FFMPEG - Encode a .wav file to G711

Question

My question is the following: how to encode a .wav file to G.711/PCM A-law?

I tried to edit this example file, but I got an EINVAL error when tried to call av_frame_get_buffer(frame, 0) with frame->nb_samples = c->frame_size which is 0 (I think this is the problem) because of pcm.c:37 (at least I think because of it).

So with which parameters allocate an audio data buffer by myself if av_frame_get_buffer() doesn't do this for me and how to use it?

Thanks for reply!

EDIT:

My code:

#include <stdio.h>
#include <stdlib.h>
#include <stdint.h>

#include <libavcodec/avcodec.h>
#include <libavutil/channel_layout.h>
#include <libavutil/common.h>
#include <libavutil/frame.h>
#include <libavutil/samplefmt.h>

/* check that a given sample format is supported by the encoder */
static int check_sample_fmt(const AVCodec *codec, enum AVSampleFormat sample_fmt)
{
    const enum AVSampleFormat *p = codec->sample_fmts;

    while (*p != AV_SAMPLE_FMT_NONE) {
        if (*p == sample_fmt)
            return 1;
        p++;
    }
    return 0;
}

/* just pick the highest supported samplerate */
static int select_sample_rate(const AVCodec *codec)
{
    const int *p;
    int best_samplerate = 0;

    if (!codec->supported_samplerates)
        return 44100;

    p = codec->supported_samplerates;
    while (*p) {
        if (!best_samplerate || abs(44100 - *p) < abs(44100 - best_samplerate))
            best_samplerate = *p;
        p++;
    }
    return best_samplerate;
}

/* select layout with the highest channel count */
static int select_channel_layout(const AVCodec *codec)
{
    const uint64_t *p;
    uint64_t best_ch_layout = 0;
    int best_nb_channels   = 0;

    if (!codec->channel_layouts)
        return AV_CH_LAYOUT_STEREO;

    p = codec->channel_layouts;
    while (*p) {
        int nb_channels = av_get_channel_layout_nb_channels(*p);

        if (nb_channels > best_nb_channels) {
            best_ch_layout    = *p;
            best_nb_channels = nb_channels;
        }
        p++;
    }
    return best_ch_layout;
}

static void encode(AVCodecContext *ctx, AVFrame *frame, AVPacket *pkt,
                   FILE *output)
{
    int ret;

    /* send the frame for encoding */
    ret = avcodec_send_frame(ctx, frame);
    if (ret < 0) {
        fprintf(stderr, "Error sending the frame to the encoder\n");
        exit(1);
    }

    /* read all the available output packets (in general there may be any
     * number of them */
    while (ret >= 0) {
        ret = avcodec_receive_packet(ctx, pkt);
        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
            return;
        else if (ret < 0) {
            fprintf(stderr, "Error encoding audio frame\n");
            exit(1);
        }

        fwrite(pkt->data, 1, pkt->size, output);
        av_packet_unref(pkt);
    }
}



int main(int argc, char **argv)
{

    const char *filename;
    const AVCodec *codec;
    AVCodecContext *c = NULL;
    AVFrame *frame;
    AVPacket *pkt;
    int i, j, k, ret;
    FILE *f;
    uint16_t *samples;
    float t, tincr;

    if (argc <= 1)
    {
      fprintf(stderr, "Usage: %s <output file>\n", argv[0]);
      return 0;
    }

    filename = argv[1];

    codec = avcodec_find_encoder(AV_CODEC_ID_PCM_ALAW);
    if (!codec)
    {
      fprintf(stderr, "Codec not found\n");
      exit(1);
    }

    c = avcodec_alloc_context3(codec);
    if (!c)
    {
      fprintf(stderr, "Could not allocate audio codec context\n");
      exit(1);
    }

    c->bit_rate = 128000;

    c->sample_fmt = AV_SAMPLE_FMT_S16;
    if (!check_sample_fmt(codec, c->sample_fmt))
    {
      fprintf(stderr, "Encoder does not support sample format %s",
              av_get_sample_fmt_name(c->sample_fmt));
      exit(1);
    }

    c->sample_rate = select_sample_rate(codec);
    c->channel_layout = select_channel_layout(codec);
    c->channels = av_get_channel_layout_nb_channels(c->channel_layout);

    if (avcodec_open2(c, codec, NULL) < 0)
    {
      fprintf(stderr, "Could not open codec\n");
      exit(1);
    }

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

    pkt = av_packet_alloc();
    if (!pkt)
    {
      fprintf(stderr, "Could not allocate the packet\n");
      exit(1);
    }

    frame = av_frame_alloc();
    if (!frame)
    {
      fprintf(stderr, "Could not allocate audio frame\n");
      exit(1);
    }

    frame->nb_samples = c->frame_size;
    frame->format = c->sample_fmt;
    frame->channel_layout = c->channel_layout;

// ERROR Here
    ret = av_frame_get_buffer(frame, 0);
    if (ret < 0)
    {
      fprintf(stderr, "Could not allocate audio data buffers\n");
      exit(1);
    }

    t = 0;
    tincr = 2 * M_PI * 440.0 / c->sample_rate;
    for (i = 0; i < 200; i++)
    {
      ret = av_frame_make_writable(frame);
      if (ret < 0)
      {
        exit(1);
      }

      samples = (uint16_t*)frame->data[0];

      for (j = 0; j < c->frame_size; j++)
      {
        samples[2*j] = (int)(sin(t)*10000);

        for (k = 1; k < c->channels; k++)
        {
          samples[2*j + k] = samples[2*j];
        }

        t += tincr;
      }
      encode(c, frame, pkt, f);
    }

    encode(c, NULL, pkt, f);

    fclose(f);

    av_frame_free(&frame);
    av_packet_free(&pkt);
    avcodec_free_context(&c);

    return 0;
}