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• lavc/libaribcaption : add MSZ character related options

  17 octobre 2023, par TADANO Tokumei

  lavc/libaribcaption : add MSZ character related options
  This patch adds two MSZ (Middle Size ; half width) character
  
  related options, mapping against newly added upstream
  
  functionality :
  * `replace_msz_japanese`, which was introduced in version 1.0.1
  
    of libaribcaption.
  
  * `replace_msz_glyph`, which was introduced in version 1.1.0
  
    of libaribcaption.
  The latter option improves bitmap type rendering if specified
  
  fonts contain half-width glyphs (e.g., BIZ UDGothic), even
  
  if both ASCII and Japanese MSZ replacement options are set
  
  to false.
  As these options require newer versions of libaribcaption, the
  
  configure requirement has been bumped accordingly.
  Signed-off-by : TADANO Tokumei <aimingoff@pc.nifty.jp>
  

  • [DH] configure
  • [DH] doc/decoders.texi
  • [DH] libavcodec/libaribcaption.c
  • [DH] libavcodec/version.h

• FFMPEG hangs (sometimes) when trying to take a photo from USB camera ?

  13 mai 2022, par dezman

  I am attempting to write a script that takes a photo every 3 seconds from camera source "1", the -pix_fmt I think is unique to my USB camera.

  


  Here is the script :

  


  #!/bin/bash

set -e

while :
do
  nice_num=$(date +%s)
  
  ffmpeg -ss 00:00:00 -f avfoundation -pix_fmt yuyv422 -r 30.000030 -i "1" -t 1 -frames:v 1 -q:v 2 "image_$nice_num_%03d.jpg"
  
  wait

  sleep 3
done


  


  The issue is half the time it successfully takes the photo, and half the time it just hangs forever echoing this (not useful) :

  


  ffmpeg version 5.0.1 Copyright (c) 2000-2022 the FFmpeg developers
  built with Apple clang version 13.1.6 (clang-1316.0.21.2)
  configuration: --prefix=/usr/local/Cellar/ffmpeg/5.0.1 --enable-shared --enable-pthreads --enable-version3 --cc=clang --host-cflags= --host-ldflags= --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libbluray --enable-libdav1d --enable-libmp3lame --enable-libopus --enable-librav1e --enable-librist --enable-librubberband --enable-libsnappy --enable-libsrt --enable-libtesseract --enable-libtheora --enable-libvidstab --enable-libvmaf --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libxvid --enable-lzma --enable-libfontconfig --enable-libfreetype --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libspeex --enable-libsoxr --enable-libzmq --enable-libzimg --disable-libjack --disable-indev=jack --enable-videotoolbox
  libavutil      57. 17.100 / 57. 17.100
  libavcodec     59. 18.100 / 59. 18.100
  libavformat    59. 16.100 / 59. 16.100
  libavdevice    59.  4.100 / 59.  4.100
  libavfilter     8. 24.100 /  8. 24.100
  libswscale      6.  4.100 /  6.  4.100
  libswresample   4.  3.100 /  4.  3.100
  libpostproc    56.  3.100 / 56.  3.100


  


  Then I CTRL-C 3 times and get :

  


  ^C^C^CReceived > 3 system signals, hard exiting


  


  Other methods of "taking a photo every n seconds from bash" are also appreciated.

  


• FFmpeg Decklink playout audio PCM

  28 décembre 2019, par Rick Jansen

  I’m working on a ffmpeg playout application for Decklink but I’m facing some audio issues. I’ve seen other questions about this topic but none of them are currently helping.

  I’ve tried Reubens code (https://stackoverflow.com/a/15372417/12610231) with the swr_convert for playing out ffmpeg/libav frames to a Decklink board (this needs to be 16 bits PCM interleaved) but the audio sounds wrong. It sounds like it’s missing samples/ only getting half of the required samples).

  When I record the samples in a raw audio file and play it out with Audacity the timeline is half the length of the actual recording and playing the samples on double speed.

  I also tried the ’manual’ conversion (https://stackoverflow.com/a/15372417/12610231) but unfortunately, not the result I was hoping for.

  Here are some snippets of my code

  swr_ctx = swr_alloc();
  
  av_opt_set_int(swr_ctx, "in_channel_count", pAudioCodecCtx->channels, 0);
  
  av_opt_set_int(swr_ctx, "in_sample_rate", pAudioCodecCtx->sample_rate, 0);
  
  av_opt_set_int(swr_ctx, "in_channel_layout", pAudioCodecCtx->channel_layout, 0);
  
  av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", pAudioCodecCtx->sample_fmt, 0);
  
  av_opt_set_int(swr_ctx, "out_channel_count", 2, 0);
  
  av_opt_set_int(swr_ctx, "out_sample_rate", 48000, 0);
  
  av_opt_set_int(swr_ctx, "out_channel_layout", AV_CH_LAYOUT_STEREO, 0);
  
  av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0);
  
  
  
  if (swr_init(swr_ctx))
  
  {
  
      printf("Error SWR");
  
  }

  ///

  ret = avcodec_decode_audio4(pAudioCodecCtx, pFrame, &frameFinished, &packet);
  
  
  
  if (ret < 0) {
  
      printf("Error in decoding audio frame.\n");
  
  }
  
  
  
  swr_convert(swr_ctx, (uint8_t**)&m_audioBuffer, pFrame->nb_samples, (const uint8_t *)pFrame->extended_data, pFrame->nb_samples);

  It also looks like that the FFmpeg packet contains out of 1 video packet en 2 audio packets, not sure what to do with the second audio packet, I already tried to combine the first and second audio package without any good result on the audio side.

  Any help is appreciated.