Improves stereo/mono audio support on Android

webrtc/modules/audio_device/android/audio_manager_unittest.cc

 /*
  *  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 118 matching lines @@
 TEST_F(AudioManagerTest, IsLowLatencyRecordSupported) {
   PRINT("%sLow latency input support: %s\n", kTag,
         audio_manager()->IsLowLatencyRecordSupported() ? "Yes" : "No");
 }
 
 TEST_F(AudioManagerTest, IsProAudioSupported) {
   PRINT("%sPro audio support: %s\n", kTag,
         audio_manager()->IsProAudioSupported() ? "Yes" : "No");
 }
 
+// Verify that playout side is configured for mono by default.
+TEST_F(AudioManagerTest, IsStereoPlayoutSupported) {
+  EXPECT_FALSE(audio_manager()->IsStereoPlayoutSupported());
+}
+
+// Verify that recording side is configured for mono by default.
+TEST_F(AudioManagerTest, IsStereoRecordSupported) {
+  EXPECT_FALSE(audio_manager()->IsStereoRecordSupported());
+}
+
 TEST_F(AudioManagerTest, ShowAudioParameterInfo) {
   const bool low_latency_out = audio_manager()->IsLowLatencyPlayoutSupported();
   const bool low_latency_in = audio_manager()->IsLowLatencyRecordSupported();
   PRINT("PLAYOUT:\n");
   PRINT("%saudio layer: %s\n", kTag,
         low_latency_out ? "Low latency OpenSL" : "Java/JNI based AudioTrack");
   PRINT("%ssample rate: %d Hz\n", kTag, playout_parameters_.sample_rate());
   PRINT("%schannels: %" PRIuS "\n", kTag, playout_parameters_.channels());
   PRINT("%sframes per buffer: %" PRIuS " <=> %.2f ms\n", kTag,
         playout_parameters_.frames_per_buffer(),
@@ skipping 73 matching lines @@
             params.frames_per_10ms_buffer());
   EXPECT_EQ(kBytesPerFrame, params.GetBytesPerFrame());
   EXPECT_EQ(kBytesPerFrame * kFramesPerBuffer, params.GetBytesPerBuffer());
   EXPECT_EQ(kBytesPerFrame * kFramesPer10msBuffer,
             params.GetBytesPer10msBuffer());
   EXPECT_EQ(kBufferSizeInMs, params.GetBufferSizeInMilliseconds());
 }
 
 }  // namespace webrtc