237 lines
7.0 KiB
C
237 lines
7.0 KiB
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 decoding with libavcodec API example
|
|
*
|
|
* @example decode_audio.c
|
|
*/
|
|
|
|
#include <stdio.h>
|
|
#include <stdlib.h>
|
|
#include <string.h>
|
|
|
|
#include <libavutil/frame.h>
|
|
#include <libavutil/mem.h>
|
|
|
|
#include <libavcodec/avcodec.h>
|
|
|
|
#define AUDIO_INBUF_SIZE 20480
|
|
#define AUDIO_REFILL_THRESH 4096
|
|
|
|
static int get_format_from_sample_fmt(const char **fmt,
|
|
enum AVSampleFormat sample_fmt)
|
|
{
|
|
int i;
|
|
struct sample_fmt_entry {
|
|
enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
|
|
} sample_fmt_entries[] = {
|
|
{ AV_SAMPLE_FMT_U8, "u8", "u8" },
|
|
{ AV_SAMPLE_FMT_S16, "s16be", "s16le" },
|
|
{ AV_SAMPLE_FMT_S32, "s32be", "s32le" },
|
|
{ AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
|
|
{ AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
|
|
};
|
|
*fmt = NULL;
|
|
|
|
for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
|
|
struct sample_fmt_entry *entry = &sample_fmt_entries[i];
|
|
if (sample_fmt == entry->sample_fmt) {
|
|
*fmt = AV_NE(entry->fmt_be, entry->fmt_le);
|
|
return 0;
|
|
}
|
|
}
|
|
|
|
fprintf(stderr,
|
|
"sample format %s is not supported as output format\n",
|
|
av_get_sample_fmt_name(sample_fmt));
|
|
return -1;
|
|
}
|
|
|
|
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame,
|
|
FILE *outfile)
|
|
{
|
|
int i, ch;
|
|
int ret, data_size;
|
|
|
|
/* send the packet with the compressed data to the decoder */
|
|
ret = avcodec_send_packet(dec_ctx, pkt);
|
|
if (ret < 0) {
|
|
fprintf(stderr, "Error submitting the packet to the decoder\n");
|
|
exit(1);
|
|
}
|
|
|
|
/* read all the output frames (in general there may be any number of them */
|
|
while (ret >= 0) {
|
|
ret = avcodec_receive_frame(dec_ctx, frame);
|
|
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
|
|
return;
|
|
else if (ret < 0) {
|
|
fprintf(stderr, "Error during decoding\n");
|
|
exit(1);
|
|
}
|
|
data_size = av_get_bytes_per_sample(dec_ctx->sample_fmt);
|
|
if (data_size < 0) {
|
|
/* This should not occur, checking just for paranoia */
|
|
fprintf(stderr, "Failed to calculate data size\n");
|
|
exit(1);
|
|
}
|
|
for (i = 0; i < frame->nb_samples; i++)
|
|
for (ch = 0; ch < dec_ctx->channels; ch++)
|
|
fwrite(frame->data[ch] + data_size*i, 1, data_size, outfile);
|
|
}
|
|
}
|
|
|
|
int main(int argc, char **argv)
|
|
{
|
|
const char *outfilename, *filename;
|
|
const AVCodec *codec;
|
|
AVCodecContext *c= NULL;
|
|
AVCodecParserContext *parser = NULL;
|
|
int len, ret;
|
|
FILE *f, *outfile;
|
|
uint8_t inbuf[AUDIO_INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
|
|
uint8_t *data;
|
|
size_t data_size;
|
|
AVPacket *pkt;
|
|
AVFrame *decoded_frame = NULL;
|
|
enum AVSampleFormat sfmt;
|
|
int n_channels = 0;
|
|
const char *fmt;
|
|
|
|
if (argc <= 2) {
|
|
fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
|
|
exit(0);
|
|
}
|
|
filename = argv[1];
|
|
outfilename = argv[2];
|
|
|
|
pkt = av_packet_alloc();
|
|
|
|
/* find the MPEG audio decoder */
|
|
codec = avcodec_find_decoder(AV_CODEC_ID_MP2);
|
|
if (!codec) {
|
|
fprintf(stderr, "Codec not found\n");
|
|
exit(1);
|
|
}
|
|
|
|
parser = av_parser_init(codec->id);
|
|
if (!parser) {
|
|
fprintf(stderr, "Parser not found\n");
|
|
exit(1);
|
|
}
|
|
|
|
c = avcodec_alloc_context3(codec);
|
|
if (!c) {
|
|
fprintf(stderr, "Could not allocate audio codec context\n");
|
|
exit(1);
|
|
}
|
|
|
|
/* open it */
|
|
if (avcodec_open2(c, codec, NULL) < 0) {
|
|
fprintf(stderr, "Could not open codec\n");
|
|
exit(1);
|
|
}
|
|
|
|
f = fopen(filename, "rb");
|
|
if (!f) {
|
|
fprintf(stderr, "Could not open %s\n", filename);
|
|
exit(1);
|
|
}
|
|
outfile = fopen(outfilename, "wb");
|
|
if (!outfile) {
|
|
av_free(c);
|
|
exit(1);
|
|
}
|
|
|
|
/* decode until eof */
|
|
data = inbuf;
|
|
data_size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
|
|
|
|
while (data_size > 0) {
|
|
if (!decoded_frame) {
|
|
if (!(decoded_frame = av_frame_alloc())) {
|
|
fprintf(stderr, "Could not allocate audio frame\n");
|
|
exit(1);
|
|
}
|
|
}
|
|
|
|
ret = av_parser_parse2(parser, c, &pkt->data, &pkt->size,
|
|
data, data_size,
|
|
AV_NOPTS_VALUE, AV_NOPTS_VALUE, 0);
|
|
if (ret < 0) {
|
|
fprintf(stderr, "Error while parsing\n");
|
|
exit(1);
|
|
}
|
|
data += ret;
|
|
data_size -= ret;
|
|
|
|
if (pkt->size)
|
|
decode(c, pkt, decoded_frame, outfile);
|
|
|
|
if (data_size < AUDIO_REFILL_THRESH) {
|
|
memmove(inbuf, data, data_size);
|
|
data = inbuf;
|
|
len = fread(data + data_size, 1,
|
|
AUDIO_INBUF_SIZE - data_size, f);
|
|
if (len > 0)
|
|
data_size += len;
|
|
}
|
|
}
|
|
|
|
/* flush the decoder */
|
|
pkt->data = NULL;
|
|
pkt->size = 0;
|
|
decode(c, pkt, decoded_frame, outfile);
|
|
|
|
/* print output pcm infomations, because there have no metadata of pcm */
|
|
sfmt = c->sample_fmt;
|
|
|
|
if (av_sample_fmt_is_planar(sfmt)) {
|
|
const char *packed = av_get_sample_fmt_name(sfmt);
|
|
printf("Warning: the sample format the decoder produced is planar "
|
|
"(%s). This example will output the first channel only.\n",
|
|
packed ? packed : "?");
|
|
sfmt = av_get_packed_sample_fmt(sfmt);
|
|
}
|
|
|
|
n_channels = c->channels;
|
|
if ((ret = get_format_from_sample_fmt(&fmt, sfmt)) < 0)
|
|
goto end;
|
|
|
|
printf("Play the output audio file with the command:\n"
|
|
"ffplay -f %s -ac %d -ar %d %s\n",
|
|
fmt, n_channels, c->sample_rate,
|
|
outfilename);
|
|
end:
|
|
fclose(outfile);
|
|
fclose(f);
|
|
|
|
avcodec_free_context(&c);
|
|
av_parser_close(parser);
|
|
av_frame_free(&decoded_frame);
|
|
av_packet_free(&pkt);
|
|
|
|
return 0;
|
|
}
|