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• How can I convert WebM file to WebP file with transparency ?

  24 août 2020, par c-an

  I tried it with ffmpeg.

  


  ffmpeg input.webm output.webp


  


  input.webm contains transparent background and But the alpha channel becomes white in webp. I think that means alpha channel doesn't come together.

  


  I extracted frames with this command :

  


  ffmpeg -i input.xxx -c:v libwebp output_%03d.webp


  


  And it also gives me webp files with white background.

  


  How can I convert it properly with alpha channel ? OR should I convert it from other format(extension) ?

  


• Streaming ffmpeg from fifo file starts only when i close the fifo file

  23 juin 2022, par tamirg

  Im starting an ffmpeg process, where the input is a FIFO file i created.
Im writing some data in a loop to the FIFO file, but the ffmpeg process doesn't start streaming until one of the two happens :

  


  

  1. i'm closing the file

  


  2. iv'e written a certain amount of data. after a while of writing, the ffmpeg process starts streaming. The more data i write, the faster it starts running. (im writing a chunk of data on each loop, if i just duplicate those chunks times 100, it starts much faster).

  


  


  What can be the reason for that ? Is there a minimum of data required for the ffmpeg process to start streaming ? How can i "force" it to start, without closing the FIFO file after writing ?

  


• store pcm data into file, but can not play that file

  25 mai 2016, par Peng Qu

  I am writing a simple program, which reads mp3 file and store its pcm data into another file. I could get that file now, but when I play that on windows, I failed. So is there any wrong in my code, or windows couldn’t play raw audio data ?

  #include 
  
  #include 
  
  #include <libavutil></libavutil>avutil.h>
  
  #include <libavformat></libavformat>avformat.h>
  
  #include <libavcodec></libavcodec>avcodec.h>
  
  
  
  int main()
  
  {
  
  int err;
  
  FILE *fout = fopen("test.wav", "wb");
  
  av_register_all();
  
  
  
  // step 1, open file and find audio stream
  
  AVFormatContext *fmtx = NULL;
  
  err = avformat_open_input(&amp;fmtx, "melodylove.mp3", NULL, NULL);
  
  assert(!err);
  
  
  
  err = avformat_find_stream_info(fmtx, NULL);
  
  assert(!err);
  
  
  
  int audio_stream_idx = -1;
  
  AVStream *st;
  
  AVCodecContext *decx;
  
  AVCodec *dec;
  
  
  
  for (int i = 0; i &lt; fmtx->nb_streams; ++i) {
  
      audio_stream_idx = i;
  
      if (fmtx->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
  
          st = fmtx->streams[i];
  
          decx = st->codec;
  
          dec = avcodec_find_decoder(decx->codec_id);
  
          decx->request_channel_layout = AV_CH_LAYOUT_STEREO_DOWNMIX;
  
          decx->request_sample_fmt = AV_SAMPLE_FMT_FLT;
  
          avcodec_open2(decx, dec, NULL);
  
          break;
  
      }
  
  }
  
  assert(audio_stream_idx != -1);
  
  
  
  int channels = decx->channels;
  
  int sample_rate = decx->sample_rate;
  
  int planar = av_sample_fmt_is_planar(decx->sample_fmt);
  
  int num_planes =  planar? decx->channels : 1;
  
  const char *sample_name = av_get_sample_fmt_name(decx->sample_fmt);
  
  printf("sample name: %s, channels: %d, sample rate: %d\n",
  
          sample_name, channels, sample_rate);
  
  printf("is planar: %d, planes: %d\n", planar, num_planes);
  
  
  
  /*
  
   * above I print some infomation about mp3 file, they are:
  
   * sample name: s16p, channels: 2, sample rate: 48000
  
   * is planar: 1, planes: 2
  
   */
  
  getchar();
  
  
  
  AVPacket pkt;
  
  av_init_packet(&amp;pkt);
  
  AVFrame *frame = av_frame_alloc();
  
  
  
  while (1) {
  
      err = av_read_frame(fmtx, &amp;pkt);
  
      if (err &lt; 0) {
  
          printf("read frame fail\n");
  
          fclose(fout);
  
          exit(-1);
  
      }
  
      if (pkt.stream_index != audio_stream_idx) {
  
          printf("we don't need this stream\n");
  
          continue;
  
      }
  
      printf("data size: %d\n", pkt.size);
  
  
  
      int got_frame = 0;
  
      int bytes = avcodec_decode_audio4(decx, frame, &amp;got_frame, &amp;pkt);
  
      if (bytes &lt; 0) {
  
          printf("decode audio fail\n");
  
          continue;
  
      }
  
      printf("frame size: %d, samples: %d\n", bytes, frame->nb_samples);
  
  
  
      if (got_frame) {
  
          int input_samples = frame->nb_samples * decx->channels;
  
          int sz = input_samples / num_planes;
  
          short buffer1[input_samples];
  
  
  
          for (int j = 0; j &lt; frame->nb_samples; ++j) {
  
              for (int i = 0; i &lt; num_planes; ++i) {
  
                  short *d = (short *)frame->data[i];
  
                  buffer1[j*2+i] = d[j];
  
              }
  
          }
  
  
  
          fwrite(buffer1, input_samples, 2, fout);
  
      } else {
  
          printf("why not get frame???");
  
      }
  
  }
  
  }