ffmpeg音频播放代码示例-avcodec_decode_audio4

一、概述

最近在学习ffmpeg解码的内容，参考了官方的教程http://dranger.com/ffmpeg/tutorial03.html，结果发现这个音频解码的教程有点问题。参考了各种博客，并同时啃ffplay.c的源码，发现avcodec_decode_audio4多了一个resample（重采样）的概念。

其解码以及播放音频的思路为：

首先，ffmpeg设置本机的audio播放参数（target format），如freq(频率）为44100，format为AV_SAMPLE_FMT_S16，channels为2。这个播放参数是SDL实际播放音频时使用的参数。

但是！但是我们的audio file（如mp3文件）的audio数据很可能有其自己的audio播放参数（source format），而这些参数不同于我们实际的SDL播放参数，于是ffmpeg在其中插入resample（重采用）的过程，将source format转换成target format。

简单的说就是一个audio参数设置思路的转变：

这个思路转变最大的好处，就是本机播放的格式可以不用再迁就audio file，而是可以根据自己的需要自行设定，缺点很显然就是ffmpeg的CPU开销会增大。

二、代码示例（源码见“附录”）

源码在官方教程基础上把其中视频部分删除，在main函数最后加上一个无限循环，并添加resample函数，最后将resample插入到sdl的回调函数之中。

源码中关于queue的代码为官网教程原版复制，其主要作用就是让main函数和SDL audio线程互斥的push queue和get queue，以下不再赘述。

1、main函数代码结构

main函数伪代码结构如下：

 SDL Initialization
 ffmpeg open audio file
 Set SDL audio parameters
 Set ffmpeg audio parameters(target format)
 while(ffmpeg_read_frame(pkt)) {
     packet_queue_put(pkt);
 }
 while() {
     sleep();
 }

ffmpeg从audio file中不停的读取数据，并将读出的packet放入queue中。此时我们要清楚，另外还有一个SDL audio线程在等待queue中的数据。

2、SDL audio线程

SDL audio线程主要执行一个回调函数，对应源码中的函数为audio_callback(void * userdata, Uint8 * stream, int len)。这个函数的使命就是将解码后的数据放入参数stream这个缓冲区中，以便SDL audio线程从stream缓冲区中获取数据play。这个缓冲区的大小为参数len，而userdata则是用户自定的参数。其伪代码结构如下：

 audio_buf_index = ;
 while(len > ) {
     audio_size = audio_decode_frame(audio_buf_tmp);
     memcpy(stream, audio_buf_tmp, audio_size);
     len -= audio_size;
     stream += audio_size;
     audio_buf_index += audio_size;
 }

其中audio_decode_frame函数会从queue中取出packet，并对packet中的frame进行解码和resample，然后将数据放入audio_buf_tmp缓冲区中。

3、Resample函数

Resample的过程和结构体SwrContext息息相关。使用这个结构体共需要2步。

1、先初始化SwrContex，指定target format和source format；

2、使用已初始化的SwrContext，对frame进行resample。

Resample的伪代码如下：

 struct SwrContext * swr_ctx = NULL;
 audio_hw_params_src = audio_hw_params_tgt
 int resample(AVFrame * af, uint8_t * audio_buf, int * audio_buf_size)
 {
     if(audio_hw_params_src != audio_hw_params(af)) {
         swr_ctx = swr_alloc_set_opts(audio_hw_params_tgt, audio_hw_params(af));
         audio_hw_params_src = audio_hw_params(af);
     }
     in = af;
     swr_convert(swr_ctx, out, in);
     audio_buf = out;
 } 

一开始，audio_hw_parames_src（source format）被初始化为target format，在resample获得第一个frame后，会从该frame中提取source format，并将其赋值给audio_hw_params_src，同时初始化SwrContext这个结构体，指定target format和source format。然后swr_convert对输入的frame进行resample(swr_convert），然后将resample后得到的数据放进resample函数指定的缓冲区（audio_buf）中。

附录：

 #include <libavcodec/avcodec.h>
 #include <libavformat/avformat.h>
 #include <libswscale/swscale.h>
 #include <libswresample/swresample.h>

 #include <SDL.h>
 #include <SDL_thread.h>

 #ifdef __MINGW32__
 #undef main /* Prevents SDL from overriding main() */
 #endif

 #include <stdio.h>
 #include <assert.h>
 #include <sys/types.h>
 #include <sys/stat.h>
 #include <fcntl.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 SDL_AUDIO_BUFFER_SIZE 1024
 #define MAX_AUDIO_FRAME_SIZE 192000

 #include <signal.h>

 typedef struct AudioParams {
     int freq;
     int channels;
     int64_t channel_layout;
     enum AVSampleFormat fmt;
     int frame_size;
     int bytes_per_sec;
 } AudioParams;
 int sample_rate, nb_channels;
 int64_t channel_layout;
 AudioParams audio_hw_params_tgt;
 AudioParams audio_hw_params_src;

 int resample(AVFrame * af, uint8_t * audio_buf, int * audio_buf_size);

 struct SwrContext * swr_ctx = NULL;

 int resample(AVFrame * af, uint8_t * audio_buf, int * audio_buf_size)
 {
     int data_size = ;
     int resampled_data_size = ;
     int64_t dec_channel_layout;
     data_size = av_samples_get_buffer_size(NULL,
             av_frame_get_channels(af),
             af->nb_samples,
             af->format,
             );

     dec_channel_layout =
         (af->channel_layout && av_frame_get_channels(af) == av_get_channel_layout_nb_channels(af->channel_layout)) ?
         af->channel_layout : av_get_default_channel_layout(av_frame_get_channels(af));
     if(     af->format              != audio_hw_params_src.fmt                 ||
             af->sample_rate     != audio_hw_params_src.freq              ||
             dec_channel_layout     != audio_hw_params_src.channel_layout     ||
             !swr_ctx) {
         swr_free(&swr_ctx);
         swr_ctx = swr_alloc_set_opts(NULL,
                                         audio_hw_params_tgt.channel_layout, audio_hw_params_tgt.fmt, audio_hw_params_tgt.freq,
                                         dec_channel_layout, af->format, af->sample_rate,
                                         , NULL);
         if (!swr_ctx || swr_init(swr_ctx) < ) {
             av_log(NULL, AV_LOG_ERROR,
                    "Cannot create sample rate converter for conversion of %d Hz %s %d channels to %d Hz %s %d channels!\n",
                     af->sample_rate, av_get_sample_fmt_name(af->format), av_frame_get_channels(af),
                     audio_hw_params_tgt.freq, av_get_sample_fmt_name(audio_hw_params_tgt.fmt), audio_hw_params_tgt.channels);
             swr_free(&swr_ctx);
             return -;
         }
         printf("swr_init\n");
         audio_hw_params_src.channels = av_frame_get_channels(af);
         audio_hw_params_src.fmt = af->format;
         audio_hw_params_src.freq = af->sample_rate;
     }

     if (swr_ctx) {
         const uint8_t **in = (const uint8_t **)af->extended_data;
         uint8_t **out = &audio_buf;
         int out_count = (int64_t)af->nb_samples * audio_hw_params_tgt.freq / af->sample_rate + ;
         int out_size  = av_samples_get_buffer_size(NULL, audio_hw_params_tgt.channels, out_count, audio_hw_params_tgt.fmt, );
         int len2;
         if (out_size < ) {
             av_log(NULL, AV_LOG_ERROR, "av_samples_get_buffer_size() failed\n");
             return -;
         }
         av_fast_malloc(&audio_buf, audio_buf_size, out_size);
         if (!audio_buf)
             return AVERROR(ENOMEM);
         len2 = swr_convert(swr_ctx, out, out_count, in, af->nb_samples);
         if (len2 < ) {
             av_log(NULL, AV_LOG_ERROR, "swr_convert() failed\n");
             return -;
         }
         if (len2 == out_count) {
             av_log(NULL, AV_LOG_WARNING, "audio buffer is probably too small\n");
             if (swr_init(swr_ctx) < )
                 swr_free(&swr_ctx);
         }
         resampled_data_size = len2 * audio_hw_params_tgt.channels * av_get_bytes_per_sample(audio_hw_params_tgt.fmt);
     } else {
         audio_buf = af->data[];
         resampled_data_size = data_size;
     }

     return resampled_data_size;
 }

 static void sigterm_handler(int sig)
 {
     exit();
 }

 typedef struct PacketQueue {
   AVPacketList *first_pkt, *last_pkt;
   int nb_packets;
   int size;
   SDL_mutex *mutex;
   SDL_cond *cond;
 } PacketQueue;

 PacketQueue audioq;

 int quit = ;

 void packet_queue_init(PacketQueue *q) {
   memset(q, , sizeof(PacketQueue));
   q->mutex = SDL_CreateMutex();
   q->cond = SDL_CreateCond();
 }
 int packet_queue_put(PacketQueue *q, AVPacket *pkt) {

   AVPacketList *pkt1;
   if(av_dup_packet(pkt) < ) {
     return -;
   }
   pkt1 = av_malloc(sizeof(AVPacketList));
   if (!pkt1)
     return -;
   pkt1->pkt = *pkt;
   pkt1->next = NULL;

   SDL_LockMutex(q->mutex);

   if (!q->last_pkt)
     q->first_pkt = pkt1;
   else
     q->last_pkt->next = pkt1;
   q->last_pkt = pkt1;
   q->nb_packets++;
   q->size += pkt1->pkt.size;
   SDL_CondSignal(q->cond);

   SDL_UnlockMutex(q->mutex);
   return ;
 }
 static int packet_queue_get(PacketQueue *q, AVPacket *pkt, int block)
 {
   AVPacketList *pkt1;
   int ret;

   SDL_LockMutex(q->mutex);

   for(;;) {

     if(quit) {
       ret = -;
       break;
     }

     pkt1 = q->first_pkt;
     if (pkt1) {
       q->first_pkt = pkt1->next;
       if (!q->first_pkt)
     q->last_pkt = NULL;
       q->nb_packets--;
       q->size -= pkt1->pkt.size;
       *pkt = pkt1->pkt;
       av_free(pkt1);
       ret = ;
       break;
     } else if (!block) {
       ret = ;
       break;
     } else {
       SDL_CondWait(q->cond, q->mutex);
     }
   }
   SDL_UnlockMutex(q->mutex);
   return ret;
 }

 AVFrame frame;
 int audio_decode_frame(AVCodecContext *aCodecCtx, uint8_t *audio_buf, int buf_size) {

     static AVPacket pkt;
     static uint8_t *audio_pkt_data = NULL;
     static int audio_pkt_size = ;

     int len1, data_size = ;

     for(;;) {
         while(audio_pkt_size > ) {
             int got_frame = ;
             len1 = avcodec_decode_audio4(aCodecCtx, &frame, &got_frame, &pkt);
             if(len1 < ) {
                 /* if error, skip frame */
                 audio_pkt_size = ;
                 break;
             }
             audio_pkt_data += len1;
             audio_pkt_size -= len1;
             data_size = ;
             if(got_frame) {
                 data_size = resample(&frame, audio_buf, &buf_size);
                 // data_size = av_samples_get_buffer_size(NULL,
                 //         aCodecCtx->channels,
                 //         frame.nb_samples,
                 //         aCodecCtx->sample_fmt,
                 //         1);
                 assert(data_size <= buf_size);
                 // memcpy(audio_buf, frame.data[0], data_size);
             }
             if(data_size <= ) {
                 /* No data yet, get more frames */
                 continue;
             }
             // memcpy(audio_buf, frame.data[0], data_size);

             /* We have data, return it and come back for more later */
             return data_size;
         }
         if(pkt.data)
             av_free_packet(&pkt);

         if(quit) {
             return -;
         }

         if(packet_queue_get(&audioq, &pkt, ) < ) {
             return -;
         }
         audio_pkt_data = pkt.data;
         audio_pkt_size = pkt.size;
     }
 }

 void audio_callback(void *userdata, Uint8 *stream, int len) {

   AVCodecContext *aCodecCtx = (AVCodecContext *)userdata;
   int len1, audio_size;

   static uint8_t audio_buf[(MAX_AUDIO_FRAME_SIZE * ) / ];
   static unsigned int audio_buf_size = ;
   static unsigned int audio_buf_index = ;

   while(len > ) {
       if(audio_buf_index >= audio_buf_size) {
           /* We have already sent all our data; get more */
           audio_size = audio_decode_frame(aCodecCtx, audio_buf, sizeof(audio_buf));
           if(audio_size < ) {
               /* If error, output silence */
               audio_buf_size = ; // arbitrary?
               memset(audio_buf, , audio_buf_size);
           } else {
               audio_buf_size = audio_size;
           }
           audio_buf_index = ;
       }
       len1 = audio_buf_size - audio_buf_index;
       if(len1 > len)
           len1 = len;
       memcpy(stream, (uint8_t *)audio_buf + audio_buf_index, len1);
       len -= len1;
       stream += len1;
       audio_buf_index += len1;
   }
 }

 int main(int argc, char *argv[]) {

   AVFormatContext *pFormatCtx = NULL;
   int             i, audioStream;
   AVPacket        packet;

   AVCodecContext  *aCodecCtxOrig = NULL;
   AVCodecContext  *aCodecCtx = NULL;
   AVCodec         *aCodec = NULL;

   SDL_Event       event;
   SDL_AudioSpec   wanted_spec, spec;

   signal(SIGINT , sigterm_handler); /* Interrupt (ANSI).    */
   signal(SIGTERM, sigterm_handler); /* Termination (ANSI).  */

   if(argc < ) {
     fprintf(stderr, "Usage: test <file>\n");
     exit();
   }
   // Register all formats and codecs
   av_register_all();

   if(SDL_Init(SDL_INIT_AUDIO)) {
     fprintf(stderr, "Could not initialize SDL - %s\n", SDL_GetError());
     exit();
   }

   // Open video file
   if(avformat_open_input(&pFormatCtx, argv[], NULL, NULL)!=)
     return -; // Couldn't open file

   // Retrieve stream information
   if(avformat_find_stream_info(pFormatCtx, NULL)<)
     return -; // Couldn't find stream information

   // Dump information about file onto standard error
   av_dump_format(pFormatCtx, , argv[], );

   // Find the first video stream
   audioStream=-;
   for(i=; i<pFormatCtx->nb_streams; i++) {
     if(pFormatCtx->streams[i]->codec->codec_type==AVMEDIA_TYPE_AUDIO &&
        audioStream < ) {
       audioStream=i;
     }
   }
   // if(videoStream==-1)
   //   return -1; // Didn't find a video stream
   if(audioStream==-)
     return -;

   aCodecCtxOrig=pFormatCtx->streams[audioStream]->codec;
   aCodec = avcodec_find_decoder(aCodecCtxOrig->codec_id);
   if(!aCodec) {
     fprintf(stderr, "Unsupported codec!\n");
     return -;
   }

   // Copy context
   aCodecCtx = avcodec_alloc_context3(aCodec);
   if(avcodec_copy_context(aCodecCtx, aCodecCtxOrig) != ) {
     fprintf(stderr, "Couldn't copy codec context");
     return -; // Error copying codec context
   }

   avcodec_open2(aCodecCtx, aCodec, NULL);

   sample_rate = aCodecCtx->sample_rate;
   nb_channels = aCodecCtx->channels;
   channel_layout = aCodecCtx->channel_layout;

   printf("channel_layout=%" PRId64 "\n", channel_layout);
   printf("nb_channels=%d\n", nb_channels);
   printf("freq=%d\n", sample_rate);

   if (!channel_layout || nb_channels != av_get_channel_layout_nb_channels(channel_layout)) {
       channel_layout = av_get_default_channel_layout(nb_channels);
       channel_layout &= ~AV_CH_LAYOUT_STEREO_DOWNMIX;
       printf("correction\n");
   }

   // Set audio settings from codec info
   wanted_spec.freq = sample_rate;
   wanted_spec.format = AUDIO_S16SYS;
   wanted_spec.channels = nb_channels;
   wanted_spec.silence = ;
   wanted_spec.samples = SDL_AUDIO_BUFFER_SIZE;
   wanted_spec.callback = audio_callback;
   wanted_spec.userdata = aCodecCtx;

   if(SDL_OpenAudio(&wanted_spec, &spec) < ) {
     fprintf(stderr, "SDL_OpenAudio: %s\n", SDL_GetError());
     return -;
   }
   printf("freq: %d\tchannels: %d\n", spec.freq, spec.channels);

   audio_hw_params_tgt.fmt = AV_SAMPLE_FMT_S16;
   audio_hw_params_tgt.freq = spec.freq;
   audio_hw_params_tgt.channel_layout = channel_layout;
   audio_hw_params_tgt.channels =  spec.channels;
   audio_hw_params_tgt.frame_size = av_samples_get_buffer_size(NULL, audio_hw_params_tgt.channels, , audio_hw_params_tgt.fmt, );
   audio_hw_params_tgt.bytes_per_sec = av_samples_get_buffer_size(NULL, audio_hw_params_tgt.channels, audio_hw_params_tgt.freq, audio_hw_params_tgt.fmt, );
   if (audio_hw_params_tgt.bytes_per_sec <=  || audio_hw_params_tgt.frame_size <= ) {
       printf("size error\n");
       return -;
   }
   audio_hw_params_src = audio_hw_params_tgt;

   // audio_st = pFormatCtx->streams[index]
   packet_queue_init(&audioq);
   SDL_PauseAudio();

   // Read frames and save first five frames to disk
   i=;
   while(av_read_frame(pFormatCtx, &packet)>=) {
     if(packet.stream_index==audioStream) {
       packet_queue_put(&audioq, &packet);
     } else {
       av_free_packet(&packet);
     }
     // Free the packet that was allocated by av_read_frame
     SDL_PollEvent(&event);
     switch(event.type) {
     case SDL_QUIT:
       quit = ;
       SDL_Quit();
       exit();
       break;
     default:
       break;
     }

   }

   while() SDL_Delay();

   // Close the codecs
   avcodec_close(aCodecCtxOrig);
   avcodec_close(aCodecCtx);

   // Close the video file
   avformat_close_input(&pFormatCtx);

   return ;
 }

下载源码以及Makefile：

http://pan.baidu.com/s/1pJUXLZP