content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc - Issue 1547073003: Switch to standard integer types in content/renderer/.

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Side by Side Diff: content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc

Issue 1547073003: Switch to standard integer types in content/renderer/. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master

Patch Set: Created 4 years, 12 months ago

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1 // Copyright 2014 The Chromium Authors. All rights reserved.	1 // Copyright 2014 The Chromium Authors. All rights reserved.

2 // Use of this source code is governed by a BSD-style license that can be	2 // Use of this source code is governed by a BSD-style license that can be

3 // found in the LICENSE file.	3 // found in the LICENSE file.

4	4

	5 #include <stddef.h>

	6

5 #include "content/renderer/media/mock_media_constraint_factory.h"	7 #include "content/renderer/media/mock_media_constraint_factory.h"

6 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"	8 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"

7 #include "content/renderer/media/webrtc_audio_capturer.h"	9 #include "content/renderer/media/webrtc_audio_capturer.h"

8 #include "content/renderer/media/webrtc_local_audio_track.h"	10 #include "content/renderer/media/webrtc_local_audio_track.h"

9 #include "testing/gmock/include/gmock/gmock.h"	11 #include "testing/gmock/include/gmock/gmock.h"

10 #include "testing/gtest/include/gtest/gtest.h"	12 #include "testing/gtest/include/gtest/gtest.h"

11 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"	13 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"

12	14

13 using ::testing::_;	15 using ::testing::_;

14 using ::testing::AnyNumber;	16 using ::testing::AnyNumber;

(...skipping 76 matching lines...) Expand 10 before \| Expand all \| Expand 10 after Loading...
91 }	93 }

92	94

93 TEST_F(WebRtcLocalAudioTrackAdapterTest, GetSignalLevel) {	95 TEST_F(WebRtcLocalAudioTrackAdapterTest, GetSignalLevel) {

94 webrtc::AudioTrackInterface* webrtc_track =	96 webrtc::AudioTrackInterface* webrtc_track =

95 static_cast<webrtc::AudioTrackInterface*>(adapter_.get());	97 static_cast<webrtc::AudioTrackInterface*>(adapter_.get());

96 int signal_level = 0;	98 int signal_level = 0;

97 EXPECT_TRUE(webrtc_track->GetSignalLevel(&signal_level));	99 EXPECT_TRUE(webrtc_track->GetSignalLevel(&signal_level));

98 }	100 }

99	101

100 } // namespace content	102 } // namespace content

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