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

Issue 1344083004: Remove the SetLocalMonitor() API. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 5 years, 3 months ago
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1 /* 1 /*
2 * libjingle 2 * libjingle
3 * Copyright 2010 Google Inc. 3 * Copyright 2010 Google Inc.
4 * 4 *
5 * Redistribution and use in source and binary forms, with or without 5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met: 6 * modification, are permitted provided that the following conditions are met:
7 * 7 *
8 * 1. Redistributions of source code must retain the above copyright notice, 8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer. 9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice, 10 * 2. Redistributions in binary form must reproduce the above copyright notice,
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288 highpass_filter_enabled_(false), 288 highpass_filter_enabled_(false),
289 stereo_swapping_enabled_(false), 289 stereo_swapping_enabled_(false),
290 typing_detection_enabled_(false), 290 typing_detection_enabled_(false),
291 ec_mode_(webrtc::kEcDefault), 291 ec_mode_(webrtc::kEcDefault),
292 aecm_mode_(webrtc::kAecmSpeakerphone), 292 aecm_mode_(webrtc::kAecmSpeakerphone),
293 ns_mode_(webrtc::kNsDefault), 293 ns_mode_(webrtc::kNsDefault),
294 agc_mode_(webrtc::kAgcDefault), 294 agc_mode_(webrtc::kAgcDefault),
295 observer_(NULL), 295 observer_(NULL),
296 playout_fail_channel_(-1), 296 playout_fail_channel_(-1),
297 send_fail_channel_(-1), 297 send_fail_channel_(-1),
298 fail_start_recording_microphone_(false),
299 recording_microphone_(false),
300 recording_sample_rate_(-1), 298 recording_sample_rate_(-1),
301 playout_sample_rate_(-1), 299 playout_sample_rate_(-1),
302 media_processor_(NULL) { 300 media_processor_(NULL) {
303 memset(&agc_config_, 0, sizeof(agc_config_)); 301 memset(&agc_config_, 0, sizeof(agc_config_));
304 } 302 }
305 ~FakeWebRtcVoiceEngine() { 303 ~FakeWebRtcVoiceEngine() {
306 // Ought to have all been deleted by the WebRtcVoiceMediaChannel 304 // Ought to have all been deleted by the WebRtcVoiceMediaChannel
307 // destructors, but just in case ... 305 // destructors, but just in case ...
308 for (std::map<int, Channel*>::const_iterator i = channels_.begin(); 306 for (std::map<int, Channel*>::const_iterator i = channels_.begin();
309 i != channels_.end(); ++i) { 307 i != channels_.end(); ++i) {
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324 } 322 }
325 return -1; 323 return -1;
326 } 324 }
327 int GetNumChannels() const { return static_cast<int>(channels_.size()); } 325 int GetNumChannels() const { return static_cast<int>(channels_.size()); }
328 bool GetPlayout(int channel) { 326 bool GetPlayout(int channel) {
329 return channels_[channel]->playout; 327 return channels_[channel]->playout;
330 } 328 }
331 bool GetSend(int channel) { 329 bool GetSend(int channel) {
332 return channels_[channel]->send; 330 return channels_[channel]->send;
333 } 331 }
334 bool GetRecordingMicrophone() {
335 return recording_microphone_;
336 }
337 bool GetVAD(int channel) { 332 bool GetVAD(int channel) {
338 return channels_[channel]->vad; 333 return channels_[channel]->vad;
339 } 334 }
340 bool GetOpusDtx(int channel) { 335 bool GetOpusDtx(int channel) {
341 return channels_[channel]->opus_dtx; 336 return channels_[channel]->opus_dtx;
342 } 337 }
343 bool GetRED(int channel) { 338 bool GetRED(int channel) {
344 return channels_[channel]->red; 339 return channels_[channel]->red;
345 } 340 }
346 bool GetCodecFEC(int channel) { 341 bool GetCodecFEC(int channel) {
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385 void TriggerCallbackOnError(int channel_num, int err_code) { 380 void TriggerCallbackOnError(int channel_num, int err_code) {
386 RTC_DCHECK(observer_ != NULL); 381 RTC_DCHECK(observer_ != NULL);
387 observer_->CallbackOnError(channel_num, err_code); 382 observer_->CallbackOnError(channel_num, err_code);
388 } 383 }
389 void set_playout_fail_channel(int channel) { 384 void set_playout_fail_channel(int channel) {
390 playout_fail_channel_ = channel; 385 playout_fail_channel_ = channel;
391 } 386 }
392 void set_send_fail_channel(int channel) { 387 void set_send_fail_channel(int channel) {
393 send_fail_channel_ = channel; 388 send_fail_channel_ = channel;
394 } 389 }
395 void set_fail_start_recording_microphone(
396 bool fail_start_recording_microphone) {
397 fail_start_recording_microphone_ = fail_start_recording_microphone;
398 }
399 void set_fail_create_channel(bool fail_create_channel) { 390 void set_fail_create_channel(bool fail_create_channel) {
400 fail_create_channel_ = fail_create_channel; 391 fail_create_channel_ = fail_create_channel;
401 } 392 }
402 void TriggerProcessPacket(MediaProcessorDirection direction) { 393 void TriggerProcessPacket(MediaProcessorDirection direction) {
403 webrtc::ProcessingTypes pt = 394 webrtc::ProcessingTypes pt =
404 (direction == cricket::MPD_TX) ? 395 (direction == cricket::MPD_TX) ?
405 webrtc::kRecordingPerChannel : webrtc::kPlaybackAllChannelsMixed; 396 webrtc::kRecordingPerChannel : webrtc::kPlaybackAllChannelsMixed;
406 if (media_processor_ != NULL) { 397 if (media_processor_ != NULL) {
407 media_processor_->Process(0, 398 media_processor_->Process(0,
408 pt, 399 pt,
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773 WEBRTC_STUB(IsPlayingFileAsMicrophone, (int channel)); 764 WEBRTC_STUB(IsPlayingFileAsMicrophone, (int channel));
774 WEBRTC_STUB(StartRecordingPlayout, (int channel, const char* fileNameUTF8, 765 WEBRTC_STUB(StartRecordingPlayout, (int channel, const char* fileNameUTF8,
775 webrtc::CodecInst* compression, 766 webrtc::CodecInst* compression,
776 int maxSizeBytes)); 767 int maxSizeBytes));
777 WEBRTC_STUB(StartRecordingPlayout, (int channel, webrtc::OutStream* stream, 768 WEBRTC_STUB(StartRecordingPlayout, (int channel, webrtc::OutStream* stream,
778 webrtc::CodecInst* compression)); 769 webrtc::CodecInst* compression));
779 WEBRTC_STUB(StopRecordingPlayout, (int channel)); 770 WEBRTC_STUB(StopRecordingPlayout, (int channel));
780 WEBRTC_FUNC(StartRecordingMicrophone, (const char* fileNameUTF8, 771 WEBRTC_FUNC(StartRecordingMicrophone, (const char* fileNameUTF8,
781 webrtc::CodecInst* compression, 772 webrtc::CodecInst* compression,
782 int maxSizeBytes)) { 773 int maxSizeBytes)) {
783 if (fail_start_recording_microphone_) {
784 return -1;
785 }
786 recording_microphone_ = true;
787 return 0; 774 return 0;
788 } 775 }
789 WEBRTC_FUNC(StartRecordingMicrophone, (webrtc::OutStream* stream, 776 WEBRTC_FUNC(StartRecordingMicrophone, (webrtc::OutStream* stream,
790 webrtc::CodecInst* compression)) { 777 webrtc::CodecInst* compression)) {
791 if (fail_start_recording_microphone_) {
792 return -1;
793 }
794 recording_microphone_ = true;
795 return 0; 778 return 0;
796 } 779 }
797 WEBRTC_FUNC(StopRecordingMicrophone, ()) { 780 WEBRTC_FUNC(StopRecordingMicrophone, ()) {
798 if (!recording_microphone_) {
799 return -1;
800 }
801 recording_microphone_ = false;
802 return 0; 781 return 0;
803 } 782 }
804 783
805 // webrtc::VoEHardware 784 // webrtc::VoEHardware
806 WEBRTC_FUNC(GetNumOfRecordingDevices, (int& num)) { 785 WEBRTC_FUNC(GetNumOfRecordingDevices, (int& num)) {
807 return GetNumDevices(num); 786 return GetNumDevices(num);
808 } 787 }
809 WEBRTC_FUNC(GetNumOfPlayoutDevices, (int& num)) { 788 WEBRTC_FUNC(GetNumOfPlayoutDevices, (int& num)) {
810 return GetNumDevices(num); 789 return GetNumDevices(num);
811 } 790 }
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1270 bool stereo_swapping_enabled_; 1249 bool stereo_swapping_enabled_;
1271 bool typing_detection_enabled_; 1250 bool typing_detection_enabled_;
1272 webrtc::EcModes ec_mode_; 1251 webrtc::EcModes ec_mode_;
1273 webrtc::AecmModes aecm_mode_; 1252 webrtc::AecmModes aecm_mode_;
1274 webrtc::NsModes ns_mode_; 1253 webrtc::NsModes ns_mode_;
1275 webrtc::AgcModes agc_mode_; 1254 webrtc::AgcModes agc_mode_;
1276 webrtc::AgcConfig agc_config_; 1255 webrtc::AgcConfig agc_config_;
1277 webrtc::VoiceEngineObserver* observer_; 1256 webrtc::VoiceEngineObserver* observer_;
1278 int playout_fail_channel_; 1257 int playout_fail_channel_;
1279 int send_fail_channel_; 1258 int send_fail_channel_;
1280 bool fail_start_recording_microphone_;
1281 bool recording_microphone_;
1282 int recording_sample_rate_; 1259 int recording_sample_rate_;
1283 int playout_sample_rate_; 1260 int playout_sample_rate_;
1284 DtmfInfo dtmf_info_; 1261 DtmfInfo dtmf_info_;
1285 webrtc::VoEMediaProcess* media_processor_; 1262 webrtc::VoEMediaProcess* media_processor_;
1286 FakeAudioProcessing audio_processing_; 1263 FakeAudioProcessing audio_processing_;
1287 }; 1264 };
1288 1265
1289 #undef WEBRTC_CHECK_HEADER_EXTENSION_ID 1266 #undef WEBRTC_CHECK_HEADER_EXTENSION_ID
1290 1267
1291 } // namespace cricket 1268 } // namespace cricket
1292 1269
1293 #endif // TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_ 1270 #endif // TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_
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