Newest 'ffmpeg' Questions - Stack Overflow

http://stackoverflow.com/questions/tagged/ffmpeg

Les articles publiés sur le site

  • How to solve issue for converting ismv file of h264 video codec and aac audio codec ?

    4 octobre 2013, par Priyal

    I want to convert transmux ismv file to mp4 format. ISMV file is encoded withe the following ffmpeg detail:

       Metadata:
   major_brand     : isml
   minor_version   : 1
   compatible_brands: piffiso2
   creation_time   : 2013-10-03 14:10:41
   Duration: 15:21:13.16, start: 0.000000, bitrate: 41 kb/s
   Stream #0:0(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 64
                     kb/s (default)
   Metadata:
   creation_time   : 2013-10-03 14:10:41
   handler_name    : Audio
   Stream #0:1(und): Video: h264 (avc1 / 0x31637661), 1280x720, 3217 kb/s, 29.9
                     7 tbr, 10000k tbn, 20000k tbc (default)
Metadata:
   creation_time   : 2013-10-03 14:10:41
   handler_name    : Video
   Stream #0:2(und): Data: none (dfxp / 0x70786664), 32 kb/s (default)
   Metadata:
   creation_time   : 2013-10-03 14:10:41
   handler_name    : Text

    I used the following ffmpeg command to transmux:

     ffmpeg -y -ss 00:00:10 -i Encoder1.ismv -vcodec libx264 -ar 44100 -t 40 -preset slow -qp 0  
 Encoder.mp4

    I got output as:

     ffmpeg version N-56010-g54d628a Copyright (c) 2000-2013 the FFmpeg developers
 built on Sep  4 2013 00:44:27 with gcc 4.7.3 (GCC)
 configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av
 isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab
 le-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetyp
 e --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --ena
 ble-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-l
 ibopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libsp
 eex --enable-libtheora --enable-libtwolame --enable-libvo-aacenc --enable-libvo-
 amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --
 enable-libxvid --enable-zlib
     libavutil      52. 43.100 / 52. 43.100
     libavcodec     55. 31.100 / 55. 31.100
     libavformat    55. 16.100 / 55. 16.100
     libavdevice    55.  3.100 / 55.  3.100
     libavfilter     3. 83.102 /  3. 83.102
     libswscale      2.  5.100 /  2.  5.100
     libswresample   0. 17.103 /  0. 17.103
     libpostproc    52.  3.100 / 52.  3.100
 [aac @ 040f1ba0] TYPE_FIL: Input buffer exhausted before END element found
 [h264 @ 00353120] AVC: nal size -729776398
 [h264 @ 00353120] no frame!
 [h264 @ 00353120] AVC: nal size -570515285
 [h264 @ 00353120] no frame!
 [h264 @ 00353120] AVC: nal size -1477874754
 [h264 @ 00353120] no frame!
 [h264 @ 00353120] AVC: nal size -712314563
 [h264 @ 00353120] no frame!
 [h264 @ 00353120] AVC: nal size -23151524
 [h264 @ 00353120] no frame!
 [h264 @ 00353120] AVC: nal size -592499201
 [h264 @ 00353120] no frame!
 [h264 @ 00353120] AVC: nal size 225768173
 [h264 @ 00353120] no frame!
 [h264 @ 00353120] AVC: nal size 698187359
 [h264 @ 00353120] AVC: nal size 635127544
 [h264 @ 00353120] no frame!
 [h264 @ 00353120] AVC: nal size -1242688339
 [h264 @ 00353120] AVC: nal size 269543071
 [h264 @ 00353120] no frame!
 [mov,mp4,m4a,3gp,3g2,mj2 @ 0035e840] decoding for stream 1 failed
 [mov,mp4,m4a,3gp,3g2,mj2 @ 0035e840] Could not find codec parameters for stream
  1 (Video: h264 (avc1 / 0x31637661), 1280x720, 3217 kb/s): unspecified pixel form
  at Consider increasing the value for the 'analyzeduration' and 'probesize' options
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'C:\inetpub\media\archives\DEFAULT WEB S
  ITE\PushToPUblishPoint\Eagan_12034_Soccer_3-10-2013_14_10_19-isml\2013-10-03-14-
  10-35-436\Segment001\Encoder1.ismv':
   Metadata:
   major_brand     : isml
   minor_version   : 1
   compatible_brands: piffiso2
   creation_time   : 2013-10-03 14:10:41
   Duration: 15:21:13.16, start: 0.000000, bitrate: 41 kb/s
   Stream #0:0(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 64
                     kb/s (default)
    Metadata:
   creation_time   : 2013-10-03 14:10:41
   handler_name    : Audio
   Stream #0:1(und): Video: h264 (avc1 / 0x31637661), 1280x720, 3217 kb/s, 29.9
                     7 tbr, 10000k tbn, 20000k tbc (default)
   Metadata:
   creation_time   : 2013-10-03 14:10:41
   handler_name    : Video
   Stream #0:2(und): Data: none (dfxp / 0x70786664), 32 kb/s (default)
   Metadata:
   creation_time   : 2013-10-03 14:10:41
   handler_name    : Text

 [buffer @ 04679480] Unable to parse option value "-1" as pixel format
               Last message repeated 1 times
 [buffer @ 04679480] Error setting option pix_fmt to value -1.
 [graph 0 input from stream 0:1 @ 040f7380] Error applying options to the filter.

 Error opening filters!

    Please suggest the solution for Windows OS

  • Improve the quality of webm video using ffmpeg

    3 octobre 2013, par Leke

    I'm trying out two codecs and was wondering if I could improve the results of the webm output ...

    ffmpeg -i test.ts -t 10 -f webm -vcodec libvpx output.webm

    to match the quality of the mp4 output...

    ffmpeg -i test.ts -t 10 -b 1500k -vcodec libx264 output.mp4

    What settings would give me the best webm rip?

    Thanks.

    Edit: Here is an error output when trying some rip improvements... 

    ffmpeg -i test.ts -c:v libvpx -crf 4 -b:v 1M -c:a libvorbis output.webm

ffmpeg version 0.8.6-4:0.8.6-0ubuntu0.12.04.1, Copyright (c) 2000-2013 the Libav developers
  built on Apr  2 2013 17:00:59 with gcc 4.6.3
*** THIS PROGRAM IS DEPRECATED ***
This program is only provided for compatibility and will be removed in a future release. Please use avconv instead.
[mpeg2video @ 0x9cde5c0] mpeg_decode_postinit() failure
    Last message repeated 4 times
[mpegts @ 0x9cda240] max_analyze_duration reached
[mpegts @ 0x9cda240] PES packet size mismatch
Input #0, mpegts, from 'test.ts':
  Duration: 00:03:59.53, start: 4630.845422, bitrate: 1977 kb/s
    Stream #0.0[0x20b]: Video: mpeg2video (Main), yuv420p, 720x576 [PAR 64:45 DAR 16:9], 15000 kb/s, 25.60 fps, 25 tbr, 90k tbn, 50 tbc
    Stream #0.1[0x28f]: Audio: mp2, 48000 Hz, stereo, s16, 224 kb/s

Unrecognized option 'c:v'
Failed to set value 'libvpx' for option 'c:v'

  • Android : How to convert WAV/PCM to another audio format without MediaCodec API

    3 octobre 2013, par khelkun

    I've searched around the web and stackoverflow during a long time about this subject. I know it has been discussed many times but I'm looking for a friendly opinion on my use case.

    I have a vocal dictation application which uses android.media.AudioRecord. It allows me to edit the recorded raw PCM buffer: insert a new record inside it, delete a certain period in the record, append a new record at the end. The result is a WAV file, but it's a bit too big file for my needs, so I want to convert it to smaller file format.

    android.media.MediaCodec could have been the solution, but it requires API level 16 which is a too high restriction in my situation.

    It sounds to me that I only have 2 options:

    1. Use the NDK and an android portage of ffmpeg like the guardian project android-ffmpeg-java. But this solution sounds a bit too hard for me so far.

    2. Recording with android.media.MediaRecorder to output an audio 3GPP file encoded with the AMR_NB encoder. Then I'll have to understand this 3GPP file header and the AMR_NB format in details to be able to perform the audio editing operations on my record.

    I must also add that creating a zip file of the WAV file is not a valid option for me.

    Does someone know another option please ? And is option 2. actually possible to achieve ?

    Thanks for your time!

  • How to i convert mp4 and other video formats to 3gpp in c# [on hold]

    3 octobre 2013, par vini

    I am not able to find any resources for conversion of mp4 and other formats into 3gpp

    I know we can use the ffmpeg.exe but can't find any proper resources would be great if some one could help out.

  • Ffmpeg decode H264 video from RTP stream very slow

    3 octobre 2013, par Dmitry Bakhtiyarov

    I have H264 RTP stream over network and I need to decode it. I use libx264 to encode stream and ffmpeg to decode. When I connect to server using VLC, it correct play video without any problem. But when I connect to server using my application I have a long period, when widget, which draw video from this stream, draw only one image.

    I check log file and found, that avcodec_decode_video2() set got_image into 1 very rarely! Decoder give me new decoded frame average every 1-2 seconds, but on the server I have 12-15 fps on encoder!

    What the reasons of this delays on decoder and how I can fix its?

    avcodec_register_all();
av_init_packet(&m_packet);
m_decoder = avcodec_find_decoder(CODEC_ID_H264);
if (!m_decoder)
{
    QString str = QString("Can't find H264 decoder!");
    emit criticalError(str);
}
m_decoderContext = avcodec_alloc_context3(m_decoder);

m_decoderContext->flags |= CODEC_FLAG_LOW_DELAY;
m_decoderContext->flags2 |= CODEC_FLAG2_CHUNKS;
AVDictionary* dictionary = nullptr;
if (avcodec_open2(m_decoderContext, m_decoder, &dictionary) < 0)
{
    QString str = QString("Failed to open decoder!");
    emit criticalError(str);
}
qDebug() << "H264 Decoder successfully opened";
m_picture = avcodec_alloc_frame();
...
    while(m_packet.size > 0)
    {
        int got_picture;
        int len = avcodec_decode_video2(m_decoderContext, m_picture, &got_picture, &m_packet);
        if (len < 0)
        {
            QString err("Decoding error");
            qDebug() << err;
            //emit criticalError(err);
            return false;
        }
        if (got_picture)
        {
                //create from frame QImage and send it into GUI thread
            qDebug() << "H264Decoder: frame decoded!";

    I try to change some options of m_decoderContext (i.e. thread_count) but this not changing anything.

Navigation

Sites Web