Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(591)

Side by Side Diff: webrtc/modules/audio_device/ios/audio_device_unittest_ios.mm

Issue 2963283002: Disable AudioDeviceTest.StartStopRecording on iOS (Closed)
Patch Set: Created 3 years, 5 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 | « no previous file | 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 (c) 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 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 584 matching lines...) Expand 10 before | Expand all | Expand 10 after
595 // Failing when running on real iOS devices: bugs.webrtc.org/6889. 595 // Failing when running on real iOS devices: bugs.webrtc.org/6889.
596 TEST_F(AudioDeviceTest, DISABLED_StartStopPlayout) { 596 TEST_F(AudioDeviceTest, DISABLED_StartStopPlayout) {
597 StartPlayout(); 597 StartPlayout();
598 StopPlayout(); 598 StopPlayout();
599 StartPlayout(); 599 StartPlayout();
600 StopPlayout(); 600 StopPlayout();
601 } 601 }
602 602
603 // Tests that recording can be initiated, started and stopped. No audio callback 603 // Tests that recording can be initiated, started and stopped. No audio callback
604 // is registered in this test. 604 // is registered in this test.
605 TEST_F(AudioDeviceTest, StartStopRecording) { 605 // Can sometimes fail when running on real devices: bugs.webrtc.org/7888.
606 TEST_F(AudioDeviceTest, DISABLED_StartStopRecording) {
607 StartRecording();
608 StopRecording();
606 StartRecording(); 609 StartRecording();
607 StopRecording(); 610 StopRecording();
608 } 611 }
609 612
610 // Verify that calling StopPlayout() will leave us in an uninitialized state 613 // Verify that calling StopPlayout() will leave us in an uninitialized state
611 // which will require a new call to InitPlayout(). This test does not call 614 // which will require a new call to InitPlayout(). This test does not call
612 // StartPlayout() while being uninitialized since doing so will hit a 615 // StartPlayout() while being uninitialized since doing so will hit a
613 // RTC_DCHECK. 616 // RTC_DCHECK.
614 TEST_F(AudioDeviceTest, StopPlayoutRequiresInitToRestart) { 617 TEST_F(AudioDeviceTest, StopPlayoutRequiresInitToRestart) {
615 EXPECT_EQ(0, audio_device()->InitPlayout()); 618 EXPECT_EQ(0, audio_device()->InitPlayout());
(...skipping 252 matching lines...) Expand 10 before | Expand all | Expand 10 after
868 // Wait for notification to propagate. 871 // Wait for notification to propagate.
869 rtc::MessageQueueManager::ProcessAllMessageQueues(); 872 rtc::MessageQueueManager::ProcessAllMessageQueues();
870 EXPECT_TRUE(audio_device->is_interrupted_); 873 EXPECT_TRUE(audio_device->is_interrupted_);
871 874
872 audio_device->Init(); 875 audio_device->Init();
873 audio_device->InitPlayout(); 876 audio_device->InitPlayout();
874 EXPECT_FALSE(audio_device->is_interrupted_); 877 EXPECT_FALSE(audio_device->is_interrupted_);
875 } 878 }
876 879
877 } // namespace webrtc 880 } // namespace webrtc
OLDNEW
« no previous file with comments | « no previous file | no next file » | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698