early-access version 1432
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pineappleEA
238
externals/ffmpeg/doc/examples/encode_audio.c
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Executable file
238
externals/ffmpeg/doc/examples/encode_audio.c
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/*
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* Copyright (c) 2001 Fabrice Bellard
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*
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* Permission is hereby granted, free of charge, to any person obtaining a copy
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* of this software and associated documentation files (the "Software"), to deal
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* in the Software without restriction, including without limitation the rights
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* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
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* copies of the Software, and to permit persons to whom the Software is
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* furnished to do so, subject to the following conditions:
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*
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* The above copyright notice and this permission notice shall be included in
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* all copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
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* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
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* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
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* THE SOFTWARE.
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*/
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/**
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* @file
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* audio encoding with libavcodec API example.
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*
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* @example encode_audio.c
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*/
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#include <stdint.h>
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#include <stdio.h>
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#include <stdlib.h>
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#include <libavcodec/avcodec.h>
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#include <libavutil/channel_layout.h>
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#include <libavutil/common.h>
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#include <libavutil/frame.h>
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#include <libavutil/samplefmt.h>
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/* check that a given sample format is supported by the encoder */
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static int check_sample_fmt(const AVCodec *codec, enum AVSampleFormat sample_fmt)
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{
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const enum AVSampleFormat *p = codec->sample_fmts;
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while (*p != AV_SAMPLE_FMT_NONE) {
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if (*p == sample_fmt)
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return 1;
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p++;
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}
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return 0;
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}
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/* just pick the highest supported samplerate */
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static int select_sample_rate(const AVCodec *codec)
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{
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const int *p;
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int best_samplerate = 0;
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if (!codec->supported_samplerates)
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return 44100;
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p = codec->supported_samplerates;
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while (*p) {
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if (!best_samplerate || abs(44100 - *p) < abs(44100 - best_samplerate))
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best_samplerate = *p;
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p++;
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}
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return best_samplerate;
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}
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/* select layout with the highest channel count */
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static int select_channel_layout(const AVCodec *codec)
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{
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const uint64_t *p;
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uint64_t best_ch_layout = 0;
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int best_nb_channels = 0;
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if (!codec->channel_layouts)
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return AV_CH_LAYOUT_STEREO;
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p = codec->channel_layouts;
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while (*p) {
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int nb_channels = av_get_channel_layout_nb_channels(*p);
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if (nb_channels > best_nb_channels) {
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best_ch_layout = *p;
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best_nb_channels = nb_channels;
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}
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p++;
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}
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return best_ch_layout;
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}
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static void encode(AVCodecContext *ctx, AVFrame *frame, AVPacket *pkt,
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FILE *output)
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{
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int ret;
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/* send the frame for encoding */
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ret = avcodec_send_frame(ctx, frame);
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if (ret < 0) {
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fprintf(stderr, "Error sending the frame to the encoder\n");
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exit(1);
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}
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/* read all the available output packets (in general there may be any
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* number of them */
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while (ret >= 0) {
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ret = avcodec_receive_packet(ctx, pkt);
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if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
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return;
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else if (ret < 0) {
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fprintf(stderr, "Error encoding audio frame\n");
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exit(1);
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}
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fwrite(pkt->data, 1, pkt->size, output);
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av_packet_unref(pkt);
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}
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}
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int main(int argc, char **argv)
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{
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const char *filename;
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const AVCodec *codec;
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AVCodecContext *c= NULL;
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AVFrame *frame;
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AVPacket *pkt;
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int i, j, k, ret;
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FILE *f;
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uint16_t *samples;
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float t, tincr;
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if (argc <= 1) {
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fprintf(stderr, "Usage: %s <output file>\n", argv[0]);
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return 0;
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}
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filename = argv[1];
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/* find the MP2 encoder */
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codec = avcodec_find_encoder(AV_CODEC_ID_MP2);
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if (!codec) {
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fprintf(stderr, "Codec not found\n");
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exit(1);
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}
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c = avcodec_alloc_context3(codec);
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if (!c) {
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fprintf(stderr, "Could not allocate audio codec context\n");
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exit(1);
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}
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/* put sample parameters */
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c->bit_rate = 64000;
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/* check that the encoder supports s16 pcm input */
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c->sample_fmt = AV_SAMPLE_FMT_S16;
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if (!check_sample_fmt(codec, c->sample_fmt)) {
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fprintf(stderr, "Encoder does not support sample format %s",
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av_get_sample_fmt_name(c->sample_fmt));
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exit(1);
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}
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/* select other audio parameters supported by the encoder */
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c->sample_rate = select_sample_rate(codec);
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c->channel_layout = select_channel_layout(codec);
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c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
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/* open it */
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if (avcodec_open2(c, codec, NULL) < 0) {
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fprintf(stderr, "Could not open codec\n");
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exit(1);
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}
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f = fopen(filename, "wb");
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if (!f) {
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fprintf(stderr, "Could not open %s\n", filename);
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exit(1);
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}
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/* packet for holding encoded output */
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pkt = av_packet_alloc();
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if (!pkt) {
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fprintf(stderr, "could not allocate the packet\n");
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exit(1);
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}
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/* frame containing input raw audio */
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frame = av_frame_alloc();
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if (!frame) {
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fprintf(stderr, "Could not allocate audio frame\n");
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exit(1);
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}
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frame->nb_samples = c->frame_size;
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frame->format = c->sample_fmt;
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frame->channel_layout = c->channel_layout;
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/* allocate the data buffers */
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ret = av_frame_get_buffer(frame, 0);
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if (ret < 0) {
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fprintf(stderr, "Could not allocate audio data buffers\n");
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exit(1);
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}
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/* encode a single tone sound */
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t = 0;
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tincr = 2 * M_PI * 440.0 / c->sample_rate;
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for (i = 0; i < 200; i++) {
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/* make sure the frame is writable -- makes a copy if the encoder
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* kept a reference internally */
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ret = av_frame_make_writable(frame);
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if (ret < 0)
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exit(1);
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samples = (uint16_t*)frame->data[0];
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for (j = 0; j < c->frame_size; j++) {
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samples[2*j] = (int)(sin(t) * 10000);
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for (k = 1; k < c->channels; k++)
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samples[2*j + k] = samples[2*j];
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t += tincr;
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}
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encode(c, frame, pkt, f);
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}
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/* flush the encoder */
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encode(c, NULL, pkt, f);
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fclose(f);
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av_frame_free(&frame);
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av_packet_free(&pkt);
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avcodec_free_context(&c);
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return 0;
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}
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