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).
