/* * 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 audio decoding API usage example * @example decode_audio.c * * Decode data from an MP2 input file and generate a raw audio file to * be played with ffplay. */ #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->ch_layout.nb_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->ch_layout.nb_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;}
/*
* 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 audio decoding API usage example
* @example decode_audio.c
*
* Decode data from an MP2 input file and generate a raw audio file to
* be played with ffplay.
*/
#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->ch_layout.nb_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->ch_layout.nb_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;
}