How to fill AVFrame with audio data

Question

I use Win32 API to implement a desktop recording program. The video capture feature works, but I have no idea to put the audio in the FFmpeg encoder's data structure.

The below code will get the data of the default output audio.

// Set wave format when sampling the audio
WAVEFORMATEX wf; 
wf->wFormatTag = WAVE_FORMAT_PCM;
wf->nChannels = 1;
wf->nSamplesPerSec = 12000;
wf->nBlockAlign = 1;
wf->wBitsPerSample = 8;
wf->cbSize = 0;
wf->nAvgBytesPerSec = nChannels * nSamplesPerSec * wBitsPerSample / 8;

// Open wave input channel
HWAVEIN hWaveIn;
OpenWaveIn(&hWaveIn,&wf);

// Prepare Wave In Header and allocate memory
WAVEHDR waveHdr;
DWORD dataSize = 240000L;
 PrepareWaveIn(&hWaveIn, &waveHdr, dataSize);

// Start recording
StartRecord(&hWaveIn);

// Stop recording
MMTIME mmt;
StopRecord(&hWaveIn, &mmt);

waveHdr->lpData is a pointer to locked data buffer.

The question is how can I implement the below function let the waveHdr->->lpData can be saved to AVFrame?

static AVFrame *get_audio_frame(OutputStream *ost)
{
    AVFrame *frame = ost->tmp_frame;
    int j, i, v;
    int16_t *q = (int16_t*)frame->data[0];
    AVRational arg;
    arg.num = 1;
    arg.den = 1;

    //the transform code should be put here

    frame->pts = ost->next_pts;
    ost->next_pts += frame->nb_samples;
    return frame;
}

Does someone know how can implement this?

Answer 1

Pasting from http://ffmpeg.org/doxygen/trunk/encoding-example_8c.html. This is the audio_encode_example() function, so it should contain everything you need to know about how to encode the audio frames. Note that they create a simple sinusoidal sound, that is, the generate the sound wave from sin().

/*
 * Audio encoding example
 */
static void audio_encode_example(const char *filename)
{
    AVCodec *codec;
    AVCodecContext *c= NULL;
    int frame_size, i, j, out_size, outbuf_size;
    FILE *f;
    short *samples;
    float t, tincr;
    uint8_t *outbuf;

    printf("Audio encoding\n");

    /* find the MP2 encoder */
    codec = avcodec_find_encoder(CODEC_ID_MP2);
    if (!codec) {
        fprintf(stderr, "codec not found\n");
        exit(1);
    }

    c= avcodec_alloc_context();

    /* put sample parameters */
    c->bit_rate = 64000;
    c->sample_rate = 44100;
    c->channels = 2;

    /* open it */
    if (avcodec_open(c, codec) < 0) {
        fprintf(stderr, "could not open codec\n");
        exit(1);
    }

    /* the codec gives us the frame size, in samples */
    frame_size = c->frame_size;
    samples = malloc(frame_size * 2 * c->channels);
    outbuf_size = 10000;
    outbuf = malloc(outbuf_size);

    f = fopen(filename, "wb");
    if (!f) {
        fprintf(stderr, "could not open %s\n", filename);
        exit(1);
    }

    /* encode a single tone sound */
    t = 0;
    tincr = 2 * M_PI * 440.0 / c->sample_rate;
    for(i=0;i<200;i++) {
        for(j=0;j<frame_size;j++) {
            samples[2*j] = (int)(sin(t) * 10000);
            samples[2*j+1] = samples[2*j];
            t += tincr;
        }
        /* encode the samples */
        out_size = avcodec_encode_audio(c, outbuf, outbuf_size, samples);
        fwrite(outbuf, 1, out_size, f);
    }
    fclose(f);
    free(outbuf);
    free(samples);

    avcodec_close(c);
    av_free(c);
}