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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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