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

Issue 2111813002: Revert of Move RtcEventLog object from inside VoiceEngine to Call. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 5 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
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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; }
275 276
276 // webrtc::VoECodec 277 // webrtc::VoECodec
277 WEBRTC_STUB(NumOfCodecs, ()); 278 WEBRTC_STUB(NumOfCodecs, ());
278 WEBRTC_STUB(GetCodec, (int index, webrtc::CodecInst& codec)); 279 WEBRTC_STUB(GetCodec, (int index, webrtc::CodecInst& codec));
279 WEBRTC_FUNC(SetSendCodec, (int channel, const webrtc::CodecInst& codec)) { 280 WEBRTC_FUNC(SetSendCodec, (int channel, const webrtc::CodecInst& codec)) {
280 WEBRTC_CHECK_CHANNEL(channel); 281 WEBRTC_CHECK_CHANNEL(channel);
281 // To match the behavior of the real implementation. 282 // To match the behavior of the real implementation.
282 if (_stricmp(codec.plname, "telephone-event") == 0 || 283 if (_stricmp(codec.plname, "telephone-event") == 0 ||
283 _stricmp(codec.plname, "audio/telephone-event") == 0 || 284 _stricmp(codec.plname, "audio/telephone-event") == 0 ||
284 _stricmp(codec.plname, "CN") == 0 || 285 _stricmp(codec.plname, "CN") == 0 ||
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583 webrtc::NsModes ns_mode_ = webrtc::kNsDefault; 584 webrtc::NsModes ns_mode_ = webrtc::kNsDefault;
584 webrtc::AgcModes agc_mode_ = webrtc::kAgcDefault; 585 webrtc::AgcModes agc_mode_ = webrtc::kAgcDefault;
585 webrtc::AgcConfig agc_config_; 586 webrtc::AgcConfig agc_config_;
586 int playout_fail_channel_ = -1; 587 int playout_fail_channel_ = -1;
587 FakeAudioProcessing audio_processing_; 588 FakeAudioProcessing audio_processing_;
588 }; 589 };
589 590
590 } // namespace cricket 591 } // namespace cricket
591 592
592 #endif // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_ 593 #endif // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_
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