Make LoadObserver settable per video send stream. Gives client flexibility and makes the implementa…

webrtc/video/call.cc

 /*
  *  Copyright (c) 2013 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 23 matching lines @@
 #include "webrtc/video/audio_receive_stream.h"
 #include "webrtc/video/video_receive_stream.h"
 #include "webrtc/video/video_send_stream.h"
 
 namespace webrtc {
 
 const int Call::Config::kDefaultStartBitrateBps = 300000;
 
 namespace internal {
 
-class CpuOveruseObserverProxy : public webrtc::CpuOveruseObserver {
- public:
-  explicit CpuOveruseObserverProxy(LoadObserver* overuse_callback)
-      : overuse_callback_(overuse_callback) {
-    DCHECK(overuse_callback != nullptr);
-  }
-
-  virtual ~CpuOveruseObserverProxy() {}
-
-  void OveruseDetected() override {
-    rtc::CritScope lock(&crit_);
-    overuse_callback_->OnLoadUpdate(LoadObserver::kOveruse);
-  }
-
-  void NormalUsage() override {
-    rtc::CritScope lock(&crit_);
-    overuse_callback_->OnLoadUpdate(LoadObserver::kUnderuse);
-  }
-
- private:
-  rtc::CriticalSection crit_;
-  LoadObserver* overuse_callback_ GUARDED_BY(crit_);
-};
-
 class Call : public webrtc::Call, public PacketReceiver {
  public:
   explicit Call(const Call::Config& config);
   virtual ~Call();
 
   PacketReceiver* Receiver() override;
 
   webrtc::AudioSendStream* CreateAudioSendStream(
       const webrtc::AudioSendStream::Config& config) override;
   void DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) override;
@@ skipping 53 matching lines @@
       GUARDED_BY(receive_crit_);
   std::set<VideoReceiveStream*> video_receive_streams_
       GUARDED_BY(receive_crit_);
   std::map<std::string, AudioReceiveStream*> sync_stream_mapping_
       GUARDED_BY(receive_crit_);
 
   rtc::scoped_ptr<RWLockWrapper> send_crit_;
   std::map<uint32_t, VideoSendStream*> video_send_ssrcs_ GUARDED_BY(send_crit_);
   std::set<VideoSendStream*> video_send_streams_ GUARDED_BY(send_crit_);
 
-  rtc::scoped_ptr<CpuOveruseObserverProxy> overuse_observer_proxy_;
-
   VideoSendStream::RtpStateMap suspended_video_send_ssrcs_;
 
   DISALLOW_COPY_AND_ASSIGN(Call);
 };
 }  // namespace internal
 
 Call* Call::Create(const Call::Config& config) {
   return new internal::Call(config);
 }
 
@@ skipping 12 matching lines @@
   DCHECK_GE(config.bitrate_config.start_bitrate_bps,
             config.bitrate_config.min_bitrate_bps);
   if (config.bitrate_config.max_bitrate_bps != -1) {
     DCHECK_GE(config.bitrate_config.max_bitrate_bps,
               config.bitrate_config.start_bitrate_bps);
   }
 
   Trace::CreateTrace();
   module_process_thread_->Start();
 
-  if (config.overuse_callback) {
-    overuse_observer_proxy_.reset(
-        new CpuOveruseObserverProxy(config.overuse_callback));
-  }
-
   SetBitrateControllerConfig(config_.bitrate_config);
 }
 
 Call::~Call() {
   CHECK_EQ(0u, video_send_ssrcs_.size());
   CHECK_EQ(0u, video_send_streams_.size());
   CHECK_EQ(0u, audio_receive_ssrcs_.size());
   CHECK_EQ(0u, video_receive_ssrcs_.size());
   CHECK_EQ(0u, video_receive_streams_.size());
 
@@ skipping 51 matching lines @@
 
 webrtc::VideoSendStream* Call::CreateVideoSendStream(
     const webrtc::VideoSendStream::Config& config,
     const VideoEncoderConfig& encoder_config) {
   TRACE_EVENT0("webrtc", "Call::CreateVideoSendStream");
   LOG(LS_INFO) << "CreateVideoSendStream: " << config.ToString();
   DCHECK(!config.rtp.ssrcs.empty());
 
   // TODO(mflodman): Base the start bitrate on a current bandwidth estimate, if
   // the call has already started.
-  VideoSendStream* send_stream = new VideoSendStream(
-      overuse_observer_proxy_.get(), num_cpu_cores_,
+  VideoSendStream* send_stream = new VideoSendStream(num_cpu_cores_,
       module_process_thread_.get(), channel_group_.get(),
       rtc::AtomicOps::Increment(&next_channel_id_), config, encoder_config,
       suspended_video_send_ssrcs_);
 
   // This needs to be taken before send_crit_ as both locks need to be held
   // while changing network state.
   rtc::CritScope lock(&network_enabled_crit_);
   WriteLockScoped write_lock(*send_crit_);
   for (uint32_t ssrc : config.rtp.ssrcs) {
     DCHECK(video_send_ssrcs_.find(ssrc) == video_send_ssrcs_.end());
@@ skipping 273 matching lines @@
                                                    const uint8_t* packet,
                                                    size_t length) {
   if (RtpHeaderParser::IsRtcp(packet, length))
     return DeliverRtcp(media_type, packet, length);
 
   return DeliverRtp(media_type, packet, length);
 }
 
 }  // namespace internal
 }  // namespace webrtc