예제 #1
0
 public void dispose() {
   if (buffers == null) return;
   if (sourceID != -1) {
     audio.freeSource(sourceID);
     sourceID = -1;
   }
   audio.getAL().alDeleteBuffers(buffers.limit(), buffers);
   buffers = null;
 }
예제 #2
0
  public void writeSamples(byte[] data, int offset, int length) {
    if (length < 0) throw new IllegalArgumentException("length cannot be < 0.");

    if (sourceID == -1) {
      sourceID = audio.obtainSource(true);
      if (sourceID == -1) return;
      if (buffers == null) {
        buffers = Buffers.newDirectIntBuffer(bufferCount);
        audio.getAL().alGenBuffers(buffers.limit(), buffers);
        if (audio.getAL().alGetError() != ALConstants.AL_NO_ERROR)
          throw new GdxRuntimeException("Unabe to allocate audio buffers.");
      }
      audio.getAL().alSourcei(sourceID, ALConstants.AL_LOOPING, ALConstants.AL_FALSE);
      audio.getAL().alSourcef(sourceID, ALConstants.AL_GAIN, volume);
      // Fill initial buffers.
      int queuedBuffers = 0;
      for (int i = 0; i < bufferCount; i++) {
        int bufferID = buffers.get(i);
        int written = Math.min(bufferSize, length);
        tempBuffer.clear();
        tempBuffer.put(data, offset, written).flip();
        audio
            .getAL()
            .alBufferData(bufferID, format, tempBuffer, tempBuffer.remaining(), sampleRate);
        ib.put(0, bufferID).rewind();
        audio.getAL().alSourceQueueBuffers(sourceID, ib.limit(), ib);
        length -= written;
        offset += written;
        queuedBuffers++;
      }
      // Queue rest of buffers, empty.
      tempBuffer.clear().flip();
      for (int i = queuedBuffers; i < bufferCount; i++) {
        int bufferID = buffers.get(i);
        audio
            .getAL()
            .alBufferData(bufferID, format, tempBuffer, tempBuffer.remaining(), sampleRate);
        audio.getAL().alSourceQueueBuffers(sourceID, ib.limit(), ib);
      }
      audio.getAL().alSourcePlay(sourceID);
      isPlaying = true;
    }

    while (length > 0) {
      int written = fillBuffer(data, offset, length);
      length -= written;
      offset += written;
    }
  }
예제 #3
0
  /** Blocks until some of the data could be buffered. */
  private int fillBuffer(byte[] data, int offset, int length) {
    int written = Math.min(bufferSize, length);

    outer:
    while (true) {
      audio.getAL().alGetSourcei(sourceID, ALConstants.AL_BUFFERS_PROCESSED, ib);
      int buffers = ib.get(0);
      while (buffers-- > 0) {
        // FIXME
        ib.put(0, buffers).rewind();
        audio.getAL().alSourceUnqueueBuffers(sourceID, ib.limit(), ib);
        int bufferID = ib.get(0);
        if (bufferID == ALConstants.AL_INVALID_VALUE) break;
        renderedSeconds += secondsPerBuffer;

        tempBuffer.clear();
        tempBuffer.put(data, offset, written).flip();
        audio
            .getAL()
            .alBufferData(bufferID, format, tempBuffer, tempBuffer.remaining(), sampleRate);
        ib.put(0, bufferID).rewind();
        audio.getAL().alSourceQueueBuffers(sourceID, ib.limit(), ib);
        break outer;
      }
      // Wait for buffer to be free.
      try {
        Thread.sleep((long) (1000 * secondsPerBuffer / bufferCount));
      } catch (InterruptedException ignored) {
      }
    }

    // A buffer underflow will cause the source to stop.
    if (!isPlaying || !isCurrentSourcePlaying()) {
      audio.getAL().alSourcePlay(sourceID);
      isPlaying = true;
    }

    return written;
  }
예제 #4
0
 public float getPosition() {
   if (sourceID == -1) return 0;
   audio.getAL().alGetSourcef(sourceID, ALConstants.AL_SEC_OFFSET, fb);
   return renderedSeconds + fb.get(0);
 }
예제 #5
0
 public void setVolume(float volume) {
   this.volume = volume;
   if (sourceID != -1) audio.getAL().alSourcef(sourceID, ALConstants.AL_GAIN, volume);
 }
예제 #6
0
 private boolean isCurrentSourcePlaying() {
   audio.getAL().alGetSourcei(sourceID, ALConstants.AL_SOURCE_STATE, ib);
   return (ib.get(0) == ALConstants.AL_PLAYING);
 }