/*
 * 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 libavcodec encoding audio API usage examples
 * @example encode_audio.c
 *
 * Generate a synthetic audio signal and encode it to an output MP2 file.
 */

#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) {
 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, AVChannelLayout *dst)
{
 const AVChannelLayout *p, *best_ch_layout;
 int best_nb_channels = 0;

 if (!codec->ch_layouts)
 return av_channel_layout_copy(dst, &(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO);

 p = codec->ch_layouts;
 while (p->nb_channels) {
 int nb_channels = p->nb_channels;

 if (nb_channels > best_nb_channels) {
 best_ch_layout = p;
 best_nb_channels = nb_channels;
 }
 p++;
 }
 return av_channel_layout_copy(dst, 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];

 /* 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);
 ret = select_channel_layout(codec, &c->ch_layout);
 if (ret < 0)
 exit(1);

 /* 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);
 }

 /* packet for holding encoded output */
 pkt = av_packet_alloc();
 if (!pkt) {
 fprintf(stderr, "could not allocate the packet\n");
 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;
 ret = av_channel_layout_copy(&frame->ch_layout, &c->ch_layout);
 if (ret < 0)
 exit(1);

 /* allocate the data buffers */
 ret = av_frame_get_buffer(frame, 0);
 if (ret < 0) {
 fprintf(stderr, "Could not allocate audio data buffers\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++) {
 /* make sure the frame is writable -- makes a copy if the encoder
 * kept a reference internally */
 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->ch_layout.nb_channels; k++)
 samples[2*j + k] = samples[2*j];
 t += tincr;
 }
 encode(c, frame, pkt, f);
 }

 /* flush the encoder */
 encode(c, NULL, pkt, f);

 fclose(f);

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

 return 0;
}