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• lavc/container_fifo : move to lavu and make public

  17 octobre 2024, par Anton Khirnov

  lavc/container_fifo : move to lavu and make public
  This can be useful in other places, e.g. it can replace objpool in
  
  fftools.
  The API is modified in the following nontrivial ways :
  
  * opaque pointers can be passed through to all user callbacks
  
  * read and write were previously separate callbacks in order to
  
    accomodate the caller wishing to write a new reference to the FIFO and
  
    keep the original one ; the two callbacks are now merged into one, and
  
    a flags argument is added that allows to request such behaviour on a
  
    per-call basis
  
  * new peek and drain functions
  

  • [DH] doc/APIchanges
  • [DH] libavcodec/Makefile
  • [DH] libavcodec/container_fifo.c
  • [DH] libavcodec/container_fifo.h
  • [DH] libavcodec/hevc/hevcdec.c
  • [DH] libavcodec/hevc/hevcdec.h
  • [DH] libavcodec/hevc/refs.c
  • [DH] libavutil/Makefile
  • [DH] libavutil/container_fifo.c
  • [DH] libavutil/container_fifo.h

• FFmpeg.swr_convert : audio to raw 16 bit pcm, to be used with xna SoundEffect. Audio cuts out when i convert

  21 mars 2019, par Robert Russell

  I want to resample mkv(vp8/ogg) and also raw 4 bit adpcm to raw 16bit pcm byte[] to be loaded into SoundEffect from xna library. So I can play it out while I’m using other code to display the frames (the video side is working).
  I can read a 16 bit wav file and play it. But when I goto resample something it doesn’t play 100%. One file is 3 mins and 15 secs. I only get 13 sec and 739 ms before it quits playing. I have been learning to do this by finding code samples in c++ and correcting it to work in c# using ffmpeg.autogen.

  the below is my best attempt at resampling.

              int nb_samples = Frame->nb_samples;
  
                      int output_nb_samples = nb_samples;
  
                      int nb_channels = ffmpeg.av_get_channel_layout_nb_channels(ffmpeg.AV_CH_LAYOUT_STEREO);
  
                      int bytes_per_sample = ffmpeg.av_get_bytes_per_sample(AVSampleFormat.AV_SAMPLE_FMT_S16) * nb_channels;
  
                      int bufsize = ffmpeg.av_samples_get_buffer_size(null, nb_channels, nb_samples,
  
                                                               AVSampleFormat.AV_SAMPLE_FMT_S16, 1);
  
  
  
                      byte*[] b = Frame->data;
  
                      fixed (byte** input = b)
  
                      {
  
                          byte* output = null;
  
                          ffmpeg.av_samples_alloc(&output, null,
  
                              nb_channels,
  
                              nb_samples,
  
                              (AVSampleFormat)Frame->format, 0);//
  
  
  
                          // Buffer input
  
  
  
                          Ret = ffmpeg.swr_convert(Swr, &output, output_nb_samples / 2, input, nb_samples);
  
                          CheckRet();
  
                          WritetoMs(output, 0, Ret * bytes_per_sample);
  
                          output_nb_samples -= Ret;
  
  
  
                          // Drain buffer
  
                          while ((Ret = ffmpeg.swr_convert(Swr, &output, output_nb_samples, null, 0)) > 0)
  
                          {
  
                              CheckRet();
  
                              WritetoMs(output, 0, Ret * bytes_per_sample);
  
                              output_nb_samples -= Ret;
  
                          }
  
                      }

  I changed that all to this but it cuts off sooner.

    Channels = ffmpeg.av_get_channel_layout_nb_channels(OutFrame->channel_layout);
  
                      int nb_channels = ffmpeg.av_get_channel_layout_nb_channels(ffmpeg.AV_CH_LAYOUT_STEREO);
  
                      int bytes_per_sample = ffmpeg.av_get_bytes_per_sample(AVSampleFormat.AV_SAMPLE_FMT_S16) * nb_channels;
  
  
  
                      if((Ret = ffmpeg.swr_convert_frame(Swr, OutFrame, Frame))>=0)
  
                          WritetoMs(*OutFrame->extended_data, 0, OutFrame->nb_samples * bytes_per_sample);
  
                      CheckRet();

  Both code use a function to set Swr it runs one time after the first frame is decoded.

          private void PrepareResampler()
  
      {
  
          ffmpeg.av_frame_copy_props(OutFrame, Frame);
  
          OutFrame->channel_layout = ffmpeg.AV_CH_LAYOUT_STEREO;
  
          OutFrame->format = (int)AVSampleFormat.AV_SAMPLE_FMT_S16;
  
          OutFrame->sample_rate = Frame->sample_rate;
  
          OutFrame->channels = 2;
  
          Swr = ffmpeg.swr_alloc();
  
          if (Swr == null)
  
              throw new Exception("SWR = Null");
  
          Ret = ffmpeg.swr_config_frame(Swr, OutFrame, Frame);
  
          CheckRet();
  
          Ret = ffmpeg.swr_init(Swr);
  
          CheckRet();
  
          Ret = ffmpeg.swr_is_initialized(Swr);
  
          CheckRet();
  
      }

  This is where I take the output and put it in the sound effect

  private void ReadAll()
  
      {
  
  
  
          using (Ms = new MemoryStream())
  
          {
  
              while (true)
  
              {
  
                  Ret = ffmpeg.av_read_frame(Format, Packet);
  
                  if (Ret == ffmpeg.AVERROR_EOF)
  
                      break;
  
                  CheckRet();
  
                  Decode();
  
              }
  
              if (Ms.Length > 0)
  
              {
  
                  se = new SoundEffect(Ms.ToArray(), 0, (int)Ms.Length, OutFrame->sample_rate, (AudioChannels)Channels, 0, 0);
  
                  //se.Duration; Stream->duration;
  
  
  
  
  
                  see = se.CreateInstance();
  
                  see.Play();
  
              } 
  
          }
  
      }

• FFmpeg 5 C api codec : how many times will EOF be returned by the receiving end ?

  3 février 2024, par Guanyuming He

  Edit : I may have written my questions unclearly at first, so I rewrote them. Sorry if you had found my questions confusing.

  


  I'm new to FFmpeg api programming (I'm using version 5.1) and am learning from the documentation and official examples.

  


  In the documentation page about send/receive encoding and decoding API overview, end of stream situation is discussed briefly :

  


  


  End of stream situations. These require "flushing" (aka draining) the codec, as the codec might buffer multiple frames or packets internally for performance or out of necessity (consider B-frames). This is handled as follows :

  


  


  


  Instead of valid input, send NULL to the avcodec_send_packet() (decoding) or avcodec_send_frame() (encoding) functions. This will enter draining mode.
Call avcodec_receive_frame() (decoding)
or avcodec_receive_packet() (encoding) in a loop until AVERROR_EOF is returned. The functions will not return AVERROR(EAGAIN), unless you forgot to enter draining mode.
Before decoding can be resumed again, the codec has to be reset with avcodec_flush_buffers().

  


  


  As I understand it, when I get AVERROR_EOF, I have reached a special point where I need to drain buffered data from the codec and finally reset the codec with avcodec_flush_buffers(). Without doing it, I cannot continue decoding/encoding.

  


  Is my understanding correct ?

  


  If so, then I have some questions :

  


  

  1. How many times at most can EOF be returned by the receiving end during one complete process, for example, decoding ?

  


  2. If the answer to the first question is infinity, then : If I received an EOF from the receiving end when I already finished sending data (e.g. when after EOF is returned by av_read_frame()), how should I tell if it's really finished ? Here there is no return code only for indicating if the receiving is finished.

  


  3. The data returned from the receive_... functions during draining, should I take them as valid ?

  


  


  I might have found answers to those in the official examples, but I'm not sure if the answer is universally true. I noticed that in some official examples, like in transcode_aac.c, draining is only done for the first EOF reached, and then after the second one is received, it is regarded that there are really nothing left. Any data received during draining is also written to the final output.

  


  Am I correct on interpreting the example ? If so, can I say that the answer to question 1 is once, and the answer to question 3 is yes ?

  


  I appreciate your response and time in advance. :)