Decoding m4a and dumping PCM data gives back noise - c++

Question

I'm using the code below (modified from the examples given in libavcodec) to decode audio files
int main(int argc, char **argv)
{
av_register_all();
avcodec_register_all();
char *filename = argv[1];
char *outfilename = argv[2];
FILE *outfile;
AVCodec *codec;
AVCodecContext *c= NULL;
AVPacket avpkt;
AVFrame *frame = av_frame_alloc();
printf("Decode audio file %s to %s\n", filename, outfilename);
outfile = fopen(outfilename, "wb");
if (!outfile) {
fprintf(stderr, "Could not write to %s\n", outfilename);
av_free(c);
exit(1);
}
AVFormatContext *format_context = NULL;
avformat_open_input(&format_context, filename, NULL, NULL);
printf("Opened format input\n");
int find_result = avformat_find_stream_info(format_context, NULL);
if (find_result < 0) {
fprintf(stderr, "Cannot find stream info\n");
avformat_close_input(&format_context);
exit(-1);
}
int audio_stream_idx = av_find_best_stream(format_context, AVMEDIA_TYPE_AUDIO, -1, -1, &codec, 0);
if (audio_stream_idx < 0) {
fprintf(stderr,"Couldn't find stream information\n");
exit(-1);
}
// Get a pointer to the codec context for the audio stream
c = format_context->streams[audio_stream_idx]->codec;
av_opt_set_int(c, "refcounted_frames", 1, 0);
if (avcodec_open2(c, codec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
exit(-1);
}
// read the audio frames
int ret, got_frame;
while (1) {
if ((ret = av_read_frame(format_context, &avpkt)) < 0)
break;
if (avpkt.stream_index == audio_stream_idx) {
avcodec_get_frame_defaults(frame);
got_frame = 0;
ret = avcodec_decode_audio4(c, frame, &got_frame, &avpkt);
if (ret < 0) {
fprintf(stderr, "Error decoding audio\n");
continue;
}
if (got_frame) {
// write to disk
fwrite(frame->extended_data[0], 1, frame->linesize[0], outfile);
}
}
av_free_packet(&avpkt);
}
fclose(outfile);
printf("Finished\n");
if (c)
avcodec_close(c);
avformat_close_input(&format_context);
av_frame_free(&frame);
}
I tried .mp3 and .m4a files; .mp3 files work fine but not for .m4a files. Any help?

Answer 1

Most aac files are in the FLOAT format i.e. AV_SAMPLE_FMT_FLTP, while libraries such as libao are only able to play audio as integer format i.e. AV_SAMPLE_FORMAT_S16.
As a result, you'd need to use libavresample to resample the music and convert the format as appropriate.