How do I use FFMPEG/libav to access the data in individual audio samples? - audio

Question

The end result is I'm trying to visualise the audio waveform to use in a DAW-like software. So I want to get each sample's value and draw it. With that in mind, I'm currently stumped by trying to gain access to the values stored in each sample. For the time being, I'm just trying to access the value in the first sample - I'll build it into a loop once I have some working code.
I started off by following the code in this example. However, LibAV/FFMPEG has been updated since then, so a lot of the code is deprecated or straight up doesn't work the same anymore.
Based on the example above, I believe the logic is as follows:
get the formatting info of the audio file
get audio stream info from the format
check that the codec required for the stream is an audio codec
get the codec context (I think this is info about the codec) - This is where it gets kinda confusing for me
create an empty packet and frame to use - packets are for holding compressed data and frames are for holding uncompressed data
the format reads the first frame from the audio file into our packet
pass that packet into the codec context to be decoded
pass our frame to the codec context to receive the uncompressed audio data of the first frame
create a buffer to hold the values and try allocating samples to it from our frame
From debugging my code, I can see that step 7 succeeds and the packet that was empty receives some data. In step 8, the frame doesn't receive any data. This is what I need help with. I get that if I get the frame, assuming a stereo audio file, I should have two samples per frame, so really I just need your help to get uncompressed data into the frame.
I've scoured through the documentation for loads of different classes and I'm pretty sure I'm using the right classes and functions to achieve my goal, but evidently not (I'm also using Qt, so I'm using qDebug throughout, and QString to hold the URL for the audio file as path). So without further ado, here's my code:
// Step 1 - get the formatting info of the audio file
AVFormatContext* format = avformat_alloc_context();
if (avformat_open_input(&format, path.toStdString().c_str(), NULL, NULL) != 0) {
qDebug() << "Could not open file " << path;
return -1;
}
// Step 2 - get audio stream info from the format
if (avformat_find_stream_info(format, NULL) < 0) {
qDebug() << "Could not retrieve stream info from file " << path;
return -1;
}
// Step 3 - check that the codec required for the stream is an audio codec
int stream_index =- 1;
for (unsigned int i=0; i<format->nb_streams; i++) {
if (format->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
stream_index = i;
break;
}
}
if (stream_index == -1) {
qDebug() << "Could not retrieve audio stream from file " << path;
return -1;
}
// Step 4 -get the codec context
const AVCodec *codec = avcodec_find_decoder(format->streams[stream_index]->codecpar->codec_id);
AVCodecContext *codecContext = avcodec_alloc_context3(codec);
avcodec_open2(codecContext, codec, NULL);
// Step 5 - create an empty packet and frame to use
AVPacket *packet = av_packet_alloc();
AVFrame *frame = av_frame_alloc();
// Step 6 - the format reads the first frame from the audio file into our packet
av_read_frame(format, packet);
// Step 7 - pass that packet into the codec context to be decoded
avcodec_send_packet(codecContext, packet);
//Step 8 - pass our frame to the codec context to receive the uncompressed audio data of the first frame
avcodec_receive_frame(codecContext, frame);
// Step 9 - create a buffer to hold the values and try allocating samples to it from our frame
double *buffer;
av_samples_alloc((uint8_t**) &buffer, NULL, 1, frame->nb_samples, AV_SAMPLE_FMT_DBL, 0);
qDebug () << "packet: " << &packet;
qDebug() << "frame: " << frame;
qDebug () << "buffer: " << buffer;
For the time being, step 9 is incomplete as you can probably tell. But for now, I need help with step 8. Am I missing a step, using the wrong function, wrong class? Cheers.