content/renderer/media/webrtc_audio_device_unittest.cc - Issue 16756003: Use a direct include of strings headers in content/renderer and content/shell.

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Side by Side Diff: content/renderer/media/webrtc_audio_device_unittest.cc

Issue 16756003: Use a direct include of strings headers in content/renderer and content/shell. (Closed) Base URL: svn://svn.chromium.org/chrome/trunk/src

Patch Set: Created 7 years, 6 months ago

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1 // Copyright (c) 2012 The Chromium Authors. All rights reserved.	1 // Copyright (c) 2012 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 "base/environment.h"	5 #include "base/environment.h"

6 #include "base/file_util.h"	6 #include "base/file_util.h"

7 #include "base/files/file_path.h"	7 #include "base/files/file_path.h"

8 #include "base/path_service.h"	8 #include "base/path_service.h"

9 #include "base/stringprintf.h"	9 #include "base/strings/stringprintf.h"

10 #include "base/test/test_timeouts.h"	10 #include "base/test/test_timeouts.h"

11 #include "content/renderer/media/webrtc_audio_capturer.h"	11 #include "content/renderer/media/webrtc_audio_capturer.h"

12 #include "content/renderer/media/webrtc_audio_device_impl.h"	12 #include "content/renderer/media/webrtc_audio_device_impl.h"

13 #include "content/renderer/media/webrtc_audio_renderer.h"	13 #include "content/renderer/media/webrtc_audio_renderer.h"

14 #include "content/renderer/render_thread_impl.h"	14 #include "content/renderer/render_thread_impl.h"

15 #include "content/test/webrtc_audio_device_test.h"	15 #include "content/test/webrtc_audio_device_test.h"

16 #include "media/audio/audio_manager_base.h"	16 #include "media/audio/audio_manager_base.h"

17 #include "media/base/audio_hardware_config.h"	17 #include "media/base/audio_hardware_config.h"

18 #include "testing/gmock/include/gmock/gmock.h"	18 #include "testing/gmock/include/gmock/gmock.h"

19 #include "third_party/webrtc/voice_engine/include/voe_audio_processing.h"	19 #include "third_party/webrtc/voice_engine/include/voe_audio_processing.h"

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905 WebRtcLoopbackTimeWithSignalProcessing	905 WebRtcLoopbackTimeWithSignalProcessing

906 #endif	906 #endif

907	907

908 TEST_F(WebRTCAudioDeviceTest, MAYBE_WebRtcLoopbackTimeWithSignalProcessing) {	908 TEST_F(WebRTCAudioDeviceTest, MAYBE_WebRtcLoopbackTimeWithSignalProcessing) {

909 int latency = RunWebRtcLoopbackTimeTest(audio_manager_.get(), true);	909 int latency = RunWebRtcLoopbackTimeTest(audio_manager_.get(), true);

910 PrintPerfResultMs("webrtc_loopback_with_signal_processing (100 packets)",	910 PrintPerfResultMs("webrtc_loopback_with_signal_processing (100 packets)",

911 "t", latency);	911 "t", latency);

912 }	912 }

913	913

914 } // namespace content	914 } // namespace content

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