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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 |
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265 } | 265 } |
266 WEBRTC_STUB(StopSend, (int channel)); | 266 WEBRTC_STUB(StopSend, (int channel)); |
267 WEBRTC_STUB(GetVersion, (char version[1024])); | 267 WEBRTC_STUB(GetVersion, (char version[1024])); |
268 WEBRTC_STUB(LastError, ()); | 268 WEBRTC_STUB(LastError, ()); |
269 WEBRTC_FUNC(AssociateSendChannel, (int channel, | 269 WEBRTC_FUNC(AssociateSendChannel, (int channel, |
270 int accociate_send_channel)) { | 270 int accociate_send_channel)) { |
271 WEBRTC_CHECK_CHANNEL(channel); | 271 WEBRTC_CHECK_CHANNEL(channel); |
272 channels_[channel]->associate_send_channel = accociate_send_channel; | 272 channels_[channel]->associate_send_channel = accociate_send_channel; |
273 return 0; | 273 return 0; |
274 } | 274 } |
275 webrtc::RtcEventLog* GetEventLog() override { return nullptr; } | |
276 | 275 |
277 // webrtc::VoECodec | 276 // webrtc::VoECodec |
278 WEBRTC_STUB(NumOfCodecs, ()); | 277 WEBRTC_STUB(NumOfCodecs, ()); |
279 WEBRTC_STUB(GetCodec, (int index, webrtc::CodecInst& codec)); | 278 WEBRTC_STUB(GetCodec, (int index, webrtc::CodecInst& codec)); |
280 WEBRTC_FUNC(SetSendCodec, (int channel, const webrtc::CodecInst& codec)) { | 279 WEBRTC_FUNC(SetSendCodec, (int channel, const webrtc::CodecInst& codec)) { |
281 WEBRTC_CHECK_CHANNEL(channel); | 280 WEBRTC_CHECK_CHANNEL(channel); |
282 // To match the behavior of the real implementation. | 281 // To match the behavior of the real implementation. |
283 if (_stricmp(codec.plname, "telephone-event") == 0 || | 282 if (_stricmp(codec.plname, "telephone-event") == 0 || |
284 _stricmp(codec.plname, "audio/telephone-event") == 0 || | 283 _stricmp(codec.plname, "audio/telephone-event") == 0 || |
285 _stricmp(codec.plname, "CN") == 0 || | 284 _stricmp(codec.plname, "CN") == 0 || |
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584 webrtc::NsModes ns_mode_ = webrtc::kNsDefault; | 583 webrtc::NsModes ns_mode_ = webrtc::kNsDefault; |
585 webrtc::AgcModes agc_mode_ = webrtc::kAgcDefault; | 584 webrtc::AgcModes agc_mode_ = webrtc::kAgcDefault; |
586 webrtc::AgcConfig agc_config_; | 585 webrtc::AgcConfig agc_config_; |
587 int playout_fail_channel_ = -1; | 586 int playout_fail_channel_ = -1; |
588 FakeAudioProcessing audio_processing_; | 587 FakeAudioProcessing audio_processing_; |
589 }; | 588 }; |
590 | 589 |
591 } // namespace cricket | 590 } // namespace cricket |
592 | 591 |
593 #endif // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_ | 592 #endif // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_ |
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