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Changes construction of AudioTrack for API 21 and higher.

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If we are on API level 21 or higher, it is possible to use a special AudioTrack
constructor that uses AudioAttributes and AudioFormat as input. It allows us to
supersede the notion of stream types for defining the behavior of audio playback,
and to allow certain platforms or routing policies to use this information for more
refined volume or routing decisions.

BUG=b/30018586

Review-Url: https://codereview.webrtc.org/2804013002
Cr-Commit-Position: refs/heads/master@{#17737}
This commit is contained in:
henrika 2017-04-18 05:07:52 -07:00 committed by Commit bot
parent d4ed42d7cf
commit 90ee13ce1c
1 changed files with 46 additions and 2 deletions
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48
webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioTrack.java
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@ -12,6 +12,7 @@ package org.webrtc.voiceengine;
import android.annotation.TargetApi;
import android.content.Context;
import android.media.AudioAttributes;
import android.media.AudioFormat;
import android.media.AudioManager;
import android.media.AudioTrack;
@ -199,8 +200,20 @@ public class WebRtcAudioTrack {
// 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, channelConfig,
AudioFormat.ENCODING_PCM_16BIT, minBufferSizeInBytes, AudioTrack.MODE_STREAM);
if (WebRtcAudioUtils.runningOnLollipopOrHigher()) {
// If we are on API level 21 or higher, it is possible to use a special AudioTrack
// constructor that uses AudioAttributes and AudioFormat as input. It allows us to
// supersede the notion of stream types for defining the behavior of audio playback,
// and to allow certain platforms or routing policies to use this information for more
// refined volume or routing decisions.
audioTrack = createAudioTrackOnLollipopOrHigher(
sampleRate, channelConfig, minBufferSizeInBytes);
} else {
// Use default constructor for API levels below 21.
// Note that, this constructor will be deprecated in API level O (25).
audioTrack = new AudioTrack(AudioManager.STREAM_VOICE_CALL, sampleRate, channelConfig,
AudioFormat.ENCODING_PCM_16BIT, minBufferSizeInBytes, AudioTrack.MODE_STREAM);
}
} catch (IllegalArgumentException e) {
Logging.d(TAG, e.getMessage());
releaseAudioResources();
@ -284,6 +297,37 @@ public class WebRtcAudioTrack {
+ "max gain: " + audioTrack.getMaxVolume());
}
// Creates and AudioTrack instance using AudioAttributes and AudioFormat as input.
// It allows certain platforms or routing policies to use this information for more
// refined volume or routing decisions.
@TargetApi(21)
private AudioTrack createAudioTrackOnLollipopOrHigher(
int sampleRateInHz, int channelConfig, int bufferSizeInBytes) {
Logging.d(TAG, "createAudioTrackOnLollipopOrHigher");
// TODO(henrika): use setPerformanceMode(int) with PERFORMANCE_MODE_LOW_LATENCY to control
// performance when Android O is supported. Add some logging in the mean time.
final int nativeOutputSampleRate =
AudioTrack.getNativeOutputSampleRate(AudioManager.STREAM_VOICE_CALL);
Logging.d(TAG, "nativeOutputSampleRate: " + nativeOutputSampleRate);
if (sampleRateInHz != nativeOutputSampleRate) {
Logging.w(TAG, "Unable to use fast mode since requested sample rate is not native");
}
// Create an audio track where the audio usage is for VoIP and the content type is speech.
return new AudioTrack(
new AudioAttributes.Builder()
.setUsage(AudioAttributes.USAGE_VOICE_COMMUNICATION)
.setContentType(AudioAttributes.CONTENT_TYPE_SPEECH)
.build(),
new AudioFormat.Builder()
.setEncoding(AudioFormat.ENCODING_PCM_16BIT)
.setSampleRate(sampleRateInHz)
.setChannelMask(channelConfig)
.build(),
bufferSizeInBytes,
AudioTrack.MODE_STREAM,
AudioManager.AUDIO_SESSION_ID_GENERATE);
}
@TargetApi(24)
private void logMainParametersExtended() {
if (WebRtcAudioUtils.runningOnMarshmallowOrHigher()) {