android-10.0.0_r47/s

/*
 *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
 *
 *  Use of this source code is governed by a BSD-style license
 *  that can be found in the LICENSE file in the root of the source
 *  tree. An additional intellectual property rights grant can be found
 *  in the file PATENTS.  All contributing project authors may
 *  be found in the AUTHORS file in the root of the source tree.
 */

package org.webrtc.voiceengine;

import java.lang.Thread;
import java.nio.ByteBuffer;

import android.annotation.TargetApi;
import android.content.Context;
import android.media.AudioFormat;
import android.media.AudioManager;
import android.media.AudioTrack;
import android.os.Process;

import org.webrtc.Logging;

class WebRtcAudioTrack {
  private static final boolean DEBUG = false;

  private static final String TAG = "WebRtcAudioTrack";

  // Default audio data format is PCM 16 bit per sample.
  // Guaranteed to be supported by all devices.
  private static final int BITS_PER_SAMPLE = 16;

  // Requested size of each recorded buffer provided to the client.
  private static final int CALLBACK_BUFFER_SIZE_MS = 10;

  // Average number of callbacks per second.
  private static final int BUFFERS_PER_SECOND = 1000 / CALLBACK_BUFFER_SIZE_MS;

  private final Context context;
  private final long nativeAudioTrack;
  private final AudioManager audioManager;

  private ByteBuffer byteBuffer;

  private AudioTrack audioTrack = null;
  private AudioTrackThread audioThread = null;

  /**
   * Audio thread which keeps calling AudioTrack.write() to stream audio.
   * Data is periodically acquired from the native WebRTC layer using the
   * nativeGetPlayoutData callback function.
   * This thread uses a Process.THREAD_PRIORITY_URGENT_AUDIO priority.
   */
  private class AudioTrackThread extends Thread {
    private volatile boolean keepAlive = true;

    public AudioTrackThread(String name) {
      super(name);
    }

    @Override
    public void run() {
      Process.setThreadPriority(Process.THREAD_PRIORITY_URGENT_AUDIO);
      Logging.d(TAG, "AudioTrackThread" + WebRtcAudioUtils.getThreadInfo());

      try {
        // In MODE_STREAM mode we can optionally prime the output buffer by
        // writing up to bufferSizeInBytes (from constructor) before starting.
        // This priming will avoid an immediate underrun, but is not required.
        // TODO(henrika): initial tests have shown that priming is not required.
        audioTrack.play();
        assertTrue(audioTrack.getPlayState() == AudioTrack.PLAYSTATE_PLAYING);
      } catch (IllegalStateException e) {
          Logging.e(TAG, "AudioTrack.play failed: " + e.getMessage());
        return;
      }

      // Fixed size in bytes of each 10ms block of audio data that we ask for
      // using callbacks to the native WebRTC client.
      final int sizeInBytes = byteBuffer.capacity();

      while (keepAlive) {
        // Get 10ms of PCM data from the native WebRTC client. Audio data is
        // written into the common ByteBuffer using the address that was
        // cached at construction.
        nativeGetPlayoutData(sizeInBytes, nativeAudioTrack);
        // Write data until all data has been written to the audio sink.
        // Upon return, the buffer position will have been advanced to reflect
        // the amount of data that was successfully written to the AudioTrack.
        assertTrue(sizeInBytes <= byteBuffer.remaining());
        int bytesWritten = 0;
        if (WebRtcAudioUtils.runningOnLollipopOrHigher()) {
          bytesWritten = writeOnLollipop(audioTrack, byteBuffer, sizeInBytes);
        } else {
          bytesWritten = writePreLollipop(audioTrack, byteBuffer, sizeInBytes);
        }
        if (bytesWritten != sizeInBytes) {
          Logging.e(TAG, "AudioTrack.write failed: " + bytesWritten);
          if (bytesWritten == AudioTrack.ERROR_INVALID_OPERATION) {
            keepAlive = false;
          }
        }
        // The byte buffer must be rewinded since byteBuffer.position() is
        // increased at each call to AudioTrack.write(). If we don't do this,
        // next call to AudioTrack.write() will fail.
        byteBuffer.rewind();

        // TODO(henrika): it is possible to create a delay estimate here by
        // counting number of written frames and subtracting the result from
        // audioTrack.getPlaybackHeadPosition().
      }

      try {
        audioTrack.stop();
      } catch (IllegalStateException e) {
        Logging.e(TAG, "AudioTrack.stop failed: " + e.getMessage());
      }
      assertTrue(audioTrack.getPlayState() == AudioTrack.PLAYSTATE_STOPPED);
      audioTrack.flush();
    }

    @TargetApi(21)
    private int writeOnLollipop(AudioTrack audioTrack, ByteBuffer byteBuffer, int sizeInBytes) {
      return audioTrack.write(byteBuffer, sizeInBytes, AudioTrack.WRITE_BLOCKING);
    }

    private int writePreLollipop(AudioTrack audioTrack, ByteBuffer byteBuffer, int sizeInBytes) {
      return audioTrack.write(byteBuffer.array(), byteBuffer.arrayOffset(), sizeInBytes);
    }

    public void joinThread() {
      keepAlive = false;
      while (isAlive()) {
        try {
          join();
        } catch (InterruptedException e) {
          // Ignore.
        }
      }
    }
  }

  WebRtcAudioTrack(Context context, long nativeAudioTrack) {
    Logging.d(TAG, "ctor" + WebRtcAudioUtils.getThreadInfo());
    this.context = context;
    this.nativeAudioTrack = nativeAudioTrack;
    audioManager = (AudioManager) context.getSystemService(
        Context.AUDIO_SERVICE);
    if (DEBUG) {
      WebRtcAudioUtils.logDeviceInfo(TAG);
    }
  }

  private void initPlayout(int sampleRate, int channels) {
    Logging.d(TAG, "initPlayout(sampleRate=" + sampleRate + ", channels="
        + channels + ")");
    final int bytesPerFrame = channels * (BITS_PER_SAMPLE / 8);
    byteBuffer = byteBuffer.allocateDirect(
        bytesPerFrame * (sampleRate / BUFFERS_PER_SECOND));
    Logging.d(TAG, "byteBuffer.capacity: " + byteBuffer.capacity());
    // Rather than passing the ByteBuffer with every callback (requiring
    // the potentially expensive GetDirectBufferAddress) we simply have the
    // the native class cache the address to the memory once.
    nativeCacheDirectBufferAddress(byteBuffer, nativeAudioTrack);

    // Get the minimum buffer size required for the successful creation of an
    // AudioTrack object to be created in the MODE_STREAM mode.
    // Note that this size doesn't guarantee a smooth playback under load.
    // TODO(henrika): should we extend the buffer size to avoid glitches?
    final int minBufferSizeInBytes = AudioTrack.getMinBufferSize(
        sampleRate,
        AudioFormat.CHANNEL_OUT_MONO,
        AudioFormat.ENCODING_PCM_16BIT);
    Logging.d(TAG, "AudioTrack.getMinBufferSize: " + minBufferSizeInBytes);
    assertTrue(audioTrack == null);

    // For the streaming mode, data must be written to the audio sink in
    // chunks of size (given by byteBuffer.capacity()) less than or equal
    // to the total buffer size |minBufferSizeInBytes|.
    assertTrue(byteBuffer.capacity() < minBufferSizeInBytes);
    try {
      // Create an AudioTrack object and initialize its associated audio buffer.
      // The size of this buffer determines how long an AudioTrack can play
      // before running out of data.
      audioTrack = new AudioTrack(AudioManager.STREAM_VOICE_CALL,
                                  sampleRate,
                                  AudioFormat.CHANNEL_OUT_MONO,
                                  AudioFormat.ENCODING_PCM_16BIT,
                                  minBufferSizeInBytes,
                                  AudioTrack.MODE_STREAM);
    } catch (IllegalArgumentException e) {
      Logging.d(TAG, e.getMessage());
      return;
    }
    assertTrue(audioTrack.getState() == AudioTrack.STATE_INITIALIZED);
    assertTrue(audioTrack.getPlayState() == AudioTrack.PLAYSTATE_STOPPED);
    assertTrue(audioTrack.getStreamType() == AudioManager.STREAM_VOICE_CALL);
  }

  private boolean startPlayout() {
    Logging.d(TAG, "startPlayout");
    assertTrue(audioTrack != null);
    assertTrue(audioThread == null);
    audioThread = new AudioTrackThread("AudioTrackJavaThread");
    audioThread.start();
    return true;
  }

  private boolean stopPlayout() {
    Logging.d(TAG, "stopPlayout");
    assertTrue(audioThread != null);
    audioThread.joinThread();
    audioThread = null;
    if (audioTrack != null) {
      audioTrack.release();
      audioTrack = null;
    }
    return true;
  }

  /** Get max possible volume index for a phone call audio stream. */
  private int getStreamMaxVolume() {
    Logging.d(TAG, "getStreamMaxVolume");
    assertTrue(audioManager != null);
    return audioManager.getStreamMaxVolume(AudioManager.STREAM_VOICE_CALL);
  }

  /** Set current volume level for a phone call audio stream. */
  private boolean setStreamVolume(int volume) {
    Logging.d(TAG, "setStreamVolume(" + volume + ")");
    assertTrue(audioManager != null);
    if (isVolumeFixed()) {
      Logging.e(TAG, "The device implements a fixed volume policy.");
      return false;
    }
    audioManager.setStreamVolume(AudioManager.STREAM_VOICE_CALL, volume, 0);
    return true;
  }

  @TargetApi(21)
  private boolean isVolumeFixed() {
    if (!WebRtcAudioUtils.runningOnLollipopOrHigher())
      return false;
    return audioManager.isVolumeFixed();
  }

  /** Get current volume level for a phone call audio stream. */
  private int getStreamVolume() {
    Logging.d(TAG, "getStreamVolume");
    assertTrue(audioManager != null);
    return audioManager.getStreamVolume(AudioManager.STREAM_VOICE_CALL);
  }

  /** Helper method which throws an exception  when an assertion has failed. */
  private static void assertTrue(boolean condition) {
    if (!condition) {
      throw new AssertionError("Expected condition to be true");
    }
  }

  private native void nativeCacheDirectBufferAddress(
      ByteBuffer byteBuffer, long nativeAudioRecord);

  private native void nativeGetPlayoutData(int bytes, long nativeAudioRecord);
}