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• FFMpeg - add sound to video that already contain sound

  24 mars 2016, par jacky brown

  Here is what I have : input1.avi - video that contain sounds. input2.avi - video that doesn’t contain sounds. music.mp3 - audio file.

  I want to add background music(music.mp3 file) to the video.

  C :\input1.avi -i C :\music.mp3 -shortest -c:v copy -c:a copy C :\output1.avi
  then output1.avi is the same as input1 - movie with sounds but without the background music (music.mp3)

  when I try to use the other file (video without sounds) :

  C :\input2.avi -i C :\music.mp3 -shortest -c:v copy -c:a copy C :\output2.avi
  then output2.avi is the same as input2 + it have the background music.

  I tried to execute this too :

  C:\ffmpeg\bin>ffmpeg -i C:\input.avi -i C:\music.mp3 -shortest -c:v copy -filter_ complex "[1]volume=1.5[1a];[0][1a]amerge[a]" -map 0:v -map "[a]" -ac 2 C:\output1.avi

  but got the next error messsage :

  ffmpeg version N-78949-g6f5048f Copyright (c) 2000-2016 the FFmpeg developers
  
    built with gcc 5.3.0 (GCC)
  
    configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av
  
  isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab
  
  le-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --
  
  enable-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm --enable-l
  
  ibilbc --enable-libmodplug --enable-libmfx --enable-libmp3lame --enable-libopenc
  
  ore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --ena
  
  ble-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable
  
  -libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --ena
  
  ble-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx
  
  264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable
  
  -lzma --enable-decklink --enable-zlib
  
    libavutil      55. 19.100 / 55. 19.100
  
    libavcodec     57. 27.101 / 57. 27.101
  
    libavformat    57. 28.100 / 57. 28.100
  
    libavdevice    57.  0.101 / 57.  0.101
  
    libavfilter     6. 39.100 /  6. 39.100
  
    libswscale      4.  0.100 /  4.  0.100
  
    libswresample   2.  0.101 /  2.  0.101
  
    libpostproc    54.  0.100 / 54.  0.100
  
  Input #0, avi, from 'C:\output1.avi':
  
    Metadata:
  
      encoder         : Lavf57.28.100
  
    Duration: 00:02:05.76, start: 0.000000, bitrate: 450 kb/s
  
      Stream #0:0: Video: mpeg4 (Simple Profile) (XVID / 0x44495658), yuv420p, 720
  
  x480 [SAR 1:1 DAR 3:2], 440 kb/s, 25 fps, 25 tbr, 25 tbn, 25 tbc
  
      Stream #0:1: Audio: mp3 (U[0][0][0] / 0x0055), 44100 Hz, stereo, s16p, 128 k
  
  b/s
  
  [mp3 @ 00000000005abc20] Skipping 0 bytes of junk at 32370.
  
  Input #1, mp3, from 'C:\music.mp3':
  
    Metadata:
  
      title           : Broadcast News Package - News Intro
  
      artist          : After Effects News Template
  
    Duration: 00:01:57.89, start: 0.025057, bitrate: 194 kb/s
  
      Stream #1:0: Audio: mp3, 44100 Hz, stereo, s16p, 192 kb/s
  
      Metadata:
  
        encoder         : Lavc56.26
  
  [Parsed_amerge_1 @ 0000000000610200] No channel layout for input 1
  
  [Parsed_amerge_1 @ 0000000000610200] No channel layout for input 2
  
  [AVFilterGraph @ 00000000005ddfe0] The following filters could not choose their
  
  formats: Parsed_amerge_1
  
  Consider inserting the (a)format filter near their input or output.
  
  Error configuring complex filters.
  
  I/O error

  So why input1 does not contain the background music ? and how can I decrease or increase the volume of music.mp3 file ?

• C++ FFmpeg distorted sound when converting audio

  21 septembre 2016, par David Andrei Norgren

  I’m using the FFmpeg library to generate MP4 files containing audio from various files, such as MP3, WAV, OGG, but I’m having some troubles (I’m also putting video in there, but for simplicity’s sake I’m omitting that for this question, since I’ve got that working). My current code opens an audio file, decodes the content and converts it into the MP4 container and finally writes it into the destination file as interleaved frames.

  It works perfectly for most MP3 files, but when inputting WAV or OGG, the audio in the resulting MP4 is slightly distorted and often plays at the wrong speed (up to many times faster or slower).

  I’ve looked at countless of examples of using the converting functions (swr_convert), but I can’t seem to get rid of the noise in the exported audio.

  Here’s how I add an audio stream to the MP4 (outContext is the AVFormatContext for the output file) :

  audioCodec = avcodec_find_encoder(outContext->oformat->audio_codec);
  
  if (!audioCodec)
  
      die("Could not find audio encoder!");
  
  
  
  
  
  // Start stream
  
  audioStream = avformat_new_stream(outContext, audioCodec);
  
  if (!audioStream)
  
      die("Could not allocate audio stream!");
  
  
  
  audioCodecContext = audioStream->codec;
  
  audioStream->id = 1;
  
  
  
  
  
  // Setup
  
  audioCodecContext->sample_fmt = AV_SAMPLE_FMT_S16;
  
  audioCodecContext->bit_rate = 128000;
  
  audioCodecContext->sample_rate = 44100;
  
  audioCodecContext->channels = 2;
  
  audioCodecContext->channel_layout = AV_CH_LAYOUT_STEREO;
  
  
  
  
  
  // Open the codec
  
  if (avcodec_open2(audioCodecContext, audioCodec, NULL) < 0)
  
      die("Could not open audio codec");

  And to open a sound file from MP3/WAV/OGG (from the filename variable)...

  // Create contex
  
  formatContext = avformat_alloc_context();
  
  if (avformat_open_input(&formatContext, filename, NULL, NULL)<0)
  
      die("Could not open file");
  
  
  
  
  
  // Find info
  
  if (avformat_find_stream_info(formatContext, 0)<0)
  
      die("Could not find file info");
  
  
  
  av_dump_format(formatContext, 0, filename, false);
  
  
  
  
  
  // Find audio stream
  
  streamId = av_find_best_stream(formatContext, AVMEDIA_TYPE_AUDIO, -1, -1, NULL, 0);
  
  if (streamId < 0)
  
      die("Could not find Audio Stream");
  
  
  
  codecContext = formatContext->streams[streamId]->codec;
  
  
  
  
  
  // Find decoder
  
  codec = avcodec_find_decoder(codecContext->codec_id);
  
  if (codec == NULL)
  
      die("cannot find codec!");
  
  
  
  
  
  // Open codec
  
  if (avcodec_open2(codecContext, codec, 0)<0)
  
      die("Codec cannot be found");
  
  
  
  
  
  // Set up resample context
  
  swrContext = swr_alloc();
  
  if (!swrContext)
  
      die("Failed to alloc swr context");
  
  
  
  av_opt_set_int(swrContext, "in_channel_count", codecContext->channels, 0);
  
  av_opt_set_int(swrContext, "in_channel_layout", codecContext->channel_layout, 0);
  
  av_opt_set_int(swrContext, "in_sample_rate", codecContext->sample_rate, 0);
  
  av_opt_set_sample_fmt(swrContext, "in_sample_fmt", codecContext->sample_fmt, 0);
  
  
  
  av_opt_set_int(swrContext, "out_channel_count", audioCodecContext->channels, 0);
  
  av_opt_set_int(swrContext, "out_channel_layout", audioCodecContext->channel_layout, 0);
  
  av_opt_set_int(swrContext, "out_sample_rate", audioCodecContext->sample_rate, 0);
  
  av_opt_set_sample_fmt(swrContext, "out_sample_fmt", audioCodecContext->sample_fmt, 0);
  
  
  
  if (swr_init(swrContext))
  
      die("Failed to init swr context");

  Finally, to decode+convert+encode...

  // Allocate and init re-usable frames
  
  audioFrameDecoded = av_frame_alloc();
  
  if (!audioFrameDecoded)
  
          die("Could not allocate audio frame");
  
  
  
  audioFrameDecoded->format = fileCodecContext->sample_fmt;
  
  audioFrameDecoded->channel_layout = fileCodecContext->channel_layout;
  
  audioFrameDecoded->channels = fileCodecContext->channels;
  
  audioFrameDecoded->sample_rate = fileCodecContext->sample_rate;
  
  
  
  audioFrameConverted = av_frame_alloc();
  
  if (!audioFrameConverted)
  
          die("Could not allocate audio frame");
  
  
  
  audioFrameConverted->nb_samples = audioCodecContext->frame_size;
  
  audioFrameConverted->format = audioCodecContext->sample_fmt;
  
  audioFrameConverted->channel_layout = audioCodecContext->channel_layout;
  
  audioFrameConverted->channels = audioCodecContext->channels;
  
  audioFrameConverted->sample_rate = audioCodecContext->sample_rate;
  
  
  
  AVPacket inPacket;
  
  av_init_packet(&inPacket);
  
  inPacket.data = NULL;
  
  inPacket.size = 0;
  
  
  
  int frameFinished = 0;
  
  
  
  while (av_read_frame(formatContext, &inPacket) >= 0) {
  
  
  
          if (inPacket.stream_index == streamId) {
  
  
  
                  int len = avcodec_decode_audio4(fileCodecContext, audioFrameDecoded, &frameFinished, &inPacket);
  
  
  
                  if (frameFinished) {
  
  
  
                          // Convert
  
  
  
                          uint8_t *convertedData=NULL;
  
  
  
                          if (av_samples_alloc(&convertedData,
  
                                               NULL,
  
                                               audioCodecContext->channels,
  
                                               audioFrameConverted->nb_samples,
  
                                               audioCodecContext->sample_fmt, 0) < 0)
  
                                  die("Could not allocate samples");
  
  
  
                          int outSamples = swr_convert(swrContext,
  
                                                       &convertedData,
  
                                                       audioFrameConverted->nb_samples,
  
                                                       (const uint8_t **)audioFrameDecoded->data,
  
                                                       audioFrameDecoded->nb_samples);
  
                          if (outSamples < 0)
  
                                  die("Could not convert");
  
  
  
                          size_t buffer_size = av_samples_get_buffer_size(NULL,
  
                                                                          audioCodecContext->channels,
  
                                                                          audioFrameConverted->nb_samples,
  
                                                                          audioCodecContext->sample_fmt,
  
                                                                          0);
  
                          if (buffer_size < 0)
  
                                  die("Invalid buffer size");
  
  
  
                          if (avcodec_fill_audio_frame(audioFrameConverted,
  
                                                       audioCodecContext->channels,
  
                                                       audioCodecContext->sample_fmt,
  
                                                       convertedData,
  
                                                       buffer_size,
  
                                                       0) < 0)
  
                                  die("Could not fill frame");
  
  
  
                          AVPacket outPacket;
  
                          av_init_packet(&outPacket);
  
                          outPacket.data = NULL;
  
                          outPacket.size = 0;
  
  
  
                          if (avcodec_encode_audio2(audioCodecContext, &outPacket, audioFrameConverted, &frameFinished) < 0)
  
                                  die("Error encoding audio frame");
  
  
  
                          if (frameFinished) {
  
                                  outPacket.stream_index = audioStream->index;
  
  
  
                                  if (av_interleaved_write_frame(outContext, &outPacket) != 0)
  
                                          die("Error while writing audio frame");
  
  
  
                                  av_free_packet(&outPacket);
  
                          }
  
                  }
  
          }
  
  }
  
  
  
  av_frame_free(&audioFrameConverted);
  
  av_frame_free(&audioFrameDecoded);
  
  av_free_packet(&inPacket);

  I have also tried setting appropriate pts values for outgoing frames, but that doesn’t seem to affect the sound quality at all.

  I’m also unsure how/if I should be allocating the converted data, can av_samples_alloc be used for this ? What about avcodec_fill_audio_frame ? Am I on the right track ?

  Any input is appreciated (I can also send the exported MP4s if necessary, if you want to hear the distortion).

• Converting FLAC to AAC outputs no sound using ffmpeg

  29 décembre 2023, par Aleksandar

  I am trying to convert a FLAC audio input file to AAC using ffmpeg, but the output .aac seems to have no sound when opening in VLC. The input container is a .mka containing only the one single audio stream.

  


  I am using this command :

  


  ffmpeg -i en.mka -c:a aac -b:a 512k -map 0:a:0 en4.aac

  


  I tried with map -0 and both 320k and 512k - nothing seems to produce an output file with sound and VLC seemingly can't even determine the length of the file constantly shifting how long it is ?