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pineappleEA
236
externals/ffmpeg/doc/examples/decode_audio.c
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Executable file
236
externals/ffmpeg/doc/examples/decode_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 decoding with libavcodec API example
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*
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* @example decode_audio.c
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*/
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#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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#include <libavutil/frame.h>
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#include <libavutil/mem.h>
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#include <libavcodec/avcodec.h>
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#define AUDIO_INBUF_SIZE 20480
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#define AUDIO_REFILL_THRESH 4096
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static int get_format_from_sample_fmt(const char **fmt,
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enum AVSampleFormat sample_fmt)
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{
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int i;
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struct sample_fmt_entry {
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enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
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} sample_fmt_entries[] = {
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{ AV_SAMPLE_FMT_U8, "u8", "u8" },
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{ AV_SAMPLE_FMT_S16, "s16be", "s16le" },
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{ AV_SAMPLE_FMT_S32, "s32be", "s32le" },
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{ AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
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{ AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
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};
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*fmt = NULL;
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for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
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struct sample_fmt_entry *entry = &sample_fmt_entries[i];
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if (sample_fmt == entry->sample_fmt) {
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*fmt = AV_NE(entry->fmt_be, entry->fmt_le);
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return 0;
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}
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}
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fprintf(stderr,
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"sample format %s is not supported as output format\n",
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av_get_sample_fmt_name(sample_fmt));
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return -1;
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}
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static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame,
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FILE *outfile)
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{
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int i, ch;
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int ret, data_size;
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/* send the packet with the compressed data to the decoder */
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ret = avcodec_send_packet(dec_ctx, pkt);
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if (ret < 0) {
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fprintf(stderr, "Error submitting the packet to the decoder\n");
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exit(1);
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}
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/* read all the output frames (in general there may be any number of them */
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while (ret >= 0) {
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ret = avcodec_receive_frame(dec_ctx, frame);
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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 during decoding\n");
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exit(1);
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}
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data_size = av_get_bytes_per_sample(dec_ctx->sample_fmt);
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if (data_size < 0) {
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/* This should not occur, checking just for paranoia */
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fprintf(stderr, "Failed to calculate data size\n");
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exit(1);
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}
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for (i = 0; i < frame->nb_samples; i++)
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for (ch = 0; ch < dec_ctx->channels; ch++)
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fwrite(frame->data[ch] + data_size*i, 1, data_size, outfile);
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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 *outfilename, *filename;
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const AVCodec *codec;
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AVCodecContext *c= NULL;
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AVCodecParserContext *parser = NULL;
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int len, ret;
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FILE *f, *outfile;
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uint8_t inbuf[AUDIO_INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
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uint8_t *data;
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size_t data_size;
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AVPacket *pkt;
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AVFrame *decoded_frame = NULL;
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enum AVSampleFormat sfmt;
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int n_channels = 0;
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const char *fmt;
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if (argc <= 2) {
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fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
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exit(0);
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}
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filename = argv[1];
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outfilename = argv[2];
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pkt = av_packet_alloc();
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/* find the MPEG audio decoder */
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codec = avcodec_find_decoder(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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parser = av_parser_init(codec->id);
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if (!parser) {
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fprintf(stderr, "Parser 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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/* 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, "rb");
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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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outfile = fopen(outfilename, "wb");
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if (!outfile) {
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av_free(c);
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exit(1);
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}
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/* decode until eof */
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data = inbuf;
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data_size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
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while (data_size > 0) {
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if (!decoded_frame) {
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if (!(decoded_frame = av_frame_alloc())) {
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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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}
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ret = av_parser_parse2(parser, c, &pkt->data, &pkt->size,
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data, data_size,
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AV_NOPTS_VALUE, AV_NOPTS_VALUE, 0);
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if (ret < 0) {
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fprintf(stderr, "Error while parsing\n");
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exit(1);
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}
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data += ret;
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data_size -= ret;
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if (pkt->size)
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decode(c, pkt, decoded_frame, outfile);
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if (data_size < AUDIO_REFILL_THRESH) {
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memmove(inbuf, data, data_size);
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data = inbuf;
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len = fread(data + data_size, 1,
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AUDIO_INBUF_SIZE - data_size, f);
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if (len > 0)
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data_size += len;
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}
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}
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/* flush the decoder */
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pkt->data = NULL;
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pkt->size = 0;
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decode(c, pkt, decoded_frame, outfile);
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/* print output pcm infomations, because there have no metadata of pcm */
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sfmt = c->sample_fmt;
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if (av_sample_fmt_is_planar(sfmt)) {
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const char *packed = av_get_sample_fmt_name(sfmt);
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printf("Warning: the sample format the decoder produced is planar "
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"(%s). This example will output the first channel only.\n",
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packed ? packed : "?");
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sfmt = av_get_packed_sample_fmt(sfmt);
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}
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n_channels = c->channels;
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if ((ret = get_format_from_sample_fmt(&fmt, sfmt)) < 0)
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goto end;
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printf("Play the output audio file with the command:\n"
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"ffplay -f %s -ac %d -ar %d %s\n",
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fmt, n_channels, c->sample_rate,
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outfilename);
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end:
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fclose(outfile);
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fclose(f);
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avcodec_free_context(&c);
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av_parser_close(parser);
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av_frame_free(&decoded_frame);
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av_packet_free(&pkt);
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return 0;
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}
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