使用 avcodec_decode_audio4() 解码 AAC 时出错
Error when decode AAC with avcodec_decode_audio4()
我正在尝试使用 FFmpeg 原生解码器解码 AAC，但遇到错误
SSR is not implemeted. Update your FFmpeg version to newest from Git. If the problem still occurs, it mean that your file has a feature which has not implemented.
函数avcodec_decode_audio4()return-1163346256。这是因为 FFmpeg 版本吗？我从 here 下载了共享和开发版本。这是最新的吗？
这里是源代码：
#include "stdafx.h"
#include "stdio.h"
#include "conio.h"
extern "C"
{
#ifndef __STDC_CONSTANT_MACROS
#define __STDC_CONSTANT_MACROS
#endif
#include <libavcodec\avcodec.h>
#include <libavformat/avformat.h>
}
// compatibility with newer API
#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(55,28,1)
#define av_frame_alloc avcodec_alloc_frame
#define av_frame_free avcodec_free_frame
#endif
#define AUDIO_INBUF_SIZE 20480
#define AUDIO_REFILL_THRESH 4096
static void audio_decode_example(const char *outfilename, const char *filename);
int main(int argc, char *argv[]) {
audio_decode_example("D:\sample.pcm","D:\sample.m4a");
getch();
return 0;
}
/*
* Audio decoding.
*/
static void audio_decode_example(const char *outfilename, const char *filename)
{
AVCodec *codec;
AVFormatContext *pFormatCtx = NULL;
AVCodecContext *pCodecCtxOrig = NULL;
AVCodecContext * pCodecCtx= NULL;
int len;
FILE *f, *outfile;
uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
AVPacket avpkt;
AVFrame *decoded_frame = NULL;
av_register_all();
av_init_packet(&avpkt);
printf("Decode audio file %s to %s\n", filename, outfilename);
// Open file to get format context
if(avformat_open_input(&pFormatCtx, filename, NULL, NULL)!=0){
printf("Couldn't open file");
return; // Couldn't open file
}
// Retrieve stream information
if(avformat_find_stream_info(pFormatCtx, NULL)<0){
printf("Couldn't find stream information");
return; // Couldn't find stream information
}
// Dump information about file onto standard error
av_dump_format(pFormatCtx, 0, filename, 0);
// Find the first audio stream
int audioStream = -1;
int i =0;
for(i=0; i<pFormatCtx->nb_streams; i++) {
if(pFormatCtx->streams[i]->codec->codec_type==AVMEDIA_TYPE_AUDIO) {
audioStream=i;
break;
}
}
if(audioStream==-1) {
printf("Didn't find a audio stream");
return; // Didn't find a audio stream
}
// Get a pointer to the codec context for the audio stream
pCodecCtxOrig=pFormatCtx->streams[audioStream]->codec;
// Find the decoder for the audio stream
codec=avcodec_find_decoder(pCodecCtxOrig->codec_id);
if(codec==NULL) {
fprintf(stderr, "Codec not found\n");
return; // Codec not found
}
pCodecCtx = avcodec_alloc_context3(codec);
if (!pCodecCtx) {
fprintf(stderr, "Could not allocate audio codec context\n");
return;
}
if(avcodec_copy_context(pCodecCtx, pCodecCtxOrig) != 0) {
fprintf(stderr, "Couldn't copy codec context");
return; // Error copying codec context
}
/* open it */
if (avcodec_open2(pCodecCtx, codec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
return;
}
f = fopen(filename, "rb");
if (!f) {
fprintf(stderr, "Could not open %s\n", filename);
return;
}
outfile = fopen(outfilename, "wb");
if (!outfile) {
av_free(pCodecCtx);
return;
}
/* decode until eof */
avpkt.data = inbuf;
avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
while (avpkt.size > 0) {
int i, ch;
int got_frame = 0;
if (!decoded_frame) {
if (!(decoded_frame = av_frame_alloc())) {
fprintf(stderr, "Could not allocate audio frame\n");
return;
}
}
len = avcodec_decode_audio4(pCodecCtx, decoded_frame, &got_frame, &avpkt);
if (len < 0) {
fprintf(stderr, "Error while decoding. len = %d \n",len);
return;
}
if (got_frame) {
/* if a frame has been decoded, output it */
int data_size = av_get_bytes_per_sample(pCodecCtx->sample_fmt);
if (data_size < 0) {
/* This should not occur, checking just for paranoia */
fprintf(stderr, "Failed to calculate data size\n");
return;
}
for (i=0; i < decoded_frame->nb_samples; i++)
for (ch=0; ch < pCodecCtx->channels; ch++)
fwrite(decoded_frame->data[ch] + data_size*i, 1, data_size, outfile);
}
avpkt.size -= len;
avpkt.data += len;
avpkt.dts =
avpkt.pts = AV_NOPTS_VALUE;
if (avpkt.size < AUDIO_REFILL_THRESH) {
/* Refill the input buffer, to avoid trying to decode
* incomplete frames. Instead of this, one could also use
* a parser, or use a proper container format through
* libavformat. */
memmove(inbuf, avpkt.data, avpkt.size);
avpkt.data = inbuf;
len = fread(avpkt.data + avpkt.size, 1,
AUDIO_INBUF_SIZE - avpkt.size, f);
if (len > 0)
avpkt.size += len;
}
}
fclose(outfile);
fclose(f);
avcodec_close(pCodecCtx);
av_free(pCodecCtx);
av_frame_free(&decoded_frame);
}
我也看过这个问题：How to decode AAC using avcodec_decode_audio4?但是没有提供解决方案。
f = fopen(filename, "rb");
if (!f) {
fprintf(stderr, "Could not open %s\n", filename);
return;
}
outfile = fopen(outfilename, "wb");
if (!outfile) {
av_free(pCodecCtx);
return;
}
/* decode until eof */
avpkt.data = inbuf;
avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
while (avpkt.size > 0) {
int i, ch;
int got_frame = 0;
是的，那是行不通的。您不能将一些随机多路复用格式（可能是 mp4）的原始字节转储到解码器中并期望它能够工作。使用 av_read_frame() to read individual audio packets from the muxing format, and feed the resulting AVPacket into the decoder using avcodec_decode_audio4(). See e.g. the dranger api tutorial. I know api-example.c uses the above code, but that unfortunately only works for a very limited subset of cases. Also see the detailed description in the API docs.
我正在尝试使用 FFmpeg 原生解码器解码 AAC，但遇到错误
SSR is not implemeted. Update your FFmpeg version to newest from Git. If the problem still occurs, it mean that your file has a feature which has not implemented.
函数avcodec_decode_audio4()return-1163346256。这是因为 FFmpeg 版本吗？我从 here 下载了共享和开发版本。这是最新的吗？
这里是源代码：
#include "stdafx.h"
#include "stdio.h"
#include "conio.h"
extern "C"
{
#ifndef __STDC_CONSTANT_MACROS
#define __STDC_CONSTANT_MACROS
#endif
#include <libavcodec\avcodec.h>
#include <libavformat/avformat.h>
}
// compatibility with newer API
#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(55,28,1)
#define av_frame_alloc avcodec_alloc_frame
#define av_frame_free avcodec_free_frame
#endif
#define AUDIO_INBUF_SIZE 20480
#define AUDIO_REFILL_THRESH 4096
static void audio_decode_example(const char *outfilename, const char *filename);
int main(int argc, char *argv[]) {
audio_decode_example("D:\sample.pcm","D:\sample.m4a");
getch();
return 0;
}
/*
* Audio decoding.
*/
static void audio_decode_example(const char *outfilename, const char *filename)
{
AVCodec *codec;
AVFormatContext *pFormatCtx = NULL;
AVCodecContext *pCodecCtxOrig = NULL;
AVCodecContext * pCodecCtx= NULL;
int len;
FILE *f, *outfile;
uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
AVPacket avpkt;
AVFrame *decoded_frame = NULL;
av_register_all();
av_init_packet(&avpkt);
printf("Decode audio file %s to %s\n", filename, outfilename);
// Open file to get format context
if(avformat_open_input(&pFormatCtx, filename, NULL, NULL)!=0){
printf("Couldn't open file");
return; // Couldn't open file
}
// Retrieve stream information
if(avformat_find_stream_info(pFormatCtx, NULL)<0){
printf("Couldn't find stream information");
return; // Couldn't find stream information
}
// Dump information about file onto standard error
av_dump_format(pFormatCtx, 0, filename, 0);
// Find the first audio stream
int audioStream = -1;
int i =0;
for(i=0; i<pFormatCtx->nb_streams; i++) {
if(pFormatCtx->streams[i]->codec->codec_type==AVMEDIA_TYPE_AUDIO) {
audioStream=i;
break;
}
}
if(audioStream==-1) {
printf("Didn't find a audio stream");
return; // Didn't find a audio stream
}
// Get a pointer to the codec context for the audio stream
pCodecCtxOrig=pFormatCtx->streams[audioStream]->codec;
// Find the decoder for the audio stream
codec=avcodec_find_decoder(pCodecCtxOrig->codec_id);
if(codec==NULL) {
fprintf(stderr, "Codec not found\n");
return; // Codec not found
}
pCodecCtx = avcodec_alloc_context3(codec);
if (!pCodecCtx) {
fprintf(stderr, "Could not allocate audio codec context\n");
return;
}
if(avcodec_copy_context(pCodecCtx, pCodecCtxOrig) != 0) {
fprintf(stderr, "Couldn't copy codec context");
return; // Error copying codec context
}
/* open it */
if (avcodec_open2(pCodecCtx, codec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
return;
}
f = fopen(filename, "rb");
if (!f) {
fprintf(stderr, "Could not open %s\n", filename);
return;
}
outfile = fopen(outfilename, "wb");
if (!outfile) {
av_free(pCodecCtx);
return;
}
/* decode until eof */
avpkt.data = inbuf;
avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
while (avpkt.size > 0) {
int i, ch;
int got_frame = 0;
if (!decoded_frame) {
if (!(decoded_frame = av_frame_alloc())) {
fprintf(stderr, "Could not allocate audio frame\n");
return;
}
}
len = avcodec_decode_audio4(pCodecCtx, decoded_frame, &got_frame, &avpkt);
if (len < 0) {
fprintf(stderr, "Error while decoding. len = %d \n",len);
return;
}
if (got_frame) {
/* if a frame has been decoded, output it */
int data_size = av_get_bytes_per_sample(pCodecCtx->sample_fmt);
if (data_size < 0) {
/* This should not occur, checking just for paranoia */
fprintf(stderr, "Failed to calculate data size\n");
return;
}
for (i=0; i < decoded_frame->nb_samples; i++)
for (ch=0; ch < pCodecCtx->channels; ch++)
fwrite(decoded_frame->data[ch] + data_size*i, 1, data_size, outfile);
}
avpkt.size -= len;
avpkt.data += len;
avpkt.dts =
avpkt.pts = AV_NOPTS_VALUE;
if (avpkt.size < AUDIO_REFILL_THRESH) {
/* Refill the input buffer, to avoid trying to decode
* incomplete frames. Instead of this, one could also use
* a parser, or use a proper container format through
* libavformat. */
memmove(inbuf, avpkt.data, avpkt.size);
avpkt.data = inbuf;
len = fread(avpkt.data + avpkt.size, 1,
AUDIO_INBUF_SIZE - avpkt.size, f);
if (len > 0)
avpkt.size += len;
}
}
fclose(outfile);
fclose(f);
avcodec_close(pCodecCtx);
av_free(pCodecCtx);
av_frame_free(&decoded_frame);
}
我也看过这个问题：How to decode AAC using avcodec_decode_audio4?但是没有提供解决方案。
f = fopen(filename, "rb"); if (!f) { fprintf(stderr, "Could not open %s\n", filename); return; } outfile = fopen(outfilename, "wb"); if (!outfile) { av_free(pCodecCtx); return; } /* decode until eof */ avpkt.data = inbuf; avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f); while (avpkt.size > 0) { int i, ch; int got_frame = 0;
是的，那是行不通的。您不能将一些随机多路复用格式（可能是 mp4）的原始字节转储到解码器中并期望它能够工作。使用 av_read_frame() to read individual audio packets from the muxing format, and feed the resulting AVPacket into the decoder using avcodec_decode_audio4(). See e.g. the dranger api tutorial. I know api-example.c uses the above code, but that unfortunately only works for a very limited subset of cases. Also see the detailed description in the API docs.