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• How to join webcam FLVs

  18 mars 2015, par Marc-André Lafortune

  I want my website to join some webcam recordings in FLV files (like this one). This needs to be done on Linux without user input. How do I do this ? For simplicity’s sake, I’ll use the same flv as both inputs in hope of getting a flv that plays the same thing twice in a row.

  That should be easy enough, right ? There’s even a full code example in the ffmpeg FAQ.

  Well, pipes seem to be giving me problems (both on my mac running Leopard and on Ubuntu 8.04) so let’s keep it simple and use normal files. Also, if I don’t specify a rate of 15 fps, the visual part plays extremely fast. The example script thus becomes :

  ffmpeg -i input.flv -vn -f u16le -acodec pcm_s16le -ac 2 -ar 44100 \
  
    - > temp.a < /dev/null
  
  ffmpeg -i input.flv -an -f yuv4mpegpipe - > temp.v < /dev/null
  
  cat temp.v temp.v > all.v
  
  cat temp.a temp.a > all.a
  
  ffmpeg -f u16le -acodec pcm_s16le -ac 2 -ar 44100 -i all.a \
  
    -f yuv4mpegpipe -i all.v -sameq -y output.flv

  Well, using this will work for the audio, but I only get the video the first time around. This seems to be the case for any flv I throw as input.flv, including the movie teasers that come with red5.

  a) Why doesn’t the example script work as advertised, in particular why do I not get all the video I’m expecting ?

  b) Why do I have to specify a framerate while Wimpy player can play the flv at the right speed ?

  The only way I found to join two flvs was to use mencoder. Problem is, mencoder doesn’t seem to join flvs :

  mencoder input.flv input.flv -o output.flv -of lavf -oac copy \
  
  -ovc lavc -lavcopts vcodec=flv

  I get a Floating point exception...

  MEncoder 1.0rc2-4.0.1 (C) 2000-2007 MPlayer Team
  
  CPU: Intel(R) Xeon(R) CPU 5150 @ 2.66GHz (Family: 6, Model: 15, Stepping: 6)
  
  CPUflags: Type: 6 MMX: 1 MMX2: 1 3DNow: 0 3DNow2: 0 SSE: 1 SSE2: 1
  
  Compiled for x86 CPU with extensions: MMX MMX2 SSE SSE2
  
  
  
  success: format: 0 data: 0x0 - 0x45b2f
  
  libavformat file format detected.
  
  [flv @ 0x697160]Unsupported audio codec (6)
  
  [flv @ 0x697160]Could not find codec parameters (Audio: 0x0006, 22050 Hz, mono)
  
  [lavf] Video stream found, -vid 0
  
  [lavf] Audio stream found, -aid 1
  
  VIDEO: [FLV1] 240x180 0bpp 1000.000 fps 0.0 kbps ( 0.0 kbyte/s)
  
  [V] filefmt:44 fourcc:0x31564C46 size:240x180 fps:1000.00 ftime:=0.0010
  
  ** MUXER_LAVF *****************************************************************
  
  REMEMBER: MEncoder's libavformat muxing is presently broken and can generate
  
  INCORRECT files in the presence of B frames. Moreover, due to bugs MPlayer
  
  will play these INCORRECT files as if nothing were wrong!
  
  *******************************************************************************
  
  OK, exit
  
  Opening video filter: [expand osd=1]
  
  Expand: -1 x -1, -1 ; -1, osd: 1, aspect: 0.000000, round: 1
  
  ==========================================================================
  
  Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec family
  
  Selected video codec: [ffflv] vfm: ffmpeg (FFmpeg Flash video)
  
  ==========================================================================
  
  audiocodec: framecopy (format=6 chans=1 rate=22050 bits=16 B/s=0 sample-0)
  
  VDec: vo config request - 240 x 180 (preferred colorspace: Planar YV12)
  
  VDec: using Planar YV12 as output csp (no 0)
  
  Movie-Aspect is undefined - no prescaling applied.
  
  videocodec: libavcodec (240x180 fourcc=31564c46 [FLV1])
  
  VIDEO CODEC ID: 22
  
  AUDIO CODEC ID: 10007, TAG: 0
  
  Writing header...
  
  [NULL @ 0x67d110]codec not compatible with flv
  
  Floating point exception

  c) Is there a way for mencoder to decode and encode flvs correctly ?

  So the only way I’ve found so far to join flvs, is to use ffmpeg to go back and forth between flv and avi, and use mencoder to join the avis :

  ffmpeg -i input.flv -vcodec rawvideo -acodec pcm_s16le -r 15 file.avi
  
  mencoder -o output.avi -oac copy -ovc copy -noskip file.avi file.avi
  
  ffmpeg -i output.avi output.flv

  d) There must be a better way to achieve this... Which one ?

  e) Because of the problem of the framerate, though, only flvs with constant framerate (like the one I recorded through facebook) will be converted correctly to avis, but this won’t work for the flvs I seem to be recording (like this one or this one). Is there a way to do this for these flvs too ?

  Any help would be very appreciated.

• How to encode 24-bit audio with libav/ffmpeg ?

  18 mars 2015, par andrewrk

  Here’s a code snippet from libavutil/samplefmt.h :

  /**
  
   * Audio Sample Formats
  
   *
  
   * @par
  
   * The data described by the sample format is always in native-endian order.
  
   * Sample values can be expressed by native C types, hence the lack of a signed
  
   * 24-bit sample format even though it is a common raw audio data format.
  
   *
  
   * @par
  
   * The floating-point formats are based on full volume being in the range
  
   * [-1.0, 1.0]. Any values outside this range are beyond full volume level.
  
   *
  
   * @par
  
   * The data layout as used in av_samples_fill_arrays() and elsewhere in Libav
  
   * (such as AVFrame in libavcodec) is as follows:
  
   *
  
   * @par
  
   * For planar sample formats, each audio channel is in a separate data plane,
  
   * and linesize is the buffer size, in bytes, for a single plane. All data
  
   * planes must be the same size. For packed sample formats, only the first data
  
   * plane is used, and samples for each channel are interleaved. In this case,
  
   * linesize is the buffer size, in bytes, for the 1 plane.
  
   */
  
  enum AVSampleFormat {
  
      AV_SAMPLE_FMT_NONE = -1,
  
      AV_SAMPLE_FMT_U8,          ///< unsigned 8 bits
  
      AV_SAMPLE_FMT_S16,         ///< signed 16 bits
  
      AV_SAMPLE_FMT_S32,         ///< signed 32 bits
  
      AV_SAMPLE_FMT_FLT,         ///< float
  
      AV_SAMPLE_FMT_DBL,         ///< double
  
  
  
      AV_SAMPLE_FMT_U8P,         ///< unsigned 8 bits, planar
  
      AV_SAMPLE_FMT_S16P,        ///< signed 16 bits, planar
  
      AV_SAMPLE_FMT_S32P,        ///< signed 32 bits, planar
  
      AV_SAMPLE_FMT_FLTP,        ///< float, planar
  
      AV_SAMPLE_FMT_DBLP,        ///< double, planar
  
  
  
      AV_SAMPLE_FMT_NB           ///< Number of sample formats. DO NOT USE if linking dynamically
  
  };

  It specifically mentions that 24-bit is missing even though it is a common raw audio data format. So if I were using libav/ffmpeg to export to an audio file, how would I use 24-bit audio ?

  Exporting an audio file looks something like this :

  AVCodec *codec = get_codec();
  
  AVOutputFormat *oformat = get_output_format();
  
  AVFormatContext *fmt_ctx = avformat_alloc_context();
  
  assert(fmt_ctx);
  
  int err = avio_open(&fmt_ctx->pb, get_output_filename(), AVIO_FLAG_WRITE);
  
  assert(err >= 0);
  
  fmt_ctx->oformat = oformat;
  
  AVStream *stream = avformat_new_stream(fmt_ctx, codec);
  
  assert(stream);
  
  AVCodecContext *codec_ctx = stream->codec;
  
  codec_ctx->bit_rate = get_export_bit_rate();
  
  
  
  // How to set this to 24 bit instead of 32?
  
  codec_ctx->sample_fmt = AV_SAMPLE_FMT_S32;
  
  
  
  codec_ctx->sample_rate = get_sample_rate();
  
  codec_ctx->channel_layout = get_channel_layout()
  
  codec_ctx->channels = get_channel_count();
  
  codec_ctx->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;

• avfilter/vf_mpdecimate : support more pixel formats, including GBRP

  4 mars 2015, par Peter Cordes

  avfilter/vf_mpdecimate : support more pixel formats, including GBRP
  Add some planar formats we can handle without distorting the meaning of
  
  hi and lo.
  Signed-off-by : Michael Niedermayer <michaelni@gmx.at>
  

  • [DH] libavfilter/vf_mpdecimate.c