Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(150)

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

Issue 227743004: Added a kEchoCancellation constraint to turn off the audio processing. (Closed) Base URL: http://git.chromium.org/chromium/src.git@master
Patch Set: Created 6 years, 7 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
OLDNEW
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 "base/command_line.h" 5 #include "base/command_line.h"
6 #include "content/public/common/content_switches.h" 6 #include "content/public/common/content_switches.h"
7 #include "content/renderer/media/mock_media_constraint_factory.h"
7 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h" 8 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
8 #include "content/renderer/media/webrtc_local_audio_track.h" 9 #include "content/renderer/media/webrtc_local_audio_track.h"
9 #include "testing/gmock/include/gmock/gmock.h" 10 #include "testing/gmock/include/gmock/gmock.h"
10 #include "testing/gtest/include/gtest/gtest.h" 11 #include "testing/gtest/include/gtest/gtest.h"
11 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" 12 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
12 13
13 using ::testing::_; 14 using ::testing::_;
14 using ::testing::AnyNumber; 15 using ::testing::AnyNumber;
15 16
16 namespace content { 17 namespace content {
(...skipping 11 matching lines...) Expand all
28 int number_of_frames)); 29 int number_of_frames));
29 }; 30 };
30 31
31 } // namespace 32 } // namespace
32 33
33 class WebRtcLocalAudioTrackAdapterTest : public ::testing::Test { 34 class WebRtcLocalAudioTrackAdapterTest : public ::testing::Test {
34 public: 35 public:
35 WebRtcLocalAudioTrackAdapterTest() 36 WebRtcLocalAudioTrackAdapterTest()
36 : params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, 37 : params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
37 media::CHANNEL_LAYOUT_STEREO, 48000, 16, 480), 38 media::CHANNEL_LAYOUT_STEREO, 48000, 16, 480),
38 adapter_(WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)), 39 adapter_(WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)) {
39 capturer_(WebRtcAudioCapturer::CreateCapturer( 40 MockMediaConstraintFactory constraint_factory;
40 -1, StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE, "", ""), 41 capturer_ = WebRtcAudioCapturer::CreateCapturer(
41 blink::WebMediaConstraints(), NULL, NULL)), 42 -1, StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE, "", ""),
42 track_(new WebRtcLocalAudioTrack(adapter_, capturer_, NULL)) {} 43 constraint_factory.CreateWebMediaConstraints(), NULL, NULL);
44 track_.reset(new WebRtcLocalAudioTrack(adapter_, capturer_, NULL));
45 }
43 46
44 protected: 47 protected:
45 virtual void SetUp() OVERRIDE { 48 virtual void SetUp() OVERRIDE {
46 track_->OnSetFormat(params_); 49 track_->OnSetFormat(params_);
47 EXPECT_TRUE(track_->GetAudioAdapter()->enabled()); 50 EXPECT_TRUE(track_->GetAudioAdapter()->enabled());
48 } 51 }
49 52
50 media::AudioParameters params_; 53 media::AudioParameters params_;
51 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_; 54 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_;
52 scoped_refptr<WebRtcAudioCapturer> capturer_; 55 scoped_refptr<WebRtcAudioCapturer> capturer_;
(...skipping 34 matching lines...) Expand 10 before | Expand all | Expand 10 after
87 int signal_level = 0; 90 int signal_level = 0;
88 EXPECT_FALSE(webrtc_track->GetSignalLevel(&signal_level)); 91 EXPECT_FALSE(webrtc_track->GetSignalLevel(&signal_level));
89 92
90 // Enable the audio processing in the audio track. 93 // Enable the audio processing in the audio track.
91 CommandLine::ForCurrentProcess()->AppendSwitch( 94 CommandLine::ForCurrentProcess()->AppendSwitch(
92 switches::kEnableAudioTrackProcessing); 95 switches::kEnableAudioTrackProcessing);
93 EXPECT_TRUE(webrtc_track->GetSignalLevel(&signal_level)); 96 EXPECT_TRUE(webrtc_track->GetSignalLevel(&signal_level));
94 } 97 }
95 98
96 } // namespace content 99 } // namespace content
OLDNEW
« no previous file with comments | « content/renderer/media/mock_media_constraint_factory.cc ('k') | content/renderer/media/webrtc_audio_capturer.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698