Removing "crypto_required" from MediaContentDescription.

webrtc/pc/channelmanager.cc

 /*
  *  Copyright 2004 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 200 matching lines @@
     DestroyVoiceChannel_w(voice_channels_.back());
   }
 }
 
 VoiceChannel* ChannelManager::CreateVoiceChannel(
     webrtc::MediaControllerInterface* media_controller,
     TransportController* transport_controller,
     const std::string& content_name,
     const std::string* bundle_transport_name,
     bool rtcp,
+    bool srtp_required,
     const AudioOptions& options) {
   return worker_thread_->Invoke<VoiceChannel*>(
       RTC_FROM_HERE, Bind(&ChannelManager::CreateVoiceChannel_w, this,
                           media_controller, transport_controller, content_name,
-                          bundle_transport_name, rtcp, options));
+                          bundle_transport_name, rtcp, srtp_required, options));
 }
 
 VoiceChannel* ChannelManager::CreateVoiceChannel_w(
     webrtc::MediaControllerInterface* media_controller,
     TransportController* transport_controller,
     const std::string& content_name,
     const std::string* bundle_transport_name,
     bool rtcp,
+    bool srtp_required,
     const AudioOptions& options) {
   ASSERT(initialized_);
   ASSERT(worker_thread_ == rtc::Thread::Current());
   ASSERT(nullptr != media_controller);
   VoiceMediaChannel* media_channel = media_engine_->CreateChannel(
       media_controller->call_w(), media_controller->config(), options);
   if (!media_channel)
     return nullptr;
 
-  VoiceChannel* voice_channel =
-      new VoiceChannel(worker_thread_, network_thread_, media_engine_.get(),
-                       media_channel, transport_controller, content_name, rtcp);
+  VoiceChannel* voice_channel = new VoiceChannel(
+      worker_thread_, network_thread_, media_engine_.get(), media_channel,
+      transport_controller, content_name, rtcp, srtp_required);
   voice_channel->SetCryptoOptions(crypto_options_);
   if (!voice_channel->Init_w(bundle_transport_name)) {
     delete voice_channel;
     return nullptr;
   }
   voice_channels_.push_back(voice_channel);
   return voice_channel;
 }
 
 void ChannelManager::DestroyVoiceChannel(VoiceChannel* voice_channel) {
@@ skipping 18 matching lines @@
   voice_channels_.erase(it);
   delete voice_channel;
 }
 
 VideoChannel* ChannelManager::CreateVideoChannel(
     webrtc::MediaControllerInterface* media_controller,
     TransportController* transport_controller,
     const std::string& content_name,
     const std::string* bundle_transport_name,
     bool rtcp,
+    bool srtp_required,
     const VideoOptions& options) {
   return worker_thread_->Invoke<VideoChannel*>(
       RTC_FROM_HERE, Bind(&ChannelManager::CreateVideoChannel_w, this,
                           media_controller, transport_controller, content_name,
-                          bundle_transport_name, rtcp, options));
+                          bundle_transport_name, rtcp, srtp_required, options));
 }
 
 VideoChannel* ChannelManager::CreateVideoChannel_w(
     webrtc::MediaControllerInterface* media_controller,
     TransportController* transport_controller,
     const std::string& content_name,
     const std::string* bundle_transport_name,
     bool rtcp,
+    bool srtp_required,
     const VideoOptions& options) {
   ASSERT(initialized_);
   ASSERT(worker_thread_ == rtc::Thread::Current());
   ASSERT(nullptr != media_controller);
   VideoMediaChannel* media_channel = media_engine_->CreateVideoChannel(
       media_controller->call_w(), media_controller->config(), options);
   if (media_channel == NULL) {
     return NULL;
   }
 
   VideoChannel* video_channel =
       new VideoChannel(worker_thread_, network_thread_, media_channel,
-                       transport_controller, content_name, rtcp);
+                       transport_controller, content_name, rtcp, srtp_required);
   video_channel->SetCryptoOptions(crypto_options_);
   if (!video_channel->Init_w(bundle_transport_name)) {
     delete video_channel;
     return NULL;
   }
   video_channels_.push_back(video_channel);
   return video_channel;
 }
 
 void ChannelManager::DestroyVideoChannel(VideoChannel* video_channel) {
@@ skipping 14 matching lines @@
       video_channels_.end(), video_channel);
   ASSERT(it != video_channels_.end());
   if (it == video_channels_.end())
     return;
 
   video_channels_.erase(it);
   delete video_channel;
 }
 
 DataChannel* ChannelManager::CreateDataChannel(
+    webrtc::MediaControllerInterface* media_controller,
     TransportController* transport_controller,
     const std::string& content_name,
     const std::string* bundle_transport_name,
     bool rtcp,
-    DataChannelType channel_type) {
-  return CreateDataChannel(transport_controller, nullptr, content_name,
-                           bundle_transport_name, rtcp, channel_type);
-}
-
-DataChannel* ChannelManager::CreateDataChannel(
-    TransportController* transport_controller,
-    webrtc::MediaControllerInterface* media_controller,
-    const std::string& content_name,
-    const std::string* bundle_transport_name,
-    bool rtcp,
+    bool srtp_required,
     DataChannelType channel_type) {
   return worker_thread_->Invoke<DataChannel*>(
       RTC_FROM_HERE,
-      Bind(&ChannelManager::CreateDataChannel_w, this, transport_controller,
-           content_name, bundle_transport_name, rtcp, channel_type,
-           media_controller));
+      Bind(&ChannelManager::CreateDataChannel_w, this, media_controller,
+           transport_controller, content_name, bundle_transport_name, rtcp,
+           srtp_required, channel_type));
 }
 
 DataChannel* ChannelManager::CreateDataChannel_w(
+    webrtc::MediaControllerInterface* media_controller,
     TransportController* transport_controller,
     const std::string& content_name,
     const std::string* bundle_transport_name,
     bool rtcp,
-    DataChannelType data_channel_type,
-    webrtc::MediaControllerInterface* media_controller) {
+    bool srtp_required,
+    DataChannelType data_channel_type) {
   // This is ok to alloc from a thread other than the worker thread.
   ASSERT(initialized_);
   MediaConfig config;
   if (media_controller) {
     config = media_controller->config();
   }
   DataMediaChannel* media_channel =
       data_media_engine_->CreateChannel(data_channel_type, config);
   if (!media_channel) {
     LOG(LS_WARNING) << "Failed to create data channel of type "
                     << data_channel_type;
     return NULL;
   }
 
+  // Only RTP data channels need SRTP.
+  srtp_required = srtp_required && data_channel_type == DCT_RTP;
   DataChannel* data_channel =
       new DataChannel(worker_thread_, network_thread_, media_channel,
-                      transport_controller, content_name, rtcp);
+                      transport_controller, content_name, rtcp, srtp_required);
   data_channel->SetCryptoOptions(crypto_options_);
   if (!data_channel->Init_w(bundle_transport_name)) {
     LOG(LS_WARNING) << "Failed to init data channel.";
     delete data_channel;
     return NULL;
   }
   data_channels_.push_back(data_channel);
   return data_channel;
 }
 
@@ skipping 27 matching lines @@
                           media_engine_.get(), file, max_size_bytes));
 }
 
 void ChannelManager::StopAecDump() {
   worker_thread_->Invoke<void>(
       RTC_FROM_HERE,
       Bind(&MediaEngineInterface::StopAecDump, media_engine_.get()));
 }
 
 }  // namespace cricket