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• C++ FFmpeg distorted sound when converting audio

  8 juin 2020, par david

  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).

  


• Automatically fade audio

  25 mars 2019, par Nadav Tasher

  I have a working script that plays a different audio every 45 minutes.

  The audio files change on a daily basis, so i cant manually fade them in/out in Audacity.

  In the script, i have the following line :

  ffplay -i /home/user/Ringtone1.mp3 -ss 00:00:7 -t 20 -nodisp -autoexit

  But every time, the audio cuts abruptly after 20 seconds, which isn’t very ear-pleasing.

  I want to make the audio fade in from 0 to max in 5 seconds and then fade out from max to 0 in 5 seconds (at the end of the 20 seconds).

  How can i do that ?

• FFmpeg muxing theora/vorbis unable to flush ?

  11 novembre 2013, par user2979732

  I'm pretty new to ffmpeg and it's confusing. I'm working on a basic muxer and have been spending over a week on this - I don't normally post as I solve 98% of my issues with google, but unable to get this one so far.

  The basis of my source is FFmpeg's own muxing.c example. When I try to force it using libvorbis for audio, and create "test.ogg" it demonstrates the same issues I'm having in my own derivation of muxing.c. The problem is with ogg/theora/vorbis. I'm forcing the use of audio codec like this :

  audio_st = add_stream(oc, &audio_codec, avcodec_find_encoder_by_name("libvorbis")->id);

  It seems the problem is in not setting audio pts in the muxing.c sample. There is a confusion in general about this, nobody apart from this guy didn't address what I am looking for http://webcache.googleusercontent.com/search?q=cache:6ml82RMN3YYJ:ffmpeg.org/pipermail/libav-user/2013-April/004304.html+&cd=4&hl=en&ct=clnk&gl=cz

  I couldn't find any answers to that naturally - like why don't they set the audio pts ? Laziness ? Not needed ? Do they believe all encoders will produce the pts for them(not true as seen below) ?

  Anyway, when you try muxing.c with mp4/libx264/forced libmp3lame all is fine, but the encoder says that "encoder did not produce valid pts, making some up.". However, it's silent with ogg/theora/vorbis, as if there were valid pts(?) but the result is no audio packets present in the stream(!), at least from what I saw using ffprobe. Which results in the video not being able to replay even, until you take out the empty audio stream. Then it plays the video, which shows that stream is fine.

  Coming to my original issue. I tried setting the pts on the audio frame you're sending to the encoder to fix that problem(this already sucks). I was unable to find a definite answer how to properly set the pts - that's the other big issue as I'm trying stuff which I'm not sure works. Anyway, in the end when setting "some" pts, this results in ogg with sound.

  if (frame->pts == AV_NOPTS_VALUE) frame->pts = audio_sync_opts;
  
  audio_sync_opts = frame->pts + frame->nb_samples;

  I'm aware I should probably use rescaling to adjust for the container time bases etc..if this was present/explained in ffmpeg's own sample I wouldn't have to guess now (as I'm stil not 100% sure about time base relationship between container and codec, I think container time base takes somehow over the codec one).

  My other problem is flushing - but that might have something to do with the screwed up pts. So I won't rather get into that in detail - the basic problem is, when I send finite number of audio frames, like 20, I get 2 packets only for example. From my understanding, I need to flush the rest of audio after all the encoding/muxing is done, which I managed to do with mp4/libx264/libmp3lame, but with ogg/theora/vorbis it doesn't flush. Why not, I have no idea.

  If someone could rework muxing.c into sending it finite number of audio / video frames - ie . not until duration > X, but until it sent 20 video & 100 audio frames(just an example). So that number of frames I have is important, not the video time I end up with. Then encode / mux all the frames - with proper video/audio pts, working with theora/ogg and flushing if needed, that would probably solve all of my issues. I'm sure for an expert ffmpeg'er modifying muxing.c addressing all those things would be a pretty quick exercise and could help more than 1 confused person.

  Thanks !