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• FFMpeg Concatenation Filters : Stream specifier ':0' in filtergraph matches no streams

  7 février 2017, par Anthony Eden

  I am developing an application that relies heavily on FFMpeg to perform various transformations on audio files. I am currently testing my FFMpeg configuration on the command line.

  I am trying to concatenate multiple audio files which are in different formats (Primarily MP3, MP2 & WAV). I have been using the official TRAC documentation (https://trac.ffmpeg.org/wiki/How%20to%20concatenate%20(join%2C%20merge)%20media%20files#differentcodec) to help me with this and have created the following command :

  ffmpeg -i OHIn.wav -i OHOut.wav -filter_complex '[0:0] [1:0] concat=n=2:a=1 [a]' -map '[a]' output.wav

  However, when I run this on Mac OS X using version 2.0.1 of FFMpeg, I get the following error message :

  Stream specifier ':0' in filtergraph description [0:0] [1:0] concat=n=2:a=1 [a] matches no streams.

  Here is my full output from the terminal :

  ~/ffmpeg -i OHIn.wav -i OHOut.wav -filter_complex '[0:0] [1:0] concat=n=2:a=1 [a]' -map '[a]' output.wav
  
  
  
  ffmpeg version 2.0.1 Copyright (c) 2000-2013 the FFmpeg developers
  
    built on Aug 15 2013 10:56:46 with llvm-gcc 4.2.1 (LLVM build 2336.11.00)
  
    configuration: --prefix=/Volumes/Ramdisk/sw --enable-gpl --enable-pthreads --enable-version3 --enable-libspeex --enable-libvpx --disable-decoder=libvpx --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libx264 --enable-avfilter --enable-libopencore_amrwb --enable-libopencore_amrnb --enable-filters --enable-libgsm --arch=x86_64 --enable-runtime-cpudetect
  
    libavutil      52. 38.100 / 52. 38.100
  
    libavcodec     55. 18.102 / 55. 18.102
  
    libavformat    55. 12.100 / 55. 12.100
  
    libavdevice    55.  3.100 / 55.  3.100
  
    libavfilter     3. 79.101 /  3. 79.101
  
    libswscale      2.  3.100 /  2.  3.100
  
    libswresample   0. 17.102 /  0. 17.102
  
    libpostproc    52.  3.100 / 52.  3.100
  
  Guessed Channel Layout for  Input Stream #0.0 : stereo
  
  Input #0, wav, from 'OHIn.wav':
  
    Duration: 00:00:06.71, bitrate: 1411 kb/s
  
      Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
  
  Guessed Channel Layout for  Input Stream #1.0 : stereo
  
  Input #1, wav, from 'OHOut.wav':
  
    Duration: 00:00:07.19, bitrate: 1411 kb/s
  
      Stream #1:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
  
  Stream specifier ':0' in filtergraph description [0:0] [1:0] concat=n=2:a=1 [a] matches no streams.

  I do not understand why this does not work. FFMpeg shows that the streams 0:0 and 1:0 exist in the source files. The only other similar problems online have surrounded the use of the single quote in Windows, however testing of this confirm it does not apply to my Mac command line.

  Any help would be much appreciated.

• ffmpeg error on decode

  25 octobre 2013, par ademar111190

  I'm developing an android app with the libav and I'm trying decode a 3gp with code below :

  #define simbiLog(...) __android_log_print(ANDROID_LOG_DEBUG, "simbiose", __VA_ARGS__)

  ...

  AVCodec *codec;
  
  AVCodecContext *c = NULL;
  
  int len;
  
  FILE *infile, *outfile;
  
  uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
  
  AVPacket avpkt;
  
  AVFrame *decoded_frame = NULL;
  
  
  
  simbiLog("inbuf size: %d", sizeof(inbuf) / sizeof(inbuf[0]));
  
  
  
  av_register_all();
  
  av_init_packet(&avpkt);
  
  
  
  codec = avcodec_find_decoder(AV_CODEC_ID_AMR_NB);
  
  if (!codec) {
  
      simbiLog("codec not found");
  
      return ERROR;
  
  }
  
  
  
  c = avcodec_alloc_context3(codec);
  
  if (!c) {
  
      simbiLog("Could not allocate audio codec context");
  
      return ERROR;
  
  }
  
  
  
  int open = avcodec_open2(c, codec, NULL);
  
  if (open < 0) {
  
      simbiLog("could not open codec %d", open);
  
      return ERROR;
  
  }
  
  
  
  infile = fopen(inputPath, "rb");
  
  if (!infile) {
  
      simbiLog("could not open %s", inputPath);
  
      return ERROR;
  
  }
  
  
  
  outfile = fopen(outputPath, "wb");
  
  if (!outfile) {
  
      simbiLog("could not open %s", outputPath);
  
      return ERROR;
  
  }
  
  
  
  avpkt.data = inbuf;
  
  avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, infile);
  
  int iterations = 0;
  
  
  
  while (avpkt.size > 0) {
  
      simbiLog("iteration %d", (++iterations));
  
      simbiLog("avpkt.size %d avpkt.data %X", avpkt.size, avpkt.data);
  
      int got_frame = 0;
  
  
  
      if (!decoded_frame) {
  
          if (!(decoded_frame = avcodec_alloc_frame())) {
  
              simbiLog("out of memory");
  
              return ERROR;
  
          }
  
      } else {
  
          avcodec_get_frame_defaults(decoded_frame);
  
      }
  
  
  
      //below the error, but it isn't occur on first time, only in 4th loop interation
  
      len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt); 
  
      if (len < 0) {
  
          simbiLog("Error while decoding error %d frame %d duration %d", len, got_frame, avpkt.duration);
  
          return ERROR;
  
      } else {
  
          simbiLog("Decoding length %d frame %d duration %d", len, got_frame, avpkt.duration);
  
      }
  
  
  
      if (got_frame) {
  
          int data_size = av_samples_get_buffer_size(NULL, c->channels, decoded_frame->nb_samples, c->sample_fmt, 1);
  
          size_t* fwrite_size = fwrite(decoded_frame->data[0], 1, data_size, outfile);
  
          simbiLog("fwrite returned %d", fwrite_size);
  
      }
  
      avpkt.size -= len;
  
      avpkt.data += len;
  
      if (avpkt.size < AUDIO_REFILL_THRESH) {
  
          memmove(inbuf, avpkt.data, avpkt.size);
  
          avpkt.data = inbuf;
  
          len = fread(avpkt.data + avpkt.size, 1, AUDIO_INBUF_SIZE - avpkt.size, infile);
  
          if (len > 0)
  
              avpkt.size += len;
  
          simbiLog("fread returned %d", len);
  
      }
  
  }
  
  
  
  fclose(outfile);
  
  fclose(infile);
  
  
  
  avcodec_close(c);
  
  av_free(c);
  
  av_free(decoded_frame);

  but I'm getting the follow log and error :

  inbuf size: 20488
  
  iteration 1
  
  avpkt.size 3305 avpkt.data BEEED40C
  
  Decoding length 13 frame 1 duration 0
  
  fwrite returned 640
  
  fread returned 0
  
  iteration 2
  
  avpkt.size 3292 avpkt.data BEEED40C
  
  Decoding length 13 frame 1 duration 0
  
  fwrite returned 640
  
  fread returned 0
  
  iteration 3
  
  avpkt.size 3279 avpkt.data BEEED40C
  
  Decoding length 14 frame 1 duration 0
  
  fwrite returned 640
  
  fread returned 0
  
  iteration 4
  
  avpkt.size 3265 avpkt.data BEEED40C
  
  Error while decoding error -1052488119 frame 0 duration 0

  the audio file I'm trying decode :

  $ avprobe blue.3gp 
  
  avprobe version 0.8.6-6:0.8.6-1ubuntu2, Copyright (c) 2007-2013 the Libav developers
  
    built on Mar 30 2013 22:23:21 with gcc 4.7.2
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'blue.3gp':
  
    Metadata:
  
      major_brand     : 3gp4
  
      minor_version   : 0
  
      compatible_brands: isom3gp4
  
      creation_time   : 2013-09-19 18:53:38
  
    Duration: 00:00:01.52, start: 0.000000, bitrate: 17 kb/s
  
      Stream #0.0(eng): Audio: amrnb, 8000 Hz, 1 channels, flt, 12 kb/s
  
      Metadata:
  
        creation_time   : 2013-09-19 18:53:38

  thanks a lot !

  ---

  EDITED

  I read on ffmper documentation about the method avcodec_decode_audio4 the follow :

  @warning The input buffer, avpkt->data must be FF_INPUT_BUFFER_PADDING_SIZE larger than the actual read bytes because some optimized bitstream readers read 32 or 64 bits at once and could read over the end.
  
  @note You might have to align the input buffer. The alignment requirements depend on the CPU and the decoder.

  and I see here a solution using posix_memalign, to android i founded a similar method called memalign, so i did the change :

  removed :

  uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];

  inserted :

  int inbufSize = sizeof(uint8_t) * (AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE);
  
  uint8_t *inbuf = memalign(FF_INPUT_BUFFER_PADDING_SIZE, inbufSize);
  
  simbiLog("inbuf size: %d", inbufSize);
  
  for (; inbufSize >= 0; inbufSize--)
  
      simbiLog("inbuf position: %d index: %p", inbufSize, &inbuf[inbufSize]);

  I'm getting the correct memory sequence position, but the error not changed.

  A piece of outpout :

  inbuf position: 37 index: 0x4e43d745
  
  inbuf position: 36 index: 0x4e43d744
  
  inbuf position: 35 index: 0x4e43d743
  
  inbuf position: 34 index: 0x4e43d742
  
  inbuf position: 33 index: 0x4e43d741
  
  inbuf position: 32 index: 0x4e43d740
  
  inbuf position: 31 index: 0x4e43d73f
  
  inbuf position: 30 index: 0x4e43d73e
  
  inbuf position: 29 index: 0x4e43d73d
  
  inbuf position: 28 index: 0x4e43d73c
  
  inbuf position: 27 index: 0x4e43d73b
  
  inbuf position: 26 index: 0x4e43d73a
  
  inbuf position: 25 index: 0x4e43d739
  
  inbuf position: 24 index: 0x4e43d738
  
  inbuf position: 23 index: 0x4e43d737
  
  inbuf position: 22 index: 0x4e43d736
  
  inbuf position: 21 index: 0x4e43d735
  
  inbuf position: 20 index: 0x4e43d734
  
  inbuf position: 19 index: 0x4e43d733

• When merging video and audio, converting audio to AAC, sound is missed at the end 3 seconds

  10 octobre 2013, par profuel

  I have to build video from images and some audio clip.
  Audio is much longer, so I have to mute last 5 seconds of audio track, cutting to video length.
  My issue is that adding AAC encoding to audio removes last 2-5 seconds of audio in resulted video.
  
  Here are my command lines :

  ffmpeg -i sound.mp3 -i video.mp4 -shortest out.mp4 -> results in correct audio in result video with played audio over 100% of movie

  ffmpeg -i sound.mp3 -i video.mp4 -acodec aac -ab 160000 -strict experimental -shortest out.mp4 -> not correct audio, gets crop at end of video for 2-5 seconds
  
  
  The problem appears for me both on Windows and on CentOS 6.4, no matter which version of ffmpeg is used.
  
  FFMPEG details (downloaded from http://ffmpeg.gusari.org/static/64bit/ffmpeg.static.64bit.2013-06-01.tar.gz )
  ffmpeg version N-53724-g716dbc7 Copyright (c) 2000-2013 the FFmpeg developers
  built on Jun 1 2013 05:26:08 with gcc 4.6 (Debian 4.6.3-1)
  configuration : —prefix=/root/ffmpeg-static/64bit —extra-cflags='-I/root/ffmpeg-static/64bit/include -static' —extra-ldflags='-L/root/ffmpeg-static/64bit/lib -static' —extra-libs='-lxml2 -lexpat -lfreetype' —enable-static —disable-shared —disable-ffserver —disable-doc —enable-bzlib —enable-zlib —enable-postproc —enable-runtime-cpudetect —enable-libx264 —enable-gpl —enable-libtheora —enable-libvorbis —enable-libmp3lame —enable-gray —enable-libass —enable-libfreetype —enable-libopenjpeg —enable-libspeex —enable-libvo-aacenc —enable-libvo-amrwbenc —enable-version3 —enable-libvpx
  libavutil 52. 34.100 / 52. 34.100
  libavcodec 55. 12.102 / 55. 12.102
  libavformat 55. 8.102 / 55. 8.102
  libavdevice 55. 2.100 / 55. 2.100
  libavfilter 3. 73.100 / 3. 73.100
  libswscale 2. 3.100 / 2. 3.100
  libswresample 0. 17.102 / 0. 17.102
  libpostproc 52. 3.100 / 52. 3.100