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

Issue 1386653002: Remove default receive channel from WVoE; baby step 0. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 5 years, 2 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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120 void SetTraceOptions(const std::string& options); 120 void SetTraceOptions(const std::string& options);
121 // Every option that is "set" will be applied. Every option not "set" will be 121 // Every option that is "set" will be applied. Every option not "set" will be
122 // ignored. This allows us to selectively turn on and off different options 122 // ignored. This allows us to selectively turn on and off different options
123 // easily at any time. 123 // easily at any time.
124 bool ApplyOptions(const AudioOptions& options); 124 bool ApplyOptions(const AudioOptions& options);
125 125
126 // webrtc::TraceCallback: 126 // webrtc::TraceCallback:
127 void Print(webrtc::TraceLevel level, const char* trace, int length) override; 127 void Print(webrtc::TraceLevel level, const char* trace, int length) override;
128 128
129 // webrtc::VoiceEngineObserver: 129 // webrtc::VoiceEngineObserver:
130 void CallbackOnError(int channel, int errCode) override; 130 void CallbackOnError(int channel_id, int errCode) override;
131 131
132 // Given the device type, name, and id, find device id. Return true and 132 // Given the device type, name, and id, find device id. Return true and
133 // set the output parameter rtc_id if successful. 133 // set the output parameter rtc_id if successful.
134 bool FindWebRtcAudioDeviceId( 134 bool FindWebRtcAudioDeviceId(
135 bool is_input, const std::string& dev_name, int dev_id, int* rtc_id); 135 bool is_input, const std::string& dev_name, int dev_id, int* rtc_id);
136 bool FindChannelAndSsrc(int channel_num,
137 WebRtcVoiceMediaChannel** channel,
138 uint32* ssrc) const;
139 136
140 void StartAecDump(const std::string& filename); 137 void StartAecDump(const std::string& filename);
141 void StopAecDump(); 138 void StopAecDump();
142 int CreateVoiceChannel(VoEWrapper* voe); 139 int CreateVoiceChannel(VoEWrapper* voe);
143 140
144 static const int kDefaultLogSeverity = rtc::LS_WARNING; 141 static const int kDefaultLogSeverity = rtc::LS_WARNING;
145 142
146 // The primary instance of WebRtc VoiceEngine. 143 // The primary instance of WebRtc VoiceEngine.
147 rtc::scoped_ptr<VoEWrapper> voe_wrapper_; 144 rtc::scoped_ptr<VoEWrapper> voe_wrapper_;
148 rtc::scoped_ptr<VoETraceWrapper> tracing_; 145 rtc::scoped_ptr<VoETraceWrapper> tracing_;
(...skipping 81 matching lines...) Expand 10 before | Expand all | Expand 10 after
230 kMaxRtpPacketLen); 227 kMaxRtpPacketLen);
231 return VoiceMediaChannel::SendPacket(&packet); 228 return VoiceMediaChannel::SendPacket(&packet);
232 } 229 }
233 230
234 bool SendRtcp(const uint8_t* data, size_t len) override { 231 bool SendRtcp(const uint8_t* data, size_t len) override {
235 rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len, 232 rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len,
236 kMaxRtpPacketLen); 233 kMaxRtpPacketLen);
237 return VoiceMediaChannel::SendRtcp(&packet); 234 return VoiceMediaChannel::SendRtcp(&packet);
238 } 235 }
239 236
240 bool FindSsrc(int channel_num, uint32* ssrc); 237 void OnError(int error);
241 void OnError(uint32 ssrc, int error);
242 238
243 int GetReceiveChannelNum(uint32 ssrc) const; 239 int GetReceiveChannelId(uint32 ssrc) const;
244 int GetSendChannelNum(uint32 ssrc) const; 240 int GetSendChannelId(uint32 ssrc) const;
245 241
246 private: 242 private:
247 bool SetSendCodecs(const std::vector<AudioCodec>& codecs); 243 bool SetSendCodecs(const std::vector<AudioCodec>& codecs);
248 bool SetSendRtpHeaderExtensions( 244 bool SetSendRtpHeaderExtensions(
249 const std::vector<RtpHeaderExtension>& extensions); 245 const std::vector<RtpHeaderExtension>& extensions);
250 bool SetOptions(const AudioOptions& options); 246 bool SetOptions(const AudioOptions& options);
251 bool SetMaxSendBandwidth(int bps); 247 bool SetMaxSendBandwidth(int bps);
252 bool SetRecvCodecs(const std::vector<AudioCodec>& codecs); 248 bool SetRecvCodecs(const std::vector<AudioCodec>& codecs);
253 bool SetRecvRtpHeaderExtensions( 249 bool SetRecvRtpHeaderExtensions(
254 const std::vector<RtpHeaderExtension>& extensions); 250 const std::vector<RtpHeaderExtension>& extensions);
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346 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; 342 std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
347 343
348 // Do not lock this on the VoE media processor thread; potential for deadlock 344 // Do not lock this on the VoE media processor thread; potential for deadlock
349 // exists. 345 // exists.
350 mutable rtc::CriticalSection receive_channels_cs_; 346 mutable rtc::CriticalSection receive_channels_cs_;
351 }; 347 };
352 348
353 } // namespace cricket 349 } // namespace cricket
354 350
355 #endif // TALK_MEDIA_WEBRTCVOICEENGINE_H_ 351 #endif // TALK_MEDIA_WEBRTCVOICEENGINE_H_
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