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Fix some lint warnings on Android.

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Bug: webrtc:6597
Change-Id: I215b319b3129949d12b74b11e6ec35eb123a1e5a
Reviewed-on: https://webrtc-review.googlesource.com/6781
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20182}
This commit is contained in:
Sami Kalliomäki
2017-10-05 15:14:06 +02:00
committed by Commit Bot
parent fb26f85b79
commit 602dc431b0
4 changed files with 43 additions and 30 deletions
Download Patch File Download Diff File Expand all files Collapse all files

26
examples/androidapp/src/org/appspot/apprtc/CallActivity.java
View File

@ -139,9 +139,8 @@ public class CallActivity extends Activity implements AppRTCClient.SignalingEven
// Peer connection statistics callback period in ms.
private static final int STAT_CALLBACK_PERIOD = 1000;
private class ProxyRenderer<T extends VideoRenderer.Callbacks & VideoSink>
implements VideoRenderer.Callbacks, VideoSink {
private T target;
private static class ProxyRenderer implements VideoRenderer.Callbacks {
private VideoRenderer.Callbacks target;
@Override
synchronized public void renderFrame(VideoRenderer.I420Frame frame) {
@ -154,6 +153,14 @@ public class CallActivity extends Activity implements AppRTCClient.SignalingEven
target.renderFrame(frame);
}
synchronized public void setTarget(VideoRenderer.Callbacks target) {
this.target = target;
}
}
private static class ProxyVideoSink implements VideoSink {
private VideoSink target;
@Override
synchronized public void onFrame(VideoFrame frame) {
if (target == null) {
@ -164,13 +171,13 @@ public class CallActivity extends Activity implements AppRTCClient.SignalingEven
target.onFrame(frame);
}
synchronized public void setTarget(T target) {
synchronized public void setTarget(VideoSink target) {
this.target = target;
}
}
private final ProxyRenderer remoteProxyRenderer = new ProxyRenderer();
private final ProxyRenderer localProxyRenderer = new ProxyRenderer();
private final ProxyVideoSink localProxyVideoSink = new ProxyVideoSink();
private PeerConnectionClient peerConnectionClient = null;
private AppRTCClient appRtcClient;
private SignalingParameters signalingParameters;
@ -211,8 +218,7 @@ public class CallActivity extends Activity implements AppRTCClient.SignalingEven
// adding content.
requestWindowFeature(Window.FEATURE_NO_TITLE);
getWindow().addFlags(LayoutParams.FLAG_FULLSCREEN | LayoutParams.FLAG_KEEP_SCREEN_ON
| LayoutParams.FLAG_DISMISS_KEYGUARD | LayoutParams.FLAG_SHOW_WHEN_LOCKED
| LayoutParams.FLAG_TURN_SCREEN_ON);
| LayoutParams.FLAG_SHOW_WHEN_LOCKED | LayoutParams.FLAG_TURN_SCREEN_ON);
getWindow().getDecorView().setSystemUiVisibility(getSystemUiVisibility());
setContentView(R.layout.activity_call);
@ -630,7 +636,7 @@ public class CallActivity extends Activity implements AppRTCClient.SignalingEven
private void disconnect() {
activityRunning = false;
remoteProxyRenderer.setTarget(null);
localProxyRenderer.setTarget(null);
localProxyVideoSink.setTarget(null);
if (appRtcClient != null) {
appRtcClient.disconnectFromRoom();
appRtcClient = null;
@ -741,7 +747,7 @@ public class CallActivity extends Activity implements AppRTCClient.SignalingEven
private void setSwappedFeeds(boolean isSwappedFeeds) {
Logging.d(TAG, "setSwappedFeeds: " + isSwappedFeeds);
this.isSwappedFeeds = isSwappedFeeds;
localProxyRenderer.setTarget(isSwappedFeeds ? fullscreenRenderer : pipRenderer);
localProxyVideoSink.setTarget(isSwappedFeeds ? fullscreenRenderer : pipRenderer);
remoteProxyRenderer.setTarget(isSwappedFeeds ? pipRenderer : fullscreenRenderer);
fullscreenRenderer.setMirror(isSwappedFeeds);
pipRenderer.setMirror(!isSwappedFeeds);
@ -760,7 +766,7 @@ public class CallActivity extends Activity implements AppRTCClient.SignalingEven
videoCapturer = createVideoCapturer();
}
peerConnectionClient.createPeerConnection(
localProxyRenderer, remoteRenderers, videoCapturer, signalingParameters);
localProxyVideoSink, remoteRenderers, videoCapturer, signalingParameters);
if (signalingParameters.initiator) {
logAndToast("Creating OFFER...");

25
examples/androidapp/src/org/appspot/apprtc/RoomParametersFetcher.java
View File

@ -10,25 +10,22 @@
package org.appspot.apprtc;
import org.appspot.apprtc.AppRTCClient.SignalingParameters;
import org.appspot.apprtc.util.AsyncHttpURLConnection;
import org.appspot.apprtc.util.AsyncHttpURLConnection.AsyncHttpEvents;
import android.util.Log;
import org.json.JSONArray;
import org.json.JSONException;
import org.json.JSONObject;
import org.webrtc.IceCandidate;
import org.webrtc.PeerConnection;
import org.webrtc.SessionDescription;
import java.io.IOException;
import java.io.InputStream;
import java.net.HttpURLConnection;
import java.net.URL;
import java.util.LinkedList;
import java.util.Scanner;
import org.appspot.apprtc.AppRTCClient.SignalingParameters;
import org.appspot.apprtc.util.AsyncHttpURLConnection;
import org.appspot.apprtc.util.AsyncHttpURLConnection.AsyncHttpEvents;
import org.json.JSONArray;
import org.json.JSONException;
import org.json.JSONObject;
import org.webrtc.IceCandidate;
import org.webrtc.PeerConnection;
import org.webrtc.SessionDescription;
/**
* AsyncTask that converts an AppRTC room URL into the set of signaling
@ -133,11 +130,13 @@ public class RoomParametersFetcher {
boolean isTurnPresent = false;
for (PeerConnection.IceServer server : iceServers) {
Log.d(TAG, "IceServer: " + server);
if (server.uri.startsWith("turn:")) {
for (String uri : server.urls) {
if (uri.startsWith("turn:")) {
isTurnPresent = true;
break;
}
}
}
// Request TURN servers.
if (!isTurnPresent && !roomJson.optString("ice_server_url").isEmpty()) {
LinkedList<PeerConnection.IceServer> turnServers =

14
modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioTrack.java
View File

@ -267,9 +267,8 @@ public class WebRtcAudioTrack {
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);
audioTrack =
createAudioTrackOnLowerThanLollipop(sampleRate, channelConfig, minBufferSizeInBytes);
}
} catch (IllegalArgumentException e) {
reportWebRtcAudioTrackInitError(e.getMessage());
@ -369,7 +368,7 @@ public class WebRtcAudioTrack {
// It allows certain platforms or routing policies to use this information for more
// refined volume or routing decisions.
@TargetApi(21)
private AudioTrack createAudioTrackOnLollipopOrHigher(
private static AudioTrack createAudioTrackOnLollipopOrHigher(
int sampleRateInHz, int channelConfig, int bufferSizeInBytes) {
Logging.d(TAG, "createAudioTrackOnLollipopOrHigher");
// TODO(henrika): use setPerformanceMode(int) with PERFORMANCE_MODE_LOW_LATENCY to control
@ -399,6 +398,13 @@ public class WebRtcAudioTrack {
AudioManager.AUDIO_SESSION_ID_GENERATE);
}
@SuppressWarnings("deprecation") // Deprecated in API level 25.
private static AudioTrack createAudioTrackOnLowerThanLollipop(
int sampleRateInHz, int channelConfig, int bufferSizeInBytes) {
return new AudioTrack(AudioManager.STREAM_VOICE_CALL, sampleRateInHz, channelConfig,
AudioFormat.ENCODING_PCM_16BIT, bufferSizeInBytes, AudioTrack.MODE_STREAM);
}
@TargetApi(24)
private void logMainParametersExtended() {
if (WebRtcAudioUtils.runningOnMarshmallowOrHigher()) {

2
sdk/android/src/java/org/webrtc/MediaCodecUtils.java
View File

@ -17,6 +17,8 @@ import android.media.MediaCodecInfo.CodecCapabilities;
/** Container class for static constants and helpers used with MediaCodec. */
@TargetApi(18)
// We are forced to use the old API because we want to support API level < 21.
@SuppressWarnings("deprecation")
class MediaCodecUtils {
private static final String TAG = "MediaCodecUtils";