audio_sdl_old.cpp

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      m_psptr->wake_sync_thread();
      return;
    }
#ifdef DEBUG_SYNC
    audio_message(LOG_DEBUG, "No buffer in audio callback %u %u", 
		  m_buffer_bytes_loaded, outBufferTotalBytes);
#endif
    m_consec_no_buffers++;
    if (m_consec_no_buffers > 10) {
      Our_SDL_PauseAudio(1);
      m_audio_paused = 1;
      m_resync_required = 1;
      m_resync_buffer = m_play_index;
      m_psptr->wake_sync_thread();
    }
    if (m_audio_waiting_buffer) {
      m_audio_waiting_buffer = 0;
      SDL_SemPost(m_audio_waiting);
      //audio_message(LOG_DEBUG, "post no data");
    }
    audio_message(LOG_DEBUG, "return - no samples");
    return;
  }

  // We have a valid buffer.  Push it to SDL.
  m_consec_no_buffers = 0;

  if (m_format == AUDIO_FMT_HW_AC3) {
    Our_SDL_MixAudio(outStream, 
		     (const unsigned char *)&m_sample_buffer[m_play_index][0],
		     m_buffer_len[m_play_index],
		     m_volume);
    if (m_buffer_bytes_loaded > m_buffer_size) {
      m_buffer_bytes_loaded -= m_buffer_size;
    } else {
      m_buffer_bytes_loaded = 0;
    }
    m_buffer_filled[m_play_index] = 0;
    m_play_index++;
    m_play_index %= DECODE_BUFFERS_MAX;
    m_play_sample_index = 0;
    freed_buffer = 1;
    if (m_resync_required) {
	// resync required from codec side.  Shut down, and notify sync task
      if (m_resync_buffer == m_play_index) {
	Our_SDL_PauseAudio(1);
	m_audio_paused = 1;
	m_psptr->wake_sync_thread();
#ifdef DEBUG_SYNC
	audio_message(LOG_DEBUG, "sempost");
#endif
	outBufferTotalBytes = 0;
      }
    }

  } else {
    while (outBufferTotalBytes > 0) {
      uint32_t outBufferSamples, outBufferBytes;
      uint32_t decodedBufferBytes, decodedBufferSamples;

      // calculate number of bytes left in the decoded bytes ring
      decodedBufferBytes = m_buffer_size - m_play_sample_index;
      // samples from bytes
      decodedBufferSamples = 
	decodedBufferBytes / (m_channels * m_bytes_per_sample_input);
    
      // See how many samples we can write into SDL buffers
      outBufferSamples = outBufferTotalBytes / 
	(m_got_channels * m_bytes_per_sample_output);

      // Adjust bytes from decoded ring accordingly
      if (outBufferSamples < decodedBufferSamples) {
	decodedBufferBytes = 
	  outBufferSamples * m_channels * m_bytes_per_sample_input;
	decodedBufferSamples = outBufferSamples;
      }
      // Adjust bytes to copy
      outBufferBytes = 
	decodedBufferSamples * m_got_channels * m_bytes_per_sample_output;
#ifdef DEBUG_SYNC
      audio_message(LOG_DEBUG, "Playing "U64" offset %d",
		    m_buffer_time[m_play_index], m_play_sample_index);
#endif
      if (m_convert_buffer) {
	// Convert the buffer based on the number of samples
	audio_convert_data(&m_sample_buffer[m_play_index][m_play_sample_index],
			   decodedBufferSamples);
	// Mix based on the number of bytes
	Our_SDL_MixAudio(outStream, (const unsigned char *)m_convert_buffer, 
			 outBufferBytes, m_volume);
      } else {
	Our_SDL_MixAudio(outStream, 
			 (const unsigned char *)&m_sample_buffer[m_play_index][m_play_sample_index],
			 outBufferBytes,
			 m_volume);
      }
      outBufferTotalBytes -= outBufferBytes;
      outStream += outBufferBytes;

      if (decodedBufferBytes <= m_buffer_bytes_loaded)
	m_buffer_bytes_loaded -= decodedBufferBytes;
      else 
	m_buffer_bytes_loaded = 0;
      m_play_sample_index += decodedBufferBytes;
      if (m_play_sample_index >= m_buffer_size) {
#ifdef DEBUG_SYNC
	audio_message(LOG_DEBUG, "finished with buffer %d %d", 
		      m_play_index, m_buffer_bytes_loaded);
#endif

	m_buffer_filled[m_play_index] = 0;
	m_play_index++;
	m_play_index %= DECODE_BUFFERS_MAX;
	m_play_sample_index = 0;
	freed_buffer = 1;
	if (m_resync_required) {
	  // resync required from codec side.  Shut down, and notify sync task
	  if (m_resync_buffer == m_play_index) {
	    Our_SDL_PauseAudio(1);
	    m_audio_paused = 1;
	    m_psptr->wake_sync_thread();
#ifdef DEBUG_SYNC
	    audio_message(LOG_DEBUG, "sempost");
#endif
	    outBufferTotalBytes = 0;
	  }
	}
      }
    }
  }    
  // Increment past this buffer.
  if (m_first_time != 0) {
    // First time through - tell the sync task we've started, so it can
    // keep sync time.
    m_first_time = 0;
    if (m_use_SDL_delay != 0) 
      m_buffer_latency = delay;
    else
      m_buffer_latency = 0;
    m_psptr->audio_is_ready(m_buffer_latency, this_time);
    m_consec_wrong_latency = 0;
    m_wrong_latency_total = 0;
  } 
  else if (m_do_sync) {
#define ALLOWED_LATENCY 2
    if (m_use_SDL_delay != 0) {
      // Here, we're using the delay value from the audio buffers,
      // rather than the calculated time...
      // Okay - now we check for latency changes.
      index_time = delay + m_play_time;

      if (this_time > index_time + ALLOWED_LATENCY || 
	  this_time < index_time - ALLOWED_LATENCY) {
#ifdef DEBUG_SYNC_CHANGES
	audio_message(LOG_DEBUG, 
		      "potential change - index time "U64" time "U64" delay %d", 
		      index_time, this_time, delay);
#endif
	if (m_consec_wrong_latency == 0) {
	  m_consec_wrong_latency = 1;
	} else {
	  m_consec_wrong_latency++;
	  int64_t test;
	  test = this_time - index_time;
	  m_wrong_latency_total += test;
	  int64_t div;
	  div = m_wrong_latency_total / (int64_t)(m_consec_wrong_latency - 1);
#ifdef DEBUG_SYNC_CHANGES
	  audio_message(LOG_DEBUG, "values are "D64" "D64" %d", 
			test, div, m_consec_wrong_latency);
#endif
	  if (test > ALLOWED_LATENCY || test < -ALLOWED_LATENCY) {
	    if (m_consec_wrong_latency > 5) {
	      m_consec_wrong_latency = 0;
	      m_wrong_latency_total = 0;
	      if (test < -10000) {
		audio_message(LOG_ERR, "Latency error - test is "D64,
			      test);
	      } else {
		m_psptr->adjust_start_time(test);
	      }
	    }
	  } else {
	    // average wrong latency is not greater than allowed latency
	    m_consec_wrong_latency = 0;
	    m_wrong_latency_total = 0;
	  }
	}
      } else {
	m_consec_wrong_latency = 0;
	m_wrong_latency_total = 0;
      }
    } else {
      // We're using the calculate latency values - they're not very
      // accurate, but better than nothing...
      // we have a latency number - see if it really is correct
      uint64_t index_time = delay + m_play_time;
#if DEBUG_SYNC
      audio_message(LOG_DEBUG, 
		    "latency - time " U64 " index " U64 " latency " U64 " %u", 
		    this_time, index_time, m_buffer_latency, m_buffer_bytes_loaded);
#endif
      if (this_time > index_time + ALLOWED_LATENCY || 
	  this_time < index_time - ALLOWED_LATENCY) {
	m_consec_wrong_latency++;
	m_wrong_latency_total += this_time - index_time;
	int64_t test;
	test = m_wrong_latency_total / m_consec_wrong_latency;
	if (test > ALLOWED_LATENCY || test < -ALLOWED_LATENCY) {
	  if (m_consec_wrong_latency > 20) {
	    m_consec_wrong_latency = 0;
	    if (test < 0 && test + m_buffer_latency > 0) {
	      m_buffer_latency = 0;
	    } else {
	      m_buffer_latency += test; 
	    }
	    m_psptr->audio_is_ready(m_buffer_latency, this_time);
	    audio_message(LOG_INFO, "Latency off by " D64 " - now is " U64, 
				 test, m_buffer_latency);
	  }
	} else {
	  // average wrong latency is not greater 5 or less -5
	  m_consec_wrong_latency = 0;
	  m_wrong_latency_total = 0;
	}
      } else {
	m_consec_wrong_latency = 0;
	m_wrong_latency_total = 0;
      }
    }
  } else {
#ifdef DEBUG_SYNC
    audio_message(LOG_DEBUG, "playing "U64" "U64" latency "U64, 
		  this_time, m_play_time, m_buffer_latency);
#endif
  }

  // If we had the decoder task waiting, signal it.
  if (freed_buffer != 0 && m_audio_waiting_buffer) {
    m_audio_waiting_buffer = 0;
    SDL_SemPost(m_audio_waiting);
    //audio_message(LOG_DEBUG, "post freed");
  }
}

void CSDLAudioSync::play_audio (void)
{
  m_first_time = 1;
  //m_resync_required = 0;
  m_audio_paused = 0;
  m_play_sample_index = 0;
  Our_SDL_PauseAudio(0);
}

// Called from the sync thread when we want to stop.  Pause the audio,
// and indicate that we're not to fill any more buffers - this should let
// the decode thread get back to receive the pause message.  Only called
// when pausing - could cause m_dont_fill race conditions if called on play
void CSDLAudioSync::flush_sync_buffers (void)
{
  // we don't need to signal the decode task right now - 
  // Go ahead 
  clear_eof();
  Our_SDL_PauseAudio(1);
  m_dont_fill = 1;
  if (m_audio_waiting_buffer) {
    m_audio_waiting_buffer = 0;
    SDL_SemPost(m_audio_waiting);
    //audio_message(LOG_DEBUG, "post flush sync");
    
  }
  //  player_debug_message("Flushed sync");
}

// this is called from the decode thread.  It gets called on entry into pause,
// and entry into play.  This way, m_dont_fill race conditions are resolved.
void CSDLAudioSync::flush_decode_buffers (void)
{
  int locked = 0;
  if (m_audio_initialized != 0) {
    locked = 1;
    Our_SDL_LockAudio();
  }
  m_dont_fill = 0;
  m_first_filled = 1;
  for (int ix = 0; ix < DECODE_BUFFERS_MAX; ix++) {
    m_buffer_filled[ix] = 0;
  }
  m_buffer_offset_on = 0;
  m_play_index = m_fill_index = 0;
  m_audio_paused = 1;
  m_resync_buffer = 0;
  m_buffer_bytes_loaded = 0;
  if (locked)
    Our_SDL_UnlockAudio();
  //player_debug_message("flushed decode");
}

void CSDLAudioSync::set_volume (int volume)
{
  m_volume = (volume * SDL_MIX_MAXVOLUME)/100;
}


static void c_audio_config (void *ifptr, int freq, 
			    int chans, audio_format_t format, uint32_t max_samples)
{
  ((CSDLAudioSync *)ifptr)->set_config(freq,
				    chans,
				    format,
				    max_samples);
}

static uint8_t *c_get_audio_buffer (void *ifptr)
{
  return ((CSDLAudioSync *)ifptr)->get_audio_buffer();
}

static void c_filled_audio_buffer (void *ifptr,
				   uint32_t freq_ts, 
				   uint64_t ts,
				   int resync_req)
{
  ((CSDLAudioSync *)ifptr)->filled_audio_buffer(ts, 
					     resync_req);
}

static void c_load_audio_buffer (void *ifptr, 
				 const uint8_t *from, 
				 uint32_t bytes, 
				 uint32_t freq_ts,
				 uint64_t ts, 
				 int resync)
{
  ((CSDLAudioSync *)ifptr)->load_audio_buffer(from,
					      bytes,
					      ts, 
					      resync);
}
  
static audio_vft_t audio_vft = {
  message,
  c_audio_config,
  c_get_audio_buffer,
  c_filled_audio_buffer,
  c_load_audio_buffer,
  NULL,
};

CAudioSync *create_audio_sync (CPlayerSession *psptr, int volume)
{
  return new CSDLAudioSync(psptr, volume);
}

audio_vft_t *get_audio_vft (void)
{
  audio_vft.pConfig = &config;
  return &audio_vft;
}

int do_we_have_audio (void) 
{
  char buffer[80];
  if (Our_SDL_AudioInit(getenv("SDL_AUDIODRIVER")) < 0) {
    return (0);
  } 
  if (Our_SDL_AudioDriverName(buffer, sizeof(buffer)) == NULL) {
    return (0);
  }
  //  Our_SDL_CloseAudio();
  return (1);
}

/* end audio.cpp */