Reduces default sample rate from 44.1kHz to 16kHz

webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java

 /*
  *  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.
  */
 
@@ skipping 24 matching lines @@
 // STREAM_VOICE_CALL-type stream.
 class WebRtcAudioManager {
   private static final boolean DEBUG = false;
 
   private static final String TAG = "WebRtcAudioManager";
 
   // 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;
 
-   // Use 44.1kHz as the default sampling rate.
-  private static final int SAMPLE_RATE_HZ = 44100;
+  // Use 16kHz as the default sample rate. A higher sample rate might prevent
+  // us from supporting communication mode on some older (e.g. ICS) devices.
+  private static final int DEFAULT_SAMPLE_RATE_HZ = 16000;
+
+  private static final int DEFAULT_FRAME_PER_BUFFER = 256;
 
   // TODO(henrika): add stereo support for playout.
   private static final int CHANNELS = 1;
 
   // List of possible audio modes.
   private static final String[] AUDIO_MODES = new String[] {
       "MODE_NORMAL",
       "MODE_RINGTONE",
       "MODE_IN_CALL",
       "MODE_IN_COMMUNICATION",
   };
 
-  private static final int DEFAULT_FRAME_PER_BUFFER = 256;
-
   private final long nativeAudioManager;
   private final Context context;
   private final AudioManager audioManager;
 
   private boolean initialized = false;
   private int nativeSampleRate;
   private int nativeChannels;
 
   private boolean hardwareAEC;
   private boolean lowLatencyOutput;
@@ skipping 88 matching lines @@
 
   // Returns the native output sample rate for this device's output stream.
   private int getNativeOutputSampleRate() {
     // 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()) {
       Logd("Running on old emulator, overriding sampling rate to 8 kHz.");
       return 8000;
     }
     if (!WebRtcAudioUtils.runningOnJellyBeanMR1OrHigher()) {
-      return SAMPLE_RATE_HZ;
+      return DEFAULT_SAMPLE_RATE_HZ;
     }
     String sampleRateString = audioManager.getProperty(
         AudioManager.PROPERTY_OUTPUT_SAMPLE_RATE);
     return (sampleRateString == null) ?
-        SAMPLE_RATE_HZ : Integer.parseInt(sampleRateString);
+        DEFAULT_SAMPLE_RATE_HZ : Integer.parseInt(sampleRateString);
   }
 
   // Returns the native output buffer size for low-latency output streams.
   private int getLowLatencyOutputFramesPerBuffer() {
     assertTrue(isLowLatencyOutputSupported());
     if (!WebRtcAudioUtils.runningOnJellyBeanMR1OrHigher()) {
       return DEFAULT_FRAME_PER_BUFFER;
     }
     String framesPerBuffer = audioManager.getProperty(
         AudioManager.PROPERTY_OUTPUT_FRAMES_PER_BUFFER);
@@ skipping 65 matching lines @@
 
   private static void Loge(String msg) {
     Logging.e(TAG, msg);
   }
 
   private native void nativeCacheAudioParameters(
     int sampleRate, int channels, boolean hardwareAEC, boolean lowLatencyOutput,
     int outputBufferSize, int inputBufferSize,
     long nativeAudioManager);
 }