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Side by Side Diff: talk/session/media/channel.h

Issue 1327033002: Remove unused TypingMonitor class. (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 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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46 #include "webrtc/base/asyncudpsocket.h" 46 #include "webrtc/base/asyncudpsocket.h"
47 #include "webrtc/base/criticalsection.h" 47 #include "webrtc/base/criticalsection.h"
48 #include "webrtc/base/network.h" 48 #include "webrtc/base/network.h"
49 #include "webrtc/base/sigslot.h" 49 #include "webrtc/base/sigslot.h"
50 #include "webrtc/base/window.h" 50 #include "webrtc/base/window.h"
51 51
52 namespace cricket { 52 namespace cricket {
53 53
54 struct CryptoParams; 54 struct CryptoParams;
55 class MediaContentDescription; 55 class MediaContentDescription;
56 struct TypingMonitorOptions;
57 class TypingMonitor;
58 struct ViewRequest; 56 struct ViewRequest;
59 57
60 enum SinkType { 58 enum SinkType {
61 SINK_PRE_CRYPTO, // Sink packets before encryption or after decryption. 59 SINK_PRE_CRYPTO, // Sink packets before encryption or after decryption.
62 SINK_POST_CRYPTO // Sink packets after encryption or before decryption. 60 SINK_POST_CRYPTO // Sink packets after encryption or before decryption.
63 }; 61 };
64 62
65 // BaseChannel contains logic common to voice and video, including 63 // BaseChannel contains logic common to voice and video, including
66 // enable/mute, marshaling calls to a worker thread, and 64 // enable/mute, marshaling calls to a worker thread, and
67 // connection and media monitors. 65 // connection and media monitors.
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156 return remote_streams_; 154 return remote_streams_;
157 } 155 }
158 156
159 sigslot::signal2<BaseChannel*, bool> SignalDtlsSetupFailure; 157 sigslot::signal2<BaseChannel*, bool> SignalDtlsSetupFailure;
160 void SignalDtlsSetupFailure_w(bool rtcp); 158 void SignalDtlsSetupFailure_w(bool rtcp);
161 void SignalDtlsSetupFailure_s(bool rtcp); 159 void SignalDtlsSetupFailure_s(bool rtcp);
162 160
163 // Used for latency measurements. 161 // Used for latency measurements.
164 sigslot::signal1<BaseChannel*> SignalFirstPacketReceived; 162 sigslot::signal1<BaseChannel*> SignalFirstPacketReceived;
165 163
166 // Used to alert UI when the muted status changes, perhaps autonomously.
167 sigslot::repeater2<BaseChannel*, bool> SignalAutoMuted;
168
169 // Made public for easier testing. 164 // Made public for easier testing.
170 void SetReadyToSend(TransportChannel* channel, bool ready); 165 void SetReadyToSend(TransportChannel* channel, bool ready);
171 166
172 // Only public for unit tests. Otherwise, consider protected. 167 // Only public for unit tests. Otherwise, consider protected.
173 virtual int SetOption(SocketType type, rtc::Socket::Option o, int val); 168 virtual int SetOption(SocketType type, rtc::Socket::Option o, int val);
174 169
175 protected: 170 protected:
176 virtual MediaChannel* media_channel() const { return media_channel_; } 171 virtual MediaChannel* media_channel() const { return media_channel_; }
177 // Sets the transport_channel_ and rtcp_transport_channel_. If 172 // Sets the transport_channel_ and rtcp_transport_channel_. If
178 // |rtcp| is false, set rtcp_transport_channel_ is set to NULL. Get 173 // |rtcp| is false, set rtcp_transport_channel_ is set to NULL. Get
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225 virtual bool WantsPacket(bool rtcp, rtc::Buffer* packet); 220 virtual bool WantsPacket(bool rtcp, rtc::Buffer* packet);
226 void HandlePacket(bool rtcp, rtc::Buffer* packet, 221 void HandlePacket(bool rtcp, rtc::Buffer* packet,
227 const rtc::PacketTime& packet_time); 222 const rtc::PacketTime& packet_time);
228 223
229 // Apply the new local/remote session description. 224 // Apply the new local/remote session description.
230 void OnNewLocalDescription(BaseSession* session, ContentAction action); 225 void OnNewLocalDescription(BaseSession* session, ContentAction action);
231 void OnNewRemoteDescription(BaseSession* session, ContentAction action); 226 void OnNewRemoteDescription(BaseSession* session, ContentAction action);
232 227
233 void EnableMedia_w(); 228 void EnableMedia_w();
234 void DisableMedia_w(); 229 void DisableMedia_w();
235 virtual bool MuteStream_w(uint32 ssrc, bool mute); 230 bool MuteStream_w(uint32 ssrc, bool mute);
236 bool IsStreamMuted_w(uint32 ssrc); 231 bool IsStreamMuted_w(uint32 ssrc);
237 void ChannelWritable_w(); 232 void ChannelWritable_w();
238 void ChannelNotWritable_w(); 233 void ChannelNotWritable_w();
239 bool AddRecvStream_w(const StreamParams& sp); 234 bool AddRecvStream_w(const StreamParams& sp);
240 bool RemoveRecvStream_w(uint32 ssrc); 235 bool RemoveRecvStream_w(uint32 ssrc);
241 bool AddSendStream_w(const StreamParams& sp); 236 bool AddSendStream_w(const StreamParams& sp);
242 bool RemoveSendStream_w(uint32 ssrc); 237 bool RemoveSendStream_w(uint32 ssrc);
243 virtual bool ShouldSetupDtlsSrtp() const; 238 virtual bool ShouldSetupDtlsSrtp() const;
244 // Do the DTLS key expansion and impose it on the SRTP/SRTCP filters. 239 // Do the DTLS key expansion and impose it on the SRTP/SRTCP filters.
245 // |rtcp_channel| indicates whether to set up the RTP or RTCP filter. 240 // |rtcp_channel| indicates whether to set up the RTP or RTCP filter.
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376 371
377 void StartMediaMonitor(int cms); 372 void StartMediaMonitor(int cms);
378 void StopMediaMonitor(); 373 void StopMediaMonitor();
379 sigslot::signal2<VoiceChannel*, const VoiceMediaInfo&> SignalMediaMonitor; 374 sigslot::signal2<VoiceChannel*, const VoiceMediaInfo&> SignalMediaMonitor;
380 375
381 void StartAudioMonitor(int cms); 376 void StartAudioMonitor(int cms);
382 void StopAudioMonitor(); 377 void StopAudioMonitor();
383 bool IsAudioMonitorRunning() const; 378 bool IsAudioMonitorRunning() const;
384 sigslot::signal2<VoiceChannel*, const AudioInfo&> SignalAudioMonitor; 379 sigslot::signal2<VoiceChannel*, const AudioInfo&> SignalAudioMonitor;
385 380
386 void StartTypingMonitor(const TypingMonitorOptions& settings);
387 void StopTypingMonitor();
388 bool IsTypingMonitorRunning() const;
389
390 // Overrides BaseChannel::MuteStream_w.
391 virtual bool MuteStream_w(uint32 ssrc, bool mute);
392
393 int GetInputLevel_w(); 381 int GetInputLevel_w();
394 int GetOutputLevel_w(); 382 int GetOutputLevel_w();
395 void GetActiveStreams_w(AudioInfo::StreamList* actives); 383 void GetActiveStreams_w(AudioInfo::StreamList* actives);
396 384
397 // Signal errors from VoiceMediaChannel. Arguments are: 385 // Signal errors from VoiceMediaChannel. Arguments are:
398 // ssrc(uint32), and error(VoiceMediaChannel::Error). 386 // ssrc(uint32), and error(VoiceMediaChannel::Error).
399 sigslot::signal3<VoiceChannel*, uint32, VoiceMediaChannel::Error> 387 sigslot::signal3<VoiceChannel*, uint32, VoiceMediaChannel::Error>
400 SignalMediaError; 388 SignalMediaError;
401 389
402 // Configuration and setting. 390 // Configuration and setting.
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432 void OnAudioMonitorUpdate(AudioMonitor* monitor, const AudioInfo& info); 420 void OnAudioMonitorUpdate(AudioMonitor* monitor, const AudioInfo& info);
433 void OnVoiceChannelError(uint32 ssrc, VoiceMediaChannel::Error error); 421 void OnVoiceChannelError(uint32 ssrc, VoiceMediaChannel::Error error);
434 void SendLastMediaError(); 422 void SendLastMediaError();
435 void OnSrtpError(uint32 ssrc, SrtpFilter::Mode mode, SrtpFilter::Error error); 423 void OnSrtpError(uint32 ssrc, SrtpFilter::Mode mode, SrtpFilter::Error error);
436 424
437 static const int kEarlyMediaTimeout = 1000; 425 static const int kEarlyMediaTimeout = 1000;
438 MediaEngineInterface* media_engine_; 426 MediaEngineInterface* media_engine_;
439 bool received_media_; 427 bool received_media_;
440 rtc::scoped_ptr<VoiceMediaMonitor> media_monitor_; 428 rtc::scoped_ptr<VoiceMediaMonitor> media_monitor_;
441 rtc::scoped_ptr<AudioMonitor> audio_monitor_; 429 rtc::scoped_ptr<AudioMonitor> audio_monitor_;
442 rtc::scoped_ptr<TypingMonitor> typing_monitor_;
443 430
444 // Last AudioSendParameters sent down to the media_channel() via 431 // Last AudioSendParameters sent down to the media_channel() via
445 // SetSendParameters. 432 // SetSendParameters.
446 AudioSendParameters last_send_params_; 433 AudioSendParameters last_send_params_;
447 // Last AudioRecvParameters sent down to the media_channel() via 434 // Last AudioRecvParameters sent down to the media_channel() via
448 // SetRecvParameters. 435 // SetRecvParameters.
449 AudioRecvParameters last_recv_params_; 436 AudioRecvParameters last_recv_params_;
450 }; 437 };
451 438
452 // VideoChannel is a specialization for video. 439 // VideoChannel is a specialization for video.
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667 // SetSendParameters. 654 // SetSendParameters.
668 DataSendParameters last_send_params_; 655 DataSendParameters last_send_params_;
669 // Last DataRecvParameters sent down to the media_channel() via 656 // Last DataRecvParameters sent down to the media_channel() via
670 // SetRecvParameters. 657 // SetRecvParameters.
671 DataRecvParameters last_recv_params_; 658 DataRecvParameters last_recv_params_;
672 }; 659 };
673 660
674 } // namespace cricket 661 } // namespace cricket
675 662
676 #endif // TALK_SESSION_MEDIA_CHANNEL_H_ 663 #endif // TALK_SESSION_MEDIA_CHANNEL_H_
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