[原]最简单的基于FFMPEG+SDL的音频播放器：拆分-解码器和播放器
leixiaohua1020发表于 2015-07-17 09:31:10
本文补充记录《最简单的基于FFMPEG+SDL的音频播放器》中的两个例子：FFmpeg音频解码器和SDL音频采样数据播放器。这两个部分是从音频播放器中拆分出来的两个例子。FFmpeg音频解码器实现了视频数据到PCM采样数据的解码，而SDL音频采样数据播放器实现了PCM数据到音频设备的播放。简而言之，原先的FFmpeg+SDL音频播放器实现了：
音频数据->PCM->音频设备
FFmpeg音频解码器实现了：
音频数据->PCM
SDL音频采样数据播放器实现了：
PCM->音频设备
FFmpeg音频解码器
源代码
/**
* 最简单的基于FFmpeg的音频解码器
* Simplest FFmpeg Audio Decoder
*
* 雷霄骅 Lei Xiaohua
* leixiaohua1020@126.com
* 中国传媒大学/数字电视技术
* Communication University of China / Digital TV Technology
* http://blog.csdn.net/leixiaohua1020
*
* 本程序可以将音频码流（mp3，AAC等）解码为PCM采样数据。
* 是最简单的FFmpeg音频解码方面的教程。
* 通过学习本例子可以了解FFmpeg的解码流程。
*
* This software decode audio streams (AAC,MP3 ...) to PCM data.
* Suitable for beginner of FFmpeg.
*
*/
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#define __STDC_CONSTANT_MACROS
#ifdef _WIN32
//Windows
extern "C"
{
#include "libavcodec/avcodec.h"
#include "libavformat/avformat.h"
#include "libswresample/swresample.h"
};
#else
//Linux...
#ifdef __cplusplus
extern "C"
{
#endif
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libswresample/swresample.h>
#ifdef __cplusplus
};
#endif
#endif
#define MAX_AUDIO_FRAME_SIZE 192000 // 1 second of 48khz 32bit audio
int main(int argc, char* argv[])
{
AVFormatContext *pFormatCtx;
int i, audioStream;
AVCodecContext *pCodecCtx;
AVCodec *pCodec;
AVPacket *packet;
uint8_t *out_buffer;
AVFrame *pFrame;
int ret;
uint32_t len = 0;
int got_picture;
int index = 0;
int64_t in_channel_layout;
struct SwrContext *au_convert_ctx;
FILE *pFile=fopen("output.pcm", "wb");
char url[]="skycity1.mp3";
av_register_all();
avformat_network_init();
pFormatCtx = avformat_alloc_context();
//Open
if(avformat_open_input(&pFormatCtx,url,NULL,NULL)!=0){
printf("Couldn't open input stream.\n");
return -1;
}
// Retrieve stream information
if(avformat_find_stream_info(pFormatCtx,NULL)<0){
printf("Couldn't find stream information.\n");
return -1;
}
// Dump valid information onto standard error
av_dump_format(pFormatCtx, 0, url, false);
// Find the first audio stream
audioStream=-1;
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.\n");
return -1;
}
// Get a pointer to the codec context for the audio stream
pCodecCtx=pFormatCtx->streams[audioStream]->codec;
// Find the decoder for the audio stream
pCodec=avcodec_find_decoder(pCodecCtx->codec_id);
if(pCodec==NULL){
printf("Codec not found.\n");
return -1;
}
// Open codec
if(avcodec_open2(pCodecCtx, pCodec,NULL)<0){
printf("Could not open codec.\n");
return -1;
}
packet=(AVPacket *)av_malloc(sizeof(AVPacket));
av_init_packet(packet);
//Out Audio Param
uint64_t out_channel_layout=AV_CH_LAYOUT_STEREO;
//nb_samples: AAC-1024 MP3-1152
int out_nb_samples=pCodecCtx->frame_size;
AVSampleFormat out_sample_fmt=AV_SAMPLE_FMT_S16;
int out_sample_rate=44100;
int out_channels=av_get_channel_layout_nb_channels(out_channel_layout);
//Out Buffer Size
int out_buffer_size=av_samples_get_buffer_size(NULL,out_channels ,out_nb_samples,out_sample_fmt, 1);
out_buffer=(uint8_t *)av_malloc(MAX_AUDIO_FRAME_SIZE*2);
pFrame=av_frame_alloc();
//FIX:Some Codec's Context Information is missing
in_channel_layout=av_get_default_channel_layout(pCodecCtx->channels);
//Swr
au_convert_ctx = swr_alloc();
au_convert_ctx=swr_alloc_set_opts(au_convert_ctx,out_channel_layout, out_sample_fmt, out_sample_rate,
in_channel_layout,pCodecCtx->sample_fmt , pCodecCtx->sample_rate,0, NULL);
swr_init(au_convert_ctx);
while(av_read_frame(pFormatCtx, packet)>=0){
if(packet->stream_index==audioStream){
ret = avcodec_decode_audio4( pCodecCtx, pFrame,&got_picture, packet);
if ( ret < 0 ) {
printf("Error in decoding audio frame.\n");
return -1;
}
if ( got_picture > 0 ){
swr_convert(au_convert_ctx,&out_buffer, MAX_AUDIO_FRAME_SIZE,(const uint8_t **)pFrame->data , pFrame->nb_samples);
printf("index:%5d\t pts:%lld\t packet size:%d\n",index,packet->pts,packet->size);
//Write PCM
fwrite(out_buffer, 1, out_buffer_size, pFile);
index++;
}
}
av_free_packet(packet);
}
swr_free(&au_convert_ctx);
fclose(pFile);
av_free(out_buffer);
// Close the codec
avcodec_close(pCodecCtx);
// Close the video file
avformat_close_input(&pFormatCtx);
return 0;
}
运行结果
程序运行后，会解码下面的音频文件。
SDL音频采样数据播放器
源代码
/**
* 最简单的SDL2播放音频的例子（SDL2播放PCM）
* Simplest Audio Play SDL2 (SDL2 play PCM)
*
* 雷霄骅 Lei Xiaohua
* leixiaohua1020@126.com
* 中国传媒大学/数字电视技术
* Communication University of China / Digital TV Technology
* http://blog.csdn.net/leixiaohua1020
*
* 本程序使用SDL2播放PCM音频采样数据。SDL实际上是对底层绘图
* API（Direct3D，OpenGL）的封装，使用起来明显简单于直接调用底层
* API。
*
* 函数调用步骤如下:
*
* [初始化]
* SDL_Init(): 初始化SDL。
* SDL_OpenAudio(): 根据参数（存储于SDL_AudioSpec）打开音频设备。
*
* [循环播放数据]
* SDL_PauseAudio(): 播放音频数据。
* SDL_Delay(): 延时等待播放完成。
*
* This software plays PCM raw audio data using SDL2.
* SDL is a wrapper of low-level API (DirectSound).
* Use SDL is much easier than directly call these low-level API.
*
* The process is shown as follows:
*
* [Init]
* SDL_Init(): Init SDL.
* SDL_OpenAudio(): Opens the audio device with the desired
* parameters (In SDL_AudioSpec).
*
* [Loop to play data]
* SDL_PauseAudio(): Play Audio.
* SDL_Delay(): Wait for completetion of playback.
*/
#include <stdio.h>
#include <tchar.h>
extern "C"
{
#include "sdl/SDL.h"
};
//Buffer:
//|-----------|-------------|
//chunk-------pos---len-----|
static Uint8 *audio_chunk;
static Uint32 audio_len;
static Uint8 *audio_pos;
/* Audio Callback
* The audio function callback takes the following parameters:
* stream: A pointer to the audio buffer to be filled
* len: The length (in bytes) of the audio buffer
*
*/
void fill_audio(void *udata,Uint8 *stream,int len){
//SDL 2.0
SDL_memset(stream, 0, len);
if(audio_len==0) /* Only play if we have data left */
return;
len=(len>audio_len?audio_len:len); /* Mix as much data as possible */
SDL_MixAudio(stream,audio_pos,len,SDL_MIX_MAXVOLUME);
audio_pos += len;
audio_len -= len;
}
int main(int argc, char* argv[])
{
//Init
if(SDL_Init(SDL_INIT_AUDIO | SDL_INIT_TIMER)) {
printf( "Could not initialize SDL - %s\n", SDL_GetError());
return -1;
}
//SDL_AudioSpec
SDL_AudioSpec wanted_spec;
wanted_spec.freq = 44100;
wanted_spec.format = AUDIO_S16SYS;
wanted_spec.channels = 2;
wanted_spec.silence = 0;
wanted_spec.samples = 1024;
wanted_spec.callback = fill_audio;
if (SDL_OpenAudio(&wanted_spec, NULL)<0){
printf("can't open audio.\n");
return -1;
}
FILE *fp=fopen("NocturneNo2inEflat_44.1k_s16le.pcm","rb+");
if(fp==NULL){
printf("cannot open this file\n");
return -1;
}
//For YUV420P
int pcm_buffer_size=4096;
char *pcm_buffer=(char *)malloc(pcm_buffer_size);
int data_count=0;
while(1){
if (fread(pcm_buffer, 1, pcm_buffer_size, fp) != pcm_buffer_size){
// Loop
fseek(fp, 0, SEEK_SET);
fread(pcm_buffer, 1, pcm_buffer_size, fp);
data_count=0;
}
printf("Now Playing %10d Bytes data.\n",data_count);
data_count+=pcm_buffer_size;
//Set audio buffer (PCM data)
audio_chunk = (Uint8 *) pcm_buffer;
//Audio buffer length
audio_len =pcm_buffer_size;
audio_pos = audio_chunk;
//Play
SDL_PauseAudio(0);
while(audio_len>0)//Wait until finish
SDL_Delay(1);
}
free(pcm_buffer);
SDL_Quit();
return 0;
}
运行结果
程序运行后，会读取程序文件夹下的一个PCM采样数据文件，内容如下所示。
下载
Simplest FFmpeg Audio Player
SourceForge：https://sourceforge.net/projects/simplestffmpegaudioplayer/
Github：https://github.com/leixiaohua1020/simplest_ffmpeg_audio_player
开源中国：http://git.oschina.net/leixiaohua1020/simplest_ffmpeg_audio_player
本程序实现了音频的解码和播放。是最简单的FFmpeg音频解码方面的教程。
通过学习本例子可以了解FFmpeg的解码流程。
项目包含3个工程：
simplest_ffmpeg_audio_player：基于FFmpeg+SDL的音频解码器
simplest_ffmpeg_audio_decoder：音频解码器。使用了libavcodec和libavformat。
simplest_audio_play_sdl2：使用SDL2播放PCM采样数据的例子。