tutorial03.java

ffmpeg开发指南

## An ffmpeg and SDL Tutorial

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## Tutorial 03: Playing Sound

Code: tutorial03.c

### Audio

So now we want to play sound. SDL also gives us methods for outputting sound.
The SDL_OpenAudio() function is used to open the audio device itself. It takes
as arguments an SDL_AudioSpec struct, which contains all the information about
the audio we are going to output. 

Before we show how you set this up, let's explain first about how audio is
handled by computers. Digital audio consists of a long stream of **samples**.
Each sample represents a value of the audio waveform. Sounds are recorded at a
certain **sample rate**, which simply says how fast to play each sample, and
is measured in number of samples per second. Example sample rates are 22,050
and 44,100 samples per second, which are the rates used for radio and CD
respectively. In addition, most audio can have more than one channel for
stereo or surround, so for example, if the sample is in stereo, the samples
will come 2 at a time. When we get data from a movie file, we don't know how
many samples we will get, but ffmpeg will not give us partial samples - that
also means that it will not split a stereo sample up, either. 

SDL's method for playing audio is this: you set up your audio options: the
sample rate (called "freq" for **frequency** in the SDL struct), number of
channels, and so forth, and we also set a callback function and userdata. When
we begin playing audio, SDL will continually call this callback function and
ask it to fill the audio buffer with a certain number of bytes. After we put
this information in the SDL_AudioSpec struct, we call SDL_OpenAudio(), which
will open the audio device and give us back _another_ AudioSpec struct. These
are the specs we will _actually_ be using -- we are not guaranteed to get what
we asked for! 

### Setting Up the Audio

Keep that all in your head for the moment, because we don't actually have any
information yet about the audio streams yet! Let's go back to the place in our
code where we found the video stream and find which stream is the audio
stream. 
    
    
    // Find the first video stream
    videoStream=-1;
    audioStream=-1;
    for(i=0; i < pFormatCtx->nb_streams; i++) {
      if(pFormatCtx->streams[i]->codec->codec_type==CODEC_TYPE_VIDEO
         &&
           videoStream < 0) {
        videoStream=i;
      }
      if(pFormatCtx->streams[i]->codec->codec_type==CODEC_TYPE_AUDIO &&
         audioStream < 0) {
        audioStream=i;
      }
    }
    if(videoStream==-1)
      return -1; // Didn't find a video stream
    if(audioStream==-1)
      return -1;
    

From here we can get all the info we want from the AVCodecContext from the
stream, just like we did with the video stream: 
    
    
    AVCodecContext *aCodecCtx;
    
    aCodecCtx=pFormatCtx->streams[audioStream]->codec;
    

Contained within this codec context is all the information we need to set up
our audio: 
    
    
    wanted_spec.freq = aCodecCtx->sample_rate;
    wanted_spec.format = AUDIO_S16SYS;
    wanted_spec.channels = aCodecCtx->channels;
    wanted_spec.silence = 0;
    wanted_spec.samples = SDL_AUDIO_BUFFER_SIZE;
    wanted_spec.callback = audio_callback;
    wanted_spec.userdata = aCodecCtx;
    
    if(SDL_OpenAudio(&wanted_spec, &spec) < 0) {
      fprintf(stderr, "SDL_OpenAudio: %s\n", SDL_GetError());
      return -1;
    }
    

Let's go through these: 

  * freq: The sample rate, as explained earlier.
  * format: This tells SDL what format we will be giving it. The "S" in
"S16SYS" stands for "signed", the 16 says that each sample is 16 bits long,
and "SYS" means that the endian-order will depend on the system you are on.
This is the format that avcodec_decode_audio2 will give us the audio in.
  * channels: Number of audio channels.
  * silence: This is the value that indicated silence. Since the audio is
signed, 0 is of course the usual value.
  * samples: This is the size of the audio buffer that we would like SDL to
give us when it asks for more audio. A good value here is between 512 and
8192; ffplay uses 1024.
  * callback: Here's where we pass the actual callback function. We'll talk
more about the callback function later.
  * userdata: SDL will give our callback a void pointer to any user data that
we want our callback function to have. We want to let it know about our codec
context; you'll see why.

Finally, we open the audio with SDL_OpenAudio. 

If you remember from the previous tutorials, we still need to open the audio
codec itself. This is straightforward: 
    
    
    AVCodec         *aCodec;
    
    aCodec = avcodec_find_decoder(aCodecCtx->codec_id);
    if(!aCodec) {
      fprintf(stderr, "Unsupported codec!\n");
      return -1;
    }
    avcodec_open(aCodecCtx, aCodec);
    

### Queues

There! Now we're ready to start pulling audio information from the stream. But
what do we do with that information? We are going to be continuously getting
packets from the movie file, but at the same time SDL is going to call the
callback function! The solution is going to be to create some kind of global
structure that we can stuff audio packets in so our audio_callback has
something to get audio data from! So what we're going to do is to create a
**queue** of packets. ffmpeg even comes with a structure to help us with this:
AVPacketList, which is just a linked list for packets. Here's our queue
structure: 
    
    
    typedef struct PacketQueue {
      AVPacketList *first_pkt, *last_pkt;
      int nb_packets;
      int size;
      SDL_mutex *mutex;
      SDL_cond *cond;
    } PacketQueue;
    

First, we should point out that nb_packets is not the same as size -- size
refers to a byte size that we get from packet->size. You'll notice that we
have a mutex and a condtion variable in there. This is because SDL is running
the audio process as a separate thread. If we don't lock the queue properly,
we could really mess up our data. We'll see how in the implementation of the
queue. Every programmer should know how to make a queue, but we're including
this so you can learn the SDL functions. 

First we make a function to initialize the queue: 
    
    
    void packet_queue_init(PacketQueue *q) {
      memset(q, 0, sizeof(PacketQueue));
      q->mutex = SDL_CreateMutex();
      q->cond = SDL_CreateCond();
    }
    

Then we will make a function to put stuff in our queue: 
    
    
    int packet_queue_put(PacketQueue *q, AVPacket *pkt) {
    
      AVPacketList *pkt1;
      if(av_dup_packet(pkt) < 0) {
        return -1;
      }
      pkt1 = av_malloc(sizeof(AVPacketList));
      if (!pkt1)
        return -1;
      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 0;
    }
    

SDL_LockMutex() locks the mutex in the queue so we can add something to it,
and then SDL_CondSignal() sends a signal to our get function (if it is
waiting) through our condition variable to tell it that there is data and it
can proceed, then unlocks the mutex to let it go. 

Here's the corresponding get function. Notice how SDL_CondWait() makes the
function **block** (i.e. pause until we get data) if we tell it to. 
    
    
    int quit = 0;
    
    static int packet_queue_get(PacketQueue *q, AVPacket *pkt, int block) {
      AVPacketList *pkt1;
      int ret;
      
      SDL_LockMutex(q->mutex);
      
      for(;;) {
        
        if(quit) {
          ret = -1;
          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 = 1;
          break;
        } else if (!block) {
          ret = 0;
          break;
        } else {
          SDL_CondWait(q->cond, q->mutex);