1 files changed, 122 insertions, 0 deletions
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioManager.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioManager.java
new file mode 100644
index 0000000000..f398602a28
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioManager.java
@@ -0,0 +1,122 @@
+/*
+ *  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.audio;
+
+import android.content.Context;
+import android.content.pm.PackageManager;
+import android.media.AudioFormat;
+import android.media.AudioManager;
+import android.media.AudioRecord;
+import android.media.AudioTrack;
+import android.os.Build;
+import org.webrtc.Logging;
+import org.webrtc.CalledByNative;
+
+/**
+ * This class contains static functions to query sample rate and input/output audio buffer sizes.
+ */
+class WebRtcAudioManager {
+  private static final String TAG = "WebRtcAudioManagerExternal";
+
+  private static final int DEFAULT_SAMPLE_RATE_HZ = 16000;
+
+  // 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;
+
+  private static final int DEFAULT_FRAME_PER_BUFFER = 256;
+
+  @CalledByNative
+  static AudioManager getAudioManager(Context context) {
+    return (AudioManager) context.getSystemService(Context.AUDIO_SERVICE);
+  }
+
+  @CalledByNative
+  static int getOutputBufferSize(
+      Context context, AudioManager audioManager, int sampleRate, int numberOfOutputChannels) {
+    return isLowLatencyOutputSupported(context)
+        ? getLowLatencyFramesPerBuffer(audioManager)
+        : getMinOutputFrameSize(sampleRate, numberOfOutputChannels);
+  }
+
+  @CalledByNative
+  static int getInputBufferSize(
+      Context context, AudioManager audioManager, int sampleRate, int numberOfInputChannels) {
+    return isLowLatencyInputSupported(context)
+        ? getLowLatencyFramesPerBuffer(audioManager)
+        : getMinInputFrameSize(sampleRate, numberOfInputChannels);
+  }
+
+  private static boolean isLowLatencyOutputSupported(Context context) {
+    return context.getPackageManager().hasSystemFeature(PackageManager.FEATURE_AUDIO_LOW_LATENCY);
+  }
+
+  private static boolean isLowLatencyInputSupported(Context context) {
+    // TODO(henrika): investigate if some sort of device list is needed here
+    // as well. The NDK doc states that: "As of API level 21, lower latency
+    // audio input is supported on select devices. To take advantage of this
+    // feature, first confirm that lower latency output is available".
+    return isLowLatencyOutputSupported(context);
+  }
+
+  /**
+   * Returns the native input/output sample rate for this device's output stream.
+   */
+  @CalledByNative
+  static int getSampleRate(AudioManager audioManager) {
+    // Override this if we're running on an old emulator image which only
+    // supports 8 kHz and doesn't support PROPERTY_OUTPUT_SAMPLE_RATE.
+    if (WebRtcAudioUtils.runningOnEmulator()) {
+      Logging.d(TAG, "Running emulator, overriding sample rate to 8 kHz.");
+      return 8000;
+    }
+    // Deliver best possible estimate based on default Android AudioManager APIs.
+    final int sampleRateHz = getSampleRateForApiLevel(audioManager);
+    Logging.d(TAG, "Sample rate is set to " + sampleRateHz + " Hz");
+    return sampleRateHz;
+  }
+
+  private static int getSampleRateForApiLevel(AudioManager audioManager) {
+    String sampleRateString = audioManager.getProperty(AudioManager.PROPERTY_OUTPUT_SAMPLE_RATE);
+    return (sampleRateString == null) ? DEFAULT_SAMPLE_RATE_HZ : Integer.parseInt(sampleRateString);
+  }
+
+  // Returns the native output buffer size for low-latency output streams.
+  private static int getLowLatencyFramesPerBuffer(AudioManager audioManager) {
+    String framesPerBuffer =
+        audioManager.getProperty(AudioManager.PROPERTY_OUTPUT_FRAMES_PER_BUFFER);
+    return framesPerBuffer == null ? DEFAULT_FRAME_PER_BUFFER : Integer.parseInt(framesPerBuffer);
+  }
+
+  // Returns the minimum output buffer size for Java based audio (AudioTrack).
+  // This size can also be used for OpenSL ES implementations on devices that
+  // lacks support of low-latency output.
+  private static int getMinOutputFrameSize(int sampleRateInHz, int numChannels) {
+    final int bytesPerFrame = numChannels * (BITS_PER_SAMPLE / 8);
+    final int channelConfig =
+        (numChannels == 1 ? AudioFormat.CHANNEL_OUT_MONO : AudioFormat.CHANNEL_OUT_STEREO);
+    return AudioTrack.getMinBufferSize(
+               sampleRateInHz, channelConfig, AudioFormat.ENCODING_PCM_16BIT)
+        / bytesPerFrame;
+  }
+
+  // Returns the minimum input buffer size for Java based audio (AudioRecord).
+  // This size can calso be used for OpenSL ES implementations on devices that
+  // lacks support of low-latency input.
+  private static int getMinInputFrameSize(int sampleRateInHz, int numChannels) {
+    final int bytesPerFrame = numChannels * (BITS_PER_SAMPLE / 8);
+    final int channelConfig =
+        (numChannels == 1 ? AudioFormat.CHANNEL_IN_MONO : AudioFormat.CHANNEL_IN_STEREO);
+    return AudioRecord.getMinBufferSize(
+               sampleRateInHz, channelConfig, AudioFormat.ENCODING_PCM_16BIT)
+        / bytesPerFrame;
+  }
+}