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Remove Legacy ADM from AppRTC mobile

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The Legacy ADM has remained to be available for testing. Now we are
ready to move on to using only the Java ADM.

Bug: webrtc:7452
Change-Id: Ic95b04b933e165f3c16b587a44384a2c965ef16c
Reviewed-on: https://webrtc-review.googlesource.com/c/123921
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26852}
This commit is contained in:
Paulina Hensman
2019-02-22 16:31:28 +01:00
committed by Commit Bot
parent 0bf4c29852
commit 22dab11270
9 changed files with 7 additions and 127 deletions
Download Patch File Download Diff File Expand all files Collapse all files

90
examples/androidapp/src/org/appspot/apprtc/PeerConnectionClient.java
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@ -75,14 +75,6 @@ import org.webrtc.audio.AudioDeviceModule;
import org.webrtc.audio.JavaAudioDeviceModule;
import org.webrtc.audio.JavaAudioDeviceModule.AudioRecordErrorCallback;
import org.webrtc.audio.JavaAudioDeviceModule.AudioTrackErrorCallback;
import org.webrtc.audio.LegacyAudioDeviceModule;
import org.webrtc.voiceengine.WebRtcAudioManager;
import org.webrtc.voiceengine.WebRtcAudioRecord;
import org.webrtc.voiceengine.WebRtcAudioRecord.AudioRecordStartErrorCode;
import org.webrtc.voiceengine.WebRtcAudioRecord.WebRtcAudioRecordErrorCallback;
import org.webrtc.voiceengine.WebRtcAudioTrack;
import org.webrtc.voiceengine.WebRtcAudioTrack.AudioTrackStartErrorCode;
import org.webrtc.voiceengine.WebRtcAudioUtils;
/**
* Peer connection client implementation.
@ -232,7 +224,6 @@ public class PeerConnectionClient {
public final boolean disableBuiltInNS;
public final boolean disableWebRtcAGCAndHPF;
public final boolean enableRtcEventLog;
public final boolean useLegacyAudioDevice;
private final DataChannelParameters dataChannelParameters;
public PeerConnectionParameters(boolean videoCallEnabled, boolean loopback, boolean tracing,
@ -241,7 +232,7 @@ public class PeerConnectionClient {
String audioCodec, boolean noAudioProcessing, boolean aecDump, boolean saveInputAudioToFile,
boolean useOpenSLES, boolean disableBuiltInAEC, boolean disableBuiltInAGC,
boolean disableBuiltInNS, boolean disableWebRtcAGCAndHPF, boolean enableRtcEventLog,
boolean useLegacyAudioDevice, DataChannelParameters dataChannelParameters) {
DataChannelParameters dataChannelParameters) {
this.videoCallEnabled = videoCallEnabled;
this.loopback = loopback;
this.tracing = tracing;
@ -263,7 +254,6 @@ public class PeerConnectionClient {
this.disableBuiltInNS = disableBuiltInNS;
this.disableWebRtcAGCAndHPF = disableWebRtcAGCAndHPF;
this.enableRtcEventLog = enableRtcEventLog;
this.useLegacyAudioDevice = useLegacyAudioDevice;
this.dataChannelParameters = dataChannelParameters;
}
}
@ -430,9 +420,7 @@ public class PeerConnectionClient {
}
}
final AudioDeviceModule adm = peerConnectionParameters.useLegacyAudioDevice
? createLegacyAudioDevice()
: createJavaAudioDevice();
final AudioDeviceModule adm = createJavaAudioDevice();
// Create peer connection factory.
if (options != null) {
@ -462,80 +450,6 @@ public class PeerConnectionClient {
adm.release();
}
AudioDeviceModule createLegacyAudioDevice() {
// Enable/disable OpenSL ES playback.
if (!peerConnectionParameters.useOpenSLES) {
Log.d(TAG, "Disable OpenSL ES audio even if device supports it");
WebRtcAudioManager.setBlacklistDeviceForOpenSLESUsage(true /* enable */);
} else {
Log.d(TAG, "Allow OpenSL ES audio if device supports it");
WebRtcAudioManager.setBlacklistDeviceForOpenSLESUsage(false);
}
if (peerConnectionParameters.disableBuiltInAEC) {
Log.d(TAG, "Disable built-in AEC even if device supports it");
WebRtcAudioUtils.setWebRtcBasedAcousticEchoCanceler(true);
} else {
Log.d(TAG, "Enable built-in AEC if device supports it");
WebRtcAudioUtils.setWebRtcBasedAcousticEchoCanceler(false);
}
if (peerConnectionParameters.disableBuiltInNS) {
Log.d(TAG, "Disable built-in NS even if device supports it");
WebRtcAudioUtils.setWebRtcBasedNoiseSuppressor(true);
} else {
Log.d(TAG, "Enable built-in NS if device supports it");
WebRtcAudioUtils.setWebRtcBasedNoiseSuppressor(false);
}
WebRtcAudioRecord.setOnAudioSamplesReady(saveRecordedAudioToFile);
// Set audio record error callbacks.
WebRtcAudioRecord.setErrorCallback(new WebRtcAudioRecordErrorCallback() {
@Override
public void onWebRtcAudioRecordInitError(String errorMessage) {
Log.e(TAG, "onWebRtcAudioRecordInitError: " + errorMessage);
reportError(errorMessage);
}
@Override
public void onWebRtcAudioRecordStartError(
AudioRecordStartErrorCode errorCode, String errorMessage) {
Log.e(TAG, "onWebRtcAudioRecordStartError: " + errorCode + ". " + errorMessage);
reportError(errorMessage);
}
@Override
public void onWebRtcAudioRecordError(String errorMessage) {
Log.e(TAG, "onWebRtcAudioRecordError: " + errorMessage);
reportError(errorMessage);
}
});
WebRtcAudioTrack.setErrorCallback(new WebRtcAudioTrack.ErrorCallback() {
@Override
public void onWebRtcAudioTrackInitError(String errorMessage) {
Log.e(TAG, "onWebRtcAudioTrackInitError: " + errorMessage);
reportError(errorMessage);
}
@Override
public void onWebRtcAudioTrackStartError(
AudioTrackStartErrorCode errorCode, String errorMessage) {
Log.e(TAG, "onWebRtcAudioTrackStartError: " + errorCode + ". " + errorMessage);
reportError(errorMessage);
}
@Override
public void onWebRtcAudioTrackError(String errorMessage) {
Log.e(TAG, "onWebRtcAudioTrackError: " + errorMessage);
reportError(errorMessage);
}
});
return new LegacyAudioDeviceModule();
}
AudioDeviceModule createJavaAudioDevice() {
// Enable/disable OpenSL ES playback.
if (!peerConnectionParameters.useOpenSLES) {