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

Issue 1269863005: MediaController/Call instantiation. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: nullptr Created 5 years, 3 months ago
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1 /* 1 /*
2 * libjingle 2 * libjingle
3 * Copyright 2004 Google Inc. 3 * Copyright 2004 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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93 93
94 public: 94 public:
95 WebRtcVoiceEngine(); 95 WebRtcVoiceEngine();
96 // Dependency injection for testing. 96 // Dependency injection for testing.
97 WebRtcVoiceEngine(VoEWrapper* voe_wrapper, VoETraceWrapper* tracing); 97 WebRtcVoiceEngine(VoEWrapper* voe_wrapper, VoETraceWrapper* tracing);
98 ~WebRtcVoiceEngine(); 98 ~WebRtcVoiceEngine();
99 bool Init(rtc::Thread* worker_thread); 99 bool Init(rtc::Thread* worker_thread);
100 void Terminate(); 100 void Terminate();
101 101
102 int GetCapabilities(); 102 int GetCapabilities();
103 VoiceMediaChannel* CreateChannel(const AudioOptions& options); 103 webrtc::VoiceEngine* GetVoE() { return voe()->engine(); }
104 VoiceMediaChannel* CreateChannel(webrtc::Call* call,
105 const AudioOptions& options);
104 106
105 AudioOptions GetOptions() const { return options_; } 107 AudioOptions GetOptions() const { return options_; }
106 bool SetOptions(const AudioOptions& options); 108 bool SetOptions(const AudioOptions& options);
107 bool SetDelayOffset(int offset); 109 bool SetDelayOffset(int offset);
108 bool SetDevices(const Device* in_device, const Device* out_device); 110 bool SetDevices(const Device* in_device, const Device* out_device);
109 bool GetOutputVolume(int* level); 111 bool GetOutputVolume(int* level);
110 bool SetOutputVolume(int level); 112 bool SetOutputVolume(int level);
111 int GetInputLevel(); 113 int GetInputLevel();
112 bool SetLocalMonitor(bool enable); 114 bool SetLocalMonitor(bool enable);
113 115
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273 Settable<bool> extended_filter_aec_; 275 Settable<bool> extended_filter_aec_;
274 Settable<bool> delay_agnostic_aec_; 276 Settable<bool> delay_agnostic_aec_;
275 Settable<bool> experimental_ns_; 277 Settable<bool> experimental_ns_;
276 }; 278 };
277 279
278 // WebRtcVoiceMediaChannel is an implementation of VoiceMediaChannel that uses 280 // WebRtcVoiceMediaChannel is an implementation of VoiceMediaChannel that uses
279 // WebRtc Voice Engine. 281 // WebRtc Voice Engine.
280 class WebRtcVoiceMediaChannel : public VoiceMediaChannel, 282 class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
281 public webrtc::Transport { 283 public webrtc::Transport {
282 public: 284 public:
283 explicit WebRtcVoiceMediaChannel(WebRtcVoiceEngine *engine); 285 explicit WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine,
286 webrtc::Call* call);
284 ~WebRtcVoiceMediaChannel() override; 287 ~WebRtcVoiceMediaChannel() override;
285 288
286 int voe_channel() const { return voe_channel_; } 289 int voe_channel() const { return voe_channel_; }
287 bool valid() const { return voe_channel_ != -1; } 290 bool valid() const { return voe_channel_ != -1; }
288 291
289 bool SetSendParameters(const AudioSendParameters& params) override; 292 bool SetSendParameters(const AudioSendParameters& params) override;
290 bool SetRecvParameters(const AudioRecvParameters& params) override; 293 bool SetRecvParameters(const AudioRecvParameters& params) override;
291 bool SetOptions(const AudioOptions& options) override; 294 bool SetOptions(const AudioOptions& options) override;
292 bool GetOptions(AudioOptions* options) const override { 295 bool GetOptions(AudioOptions* options) const override {
293 *options = options_; 296 *options = options_;
(...skipping 59 matching lines...) Expand 10 before | Expand all | Expand 10 after
353 return VoiceMediaChannel::SendRtcp(&packet) ? static_cast<int>(len) : -1; 356 return VoiceMediaChannel::SendRtcp(&packet) ? static_cast<int>(len) : -1;
354 } 357 }
355 358
356 bool FindSsrc(int channel_num, uint32* ssrc); 359 bool FindSsrc(int channel_num, uint32* ssrc);
357 void OnError(uint32 ssrc, int error); 360 void OnError(uint32 ssrc, int error);
358 361
359 bool sending() const { return send_ != SEND_NOTHING; } 362 bool sending() const { return send_ != SEND_NOTHING; }
360 int GetReceiveChannelNum(uint32 ssrc) const; 363 int GetReceiveChannelNum(uint32 ssrc) const;
361 int GetSendChannelNum(uint32 ssrc) const; 364 int GetSendChannelNum(uint32 ssrc) const;
362 365
363 void SetCall(webrtc::Call* call);
364
365 private: 366 private:
366 WebRtcVoiceEngine* engine() { return engine_; } 367 WebRtcVoiceEngine* engine() { return engine_; }
367 int GetLastEngineError() { return engine()->GetLastEngineError(); } 368 int GetLastEngineError() { return engine()->GetLastEngineError(); }
368 int GetOutputLevel(int channel); 369 int GetOutputLevel(int channel);
369 bool GetRedSendCodec(const AudioCodec& red_codec, 370 bool GetRedSendCodec(const AudioCodec& red_codec,
370 const std::vector<AudioCodec>& all_codecs, 371 const std::vector<AudioCodec>& all_codecs,
371 webrtc::CodecInst* send_codec); 372 webrtc::CodecInst* send_codec);
372 bool EnableRtcp(int channel); 373 bool EnableRtcp(int channel);
373 bool ResetRecvCodecs(int channel); 374 bool ResetRecvCodecs(int channel);
374 bool SetPlayout(int channel, bool playout); 375 bool SetPlayout(int channel, bool playout);
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397 return options_.conference_mode.GetWithDefaultIfUnset(false); 398 return options_.conference_mode.GetWithDefaultIfUnset(false);
398 } 399 }
399 bool IsDefaultChannel(int channel_id) const { 400 bool IsDefaultChannel(int channel_id) const {
400 return channel_id == voe_channel(); 401 return channel_id == voe_channel();
401 } 402 }
402 bool SetSendCodecs(int channel, const std::vector<AudioCodec>& codecs); 403 bool SetSendCodecs(int channel, const std::vector<AudioCodec>& codecs);
403 bool SetSendBitrateInternal(int bps); 404 bool SetSendBitrateInternal(int bps);
404 405
405 bool SetHeaderExtension(ExtensionSetterFunction setter, int channel_id, 406 bool SetHeaderExtension(ExtensionSetterFunction setter, int channel_id,
406 const RtpHeaderExtension* extension); 407 const RtpHeaderExtension* extension);
407 void TryAddAudioRecvStream(uint32 ssrc); 408 void RecreateCallStreams();
408 void TryRemoveAudioRecvStream(uint32 ssrc); 409 void AddAudioRecvStream(uint32 ssrc);
410 void RemoveAudioRecvStream(uint32 ssrc);
409 bool SetRecvCodecsInternal(const std::vector<AudioCodec>& new_codecs); 411 bool SetRecvCodecsInternal(const std::vector<AudioCodec>& new_codecs);
410 412
411 bool SetChannelRecvRtpHeaderExtensions( 413 bool SetChannelRecvRtpHeaderExtensions(
412 int channel_id, 414 int channel_id,
413 const std::vector<RtpHeaderExtension>& extensions); 415 const std::vector<RtpHeaderExtension>& extensions);
414 bool SetChannelSendRtpHeaderExtensions( 416 bool SetChannelSendRtpHeaderExtensions(
415 int channel_id, 417 int channel_id,
416 const std::vector<RtpHeaderExtension>& extensions); 418 const std::vector<RtpHeaderExtension>& extensions);
417 419
418 rtc::ThreadChecker thread_checker_; 420 rtc::ThreadChecker thread_checker_;
(...skipping 36 matching lines...) Expand 10 before | Expand all | Expand 10 after
455 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; 457 std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
456 458
457 // Do not lock this on the VoE media processor thread; potential for deadlock 459 // Do not lock this on the VoE media processor thread; potential for deadlock
458 // exists. 460 // exists.
459 mutable rtc::CriticalSection receive_channels_cs_; 461 mutable rtc::CriticalSection receive_channels_cs_;
460 }; 462 };
461 463
462 } // namespace cricket 464 } // namespace cricket
463 465
464 #endif // TALK_MEDIA_WEBRTCVOICEENGINE_H_ 466 #endif // TALK_MEDIA_WEBRTCVOICEENGINE_H_
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