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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" |  | 
| 21 #include "webrtc/base/stringutils.h" | 20 #include "webrtc/base/stringutils.h" | 
| 22 #include "webrtc/config.h" | 21 #include "webrtc/config.h" | 
| 23 #include "webrtc/media/base/codec.h" | 22 #include "webrtc/media/base/codec.h" | 
| 24 #include "webrtc/media/base/rtputils.h" | 23 #include "webrtc/media/base/rtputils.h" | 
| 25 #include "webrtc/media/engine/fakewebrtccommon.h" |  | 
| 26 #include "webrtc/media/engine/webrtcvoe.h" | 24 #include "webrtc/media/engine/webrtcvoe.h" | 
| 27 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h" | 25 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h" | 
| 28 #include "webrtc/modules/audio_processing/include/audio_processing.h" | 26 #include "webrtc/modules/audio_processing/include/audio_processing.h" | 
| 29 | 27 | 
| 30 namespace cricket { | 28 namespace cricket { | 
| 31 | 29 | 
| 32 static const int kOpusBandwidthNb = 4000; | 30 static const int kOpusBandwidthNb = 4000; | 
| 33 static const int kOpusBandwidthMb = 6000; | 31 static const int kOpusBandwidthMb = 6000; | 
| 34 static const int kOpusBandwidthWb = 8000; | 32 static const int kOpusBandwidthWb = 8000; | 
| 35 static const int kOpusBandwidthSwb = 12000; | 33 static const int kOpusBandwidthSwb = 12000; | 
| 36 static const int kOpusBandwidthFb = 20000; | 34 static const int kOpusBandwidthFb = 20000; | 
| 37 | 35 | 
| 38 #define WEBRTC_CHECK_CHANNEL(channel) \ | 36 #define WEBRTC_CHECK_CHANNEL(channel) \ | 
| 39   if (channels_.find(channel) == channels_.end()) return -1; | 37   if (channels_.find(channel) == channels_.end()) return -1; | 
| 40 | 38 | 
|  | 39 #define WEBRTC_STUB(method, args) \ | 
|  | 40   int method args override { return 0; } | 
|  | 41 | 
|  | 42 #define WEBRTC_STUB_CONST(method, args) \ | 
|  | 43   int method args const override { return 0; } | 
|  | 44 | 
|  | 45 #define WEBRTC_BOOL_STUB(method, args) \ | 
|  | 46   bool method args override { return true; } | 
|  | 47 | 
|  | 48 #define WEBRTC_BOOL_STUB_CONST(method, args) \ | 
|  | 49   bool method args const override { return true; } | 
|  | 50 | 
|  | 51 #define WEBRTC_VOID_STUB(method, args) \ | 
|  | 52   void method args override {} | 
|  | 53 | 
|  | 54 #define WEBRTC_FUNC(method, args) int method args override | 
|  | 55 | 
|  | 56 #define WEBRTC_VOID_FUNC(method, args) void method args override | 
|  | 57 | 
| 41 class FakeAudioProcessing : public webrtc::AudioProcessing { | 58 class FakeAudioProcessing : public webrtc::AudioProcessing { | 
| 42  public: | 59  public: | 
| 43   FakeAudioProcessing() : experimental_ns_enabled_(false) {} | 60   FakeAudioProcessing() : experimental_ns_enabled_(false) {} | 
| 44 | 61 | 
| 45   WEBRTC_STUB(Initialize, ()) | 62   WEBRTC_STUB(Initialize, ()) | 
| 46   WEBRTC_STUB(Initialize, ( | 63   WEBRTC_STUB(Initialize, ( | 
| 47       int input_sample_rate_hz, | 64       int input_sample_rate_hz, | 
| 48       int output_sample_rate_hz, | 65       int output_sample_rate_hz, | 
| 49       int reverse_sample_rate_hz, | 66       int reverse_sample_rate_hz, | 
| 50       webrtc::AudioProcessing::ChannelLayout input_layout, | 67       webrtc::AudioProcessing::ChannelLayout input_layout, | 
| (...skipping 508 matching lines...) Expand 10 before | Expand all | Expand 10 after  Loading... | 
| 559   webrtc::AecmModes aecm_mode_ = webrtc::kAecmSpeakerphone; | 576   webrtc::AecmModes aecm_mode_ = webrtc::kAecmSpeakerphone; | 
| 560   webrtc::NsModes ns_mode_ = webrtc::kNsDefault; | 577   webrtc::NsModes ns_mode_ = webrtc::kNsDefault; | 
| 561   webrtc::AgcModes agc_mode_ = webrtc::kAgcDefault; | 578   webrtc::AgcModes agc_mode_ = webrtc::kAgcDefault; | 
| 562   webrtc::AgcConfig agc_config_; | 579   webrtc::AgcConfig agc_config_; | 
| 563   FakeAudioProcessing audio_processing_; | 580   FakeAudioProcessing audio_processing_; | 
| 564 }; | 581 }; | 
| 565 | 582 | 
| 566 }  // namespace cricket | 583 }  // namespace cricket | 
| 567 | 584 | 
| 568 #endif  // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_ | 585 #endif  // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_ | 
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