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Side by Side Diff: webrtc/media/engine/fakewebrtcvoiceengine.h

Issue 1830213002: Remove WVoE::SetAudioDeviceModule() - it is now set in ctor. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: removed Construct() method Created 4 years, 9 months ago
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1 /* 1 /*
2 * Copyright (c) 2010 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2010 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
11 #ifndef WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_ 11 #ifndef WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_
12 #define WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_ 12 #define WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_
13 13
14 #include <list> 14 #include <list>
15 #include <map> 15 #include <map>
16 #include <vector> 16 #include <vector>
17 17
18 #include "webrtc/base/basictypes.h" 18 #include "webrtc/base/basictypes.h"
19 #include "webrtc/base/checks.h" 19 #include "webrtc/base/checks.h"
20 #include "webrtc/base/gunit.h" 20 #include "webrtc/base/gunit.h"
21 #include "webrtc/base/stringutils.h" 21 #include "webrtc/base/stringutils.h"
22 #include "webrtc/config.h" 22 #include "webrtc/config.h"
23 #include "webrtc/media/base/codec.h" 23 #include "webrtc/media/base/codec.h"
24 #include "webrtc/media/base/rtputils.h" 24 #include "webrtc/media/base/rtputils.h"
25 #include "webrtc/media/engine/fakewebrtccommon.h" 25 #include "webrtc/media/engine/fakewebrtccommon.h"
26 #include "webrtc/media/engine/webrtcvoe.h" 26 #include "webrtc/media/engine/webrtcvoe.h"
27 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h" 27 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
28 #include "webrtc/modules/audio_device/include/fake_audio_device.h"
28 #include "webrtc/modules/audio_processing/include/audio_processing.h" 29 #include "webrtc/modules/audio_processing/include/audio_processing.h"
29 30
30 namespace cricket { 31 namespace cricket {
31 32
32 static const int kOpusBandwidthNb = 4000; 33 static const int kOpusBandwidthNb = 4000;
33 static const int kOpusBandwidthMb = 6000; 34 static const int kOpusBandwidthMb = 6000;
34 static const int kOpusBandwidthWb = 8000; 35 static const int kOpusBandwidthWb = 8000;
35 static const int kOpusBandwidthSwb = 12000; 36 static const int kOpusBandwidthSwb = 12000;
36 static const int kOpusBandwidthFb = 20000; 37 static const int kOpusBandwidthFb = 20000;
37 38
(...skipping 261 matching lines...) Expand 10 before | Expand all | Expand 10 after
299 inited_ = true; 300 inited_ = true;
300 return 0; 301 return 0;
301 } 302 }
302 WEBRTC_FUNC(Terminate, ()) { 303 WEBRTC_FUNC(Terminate, ()) {
303 inited_ = false; 304 inited_ = false;
304 return 0; 305 return 0;
305 } 306 }
306 webrtc::AudioProcessing* audio_processing() override { 307 webrtc::AudioProcessing* audio_processing() override {
307 return &audio_processing_; 308 return &audio_processing_;
308 } 309 }
310 webrtc::AudioDeviceModule* audio_device_module() override {
311 return &audio_device_module_;
312 }
309 WEBRTC_FUNC(CreateChannel, ()) { 313 WEBRTC_FUNC(CreateChannel, ()) {
310 webrtc::Config empty_config; 314 webrtc::Config empty_config;
311 return AddChannel(empty_config); 315 return AddChannel(empty_config);
312 } 316 }
313 WEBRTC_FUNC(CreateChannel, (const webrtc::Config& config)) { 317 WEBRTC_FUNC(CreateChannel, (const webrtc::Config& config)) {
314 return AddChannel(config); 318 return AddChannel(config);
315 } 319 }
316 WEBRTC_FUNC(DeleteChannel, (int channel)) { 320 WEBRTC_FUNC(DeleteChannel, (int channel)) {
317 WEBRTC_CHECK_CHANNEL(channel); 321 WEBRTC_CHECK_CHANNEL(channel);
318 for (const auto& ch : channels_) { 322 for (const auto& ch : channels_) {
(...skipping 449 matching lines...) Expand 10 before | Expand all | Expand 10 after
768 webrtc::EcModes ec_mode_; 772 webrtc::EcModes ec_mode_;
769 webrtc::AecmModes aecm_mode_; 773 webrtc::AecmModes aecm_mode_;
770 webrtc::NsModes ns_mode_; 774 webrtc::NsModes ns_mode_;
771 webrtc::AgcModes agc_mode_; 775 webrtc::AgcModes agc_mode_;
772 webrtc::AgcConfig agc_config_; 776 webrtc::AgcConfig agc_config_;
773 webrtc::VoiceEngineObserver* observer_; 777 webrtc::VoiceEngineObserver* observer_;
774 int playout_fail_channel_; 778 int playout_fail_channel_;
775 int recording_sample_rate_; 779 int recording_sample_rate_;
776 int playout_sample_rate_; 780 int playout_sample_rate_;
777 FakeAudioProcessing audio_processing_; 781 FakeAudioProcessing audio_processing_;
782 webrtc::FakeAudioDeviceModule audio_device_module_;
778 }; 783 };
779 784
780 } // namespace cricket 785 } // namespace cricket
781 786
782 #endif // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_ 787 #endif // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_
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