FFmpeg/doc/examples/encode_audio.c

/*
 * Copyright (c) 2001 Fabrice Bellard
 *
 * Permission is hereby granted, free of charge, to any person obtaining a copy
 * of this software and associated documentation files (the "Software"), to deal
 * in the Software without restriction, including without limitation the rights
 * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
 * copies of the Software, and to permit persons to whom the Software is
 * furnished to do so, subject to the following conditions:
 *
 * The above copyright notice and this permission notice shall be included in
 * all copies or substantial portions of the Software.
 *
 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
 * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
 * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
 * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
 * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
 * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
 * THE SOFTWARE.
 */

/**
 * @file
 * audio encoding with libavcodec API example.
 *
 * @example encode_audio.c
 */

#include <stdint.h>
#include <stdio.h>
#include <stdlib.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) {
        best_samplerate = FFMAX(*p, best_samplerate);
        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;
}

int main(int argc, char **argv)
{
    const char *filename;
    const AVCodec *codec;
    AVCodecContext *c= NULL;
    AVFrame *frame;
    AVPacket pkt;
    int i, j, k, ret, got_output;
    int buffer_size;
    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];

    /* register all the codecs */
    avcodec_register_all();

    /* find the MP2 encoder */
    codec = avcodec_find_encoder(AV_CODEC_ID_MP2);
    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);
    }

    /* put sample parameters */
    c->bit_rate = 64000;

    /* check that the encoder supports s16 pcm input */
    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);
    }

    /* select other audio parameters supported by the encoder */
    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);

    /* open it */
    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);
    }

    /* frame containing input raw audio */
    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;

    /* the codec gives us the frame size, in samples,
     * we calculate the size of the samples buffer in bytes */
    buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size,
                                             c->sample_fmt, 0);
    if (buffer_size < 0) {
        fprintf(stderr, "Could not get sample buffer size\n");
        exit(1);
    }
    samples = av_malloc(buffer_size);
    if (!samples) {
        fprintf(stderr, "Could not allocate %d bytes for samples buffer\n",
                buffer_size);
        exit(1);
    }
    /* setup the data pointers in the AVFrame */
    ret = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
                                   (const uint8_t*)samples, buffer_size, 0);
    if (ret < 0) {
        fprintf(stderr, "Could not setup audio frame\n");
        exit(1);
    }

    /* encode a single tone sound */
    t = 0;
    tincr = 2 * M_PI * 440.0 / c->sample_rate;
    for (i = 0; i < 200; i++) {
        av_init_packet(&pkt);
        pkt.data = NULL; // packet data will be allocated by the encoder
        pkt.size = 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 the samples */
        ret = avcodec_encode_audio2(c, &pkt, frame, &got_output);
        if (ret < 0) {
            fprintf(stderr, "Error encoding audio frame\n");
            exit(1);
        }
        if (got_output) {
            fwrite(pkt.data, 1, pkt.size, f);
            av_packet_unref(&pkt);
        }
    }

    /* get the delayed frames */
    for (got_output = 1; got_output; i++) {
        ret = avcodec_encode_audio2(c, &pkt, NULL, &got_output);
        if (ret < 0) {
            fprintf(stderr, "Error encoding frame\n");
            exit(1);
        }

        if (got_output) {
            fwrite(pkt.data, 1, pkt.size, f);
            av_packet_unref(&pkt);
        }
    }
    fclose(f);

    av_freep(&samples);
    av_frame_free(&frame);
    avcodec_free_context(&c);
}