Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(26)

Side by Side Diff: webrtc/examples/androidapp/src/org/appspot/apprtc/CallActivity.java

Issue 1312523004: Fix AppRTCDemo crash when room is connected after PC is destroyed. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master
Patch Set: Created 5 years, 3 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
« no previous file with comments | « talk/app/webrtc/java/android/org/webrtc/VideoRendererGui.java ('k') | no next file » | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
1 /* 1 /*
2 * Copyright 2015 The WebRTC Project Authors. All rights reserved. 2 * Copyright 2015 The WebRTC Project Authors. All rights reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
(...skipping 351 matching lines...) Expand 10 before | Expand all | Expand 10 after
362 // Store existing audio settings and change audio mode to 362 // Store existing audio settings and change audio mode to
363 // MODE_IN_COMMUNICATION for best possible VoIP performance. 363 // MODE_IN_COMMUNICATION for best possible VoIP performance.
364 Log.d(TAG, "Initializing the audio manager..."); 364 Log.d(TAG, "Initializing the audio manager...");
365 audioManager.init(); 365 audioManager.init();
366 } 366 }
367 367
368 // Should be called from UI thread 368 // Should be called from UI thread
369 private void callConnected() { 369 private void callConnected() {
370 final long delta = System.currentTimeMillis() - callStartedTimeMs; 370 final long delta = System.currentTimeMillis() - callStartedTimeMs;
371 Log.i(TAG, "Call connected: delay=" + delta + "ms"); 371 Log.i(TAG, "Call connected: delay=" + delta + "ms");
372 372 if (peerConnectionClient == null || isError) {
373 Log.w(TAG, "Call is connected in closed or error state");
374 return;
375 }
373 // Update video view. 376 // Update video view.
374 updateVideoView(); 377 updateVideoView();
375 // Enable statistics callback. 378 // Enable statistics callback.
376 peerConnectionClient.enableStatsEvents(true, STAT_CALLBACK_PERIOD); 379 peerConnectionClient.enableStatsEvents(true, STAT_CALLBACK_PERIOD);
377 } 380 }
378 381
379 private void onAudioManagerChangedState() { 382 private void onAudioManagerChangedState() {
380 // TODO(henrika): disable video if AppRTCAudioManager.AudioDevice.EARPIECE 383 // TODO(henrika): disable video if AppRTCAudioManager.AudioDevice.EARPIECE
381 // is active. 384 // is active.
382 } 385 }
(...skipping 256 matching lines...) Expand 10 before | Expand all | Expand 10 after
639 } 642 }
640 } 643 }
641 }); 644 });
642 } 645 }
643 646
644 @Override 647 @Override
645 public void onPeerConnectionError(final String description) { 648 public void onPeerConnectionError(final String description) {
646 reportError(description); 649 reportError(description);
647 } 650 }
648 } 651 }
OLDNEW
« no previous file with comments | « talk/app/webrtc/java/android/org/webrtc/VideoRendererGui.java ('k') | no next file » | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698