amp;pkt);
               }
               if (ret == AVERROR(EAGAIN))
                   break;
               else if (ret &lt; 0)
                   return ERROR_ENCODING_FRAME_RECEIVE;
               else
                   break;
           }

           return ENCODED_AUDIO;
       }

       int finish_audio_encoding()
       {
           AVPacket pkt;
           av_init_packet(&amp;pkt);
           pkt.data = NULL;
           pkt.size = 0;

           fflush(stdout);

           int ret = avcodec_send_frame(aud_codec_context, NULL);
           if (ret &lt; 0)
               return ERROR_ENCODING_FRAME_SEND;

           while (true)
           {
               ret = avcodec_receive_packet(aud_codec_context, &amp;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, &amp;pkt);
                   av_packet_unref(&amp;pkt);
               }
               if (ret == -AVERROR(AVERROR_EOF))
                   break;
               else if (ret &lt; 0)
                   return ERROR_ENCODING_FRAME_RECEIVE;
           }

           av_write_trailer(outctx);

           return 0;
       }

       void cleanup()
       {
           if (aud_frame)
           {
               av_frame_free(&amp;aud_frame);
           }
           if (outctx)
           {
               for (int i = 0; i &lt; outctx->nb_streams; i++)
                   av_freep(&amp;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 &lt; nb_samples; i++) {
               *dst = sin(c * *t);
               for (j = 1; j &lt; 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(&amp;src_data, &amp;src_samples_linesize, src_channels,
               src_nb_samples, AV_SAMPLE_FMT_FLT, 0);

           for (size_t i = 0; i &lt; frame_rate * sec; i++)
           {
                   fill_samples((float *)src_data[0], src_nb_samples, src_channels, sample_rate, &amp;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=sharing

    The ogg audio file that works :
    https://drive.google.com/file/d/0B16rIXjPXJCqMUZhbW0tTDFjT1E/view?usp=sharing

    Video 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 !

  • no video above 360P with sound ytdl-core

    26 juin 2021, par Ashish

    I am able to download youtube separate videos and audios. but merging not working. 0 KB file is downloading

    &#xA;

    let info = await ytdl.getInfo(req.body.videoId);&#xA;let formatVideo = ytdl.chooseFormat(info.formats, { quality: req.body.format.itag, filter: &#x27;videoonly&#x27; });&#xA;let formatAudio = ytdl.chooseFormat(info.formats, { quality: req.body.audioTag });&#xA;// res.json({video: formatVideo, audio: formatAudio});&#xA;const tracker = {&#xA;  start: Date.now(),&#xA;  audio: { downloaded: 0, total: Infinity },&#xA;  video: { downloaded: 0, total: Infinity },&#xA;  merged: { frame: 0, speed: &#x27;0x&#x27;, fps: 0 },&#xA;};&#xA;res.header(&#x27;Content-Disposition&#x27;, `attachment; filename=${fileName}`);&#xA;&#xA;cp.exec(`ffmpeg -i ${videoStream} -i ${audioStream} -c copy ${fileName}`, (error, success) => {&#xA;  if (!error) {&#xA;    console.log(success);&#xA;  } else {&#xA;    console.log(error);&#xA;  }&#xA;});&#xA;// Prepare the progress bar&#xA;let progressbarHandle = null;&#xA;const progressbarInterval = 1000;&#xA;const showProgress = () => {&#xA;  console.log(&#x27;test&#x27;);&#xA;  readline.cursorTo(process.stdout, 0);&#xA;  const toMB = i => (i / 1024 / 1024).toFixed(2);&#xA;&#xA;  process.stdout.write(`Audio  | ${(tracker.audio.downloaded / tracker.audio.total * 100).toFixed(2)}% processed `);&#xA;  process.stdout.write(`(${toMB(tracker.audio.downloaded)}MB of ${toMB(tracker.audio.total)}MB).${&#x27; &#x27;.repeat(10)}\n`);&#xA;&#xA;  process.stdout.write(`Video  | ${(tracker.video.downloaded / tracker.video.total * 100).toFixed(2)}% processed `);&#xA;  process.stdout.write(`(${toMB(tracker.video.downloaded)}MB of ${toMB(tracker.video.total)}MB).${&#x27; &#x27;.repeat(10)}\n`);&#xA;&#xA;  process.stdout.write(`Merged | processing frame ${tracker.merged.frame} `);&#xA;  process.stdout.write(`(at ${tracker.merged.fps} fps => ${tracker.merged.speed}).${&#x27; &#x27;.repeat(10)}\n`);&#xA;&#xA;  process.stdout.write(`running for: ${((Date.now() - tracker.start) / 1000 / 60).toFixed(2)} Minutes.`);&#xA;  readline.moveCursor(process.stdout, 0, -3);&#xA;};&#xA;&#xA;// Start the ffmpeg child process&#xA;const ffmpegProcess = cp.spawn(ffmpeg, [&#xA;  // Remove ffmpeg&#x27;s console spamming&#xA;  &#x27;-loglevel&#x27;, &#x27;8&#x27;, &#x27;-hide_banner&#x27;,&#xA;  // Redirect/Enable progress messages&#xA;  &#x27;-progress&#x27;, &#x27;pipe:3&#x27;,&#xA;  // Set inputs&#xA;  &#x27;-i&#x27;, &#x27;pipe:4&#x27;,&#xA;  &#x27;-i&#x27;, &#x27;pipe:5&#x27;,&#xA;  // Map audio &amp; video from streams&#xA;  &#x27;-map&#x27;, &#x27;0:a&#x27;,&#xA;  &#x27;-map&#x27;, &#x27;1:v&#x27;,&#xA;  // Keep encoding&#xA;  &#x27;-c:v&#x27;, &#x27;copy&#x27;,&#xA;  // Define output file&#xA;  `${fileName}`,&#xA;], {&#xA;  windowsHide: true,&#xA;  stdio: [&#xA;    /* Standard: stdin, stdout, stderr */&#xA;    &#x27;inherit&#x27;, &#x27;inherit&#x27;, &#x27;inherit&#x27;,&#xA;    /* Custom: pipe:3, pipe:4, pipe:5 */&#xA;    &#x27;pipe&#x27;, &#x27;pipe&#x27;, &#x27;pipe&#x27;, &#x27;pipe&#x27;&#xA;  ],&#xA;});&#xA;ffmpegProcess.on(&#x27;close&#x27;, () => {&#xA;  console.log(&#x27;done&#x27;);&#xA;  // Cleanup&#xA;  process.stdout.write(&#x27;\n\n\n\n&#x27;);&#xA;  clearInterval(progressbarHandle);&#xA;  console.log(tracker, &#x27;146&#x27;);&#xA;});&#xA;&#xA;// Link streams&#xA;// FFmpeg creates the transformer streams and we just have to insert / read data&#xA;ffmpegProcess.stdio[3].on(&#x27;data&#x27;, chunk => {&#xA;  console.log(chunk, &#x27;152&#x27;);&#xA;  // Start the progress bar&#xA;  if (!progressbarHandle) progressbarHandle = setInterval(showProgress, progressbarInterval);&#xA;  // Parse the param=value list returned by ffmpeg&#xA;  const lines = chunk.toString().trim().split(&#x27;\n&#x27;);&#xA;  const args = {};&#xA;  for (const l of lines) {&#xA;    const [key, value] = l.split(&#x27;=&#x27;);&#xA;    args[key.trim()] = value.trim();&#xA;  }&#xA;  tracker.merged = args;&#xA;  // console.log(tracker.merged, &#x27;162&#x27;);&#xA;});&#xA;// audio.pipe(ffmpegProcess.stdio[4]);&#xA;// video.pipe(ffmpegProcess.stdio[5]);&#xA;const audioStream = await requestObj.get(formatAudio.url);&#xA;const videoStream = await requestObj.get(formatVideo.url);&#xA;audioStream.pipe(ffmpegProcess.stdio[4]);&#xA;videoStream.pipe(ffmpegProcess.stdio[5]);&#xA;ffmpegProcess.stdio[6].pipe(res);&#xA;

    &#xA;

    here requestObj is a node js 'request' module

    &#xA;

  • FFmpeg AVI to MP4 preserves sound, but not images [migrated]

    21 février 2013, par user1711384

    Working 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>&amp;1

    With 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 &#39;testdatei.avi&#39;:
     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 &#39;logo.jpg&#39;:
     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 &#39;xxx.mp4&#39;:
     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:895.99

    Output2 :

    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 &#39;testdatei.avi&#39;:
     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 &#39;logo.jpg&#39;:
     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 &#39;xxx.mp4&#39;:
     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:895.99

    Do you have a idea why AVI makes problems ? What could be the solution ?