Playing audio using ffmpeg and AVAudioPlayer - ios

Question

I am trying to read an audio file (that is not supported by iOS) with ffmpeg and then play it using AVAudioPlayer. It took me a while to get ffmpeg built inside an iOS project, but I finally did using kewlbear/FFmpeg-iOS-build-script.
This is the snippet I have right now, after a lot of searching on the web, including stackoverflow. One of the best examples I found was here.
I believe this is all the relevant code. I added comments to let you know what I'm doing and where I need something clever to happen.
#import "FFmpegWrapper.h"
#import <AVFoundation/AVFoundation.h>
AVFormatContext *formatContext = NULL;
AVStream *audioStream = NULL;
av_register_all();
avformat_network_init();
avcodec_register_all();
// this is a file locacted on my NAS
int opened = avformat_open_input(&formatContext, #"http://192.168.1.70:50002/m/NDLNA/43729.flac", NULL, NULL);
// can't open file
if(opened == 1) {
avformat_close_input(&formatContext);
}
int streamInfoValue = avformat_find_stream_info(formatContext, NULL);
// can't open stream
if (streamInfoValue < 0)
{
avformat_close_input(&formatContext);
}
// number of streams available
int inputStreamCount = formatContext->nb_streams;
for(unsigned int i = 0; i<inputStreamCount; i++)
{
// I'm only interested in the audio stream
if(formatContext->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO)
{
// found audio stream
audioStream = formatContext->streams[i];
}
}
if(audioStream == NULL) {
// no audio stream
}
AVFrame* frame = av_frame_alloc();
AVCodecContext* codecContext = audioStream->codec;
codecContext->codec = avcodec_find_decoder(codecContext->codec_id);
if (codecContext->codec == NULL)
{
av_free(frame);
avformat_close_input(&formatContext);
// no proper codec found
}
else if (avcodec_open2(codecContext, codecContext->codec, NULL) != 0)
{
av_free(frame);
avformat_close_input(&formatContext);
// could not open the context with the decoder
}
// this is displaying: This stream has 2 channels and a sample rate of 44100Hz
// which makes sense
NSLog(#"This stream has %d channels and a sample rate of %dHz", codecContext->channels, codecContext->sample_rate);
AVPacket packet;
av_init_packet(&packet);
// this is where I try to store in the sound data
NSMutableData *soundData = [[NSMutableData alloc] init];
while (av_read_frame(formatContext, &packet) == 0)
{
if (packet.stream_index == audioStream->index)
{
// Try to decode the packet into a frame
int frameFinished = 0;
avcodec_decode_audio4(codecContext, frame, &frameFinished, &packet);
// Some frames rely on multiple packets, so we have to make sure the frame is finished before
// we can use it
if (frameFinished)
{
// this is where I think something clever needs to be done
// I need to store some bytes, but I can't figure out what exactly and what length?
// should the length be multiplied by the of the number of channels?
NSData *frameData = [[NSData alloc] initWithBytes:packet.buf->data length:packet.buf->size];
[soundData appendData: frameData];
}
}
// You *must* call av_free_packet() after each call to av_read_frame() or else you'll leak memory
av_free_packet(&packet);
}
// first try to write it to a file, see if that works
// this is indeed writing bytes, but it is unplayable
[soundData writeToFile:#"output.wav" atomically:YES];
NSError *error;
// this is my final goal, playing it with the AVAudioPlayer, but this is giving unclear errors
AVAudioPlayer *player = [[AVAudioPlayer alloc] initWithData:soundData error:&error];
if(player == nil) {
NSLog(error.description); // Domain=NSOSStatusErrorDomain Code=1954115647 "(null)"
} else {
[player prepareToPlay];
[player play];
}
// Some codecs will cause frames to be buffered up in the decoding process. If the CODEC_CAP_DELAY flag
// is set, there can be buffered up frames that need to be flushed, so we'll do that
if (codecContext->codec->capabilities & CODEC_CAP_DELAY)
{
av_init_packet(&packet);
// Decode all the remaining frames in the buffer, until the end is reached
int frameFinished = 0;
while (avcodec_decode_audio4(codecContext, frame, &frameFinished, &packet) >= 0 && frameFinished)
{
}
}
av_free(frame);
avcodec_close(codecContext);
avformat_close_input(&formatContext);

Answer 1

Not really found a solution to this specific problem, but ended up using ap4y/OrigamiEngine instead.
My main reason I wanted to use FFmpeg is to play unsupported audio files (FLAC/OGG) on iOS and tvOS and OrigamiEngine does the job just fine.