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• ffmpeg to lower/fade audio volume of one audio stream when microphone voice detected ?

  11 juin 2021, par Lectos Lacious

  I want to do live audio translation via microphone, to get streamed live vid/audio from Facebook, plug the mic into laptop and do live translation by mixing existing audio stream with one coming from the mic (translation). This is OK, somehow I got this part by using audio filter "amix" and mix two audio streams together into one. Now I want to add more perfection to it, is it possible to (probably is) upon mic voice detection to automatically decrease/fade down 20% volume of input/original audio stream to hear translation (mic audio) more loudly and then when mic action/voice stops for lets say 3-5 seconds the volume of original audio stream fades up/goes up to normal volume... is this too much, i can play with sox or similar ?

  


• FFmpeg RTP payload 96 instead of 97

  26 octobre 2016, par bot1131357

  I am trying to create an rtp audio stream with ffmpeg. The application output and SDP file configuration are as follows :

  Output #0, rtp, to 'rtp://127.0.0.1:8554':
  
      Stream #0:0: Audio: pcm_s16be, 8000 Hz, stereo, s16, 256 kb/s
  
  
  
  SDP:    
  
  v=0
  
  o=- 0 0 IN IP4 127.0.0.1
  
  s=No Name
  
  c=IN IP4 127.0.0.1
  
  t=0 0
  
  a=tool:libavformat 57.25.101
  
  m=audio 8554 RTP/AVP 96
  
  b=AS:256
  
  a=rtpmap:96 L16/8000/2

  However, when I try to read it with ffplay -i test.sdp -protocol_whitelist file,udp,rtp, it fails,shows the following :

  ffplay version N-78598-g98a0053 Copyright (c) 2003-2016 the FFmpeg developers
  
    built with gcc 5.3.0 (GCC)
  
    configuration: --disable-static --enable-shared --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-decklink --enable-zlib
  
    libavutil      55. 18.100 / 55. 18.100
  
    libavcodec     57. 24.103 / 57. 24.103
  
    libavformat    57. 25.101 / 57. 25.101
  
    libavdevice    57.  0.101 / 57.  0.101
  
    libavfilter     6. 34.100 /  6. 34.100
  
    libswscale      4.  0.100 /  4.  0.100
  
    libswresample   2.  0.101 /  2.  0.101
  
    libpostproc    54.  0.100 / 54.  0.100
  
      nan    :  0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0
  
      (...waits indefinitely.)

  The only way to make it work again is to modify the payload type in the SDP file from 96 to 97. Can someone tell me why ? Where is this number defined ?

  Here is my source. See if you can replicate it.

  #include 
  
  extern "C"
  
  {
  
  #include <libavutil></libavutil>opt.h>
  
  #include <libavcodec></libavcodec>avcodec.h>
  
  #include <libavutil></libavutil>channel_layout.h>
  
  #include <libavutil></libavutil>common.h>
  
  #include <libavutil></libavutil>imgutils.h>
  
  #include <libavutil></libavutil>mathematics.h>
  
  #include <libavutil></libavutil>samplefmt.h>
  
  #include <libavformat></libavformat>avformat.h>
  
  }
  
  
  
  
  
  static int write_frame(AVFormatContext *fmt_ctx, const AVRational *time_base, AVStream *st, AVPacket *pkt)
  
  {
  
      /* rescale output packet timestamp values from codec to stream timebase */
  
      av_packet_rescale_ts(pkt, *time_base, st->time_base);
  
  
  
      /* Write the compressed frame to the media file. */
  
      return av_interleaved_write_frame(fmt_ctx, pkt);
  
  }
  
  
  
  /*
  
  * Audio encoding example
  
  */
  
  static void audio_encode_example(const char *filename)
  
  {
  
      AVPacket pkt;
  
      int i, j, k, ret, got_output;
  
      int buffer_size;
  
  
  
      uint16_t *samples;
  
      float t, tincr;
  
  
  
      AVCodec *outCodec = NULL;
  
      AVCodecContext *outCodecCtx = NULL;
  
      AVFormatContext *outFormatCtx = NULL;
  
      AVStream * outAudioStream = NULL;
  
      AVFrame *outFrame = NULL;
  
  
  
      ret = avformat_alloc_output_context2(&amp;outFormatCtx, NULL, "rtp", filename);
  
      if (!outFormatCtx || ret &lt; 0)
  
      {
  
          fprintf(stderr, "Could not allocate output context");
  
      }
  
  
  
      outFormatCtx->flags |= AVFMT_FLAG_NOBUFFER | AVFMT_FLAG_FLUSH_PACKETS;
  
      outFormatCtx->oformat->audio_codec = AV_CODEC_ID_PCM_S16BE;
  
  
  
      /* find the encoder */
  
      outCodec = avcodec_find_encoder(outFormatCtx->oformat->audio_codec);
  
      if (!outCodec) {
  
          fprintf(stderr, "Codec not found\n");
  
          exit(1);
  
      }
  
  
  
      outAudioStream = avformat_new_stream(outFormatCtx, outCodec);
  
      if (!outAudioStream)
  
      {
  
          fprintf(stderr, "Cannot add new audio stream\n");
  
          exit(1);
  
      }
  
  
  
      outAudioStream->id = outFormatCtx->nb_streams - 1;
  
      outCodecCtx = outAudioStream->codec;
  
      outCodecCtx->sample_fmt = AV_SAMPLE_FMT_S16;
  
  
  
      /* select other audio parameters supported by the encoder */
  
      outCodecCtx->sample_rate = 8000;
  
      outCodecCtx->channel_layout = AV_CH_LAYOUT_STEREO;
  
      outCodecCtx->channels = 2;
  
  
  
      /* open it */
  
      if (avcodec_open2(outCodecCtx, outCodec, NULL) &lt; 0) {
  
          fprintf(stderr, "Could not open codec\n");
  
          exit(1);
  
      }
  
  
  
      // PCM has no frame, so we have to explicitly specify
  
      outCodecCtx->frame_size = 1152;
  
  
  
      av_dump_format(outFormatCtx, 0, filename, 1);
  
  
  
      char buff[10000] = { 0 };
  
      ret = av_sdp_create(&amp;outFormatCtx, 1, buff, sizeof(buff));
  
      printf("%s", buff);
  
  
  
      ret = avio_open2(&amp;outFormatCtx->pb, filename, AVIO_FLAG_WRITE, NULL, NULL);
  
      ret = avformat_write_header(outFormatCtx, NULL);
  
      printf("ret = %d\n", ret);
  
      if (ret &lt;0) {
  
          exit(1);
  
      }
  
  
  
      /* frame containing input audio */
  
      outFrame = av_frame_alloc();
  
      if (!outFrame) {
  
          fprintf(stderr, "Could not allocate audio frame\n");
  
          exit(1);
  
      }
  
  
  
      outFrame->nb_samples = outCodecCtx->frame_size;
  
      outFrame->format = outCodecCtx->sample_fmt;
  
      outFrame->channel_layout = outCodecCtx->channel_layout;
  
  
  
      /* we calculate the size of the samples buffer in bytes */
  
      buffer_size = av_samples_get_buffer_size(NULL, outCodecCtx->channels, outCodecCtx->frame_size,
  
          outCodecCtx->sample_fmt, 0);
  
      if (buffer_size &lt; 0) {
  
          fprintf(stderr, "Could not get sample buffer size\n");
  
          exit(1);
  
      }
  
      samples = (uint16_t*)av_malloc(buffer_size);
  
      if (!samples) {
  
          fprintf(stderr, "Could not allocate %d bytes for samples buffer\n",
  
              buffer_size);
  
          exit(1);
  
      }
  
      /* setup the data pointers in the AVFrame */
  
      ret = avcodec_fill_audio_frame(outFrame, outCodecCtx->channels, outCodecCtx->sample_fmt,
  
          (const uint8_t*)samples, buffer_size, 0);
  
      if (ret &lt; 0) {
  
          fprintf(stderr, "Could not setup audio frame\n");
  
          exit(1);
  
      }
  
  
  
      /* encode a single tone sound */
  
      t = 0;
  
      int next_pts = 0;
  
      tincr = 2 * M_PI * 440.0 / outCodecCtx->sample_rate;
  
      for (i = 0; i &lt; 400000; i++) {
  
          av_init_packet(&amp;pkt);
  
          pkt.data = NULL; // packet data will be allocated by the encoder
  
          pkt.size = 0;
  
  
  
          for (j = 0; j &lt; outCodecCtx->frame_size; j++) {
  
              samples[2 * j] = (uint16_t)(sin(t) * 10000);
  
  
  
              for (k = 1; k &lt; outCodecCtx->channels; k++)
  
                  samples[2 * j + k] = samples[2 * j];
  
              t += tincr;
  
          }
  
          t = (t > 50000) ? 0 : t;
  
  
  
          // Sets time stamp
  
          next_pts += outFrame->nb_samples;
  
          outFrame->pts = next_pts;
  
  
  
          /* encode the samples */
  
          ret = avcodec_encode_audio2(outCodecCtx, &amp;pkt, outFrame, &amp;got_output);
  
          if (ret &lt; 0) {
  
              fprintf(stderr, "Error encoding audio frame\n");
  
              exit(1);
  
          }
  
          if (got_output) {
  
              write_frame(outFormatCtx, &amp;outCodecCtx->time_base, outAudioStream, &amp;pkt);
  
              av_packet_unref(&amp;pkt);
  
          }
  
  
  
          printf("i:%d\n", i); // waste some time to avoid over-filling jitter buffer
  
          printf("Audio: %d\t%d\n", samples[0], samples[1]); // waste some time to avoid over-filling jitter buffer
  
          printf("t: %f\n", t); // waste some time to avoid over-filling jitter buffer
  
      }
  
  
  
      /* get the delayed frames */
  
      for (got_output = 1; got_output; i++) {
  
          ret = avcodec_encode_audio2(outCodecCtx, &amp;pkt, NULL, &amp;got_output);
  
          if (ret &lt; 0) {
  
              fprintf(stderr, "Error encoding frame\n");
  
              exit(1);
  
          }
  
  
  
          if (got_output) {
  
              pkt.pts = AV_NOPTS_VALUE;
  
              write_frame(outFormatCtx, &amp;outCodecCtx->time_base, outAudioStream, &amp;pkt);
  
              av_packet_unref(&amp;pkt);
  
          }
  
      }
  
  
  
      av_freep(&amp;samples);
  
      av_frame_free(&amp;outFrame);
  
      avcodec_close(outCodecCtx);
  
      av_free(outCodecCtx);
  
  }
  
  
  
  
  
  int main(int argc, char **argv)
  
  {
  
      const char *output;
  
  
  
      av_register_all();
  
      avformat_network_init(); // for network streaming
  
  
  
      audio_encode_example("rtp://127.0.0.1:8554");
  
  
  
      return 0;
  
  }

  Update

  Curiously, running on Linux Ubuntu gives me the following instead :

  Output #0, rtp, to 'rtp://127.0.0.1:8554':
  
      Stream #0:0: Unknown: none (pcm_s16be)
  
  v=0
  
  o=- 0 0 IN IP4 127.0.0.1
  
  s=No Name
  
  c=IN IP4 127.0.0.1
  
  t=0 0
  
  a=tool:libavformat 57.48.100
  
  m=application 8554 RTP/AVP 3

  Does anyone know why the stream has been changed from audio to application ?

• Finding File Matches & Variable Assignment using a .BAT Script

  6 octobre 2019, par A Person — batch-file, ffmpeg

  I am trying to assign a file to a variable in a batch-file and then also assign another 2 files into anoter variable.

  However, I am having an issue.

  From research, I found how I can do the assigning but does anyone know how I can do the below.

  From a folder or text file, (either is fine), find the .m2v video file and assign that to Var1 then find matching audio in .wav and put that in Var2 and the third is also an audio .wav with mathcing name and assign that to Var3.

  The problem I am having is trying to find the matching 2 audio files to the video.

  The video file is named as :

  PAV_PRG_13683Highc450277201906251802090353.m2v

  Audio 1 is :

  PAV_PRG_13683High01c450211201906251802090376.wav

  Audio 2 is :

  PAV_PRG_13683High00c450211201906251802090368.wav

  The file name matches until it sees the word High. Everything after High is not needed, (it is just a random string), so trying to match is an issue.

  Is there a way to find the match by comparing everything before High.

  Also as I will be using the variable and putting them through ffmpeg to merge, is there way to do it so that when the ffmpeg command has completed, it moves to the next matching files and assigns them to the variable.

  The files are store in 2 folders, one folder has all the video files *.m2v and another folder has all the *.wav audio files in pairs of 2. Each video has exactly 2 audios, (left right).

  is there any help on this subject, I have already come up empty in my research and have been checking for this over the last week spent almost 30 hours.