int initialize_encoding_audio(const char *filename)
    {
        int ret;
        AVCodecID aud_codec_id = AV_CODEC_ID_AAC;
        AVSampleFormat sample_fmt = AV_SAMPLE_FMT_FLTP;

        avcodec_register_all();
        av_register_all();

        aud_codec = avcodec_find_encoder(aud_codec_id);
        avcodec_register(aud_codec);

        if (!aud_codec)
            return COULD_NOT_FIND_AUD_CODEC;

        aud_codec_context = avcodec_alloc_context3(aud_codec);
        if (!aud_codec_context)
            return CONTEXT_CREATION_ERROR;

        aud_codec_context->bit_rate = 192000;
        aud_codec_context->sample_rate = select_sample_rate(aud_codec);
        aud_codec_context->sample_fmt = sample_fmt;
        aud_codec_context->channel_layout = AV_CH_LAYOUT_STEREO;
        aud_codec_context->channels = av_get_channel_layout_nb_channels(aud_codec_context->channel_layout);

        aud_codec_context->codec = aud_codec;
        aud_codec_context->codec_id = aud_codec_id;

        ret = avcodec_open2(aud_codec_context, aud_codec, NULL);

        if (ret < 0)
            return COULD_NOT_OPEN_AUD_CODEC;

        outctx = avformat_alloc_context();
        ret = avformat_alloc_output_context2(&outctx, NULL, "mp4", filename);

        outctx->audio_codec = aud_codec;
        outctx->audio_codec_id = aud_codec_id;

        audio_st = avformat_new_stream(outctx, aud_codec);

        audio_st->codecpar->bit_rate = aud_codec_context->bit_rate;
        audio_st->codecpar->sample_rate = aud_codec_context->sample_rate;
        audio_st->codecpar->channels = aud_codec_context->channels;
        audio_st->codecpar->channel_layout = aud_codec_context->channel_layout;
        audio_st->codecpar->codec_id = aud_codec_id;
        audio_st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
        audio_st->codecpar->format = sample_fmt;
        audio_st->codecpar->frame_size = aud_codec_context->frame_size;
        audio_st->codecpar->block_align = aud_codec_context->block_align;
        audio_st->codecpar->initial_padding = aud_codec_context->initial_padding;

        outctx->streams = new AVStream*[1];
        outctx->streams[0] = audio_st;

        av_dump_format(outctx, 0, filename, 1);

        if (!(outctx->oformat->flags & AVFMT_NOFILE))
        {
            if (avio_open(&outctx->pb, filename, AVIO_FLAG_WRITE) < 0)
                return COULD_NOT_OPEN_FILE;
        }

        ret = avformat_write_header(outctx, NULL);

        aud_frame = av_frame_alloc();
        aud_frame->nb_samples = aud_codec_context->frame_size;
        aud_frame->format = aud_codec_context->sample_fmt;
        aud_frame->channel_layout = aud_codec_context->channel_layout;

        int buffer_size = av_samples_get_buffer_size(NULL, aud_codec_context->channels, aud_codec_context->frame_size,
            aud_codec_context->sample_fmt, 0);

        av_frame_get_buffer(aud_frame, buffer_size / 2);

        if (!aud_frame)
            return COULD_NOT_ALLOCATE_FRAME;

        aud_frame_counter = 0;

        return 0;
    }

    float t, tincr, tincr2;
    void get_audio_frame(float_t *samples, int frame_size)
    {
        int j, i;
        float v;
        for (j = 0; j < frame_size; j++)
        {
            v = sin(t);

            *samples = v;
            samples++;

            t += tincr;
            tincr += tincr2;
        }
    }
    int encode_audio_samples(uint8_t **aud_samples)
    {
        int ret;

        int buffer_size = av_samples_get_buffer_size(NULL, aud_codec_context->channels, aud_codec_context->frame_size,
            aud_codec_context->sample_fmt, 0);

        //This looks bad to me, but it's been so long since I've coded in c++ (work with c#) so I'm just trying
        for (size_t i = 0; i < buffer_size / 2; i++)
        {
            aud_frame->data[0][i] = aud_samples[0][i];
            aud_frame->data[1][i] = aud_samples[1][i];
        }

        aud_frame->pts = aud_frame_counter++;

        ret = avcodec_send_frame(aud_codec_context, aud_frame);
        if (ret < 0)
            return ERROR_ENCODING_SAMPLES_SEND;

        AVPacket pkt;
        av_init_packet(&pkt);
        pkt.data = NULL;
        pkt.size = 0;

        while (true)
        {
            ret = avcodec_receive_packet(aud_codec_context, &pkt);
            if (!ret)
            {
                av_packet_rescale_ts(&pkt, aud_codec_context->time_base, audio_st->time_base);

                pkt.stream_index = audio_st->index;
                av_write_frame(outctx, &pkt);
                av_packet_unref(&pkt);
            }
            if (ret == AVERROR(EAGAIN))
                break;
            else if (ret < 0)
                return ERROR_ENCODING_SAMPLES_RECEIVE;
            else
                break;
        }

        return 0;
    }


    int finish_audio_encoding()
    {
        AVPacket pkt;
        av_init_packet(&pkt);
        pkt.data = NULL;
        pkt.size = 0;

        fflush(stdout);

        int ret = avcodec_send_frame(aud_codec_context, NULL);
        if (ret < 0)
            return ERROR_ENCODING_FRAME_SEND;

        while (true)
        {
            ret = avcodec_receive_packet(aud_codec_context, &pkt);
            if (!ret)
            {
                if (pkt.pts != AV_NOPTS_VALUE)
                    pkt.pts = av_rescale_q(pkt.pts, aud_codec_context->time_base, audio_st->time_base);
                if (pkt.dts != AV_NOPTS_VALUE)
                    pkt.dts = av_rescale_q(pkt.dts, aud_codec_context->time_base, audio_st->time_base);

                av_write_frame(outctx, &pkt);
                av_packet_unref(&pkt);
            }
            if (ret == -AVERROR(AVERROR_EOF))
                break;
            else if (ret < 0)
                return ERROR_ENCODING_FRAME_RECEIVE;
        }

        av_write_trailer(outctx);
    }

void get_audio_frame(float_t *samples, int frame_size, float* t, float* tincr, float* tincr2)
{
    int j, i;
    float v;
    for (j = 0; j < frame_size; j++)
    {
        v = sin(*t);

        *samples = v;
        samples++;

        *t += *tincr;
        *tincr += *tincr2;
    }
}

int main()
{
    int frame_rate = 30; // this should be like 96000 / 1024 or somthing i guess?
    float t, tincr, tincr2;

    initialize_encoding_audio("audio.mp4");

    int sec = 10;

    float_t** aud_samples;
    int src_nb_samples = 1024;
    int src_channels = 2;

    int ret = av_samples_alloc_array_and_samples((uint8_t***)&aud_samples, &src_samples_linesize, src_channels,
        src_nb_samples, AV_SAMPLE_FMT_FLTP, 0);

    for (size_t i = 0; i < frame_rate * sec; i++)
    {
        get_audio_frame(aud_samples[0], src_nb_samples, &t, &tincr, &tincr2);
        get_audio_frame(aud_samples[1], src_nb_samples, &t, &tincr, &tincr2);

        encode_audio_samples((uint8_t **)aud_samples);

    }

    finish_audio_encoding();
    cleanup();

    return 0;
}