ao_dsound.c

自己移植的linux下的流媒体播放器原代码,支持mms协议,支持ftp和http协议.

	
  // Lock the buffer
  res = IDirectSoundBuffer_Lock(hdsbuf,write_offset, len, &lpvPtr1, &dwBytes1, &lpvPtr2, &dwBytes2, 0); 
  // If the buffer was lost, restore and retry lock. 
  if (DSERR_BUFFERLOST == res) 
  { 
    IDirectSoundBuffer_Restore(hdsbuf);
	res = IDirectSoundBuffer_Lock(hdsbuf,write_offset, len, &lpvPtr1, &dwBytes1, &lpvPtr2, &dwBytes2, 0);
  }
 
  
  if (SUCCEEDED(res)) 
  {
  	if( (ao_data.channels == 6) && (ao_data.format!=AF_FORMAT_AC3) ) {
  	    // reorder channels while writing to pointers.
  	    // it's this easy because buffer size and len are always
  	    // aligned to multiples of channels*bytespersample
  	    // there's probably some room for speed improvements here
  	    const int chantable[6] = {0, 1, 4, 5, 2, 3}; // reorder "matrix"
  	    int i, j;
  	    int numsamp,sampsize;

  	    sampsize = af_fmt2bits(ao_data.format)>>3; // bytes per sample
  	    numsamp = dwBytes1 / (ao_data.channels * sampsize);  // number of samples for each channel in this buffer

  	    for( i = 0; i < numsamp; i++ ) for( j = 0; j < ao_data.channels; j++ ) {
  	        memcpy(lpvPtr1+(i*ao_data.channels*sampsize)+(chantable[j]*sampsize),data+(i*ao_data.channels*sampsize)+(j*sampsize),sampsize);
  	    }

  	    if (NULL != lpvPtr2 )
  	    {
  	        numsamp = dwBytes2 / (ao_data.channels * sampsize);
  	        for( i = 0; i < numsamp; i++ ) for( j = 0; j < ao_data.channels; j++ ) {
  	            memcpy(lpvPtr2+(i*ao_data.channels*sampsize)+(chantable[j]*sampsize),data+dwBytes1+(i*ao_data.channels*sampsize)+(j*sampsize),sampsize);
  	        }
  	    }

  	    write_offset+=dwBytes1+dwBytes2;
  	    if(write_offset>=buffer_size)write_offset=dwBytes2;
  	} else {
  	    // Write to pointers without reordering. 
	memcpy(lpvPtr1,data,dwBytes1);
    if (NULL != lpvPtr2 )memcpy(lpvPtr2,data+dwBytes1,dwBytes2);
	write_offset+=dwBytes1+dwBytes2;
    if(write_offset>=buffer_size)write_offset=dwBytes2;
  	}
	
   // Release the data back to DirectSound. 
    res = IDirectSoundBuffer_Unlock(hdsbuf,lpvPtr1,dwBytes1,lpvPtr2,dwBytes2);
    if (SUCCEEDED(res)) 
    { 
	  // Success. 
	  DWORD status;
	  IDirectSoundBuffer_GetStatus(hdsbuf, &status);
      if (!(status & DSBSTATUS_PLAYING)){
	    res = IDirectSoundBuffer_Play(hdsbuf, 0, 0, DSBPLAY_LOOPING);
	  }
	  return dwBytes1+dwBytes2; 
    } 
  } 
  // Lock, Unlock, or Restore failed. 
  return 0;
}

/***************************************************************************************/

/**
\brief handle control commands
\param cmd command
\param arg argument
\return CONTROL_OK or -1 in case the command can't be handled
*/
static int control(int cmd, void *arg)
{
	DWORD volume;
	switch (cmd) {
		case AOCONTROL_GET_VOLUME: {
			ao_control_vol_t* vol = (ao_control_vol_t*)arg;
			IDirectSoundBuffer_GetVolume(hdsbuf, &volume);
			vol->left = vol->right = (float)(volume+10000) / 100.0;
			//printf("ao_dsound: volume: %f\n",vol->left);
			return CONTROL_OK;
		}
		case AOCONTROL_SET_VOLUME: {
			ao_control_vol_t* vol = (ao_control_vol_t*)arg;
			volume = (vol->right * 100.0)-10000;
			IDirectSoundBuffer_SetVolume(hdsbuf, volume);
			//printf("ao_dsound: volume: %f\n",vol->left);
			return CONTROL_OK;
		}
	}
	return -1;
}

/** 
\brief setup sound device
\param rate samplerate
\param channels number of channels
\param format format
\param flags unused
\return 1=success 0=fail
*/
static int init(int rate, int channels, int format, int flags)
{
    int res;
	if (!InitDirectSound()) return 0;

	// ok, now create the buffers
	WAVEFORMATEXTENSIBLE wformat;
	DSBUFFERDESC dsbpridesc;
	DSBUFFERDESC dsbdesc;

	//check if the format is supported in general
	switch(format){
		case AF_FORMAT_AC3:
		case AF_FORMAT_S24_LE:
		case AF_FORMAT_S16_LE:
		case AF_FORMAT_S8:
			break;
		default:
			mp_msg(MSGT_AO, MSGL_V,"ao_dsound: format %s not supported defaulting to Signed 16-bit Little-Endian\n",af_fmt2str_short(format));
			format=AF_FORMAT_S16_LE;
	}   	
	//fill global ao_data
	ao_data.channels = channels;
	ao_data.samplerate = rate;
	ao_data.format = format;
	ao_data.bps = channels * rate * (af_fmt2bits(format)>>3);
	if(ao_data.buffersize==-1) ao_data.buffersize = ao_data.bps; // space for 1 sec
	mp_msg(MSGT_AO, MSGL_V,"ao_dsound: Samplerate:%iHz Channels:%i Format:%s\n", rate, channels, af_fmt2str_short(format));
	mp_msg(MSGT_AO, MSGL_V,"ao_dsound: Buffersize:%d bytes (%d msec)\n", ao_data.buffersize, ao_data.buffersize / ao_data.bps * 1000);

	//fill waveformatex
	ZeroMemory(&wformat, sizeof(WAVEFORMATEXTENSIBLE));
	wformat.Format.cbSize          = (channels > 2) ? sizeof(WAVEFORMATEXTENSIBLE)-sizeof(WAVEFORMATEX) : 0;
	wformat.Format.nChannels       = channels;
	wformat.Format.nSamplesPerSec  = rate;
	if (format == AF_FORMAT_AC3) {
		wformat.Format.wFormatTag      = WAVE_FORMAT_DOLBY_AC3_SPDIF;
		wformat.Format.wBitsPerSample  = 16;
		wformat.Format.nBlockAlign     = 4;
	} else {
		wformat.Format.wFormatTag      = (channels > 2) ? WAVE_FORMAT_EXTENSIBLE : WAVE_FORMAT_PCM;
		wformat.Format.wBitsPerSample  = af_fmt2bits(format);
		wformat.Format.nBlockAlign     = wformat.Format.nChannels * (wformat.Format.wBitsPerSample >> 3);
	}

	// fill in primary sound buffer descriptor
	memset(&dsbpridesc, 0, sizeof(DSBUFFERDESC));
	dsbpridesc.dwSize = sizeof(DSBUFFERDESC);
	dsbpridesc.dwFlags       = DSBCAPS_PRIMARYBUFFER;
	dsbpridesc.dwBufferBytes = 0;
	dsbpridesc.lpwfxFormat   = NULL;


	// fill in the secondary sound buffer (=stream buffer) descriptor
	memset(&dsbdesc, 0, sizeof(DSBUFFERDESC));
	dsbdesc.dwSize = sizeof(DSBUFFERDESC);
	dsbdesc.dwFlags = DSBCAPS_GETCURRENTPOSITION2 /** Better position accuracy */
	                | DSBCAPS_GLOBALFOCUS         /** Allows background playing */
	                | DSBCAPS_CTRLVOLUME;         /** volume control enabled */

	if (channels > 2) {
		wformat.dwChannelMask = channel_mask[channels - 3];
		wformat.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
		wformat.Samples.wValidBitsPerSample = wformat.Format.wBitsPerSample;
		// Needed for 5.1 on emu101k - shit soundblaster
		dsbdesc.dwFlags |= DSBCAPS_LOCHARDWARE;
	}
	wformat.Format.nAvgBytesPerSec = wformat.Format.nSamplesPerSec * wformat.Format.nBlockAlign;

	dsbdesc.dwBufferBytes = ao_data.buffersize;
	dsbdesc.lpwfxFormat = (WAVEFORMATEX *)&wformat;
	buffer_size = dsbdesc.dwBufferBytes;
	write_offset = 0;
	min_free_space = wformat.Format.nBlockAlign;
	ao_data.outburst = wformat.Format.nBlockAlign * 512;

	// create primary buffer and set its format
    
	res = IDirectSound_CreateSoundBuffer( hds, &dsbpridesc, &hdspribuf, NULL );
	if ( res != DS_OK ) {
		UninitDirectSound();
		mp_msg(MSGT_AO, MSGL_ERR,"ao_dsound: cannot create primary buffer (%s)\n", dserr2str(res));
		return 0;
	}
	res = IDirectSoundBuffer_SetFormat( hdspribuf, (WAVEFORMATEX *)&wformat );
	if ( res != DS_OK ) mp_msg(MSGT_AO, MSGL_WARN,"ao_dsound: cannot set primary buffer format (%s), using standard setting (bad quality)", dserr2str(res));

	mp_msg(MSGT_AO, MSGL_V, "ao_dsound: primary buffer created\n");

	// now create the stream buffer

	res = IDirectSound_CreateSoundBuffer(hds, &dsbdesc, &hdsbuf, NULL);
	if (res != DS_OK) {
		if (dsbdesc.dwFlags & DSBCAPS_LOCHARDWARE) {
			// Try without DSBCAPS_LOCHARDWARE
			dsbdesc.dwFlags &= ~DSBCAPS_LOCHARDWARE;
			res = IDirectSound_CreateSoundBuffer(hds, &dsbdesc, &hdsbuf, NULL);
		}
		if (res != DS_OK) {
			UninitDirectSound();
			mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: cannot create secondary (stream)buffer (%s)\n", dserr2str(res));
			return 0;
		}
	}
	mp_msg(MSGT_AO, MSGL_V, "ao_dsound: secondary (stream)buffer created\n");
	return 1;
}



/**
\brief stop playing and empty buffers (for seeking/pause)
*/
static void reset()
{
	IDirectSoundBuffer_Stop(hdsbuf);
	// reset directsound buffer
	IDirectSoundBuffer_SetCurrentPosition(hdsbuf, 0);
	write_offset=0;
}

/**
\brief stop playing, keep buffers (for pause)
*/
static void audio_pause()
{
	IDirectSoundBuffer_Stop(hdsbuf);
}

/**
\brief resume playing, after audio_pause()
*/
static void audio_resume()
{
	IDirectSoundBuffer_Play(hdsbuf, 0, 0, DSBPLAY_LOOPING);
}

/** 
\brief close audio device
\param immed stop playback immediately
*/
static void uninit(int immed)
{
	if(immed)reset();
	else{
		DWORD status;
		IDirectSoundBuffer_Play(hdsbuf, 0, 0, 0);
		while(!IDirectSoundBuffer_GetStatus(hdsbuf,&status) && (status&DSBSTATUS_PLAYING))
			usec_sleep(20000);
	}
	DestroyBuffer();
	UninitDirectSound();
}

/**
\brief find out how many bytes can be written into the audio buffer without
\return free space in bytes, has to return 0 if the buffer is almost full
*/
static int get_space()
{
	int space;
	DWORD play_offset;
	IDirectSoundBuffer_GetCurrentPosition(hdsbuf,&play_offset,NULL);
	space=buffer_size-(write_offset-play_offset);                                             
	//                |                                                      | <-- const --> |                |                 |
	//                buffer start                                           play_cursor     write_cursor     write_offset      buffer end
	// play_cursor is the actual postion of the play cursor
	// write_cursor is the position after which it is assumed to be save to write data
	// write_offset is the postion where we actually write the data to
	if(space > buffer_size)space -= buffer_size; // write_offset < play_offset
	if(space < min_free_space)return 0;
	return space-min_free_space;
}

/**
\brief play 'len' bytes of 'data'
\param data pointer to the data to play
\param len size in bytes of the data buffer, gets rounded down to outburst*n
\param flags currently unused
\return number of played bytes
*/
static int play(void* data, int len, int flags)
{
	DWORD play_offset;
	int space;
  
	// make sure we have enough space to write data
	IDirectSoundBuffer_GetCurrentPosition(hdsbuf,&play_offset,NULL);
	space=buffer_size-(write_offset-play_offset);                                             
	if(space > buffer_size)space -= buffer_size; // write_offset < play_offset
	if(space < len) len = space;

	len = (len / ao_data.outburst) * ao_data.outburst;
	return write_buffer(data, len);
}

/**
\brief get the delay between the first and last sample in the buffer
\return delay in seconds
*/
static float get_delay()
{
	DWORD play_offset;
	int space;
	IDirectSoundBuffer_GetCurrentPosition(hdsbuf,&play_offset,NULL);
	space=play_offset-write_offset;                                             
	if(space <= 0)space += buffer_size;
	return (float)(buffer_size - space) / (float)ao_data.bps;
}