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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 |
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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()); |
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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 |
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