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• Streaming protocol relay without involving codec

  4 décembre 2015, par kiran_g

  I am trying to use libav to relay an RTSP stream. It involves PULLing the stream from an IP camera and then PUSHing to wowza.

  The video encoding in the IP camera stream is h264. To enable h264 in my libav application I need to enable x264. But as x264 is GPL, it will not work with my business plan.

  My questions is whether libav (ffmpeg) can be made to work like a dumb relay which is encoding-agnostic? so that I dont need to integrate x264 with ffmpeg.

  This SO post says that I can use the "copy" argument, but does that allow me to exclude x264?

  BTW, is x264 actually needed by ffmpeg for decoding h264? Is x264 only used in encoding?

  See here for my current code.

• avconv complains "non monotonically increasing dts to muxer in stream" when piping from libx264 on 14.04

  1er décembre 2015, par Mike

  I have a piece of code that takes a stream of images and writes them to a file using x264 to encode and avconv. The relevant bits are as follows

  // Setup encoder
  sprintf(cmd, "avconv -i pipe: -vcodec copy %s", filename);
  fp = popen(cmd, "w");
  x264_param_default_preset(&params, "ultrafast", "stillimage,zerolatency");
  params.i_fps_num = fps;
  params.i_fps_den = 1;
  x264_picture_alloc(&in, X264_CSP_I420, width, height);
  params.i_csp = X264_CSP_I420;
  in.img.i_csp = X264_CSP_I420;
  x.params.b_vfr_input = 0;
  in.i_pts = -1;
  out.i_pts = -1;
  params.i_width = width;
  params.i_height = height;
  encoder = x264_encoder_open(&params);
  
  in.img.i_plane = 1;
  in.img.i_stride[0] = width;
  ret = x264_encoder_headers(encoder, &nals, &nheader);
  header_size = nals[0].i_payload + nals[1].i_payload + nals[2].i_payload;
  fwrite(nals[0].p_payload, header_size, 1, fp);
  
  ...
  // Loop body
  ++in.i_pts;
  frame_size = x264_encoder_encode(encoder, &nals, &num_nals, &in, &out);
  fwrite(nals[0].p_payload, frame_size, 1, fp);
  

  I've left out error checking for clarity, but no error is ever returned. Indeed, until I upgraded my system to 14.04, this code worked perfectly (and continues to work perfectly for colleagues who've yet to upgrade).

  Only now running Ubuntu 14.04, I get this output from avconv

  [h264 @ 0x98dec0] Estimating duration from bitrate, this may be inaccurate
  Input #0, h264, from 'pipe:'
    Duration: N/A, bitrate: N/A
      Stream #0.0: Video: h264 (Constrained Baseline), yuv420p, 640x480, 25 fps, 25 tbr, 25 tbn, 60 tbc
  Output #0, mp4, to '../reel/test.mp4':
    Metadata:
      encoder         : Lavf54.20.3
      Stream #0.0: Video: libx264, yuv420p, 640x480, q=2-31, 25 tbn, 25 tbc
  Stream mapping:
    Stream #0:0 -> #0:0 (copy)
  [mp4 @ 0x1347800] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 2 >= 2
  av_interleaved_write_frame(): Invalid argument
  

  Despite the error, out.i_dts does monotonically increase over the whole duration of the video (it's always equal to in.i_pts).

  My general assumption is that newer versions of avconv are more strict with the parameters of the input stream, so while my choice of parameters may have worked before, they do not work now.

  Another curiousity, which might be related, is avconv detecting that the stream is 25 FPS, even though x264 is configured with an input FPS of 30.

  Edit: Some further information, the same dts error (2>=2) happens regardless of the initial pts (with the output dts matching as expected).

• Building x264 on a Mac for android “No working C compiler found”

  20 novembre 2015, par LostPuppy

  I am trying to build x264 and eventually use it to build ffmpeg the following is my script:

  #!/bin/bash
  
  NDK="/Users/account/Documents/FFmpeg/android-ndk-r10e"
  TOOL_DIR_ARM="/Users/account/Documents/FFmpeg/android-ndk-r10e/sources/ffmpeg-2.8.2/my-android-toolchain"
  SYSROOT=$NDK/platforms/platforms/android-21/arch-arm
  echo "removing tool chain directory if it exists"
  rm -r $TOOL_DIR_ARM
  
  $NDK/build/tools/make-standalone-toolchain.sh \
      --platform=android-21 \
      --arch=arm \
      --install-dir=$TOOL_DIR_ARM \
      --toolchain=arm-linux-androideabi-clang3.6\
      --llvm-version=3.6 \
      --system=darwin-x86_64
  
  if [ -d $SYSROOT ]; then
      echo "$SYSROOT is a valid directory"
  fi
  
  function build_x264
  {
  cd x264
  
  # another build for phone
  ./configure \
  --cross-prefix=arm-linux-androideabi- \
  --sysroot=$SYSROOT \
  --host=arm-linux \
  --enable-static \
  --enable-pic \
  --disable-cli 
  
  make -j4
  make install
  
  cd ../
  }
  
  
  function build_one
  {
  cd 
  
  # another build for phone
  ./configure \
  --sysroot=$SYSROOT \
  --target-os=linux \
  --arch=arm \
  --cross-prefix=$TOOL_DIR_ARM/bin/arm-linux-androideabi- \
  --cc=$TOOL_DIR_ARM/bin/clang \
  --enable-cross-compile \
  --enable-encoder=mpeg4 \
  --enable-pthreads \
  --enable-gpl \
  --enable-libx264 \
  --enable-pic \
  --extra-cflags='-I/usr/local/include' \
  --extra-ldflags='-L/usr/local/lib'
  
  make clean
  make -j4
  make install
  
  }
  echo "Building x_264 library"
  build_x264
  echo "Building FFmpeg"
  build_one
  

  I get an error saying

  No working C compiler found.
  Makefile:3: config.mak: No such file or directory
  

  The following is the output from the config.log file

  checking for -mdynamic-no-pic... yes
  checking for -arch x86_64... yes
  x264 configure script
  
  checking whether gcc works... yes
  checking whether gcc supports for( int i = 0; i < 9; i++ ); with -std=gnu99... yes
  checking for -mpreferred-stack-boundary=5... no
  Failed commandline was:
  --------------------------------------------------
  gcc conftest.c -m64 -Qunused-arguments -Wall -I. -I$(SRCPATH) -mdynamic-no-pic -arch x86_64 -std=gnu99  -mpreferred-stack-boundary=5  -m64  -lm -arch x86_64$
  clang: error: unknown argument: '-mpreferred-stack-boundary=5'
  --------------------------------------------------
  Failed program was:
  --------------------------------------------------
  int main (void) {  return 0; }
  --------------------------------------------------
  checking whether yasm supports vpmovzxwd ymm0, xmm0... yes
  

  Solutions that I tried:

  I made sure I have a working GCC

  I have the update to date version of clang

  I have the update to date version of LLVM

• High Quality Screen Capture with FFmpeg

  18 novembre 2015, par plumzl

  I'm trying to figure out a command for high quality screen capture on REHL.

  I have tried various options, but none really worked. The quality is not good enough. I'm quite new to ffmpeg and doesn't quite understand all the different flags. So any help would be appreciated!

  The commands I have tried:

  ffmpeg -y -f x11grab -r 30 -s 1920x1080 -i :0.0+0,0 -vcodec libx264 -strict -2 -b:v 16M output.mov
  

  The first command I tried. The weird thing is it doesn't seem to preserve the bitrate specified -b:v 16M. The output video's bitrate is only 2mb/s. Is that not the correct way to specify bitrate when doing screen capture?

  ffmpeg -f x11grab -s 1920x1080 -r 30 -i :0.0 -qscale 0 -vcodec huffyuv output.mov
  

  The huffyuv codec is supposed to be lossless, but the quality doesn't seem to be any better than the libx264 output.

  ffmpeg -f x11grab -r 30 -s 1920x1080 -i :0.0+0,0 -vcodec copy -pix_fmt yuv420p -qscale 0 -an output.mkv
  ffmpeg -f rawvideo -pix_fmt yuv420p -s:v 1920x1080 -r 30 -i output.mkv -an -c:v libx264 -qscale 0 result.mov 
  

  I'm even trying to capture rawvideo and then convert it to libx264 codec viedeo. But the output is corrupted.

  I'm using the version of ffmpeg that's built on April 23rd this year. It doesn't have the preset like -vpre lossless_ultrafast which I see mentioned in a lot posts.

  Edit:

  After more research, this is the command I end up using. I have to use h264 codec and yuv420p as pixel_fmt since I need to be able to play the quicktimes using quicktime.

  ffmpeg -y -f x11grab -framerate 30 -video_size 1920x1080 -i :0.0+0,0 -c:v libx264 -pix_fmt yuv420p -qp 0 -preset ultrafast output.mov
  

  The quality doesn't seem to be much improved even though -qp is set to 0. But it could also because of the rgb conversion to yuv color space according to this post.

  https://trac.ffmpeg.org/wiki/Encode/H.264

• Volume adjust and channel merge on video using FFMPEG

  6 novembre 2015, par user1566352

  I'm trying to convert a recorded AVI (1 video stream, 2 audio streams[1st is stereo, 2nd is mono]) video file to H264/AAC.

  I want the second audio stream to be at 60% volume and the first at 100%.

  I also want to merge the first and second audio stream.

  Output should be H264 with AAC audio.

  The command I tried to use is:

  "ffmpeg.exe" 
    -i "input.avi" 
    -filter_complex " 
      [0:a:0]aformat=channel_layouts=stereo,volume=1.0[a1]; 
      [0:a:1]aformat=channel_layouts=mono,volume=0.6[a2]; 
      [a1][a2]amerge,pan=stereo|c0code>

  But I get the error: [AVFilterGraph @ 0000000ceb630c00] The following filters could not choose their formats: Parsed_amerge_4 Consider inserting the (a)format filter near their input or output. Error configuring complex filters. Error number -5 occurred

  What am I doing wrong and how can it be fixed?

  Complete Console Log:

  ffmpeg version N-76456-g6df2c94 Copyright (c) 2000-2015 the FFmpeg developers
    built with gcc 5.2.0 (GCC)
    configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-lzma --enable-decklink --enable-zlib
    libavutil      55.  5.100 / 55.  5.100
    libavcodec     57. 14.100 / 57. 14.100
    libavformat    57. 14.100 / 57. 14.100
    libavdevice    57.  0.100 / 57.  0.100
    libavfilter     6. 14.101 /  6. 14.101
    libswscale      4.  0.100 /  4.  0.100
    libswresample   2.  0.100 /  2.  0.100
    libpostproc    54.  0.100 / 54.  0.100
  [avi @ 000000b1cfa6bc20] non-interleaved AVI
  Guessed Channel Layout for  Input Stream #0.1 : stereo
  Guessed Channel Layout for  Input Stream #0.2 : mono
  Input #0, avi, from 'input.avi':
    Metadata:
      encoder         : DxtoryCore ver2.0.0.122
      ISRC            : Video:Lagarith Lossless Codec Audio0:Lautsprecher (2- USB Audio CODEC ) Audio1:Mikrofon (2- USB Audio CODEC )
    Duration: 00:27:55.60, start: 0.000000, bitrate: 181780 kb/s
      Stream #0:0: Video: lagarith (LAGS / 0x5347414C), rgb24, 1920x1017, 179663 kb/s, 30 fps, 30 tbr, 30 tbn, 30 tbc
      Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, 2 channels, s16, 1411 kb/s
      Stream #0:2: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, 1 channels, s16, 705 kb/s
  [Parsed_amerge_4 @ 000000b1cfaf3200] No channel layout for input 1
  Last message repeated 1 times
  [AVFilterGraph @ 000000b1cfa70c00] The following filters could not choose their formats: Parsed_amerge_4
  Consider inserting the (a)format filter near their input or output.
  Error configuring complex filters.
  Error number -5 occurred