Reduces default sample rate from 44.1kHz to 16kHz to ensure
that we can open up audio in communication mode also on older
devices that only supports it in combination with 16kHz.
BUG=webrtc:4756
R=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1347243003 .
Cr-Commit-Position: refs/heads/master@{#9971}
diff --git a/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java b/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java
index 81a2bdd..2bd8400 100644
--- a/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java
+++ b/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java
@@ -42,8 +42,11 @@
// 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;
@@ -56,8 +59,6 @@
"MODE_IN_COMMUNICATION",
};
- private static final int DEFAULT_FRAME_PER_BUFFER = 256;
-
private final long nativeAudioManager;
private final Context context;
private final AudioManager audioManager;
@@ -166,12 +167,12 @@
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.
