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• libav ffmpeg - streaming from both a mkv and input stream

  20 janvier 2020, par kealist

  I am trying to use ffmpeg libraries in C# with AutoGen bindings. The overall issue is that I am taking a collection of sources, some streams, and some .mkv containing recordings of a stream. As for now, they are all h264 and only video. For input streams, I am able to adjust the packets and broad cast them and that works fine, but any time I try to call av_interleaved_write_frame with packets from the MKV file, I get the error Error occurred: Invalid data found when processing input.

  Here is the main loop, where the error happens for mkv files. Is there an extra step ?

  /* read all packets */
  
  while (true)
  
  {
  
      if ((ret = ffmpeg.av_read_frame(ifmt_ctx, &packet)) < 0)
  
      {
  
          Console.WriteLine("Unable to read packet");
  
          break;
  
      }
  
      stream_index = (uint)packet.stream_index;
  
      type = ifmt_ctx->streams[packet.stream_index]->codecpar->codec_type;
  
      Console.WriteLine($"Demuxer gave frame of stream_index %{stream_index}");
  
  
  
  
  
      /* remux this frame without reencoding */
  
      ffmpeg.av_packet_rescale_ts(&packet,
  
          ifmt_ctx->streams[stream_index]->time_base,
  
          ofmt_ctx->streams[stream_index]->time_base);
  
  
  
      if (packet.stream_index < 0)
  
      {
  
          Console.WriteLine("Packet stream error");
  
      }
  
  
  
  
  
  
  
      ret = ffmpeg.av_write_frame(ofmt_ctx, &packet);
  
      if (ret < 0)
  
      {
  
          goto end;
  
      }
  
      else
  
      {
  
          ffmpeg.av_packet_unref(&packet);
  
      }
  
  }

  Anything need to be different for MKV files ?

  I get some contradictory error output where it claims it is annex b but also isn’t :

  [AVBSFContext @ 00000220eb657080] The input looks like it is Annex B already
  
  Automatically inserted bitstream filter 'h264_mp4toannexb'; args=''
  
  [mpegts @ 00000220ebace300] H.264 bitstream malformed, no startcode found, use the video bitstream filter 'h264_mp4toannexb' to fix it ('-bsf:v h264_mp4toannexb' option with ffmpeg)

  Verbose output from ffplay from an MKV file :

  ffplay version git-2020-01-13-7225479 Copyright (c) 2003-2020 the FFmpeg developers
  
    built with gcc 9.2.1 (GCC) 20200111
  
    configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt --enable-amf
  
    libavutil      56. 38.100 / 56. 38.100
  
    libavcodec     58. 65.103 / 58. 65.103
  
    libavformat    58. 35.102 / 58. 35.102
  
    libavdevice    58.  9.103 / 58.  9.103
  
    libavfilter     7. 71.100 /  7. 71.100
  
    libswscale      5.  6.100 /  5.  6.100
  
    libswresample   3.  6.100 /  3.  6.100
  
    libpostproc    55.  6.100 / 55.  6.100
  
  Initialized direct3d renderer.
  
  [h264 @ 00000165ed18d140] Reinit context to 640x480, pix_fmt: yuv444p
  
  Input #0, matroska,webm, from '.\webcam_14_Test1.mkv':   0B f=0/0
  
    Metadata:
  
      ENCODER         : Lavf58.12.100
  
    Duration: 00:00:39.30, start: 0.000000, bitrate: 1943 kb/s
  
      Stream #0:0: Video: h264 (High 4:4:4 Predictive), 1 reference frame, yuv444p(progressive, left), 640x480 [SAR 1:1 DAR 4:3], 1k fps, 30 tbr, 1k tbn, 60 tbc (default)
  
      Metadata:
  
        DURATION        : 00:00:39.299000000
  
  [h264 @ 00000165f424e200] Reinit context to 640x480, pix_fmt: yuv444p
  
  [ffplay_buffer @ 00000165f52ea840] w:640 h:480 pixfmt:yuv444p tb:1/1000 fr:30/1 sar:1/1
  
  [auto_scaler_0 @ 00000165ed1d2c80] w:iw h:ih flags:'bicubic' interl:0
  
  [ffplay_buffersink @ 00000165f424ef00] auto-inserting filter 'auto_scaler_0' between the filter 'ffplay_buffer' and the filter 'ffplay_buffersink'
  
  [auto_scaler_0 @ 00000165ed1d2c80] w:640 h:480 fmt:yuv444p sar:1/1 -> w:640 h:480 fmt:yuv420p sar:1/1 flags:0x4
  
  Created 640x480 texture with SDL_PIXELFORMAT_IYUV.
  
  [AVIOContext @ 00000165ed179a40] Statistics: 9547965 bytes read, 0 seeks

• FFmpeg / libmp3lame crash while converting from .wav to .mp3 with vibrato

  1er juillet 2020, par Chitrang

  I have integrated mobile-ffmpeg-full-gpl:4.3.1.LTS library in my android app. And trying to convert .wav file to .mp3 format with vibrato option and libmp3lame encoder.

  


  ffmpegCommand = "-i input.wav " +
                "-af vibrato=f=4 " +
                "-c:a libmp3lame " +
                "-b:a 96k " +
                "-ac 1 " +
                "-ar 44100 " +
                "-y output.mp3"


  


  FFmpeg Logs :

  


  a.b.com I/mobile-ffmpeg: Loading mobile-ffmpeg.
a.b.com I/mobile-ffmpeg: Loaded mobile-ffmpeg-full-gpl-arm64-v8a-4.3.1-lts-20200125.
a.b.com D/mobile-ffmpeg: Callback thread started.
a.b.com I/mobile-ffmpeg: ffmpeg version git-2020-01-25-fd11dd500
a.b.com I/mobile-ffmpeg:  Copyright (c) 2000-2020 the FFmpeg developers
a.b.com I/mobile-ffmpeg:   built with Android (5220042 based on r346389c) clang version 8.0.7 (https://android.googlesource.com/toolchain/clang b55f2d4ebfd35bf643d27dbca1bb228957008617) (https://android.googlesource.com/toolchain/llvm 3c393fe7a7e13b0fba4ac75a01aa683d7a5b11cd) (based on LLVM 8.0.7svn)
a.b.com I/mobile-ffmpeg:   configuration: --cross-prefix=aarch64-linux-android- --sysroot=/files/android-sdk/ndk-bundle/toolchains/llvm/prebuilt/linux-x86_64/sysroot --prefix=/home/taner/Projects/mobile-ffmpeg/prebuilt/android-arm64/ffmpeg --pkg-config=/usr/bin/pkg-config --enable-version3 --arch=aarch64 --cpu=armv8-a --cc=aarch64-linux-android21-clang --cxx=aarch64-linux-android21-clang++ --target-os=android --enable-neon --enable-asm --enable-inline-asm --enable-cross-compile --enable-pic --enable-jni --enable-optimizations --enable-swscale --enable-shared --disable-v4l2-m2m --disable-outdev=v4l2 --disable-outdev=fbdev --disable-indev=v4l2 --disable-indev=fbdev --enable-small --disable-openssl --disable-xmm-clobber-test --disable-debug --enable-lto --disable-neon-clobber-test --disable-programs --disable-postproc --disable-doc --disable-htmlpages --disable-manpages --disable-podpages --disable-txtpages --disable-static --disable-sndio --disable-schannel --disable-securetransport --disable-xlib --disable-cuda --disable-cuvid --disa
a.b.com I/mobile-ffmpeg:   libavutil      56. 38.100 / 56. 38.100
a.b.com I/mobile-ffmpeg:   libavcodec     58. 65.102 / 58. 65.102
a.b.com I/mobile-ffmpeg:   libavformat    58. 35.101 / 58. 35.101
a.b.com I/mobile-ffmpeg:   libavdevice    58.  9.103 / 58.  9.103
a.b.com I/mobile-ffmpeg:   libavfilter     7. 70.101 /  7. 70.101
a.b.com I/mobile-ffmpeg:   libswscale      5.  6.100 /  5.  6.100
a.b.com I/mobile-ffmpeg:   libswresample   3.  6.100 /  3.  6.100
a.b.com W/mobile-ffmpeg: [wav @ 0x7294a86600] Estimating duration from bitrate, this may be inaccurate
a.b.com W/mobile-ffmpeg: Guessed Channel Layout for Input Stream #0.0 : mono
a.b.com I/mobile-ffmpeg: Input #0, wav, from '/data/user/0/a.b.com/cache/creation/input.wav':
a.b.com I/mobile-ffmpeg:   Duration: 
a.b.com I/mobile-ffmpeg: 00:00:07.15
a.b.com I/mobile-ffmpeg: , bitrate: 
a.b.com I/mobile-ffmpeg: 705 kb/s
a.b.com I/mobile-ffmpeg:     Stream #0:0
a.b.com I/mobile-ffmpeg: : Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, mono, s16, 705 kb/s
a.b.com I/mobile-ffmpeg: Stream mapping:
a.b.com I/mobile-ffmpeg:   Stream #0:0 -> #0:0
a.b.com I/mobile-ffmpeg:  (pcm_s16le (native) -> mp3 (libmp3lame))
a.b.com I/mobile-ffmpeg: Press [q] to stop, [?] for help
a.b.com I/mobile-ffmpeg: Output #0, mp3, to '/data/user/0/a.b.com/cache/creation/output.mp3':
a.b.com I/mobile-ffmpeg:   Metadata:
a.b.com I/mobile-ffmpeg:     TSSE            : 
a.b.com I/mobile-ffmpeg: Lavf58.35.101
a.b.com I/mobile-ffmpeg:     Stream #0:0
a.b.com I/mobile-ffmpeg: : Audio: mp3 (libmp3lame), 44100 Hz, mono, fltp, 96 kb/s
a.b.com I/mobile-ffmpeg:     Metadata:
a.b.com I/mobile-ffmpeg:       encoder         : 
a.b.com I/mobile-ffmpeg: Lavc58.65.102 libmp3lame
a.b.com I/mobile-ffmpeg: --------- beginning of crash
a.b.com A/libc: psymodel.c:576: void calc_energy(const PsyConst_CB2SB_t *, const FLOAT *, FLOAT *, FLOAT *, FLOAT *): assertion "el >= 0" failed
a.b.com A/libc: Fatal signal 6 (SIGABRT), code -6 (SI_TKILL) in tid 25800 (a.b.com), pid 25800 (a.b.com)


  


  Crash :

  


  --------- beginning of crash
 A/libc: psymodel.c:576: void calc_energy(const PsyConst_CB2SB_t *, const FLOAT *, FLOAT *, FLOAT *, FLOAT *): assertion "el >= 0" failed

? A/DEBUG: *** *** *** *** *** *** *** *** *** *** *** *** *** *** *** ***
? A/DEBUG: Build fingerprint: 'samsung/star2qltecs/star2qltecs:10/QP1A.190711.020/G965WVLS7DTE1:user/release-keys'
? A/DEBUG: Revision: '14'
? A/DEBUG: ABI: 'arm64'
? A/DEBUG: Timestamp: 2020-06-29 15:13:17-0400
? A/DEBUG: pid: 1849, tid: 1849, name: a.b.com  >>> a.b.com <<<
? A/DEBUG: uid: 12171
? A/DEBUG: signal 6 (SIGABRT), code -6 (SI_TKILL), fault addr --------
? A/DEBUG: Abort message: 'psymodel.c:576: void calc_energy(const PsyConst_CB2SB_t *, const FLOAT *, FLOAT *, FLOAT *, FLOAT *): assertion "el >= 0" failed'
? A/DEBUG:     x0  0000000000000000  x1  0000000000000739  x2  0000000000000006  x3  0000007fd65f7bb0
? A/DEBUG:     x4  0000000000000000  x5  0000000000000000  x6  0000000000000000  x7  0000000000000008
? A/DEBUG:     x8  00000000000000f0  x9  7f96d7a39856d151  x10 0000000000000001  x11 0000000000000000
? A/DEBUG:     x12 fffffff0fffffbdf  x13 000000005efa3d4c  x14 001c23c1a79207f5  x15 000079d970d48db2
? A/DEBUG:     x16 00000073e009e8c0  x17 00000073e007afe0  x18 00000073e492c000  x19 0000000000000739
? A/DEBUG:     x20 0000000000000739  x21 00000000ffffffff  x22 0000007fd65fc44c  x23 0000007fd65f8640
? A/DEBUG:     x24 0000007fd65fd120  x25 0000007fd65fd3a8  x26 0000007fd65f8240  x27 0000007fd65f9e40
? A/DEBUG:     x28 00000071f9f60900  x29 0000007fd65f7c50
? A/DEBUG:     sp  0000007fd65f7b90  lr  00000073e002c27c  pc  00000073e002c2a8
? A/DEBUG: backtrace:
? A/DEBUG:       #00 pc 00000000000832a8  /apex/com.android.runtime/lib64/bionic/libc.so (abort+160) (BuildId: 55ce0a7d78144b0290f9746ed1615719)
? A/DEBUG:       #01 pc 00000000000839e8  /apex/com.android.runtime/lib64/bionic/libc.so (__assert2+36) (BuildId: 55ce0a7d78144b0290f9746ed1615719)
? A/DEBUG:       #02 pc 0000000000969c60  /data/app/a.b.com-jXqE8oxytEkfSsn6pcdloQ==/lib/arm64/libavcodec.so


  


  I referred link1, link2 to understand the problem but could not find a solution.

  


• Convert SDR-JPEG to HDR-AVIF [closed]

  27 avril 2024, par Jonas Janzen

  I would like to convert a jpg file into an avif file that is to be saved in HDR10-capable metadata (PQ curve, 2020 color space, 10 bit).

  


  The idea is to save normal SDR images in HDR-capable containers so that they can be displayed in all their glory on HDR-capable displays.

  


  I want to play with inverse tone mapping, to manipulate the output, so I implemented in Python via subprocess.

  


  So far I just want the input image to be saved in AVIF as HDR and look the same at the end as before, so that I can then make changes in the next step.

  


  I used the following command for this :

  


  ffmpeg_command = [
'ffmpeg',

  


  Input File
'-i', temp_file,

  


  Used Library
'-c', 'libaom-av1',

  


  '-still-picture', '1',

  


  Output Metadata
'-pix_fmt', 'yuv420p10le',
'-strict', 'experimental',
'-color_primaries', 'bt2020',
'-color_trc', 'smpte2084',
'-colorspace', 'bt2020nc',
'-color_range', 'pc',

  


  Output File
output_file
]

  


  So far my attempts have only been successful with the HLG characteristic. Here you can see that the images are really brighter in the peaks on my HDR monitor.

  


  With the PQ characteristic curve, the images are far too oversaturated.

  


  I guess this is because the HLG curve is compatible with the gamma curve, but PQ is not.

  


  Now my question is what I need to change.

  


  Which curve does FFMpeg expect as input.

  


  In Python I can change the images mathematically without any problems.

  


  The Example Images are again tone mapped down to jpg, to show what happened.

  


  enter image description here
enter image description here