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• libavcodec/ffmpeg reports two channels when decoding, but segfaults when trying to access the second

  1er décembre 2017, par Astrognome

  I am attempting to decode an audio file into raw pcm data using FFMpeg’s libavcodec. I have the following code so far, but it segfaults when trying to access the second audio channel for some reason.

  #include 
  
  #include 
  
  #include <ao></ao>ao.h>
  
  #include 
  
  
  
  #include <libavutil></libavutil>opt.h>
  
  #include <libavcodec></libavcodec>avcodec.h>
  
  #include <libavformat></libavformat>avformat.h>
  
  #include <libswresample></libswresample>swresample.h>
  
  
  
  #define BUF_SIZE 4096
  
  
  
  
  
  #define AUDIO_INBUF_SIZE 20480
  
  #define AUDIO_REFILL_THRESH 4096
  
  
  
  int decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, uint8_t* out_data, int* index) {
  
  
  
      int i, ch;
  
      int ret, data_size;
  
  
  
      ret = avcodec_send_packet(dec_ctx, pkt);
  
      if (ret &lt; 0) {
  
          fprintf(stderr, "Error submitting packet to decoder\n");
  
          exit(1);
  
      }
  
  
  
      while (ret >= 0) {
  
          ret = avcodec_receive_frame(dec_ctx, frame);
  
          if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
  
              return 0;
  
          }
  
          else if (ret &lt; 0) {
  
              fprintf(stderr, "Error during decoding\n");
  
              exit(1);
  
          }
  
  
  
          data_size = av_get_bytes_per_sample(dec_ctx->sample_fmt);
  
          if (data_size &lt; 0) {
  
              fprintf(stderr, "Failed to calculate data size\n");
  
              exit(1);
  
          }
  
          for (i = 0; i &lt; frame->nb_samples; ++i)
  
              for (ch = 0; ch &lt; dec_ctx->channels; ++ch) {
  
  
  
                  //Things go wrong here...
  
                  memcpy(out_data + *index, frame->data[ch] + (data_size*i), data_size);
  
                  *index += data_size;
  
              }
  
      }
  
  
  
      return 0;
  
  }
  
  
  
  int decode_audio_file(const char* path, uint8_t* out_data, int size) {
  
      const AVCodec *codec;
  
      AVCodecContext *c = NULL;
  
      AVCodecParserContext *parser = NULL;
  
      int len, ret;
  
      FILE *f;
  
      uint8_t inbuf[AUDIO_INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
  
      uint8_t *data;
  
      size_t data_size;
  
      AVPacket *pkt;
  
      AVFrame *decoded_frame = NULL;
  
  
  
  
  
      avcodec_register_all();
  
      pkt = av_packet_alloc();
  
      codec = avcodec_find_decoder(AV_CODEC_ID_FLAC);
  
      if (!codec) {
  
          fprintf(stderr, "Codec not found!\n");
  
          return -1;
  
      }
  
  
  
      parser = av_parser_init(codec->id);
  
      if (!parser) {
  
          fprintf(stderr, "Parser not found!\n");
  
          return -1;
  
      }
  
  
  
      c = avcodec_alloc_context3(codec);
  
      if (!c) {
  
          fprintf(stderr, "Could not allocate audio context!\n");
  
          return -1;
  
      }
  
  
  
      if(avcodec_open2(c, codec, NULL) &lt; 0) {
  
          fprintf(stderr, "Could not open codec\n");
  
          exit(1);
  
      }
  
  
  
      f = fopen(path, "rb");
  
      if(!f) {
  
          fprintf(stderr, "Could not open %s\n", path);
  
          exit(1);
  
      }
  
  
  
      data = inbuf;
  
      data_size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
  
  
  
      int index = 0;  
  
      while (data_size > 0) {
  
          if (!decoded_frame) {
  
              if (!(decoded_frame = av_frame_alloc())) {
  
                  fprintf(stderr, "Could not allocate audio frame\n");
  
                  exit(1);
  
              }
  
          }
  
  
  
          ret = av_parser_parse2(parser, c, &amp;pkt->data, &amp;pkt->size, data, data_size, AV_NOPTS_VALUE, AV_NOPTS_VALUE, 0);
  
          if (ret &lt; 0) {
  
              fprintf(stderr, "Error while parsing\n");
  
              exit(1);
  
          }
  
          data += ret;
  
          data_size -= ret;
  
  
  
          if (pkt->size)
  
              decode(c, pkt, decoded_frame, out_data, &amp;index);
  
          if (data_size &lt; AUDIO_REFILL_THRESH) {
  
              memmove(inbuf, data, data_size);
  
              data = inbuf;
  
              len = fread(data + data_size, 1, AUDIO_INBUF_SIZE - data_size, f);
  
              if (len > 0)
  
                  data_size += len;
  
          }
  
      }
  
      avcodec_free_context(&amp;c);
  
      av_parser_close(parser);
  
      av_frame_free(&amp;decoded_frame);
  
      av_packet_free(&amp;pkt);
  
  
  
      return 0;
  
  }
  
  
  
  
  
  int main(int argc, char **argv)
  
  {
  
      int bsize = 1024*1024*60;
  
      uint8_t* audio_buffer = calloc(bsize, sizeof(uint8_t));
  
      decode_audio_file("testsong.flac", audio_buffer, bsize);
  
  }

  Here is the offending line :

  memcpy(out_data + *index, frame->data[ch] + (data_size*i), data_size);

  If you access frame->data[0] it’s fine, but if you attempt to access frame->data[1] it segfaults. The context reports two channels and the file contains two channels (tried with several different tracks).
  But it gets even weirder. If I switch the offending line to

  memcpy(out_data + *index, frame->data[0] + (data_size*i*2 + ch), data_size);

  it will have the left channel with correct data and the right channel is white noise. This tells me (and I could be wrong) that the first channel actually contains 2 channels of audio interleaved, the actual left channel, plus a junk channel of white noise.
  I was wrong, I forgot to multiple ch by data_size. If the line is memcpy(out_data + *index, frame->data[0] + (data_size*i*2 + ch*data_size), data_size); things seem to work as expected. This all still seems very sketchy though so if anyone can thoroughly explain what’s happening here, that would be nice.

  I have been working off the decode_audio example available in the ffmpeg documentation here although modified slightly.

  This ALSO happens if I use the vanilla example with the only change being setting the codec to FLAC, so I assume it’s a bug in the library. In that case, how can I work around it.

• ffmpeg commad with colorchannelmixer

  12 décembre 2013, par bindal

  I am using following command :
  command = "ffmpeg -y -i "+strFinalPath+" -vf colorchannelmixer=.3 :.4 :.3:0 :.3 :.4 :.3:0 :.3 :.4 :.3 "+strRotatePath ;

  But this command exit with error code 1

  is there any mistake in this command

  Please suggest me

  Thanks

• ffprobe to bitrate variable stopped working

  6 novembre 2023, par Bricktop

  I have a simple script to encode a video using the same bitrate as the original. I use ffprobe to fetch the bitrate like this :

  


  ffprobe "%file%" -v 0 -select_streams v:0 -show_entries stream=bit_rate -print_format compact=p=0:nokey=1 >%temp%\bitrate.txt


  


  However, while fixing a but in the script where I had an odd number of " marks, I suddenly ran into this problem with ffprobe :

  


  Argument ' -v 0 -select_streams v:0 -show_entries stream=bit_rate -print_format compact=p=0:nokey=1 >C:\Users\ADMINI~1\AppData\Local\Temp\bitrate.txt' provided as input filename, but 'D:\VIDEO\AMBIANCE\SCOPITONE\MUSIC TELEVISION\This Here - Calm - OFFICIAL VIDEO (1080p 25fps AV1-128kbit AAC).mp4' was already specified.


  


  I am trying to understand this, scanning insanely for yet another " or something in my code but can't figure it out. Here is the full code :

  


  :: write file to queue (first)
move /y "%~dpn0.txt" "%temp%\%~n0.tmp" >nul
echo "%~1" >"%~dpn0.txt"
type "%temp%\%~n0.tmp" >>"%~dpn0.txt"

:: desyncronize instances (todo: try support for adding 9 files at a time)
timeout /t %time:~9,1% /nobreak
:: if not first instance exit
tasklist /fi "imagename eq handbrakecli.exe" | find /i "handbrakecli" && exit
title Transcode

:: delegate queue
for /f "delims=" %%f in (%~dpn0.txt) do (
    set "name=%%~nf"
    set "file=%%~f"
    rem todo: if file has x264 or other video codec mentioned, change to x265
    set "code=%%~dpnf (x265 transcoded)%%~xf"
    call :transcode
)
echo all done!
exit /b

:transcode
title "%name%"
if not exist "%file%" echo %date% %time% source file missing %file% >>%~dpn0.log & goto cleanup
if exist "%code%" echo %date% %time% target file exists %file% >>%~dpn0.log & goto cleanup

:: determine appropriate bitrate (does not seem to work on .webm files, closing the script as a result)
%~dp0ffprobe "%file%" -v 0 -select_streams v:0 -show_entries stream=bit_rate -print_format compact=p=0:nokey=1 >%temp%\bitrate.txt
set /p bitrate=<%temp%\bitrate.txt
:: reduce to full kilobytes
set "bitrate=%bitrate:~0,-3%"
if not defined bitrate echo failed to fetch bitrate & echo %date% %time% no bitrate for %file% >>%~dpn0.log & exit /b
if %bitrate% gtr 7000 set bitrate=7000

:: transcode
%~dp0HandBrakeCLI -i "%file%" -o "%code%" --encoder x265_10bit --encoder-preset slow --encoder-profile main444-10 --vb %bitrate% --two-pass --turbo --audio 1-9 --aencoder copy --audio-copy-mask aac,ac3,mp2,mp3,opus --audio-fallback opus --ab 160 --drc 2.0

:: remove current file from queue, regardless
:cleanup
findstr /v /c:"%file%" "%~dpn0.txt" >"%temp%\%~n0.tmp"
move /y "%temp%\%~n0.tmp" "%~dpn0.txt"


  


  It appears that the set "file=%%~f" is the problem, somehow it shows up as set "file=D:\VIDEO\this video here.mp4" " where the last two characters " should not belong, and I don't know what to change to fix this.

  


  Every type of improvement to the script is very welcomed !