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• Produce waveform video from audio using FFMPEG

  27 avril 2017, par RhythmicDevil

  I am trying to create a waveform video from audio. My goal is to produce a video that looks something like this

  

  For my test I have an mp3 that plays a short clipped sound. There are 4 bars of 1/4 notes and 4 bars of 1/8 notes played at 120bpm. I am having some trouble coming up with the right combination of preprocessing and filtering to produce a video that looks like the image. The colors dont have to be exact, I am more concerned with the shape of the beats. I tried a couple of different approaches using showwaves and showspectrum. I cant quite wrap my head around why when using showwaves the beats go past so quickly, but using showspectrum produces a video where I can see each individual beat.

  ShowWaves

  ffmpeg -i beat_test.mp3 -filter_complex "[0:a]showwaves=s=1280x100:mode=cline:rate=25:scale=sqrt,format=yuv420p[v]" -map "[v]" -map 0:a output_wav.mp4

  This link will download the output of that command.

  ShowSpectrum

  ffmpeg -i beat_test.mp3 -filter_complex "[0:a]showspectrum=s=1280x100:mode=combined:color=intensity:saturation=5:slide=1:scale=cbrt,format=yuv420p[v]" -map "[v]" -an -map 0:a output_spec.mp4

  This link will download the output of that command.

  I posted the simple examples because I didn’t want to confuse the issue by adding all the variations I have tried.

  In practice I suppose I can get away with the output from showspectrum but I’d like to understand where/how I am thinking about this incorrectly. Thanks for any advice.

  Here is a link to the source audio file.

• Mix 2 mp3 files with FFmpeg for adroid

  1er avril 2017, par Thankgod Richard

  I am developing something similar to a karaoke app. I have a sound file(beat) and a recorded file(voice) from android mediarecorder. I am trying to mix the two files to become one file that plays the same time. I have set up ffmpeg provided by writingminds and everything is working fine. I am not familiar with ffmpeg commands so searched for and got few commands that can achieve the task for me. This is the code i got :

    String files = "-i " + currentFile + " -i " + someFile.getAbsolutePath();
  
              String output = mFileName + "/recorded/test.mp3";
  
              String cmd = "ffmpeg "+files+
  
                      " -filter_complex \"aevalsrc=0:d=10[s1];[s1][1:a]concat=n=2:v=0:a=1[ac1];[0:a][ac1]amix[aout]\" -map [aout] -c:a libmp3lame " + output;

  But it not working. this is the error i got :

  04-01 12:09:26.598 21300-21300/com.shixels.thankgodrichard.mixer I/fpeg : ffmpeg version n3.0.1 Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 4.8 (GCC)
  configuration : —target-os=linux —cross-prefix=/home/vagrant/SourceCode/ffmpeg-android/toolchain-android/bin/arm-linux-androideabi- —arch=arm —cpu=cortex-a8 —enable-runtime-cpudetect —sysroot=/home/vagrant/SourceCode/ffmpeg-android/toolchain-android/sysroot —enable-pic —enable-libx264 —enable-libass —enable-libfreetype —enable-libfribidi —enable-libmp3lame —enable-fontconfig —enable-pthreads —disable-debug —disable-ffserver —enable-version3 —enable-hardcoded-tables —disable-ffplay —disable-ffprobe —enable-gpl —enable-yasm —disable-doc —disable-shared —enable-static —pkg-config=/home/vagrant/SourceCode/ffmpeg-android/ffmpeg-pkg-config —prefix=/home/vagrant/SourceCode/ffmpeg-android/build/armeabi-v7a —extra-cflags=’-I/home/vagrant/SourceCode/ffmpeg-android/toolchain-android/include -U_FORTIFY_SOURCE -D_FORTIFY_SOURCE=2 -fno-strict-overflow -fstack-protector-all’ —extra-ldflags=’-L/home/vagrant/SourceCode/ffmpeg-android/toolchain-android/lib -Wl,-z,relro -Wl,-z,now -pie’ —extra-libs=’-lpng -lexpat -lm’ —extra-cxxflags=
  libavutil 55. 17.103 / 55. 17.103
  libavcodec 57. 24.102 / 57. 24.102
  libavformat 57. 25.100 / 57. 25.100
  libavdevice 57. 0.101 / 57. 0.101
  libavfilter 6. 31.100 / 6. 31.100
  libswscale 4. 0.100 / 4. 0.100
  libswresample 2. 0.101 / 2. 0.101
  libpostproc 54. 0.100 / 54. 0.100
  Output #0, mp3, to ’ffmpeg -i /storage/emulated/0/MyStudio/temps/1491044929409.mp3 -i /storage/emulated/0/temp.mp3 -filter_complex "aevalsrc=0:d=10[s1] ;[s1][1:a]concat=n=2:v=0:a=1[ac1] ;[0:a][ac1]amix[aout]" -map [aout] -c:a libmp3lame /storage/emulated/0/recorded/test.mp3’ :
  Output file #0 does not contain any stream
  04-01 12:09:26.599 21300-21300/com.shixels.thankgodrichard.mixer I/fpeg : finished

  I read some answers here saying it depends on the version of the ffmpeg your using. I am using the latest ffmpeg for android by writingminds.

• encode .wav file using ffmpeg in objective c or c

  15 février 2017, par deshu

  I have to encode .wav file and write it into same file,or other file using

  ffmpeg library,here is my code for encoding

  -(void)audioencode:(const char *)fileName
  
  {
  
  
  
      AVFrame *frame;
  
      AVPacket pkt;
  
      int i, j, k, ret, got_output;
  
      int buffer_size;
  
      FILE *f;
  
      uint16_t *samples;
  
  
  
      const char *format_name = "wav", 
  
  
  
  
  
  const char *file_url = "/Users/xxxx/Downloads/simple-drum-beat.wav";
  
  
  
      avcodec_register_all();
  
  
  
      av_register_all();
  
  
  
      AVOutputFormat *format = NULL;
  
  
  
      for (AVOutputFormat *formatIter = av_oformat_next(NULL); formatIter != NULL; formatIter = av_oformat_next(formatIter) 
  
  {
  
          int hasEncoder = NULL != avcodec_find_encoder(formatIter->audio_codec);
  
          if (0 == strcmp(format_name, formatIter->name)) {
  
              format = formatIter;
  
              break;
  
          }
  
      }
  
  
  
  
  
      AVCodec *codec = avcodec_find_encoder(format->audio_codec);
  
  
  
      NSLog(@"tet test tststs");
  
  
  
      AVCodecContext *c;
  
  
  
          c = avcodec_alloc_context3(codec);
  
  
  
          if (!c) {
  
              fprintf(stderr, "Could not allocate audio codec context\n");
  
              exit(1);
  
          }
  
  
  
          c->sample_fmt = AV_SAMPLE_FMT_S16;
  
          if (!check_sample_fmt(codec, c->sample_fmt)) {
  
              fprintf(stderr, "Encoder does not support sample format %s",
  
                      av_get_sample_fmt_name(c->sample_fmt));
  
              exit(1);
  
          }
  
      c->bit_rate = 64000;//705600;
  
     c->sample_rate    = select_sample_rate(codec);
  
      c->channel_layout = select_channel_layout(codec);
  
      c->channels       = av_get_channel_layout_nb_channels(c->channel_layout);
  
      c->frame_size = av_get_audio_frame_duration(c, 16);
  
  
  
      int bits_per_sample = av_get_bits_per_sample(c->codec_id);
  
  
  
  
  
      int frameSize   = av_get_audio_frame_duration(c,16);
  
  
  
          /* open it */
  
          if (avcodec_open2(c, codec, NULL) < 0) {
  
              fprintf(stderr, "Could not open codec\n");
  
              exit(1);
  
          }
  
  
  
          f = fopen(fileName, "wb");
  
          if (!f) {
  
              fprintf(stderr, "Could not open %s\n", fileName);
  
              exit(1);
  
          }
  
  
  
          /* frame containing input raw audio */
  
          frame = av_frame_alloc();
  
          if (!frame) {
  
              fprintf(stderr, "Could not allocate audio frame\n");
  
              exit(1);
  
          }
  
  
  
          frame->nb_samples     = frameSize/*c->frame_size*/;
  
          frame->format         = c->sample_fmt;
  
          frame->channel_layout = c->channel_layout;
  
  
  
          buffer_size = av_samples_get_buffer_size(NULL, c->channels,frameSize /*c->frame_size*/,
  
                                                   c->sample_fmt, 0);
  
          samples = 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(frame, c->channels, c->sample_fmt,
  
                                         (const uint8_t*)samples, buffer_size, 0);
  
          if (ret < 0) {
  
              fprintf(stderr, "Could not setup audio frame\n");
  
              exit(1);
  
          }
  
  
  
      float t, tincr;
  
          /* encode a single tone sound */
  
          t = 0;
  
          tincr = 2 * M_PI * 440.0 / c->sample_rate;
  
          for(i=0;i<800;i++) {
  
              av_init_packet(&pkt);
  
              pkt.data = NULL; // packet data will be allocated by the encoder
  
              pkt.size = 0;
  
  
  
              for (j = 0; j < frameSize/*c->frame_size*/; j++) {
  
                  samples[2*j] = (int)(sin(t) * 10000);
  
  
  
                  for (k = 1; k < c->channels; k++)
  
                      samples[2*j + k] = samples[2*j];
  
                  t += tincr;
  
              }
  
              /* encode the samples */
  
              ret = avcodec_encode_audio2(c, &pkt, frame, &got_output);
  
              if (ret < 0) {
  
                  fprintf(stderr, "Error encoding audio frame\n");
  
                  exit(1);
  
              }
  
              if (got_output) {
  
                  fwrite(pkt.data, 1, pkt.size, f);
  
                  av_free_packet(&pkt);
  
              }
  
          }
  
  }

  but after encoded file size is zero,
  Please suggest what m doing wrong,any help will be appreciate, thanks in advance