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|    1 /* |    1 /* | 
|    2  *  Copyright 2004 The WebRTC project authors. All Rights Reserved. |    2  *  Copyright 2004 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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|  103   VideoChannel* CreateVideoChannel( |  103   VideoChannel* CreateVideoChannel( | 
|  104       webrtc::MediaControllerInterface* media_controller, |  104       webrtc::MediaControllerInterface* media_controller, | 
|  105       TransportController* transport_controller, |  105       TransportController* transport_controller, | 
|  106       const std::string& content_name, |  106       const std::string& content_name, | 
|  107       const std::string* bundle_transport_name, |  107       const std::string* bundle_transport_name, | 
|  108       bool rtcp, |  108       bool rtcp, | 
|  109       bool srtp_required, |  109       bool srtp_required, | 
|  110       const VideoOptions& options); |  110       const VideoOptions& options); | 
|  111   // Destroys a video channel created with the Create API. |  111   // Destroys a video channel created with the Create API. | 
|  112   void DestroyVideoChannel(VideoChannel* video_channel); |  112   void DestroyVideoChannel(VideoChannel* video_channel); | 
|  113   DataChannel* CreateDataChannel( |  113   RtpDataChannel* CreateRtpDataChannel( | 
|  114       webrtc::MediaControllerInterface* media_controller, |  114       webrtc::MediaControllerInterface* media_controller, | 
|  115       TransportController* transport_controller, |  115       TransportController* transport_controller, | 
|  116       const std::string& content_name, |  116       const std::string& content_name, | 
|  117       const std::string* bundle_transport_name, |  117       const std::string* bundle_transport_name, | 
|  118       bool rtcp, |  118       bool rtcp, | 
|  119       bool srtp_required, |  119       bool srtp_required); | 
|  120       DataChannelType data_channel_type); |  | 
|  121   // Destroys a data channel created with the Create API. |  120   // Destroys a data channel created with the Create API. | 
|  122   void DestroyDataChannel(DataChannel* data_channel); |  121   void DestroyRtpDataChannel(RtpDataChannel* data_channel); | 
|  123  |  122  | 
|  124   // Indicates whether any channels exist. |  123   // Indicates whether any channels exist. | 
|  125   bool has_channels() const { |  124   bool has_channels() const { | 
|  126     return (!voice_channels_.empty() || !video_channels_.empty()); |  125     return (!voice_channels_.empty() || !video_channels_.empty()); | 
|  127   } |  126   } | 
|  128  |  127  | 
|  129   // RTX will be enabled/disabled in engines that support it. The supporting |  128   // RTX will be enabled/disabled in engines that support it. The supporting | 
|  130   // engines will start offering an RTX codec. Must be called before Init(). |  129   // engines will start offering an RTX codec. Must be called before Init(). | 
|  131   bool SetVideoRtxEnabled(bool enable); |  130   bool SetVideoRtxEnabled(bool enable); | 
|  132  |  131  | 
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|  143   // bytes. When the limit is reached, logging will stop and the file will be |  142   // bytes. When the limit is reached, logging will stop and the file will be | 
|  144   // closed. If max_size_bytes is set to <= 0, no limit will be used. |  143   // closed. If max_size_bytes is set to <= 0, no limit will be used. | 
|  145   bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes); |  144   bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes); | 
|  146  |  145  | 
|  147   // Stops recording AEC dump. |  146   // Stops recording AEC dump. | 
|  148   void StopAecDump(); |  147   void StopAecDump(); | 
|  149  |  148  | 
|  150  private: |  149  private: | 
|  151   typedef std::vector<VoiceChannel*> VoiceChannels; |  150   typedef std::vector<VoiceChannel*> VoiceChannels; | 
|  152   typedef std::vector<VideoChannel*> VideoChannels; |  151   typedef std::vector<VideoChannel*> VideoChannels; | 
|  153   typedef std::vector<DataChannel*> DataChannels; |  152   typedef std::vector<RtpDataChannel*> RtpDataChannels; | 
|  154  |  153  | 
|  155   void Construct(MediaEngineInterface* me, |  154   void Construct(MediaEngineInterface* me, | 
|  156                  DataEngineInterface* dme, |  155                  DataEngineInterface* dme, | 
|  157                  rtc::Thread* worker_thread, |  156                  rtc::Thread* worker_thread, | 
|  158                  rtc::Thread* network_thread); |  157                  rtc::Thread* network_thread); | 
|  159   bool InitMediaEngine_w(); |  158   bool InitMediaEngine_w(); | 
|  160   void DestructorDeletes_w(); |  159   void DestructorDeletes_w(); | 
|  161   void Terminate_w(); |  160   void Terminate_w(); | 
|  162   bool SetCryptoOptions_w(const rtc::CryptoOptions& crypto_options); |  161   bool SetCryptoOptions_w(const rtc::CryptoOptions& crypto_options); | 
|  163   VoiceChannel* CreateVoiceChannel_w( |  162   VoiceChannel* CreateVoiceChannel_w( | 
|  164       webrtc::MediaControllerInterface* media_controller, |  163       webrtc::MediaControllerInterface* media_controller, | 
|  165       TransportController* transport_controller, |  164       TransportController* transport_controller, | 
|  166       const std::string& content_name, |  165       const std::string& content_name, | 
|  167       const std::string* bundle_transport_name, |  166       const std::string* bundle_transport_name, | 
|  168       bool rtcp, |  167       bool rtcp, | 
|  169       bool srtp_required, |  168       bool srtp_required, | 
|  170       const AudioOptions& options); |  169       const AudioOptions& options); | 
|  171   void DestroyVoiceChannel_w(VoiceChannel* voice_channel); |  170   void DestroyVoiceChannel_w(VoiceChannel* voice_channel); | 
|  172   VideoChannel* CreateVideoChannel_w( |  171   VideoChannel* CreateVideoChannel_w( | 
|  173       webrtc::MediaControllerInterface* media_controller, |  172       webrtc::MediaControllerInterface* media_controller, | 
|  174       TransportController* transport_controller, |  173       TransportController* transport_controller, | 
|  175       const std::string& content_name, |  174       const std::string& content_name, | 
|  176       const std::string* bundle_transport_name, |  175       const std::string* bundle_transport_name, | 
|  177       bool rtcp, |  176       bool rtcp, | 
|  178       bool srtp_required, |  177       bool srtp_required, | 
|  179       const VideoOptions& options); |  178       const VideoOptions& options); | 
|  180   void DestroyVideoChannel_w(VideoChannel* video_channel); |  179   void DestroyVideoChannel_w(VideoChannel* video_channel); | 
|  181   DataChannel* CreateDataChannel_w( |  180   RtpDataChannel* CreateRtpDataChannel_w( | 
|  182       webrtc::MediaControllerInterface* media_controller, |  181       webrtc::MediaControllerInterface* media_controller, | 
|  183       TransportController* transport_controller, |  182       TransportController* transport_controller, | 
|  184       const std::string& content_name, |  183       const std::string& content_name, | 
|  185       const std::string* bundle_transport_name, |  184       const std::string* bundle_transport_name, | 
|  186       bool rtcp, |  185       bool rtcp, | 
|  187       bool srtp_required, |  186       bool srtp_required); | 
|  188       DataChannelType data_channel_type); |  187   void DestroyRtpDataChannel_w(RtpDataChannel* data_channel); | 
|  189   void DestroyDataChannel_w(DataChannel* data_channel); |  | 
|  190  |  188  | 
|  191   std::unique_ptr<MediaEngineInterface> media_engine_; |  189   std::unique_ptr<MediaEngineInterface> media_engine_; | 
|  192   std::unique_ptr<DataEngineInterface> data_media_engine_; |  190   std::unique_ptr<DataEngineInterface> data_media_engine_; | 
|  193   bool initialized_; |  191   bool initialized_; | 
|  194   rtc::Thread* main_thread_; |  192   rtc::Thread* main_thread_; | 
|  195   rtc::Thread* worker_thread_; |  193   rtc::Thread* worker_thread_; | 
|  196   rtc::Thread* network_thread_; |  194   rtc::Thread* network_thread_; | 
|  197  |  195  | 
|  198   VoiceChannels voice_channels_; |  196   VoiceChannels voice_channels_; | 
|  199   VideoChannels video_channels_; |  197   VideoChannels video_channels_; | 
|  200   DataChannels data_channels_; |  198   RtpDataChannels data_channels_; | 
|  201  |  199  | 
|  202   bool enable_rtx_; |  200   bool enable_rtx_; | 
|  203   rtc::CryptoOptions crypto_options_; |  201   rtc::CryptoOptions crypto_options_; | 
|  204  |  202  | 
|  205   bool capturing_; |  203   bool capturing_; | 
|  206 }; |  204 }; | 
|  207  |  205  | 
|  208 }  // namespace cricket |  206 }  // namespace cricket | 
|  209  |  207  | 
|  210 #endif  // WEBRTC_PC_CHANNELMANAGER_H_ |  208 #endif  // WEBRTC_PC_CHANNELMANAGER_H_ | 
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