Add configuration for cpu overuse detection to video send stream.

BUG=2422
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5468 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/call.h

@@ -31,6 +31,19 @@ class PacketReceiver {
   virtual ~PacketReceiver() {}
 };
 
+// Callback interface for reporting when a system overuse is detected.
+// The detection is based on the jitter of incoming captured frames.
+class OveruseCallback {
+ public:
+  // Called as soon as an overuse is detected.
+  virtual void OnOveruse() = 0;
+  // Called periodically when the system is not overused any longer.
+  virtual void OnNormalUse() = 0;
+
+ protected:
+  virtual ~OveruseCallback() {}
+};
+
 // A Call instance can contain several send and/or receive streams. All streams
 // are assumed to have the same remote endpoint and will share bitrate estimates
 // etc.
@@ -40,21 +53,24 @@ class Call {
     explicit Config(newapi::Transport* send_transport)
         : webrtc_config(NULL),
           send_transport(send_transport),
-          overuse_detection(false),
           voice_engine(NULL),
           trace_callback(NULL),
-          trace_filter(kTraceDefault) {}
+          trace_filter(kTraceDefault),
+          overuse_callback(NULL) {}
 
     webrtc::Config* webrtc_config;
 
     newapi::Transport* send_transport;
-    bool overuse_detection;
 
     // VoiceEngine used for audio/video synchronization for this Call.
     VoiceEngine* voice_engine;
 
     TraceCallback* trace_callback;
     uint32_t trace_filter;
+
+    // Callback for overuse and normal usage based on the jitter of incoming
+    // captured frames. 'NULL' disables the callback.
+    OveruseCallback* overuse_callback;
   };
 
   static Call* Create(const Call::Config& config);

webrtc/video/call.cc

@@ -33,6 +33,32 @@ const char* RtpExtension::kTOffset = "urn:ietf:params:rtp-hdrext:toffset";
 const char* RtpExtension::kAbsSendTime =
     "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time";
 namespace internal {
+
+class CpuOveruseObserverProxy : public webrtc::CpuOveruseObserver {
+ public:
+  CpuOveruseObserverProxy(OveruseCallback* overuse_callback)
+      : crit_(CriticalSectionWrapper::CreateCriticalSection()),
+        overuse_callback_(overuse_callback) {
+    assert(overuse_callback != NULL);
+  }
+
+  virtual ~CpuOveruseObserverProxy() {}
+
+  virtual void OveruseDetected() OVERRIDE {
+    CriticalSectionScoped cs(crit_.get());
+    overuse_callback_->OnOveruse();
+  }
+
+  virtual void NormalUsage() OVERRIDE {
+    CriticalSectionScoped cs(crit_.get());
+    overuse_callback_->OnNormalUse();
+  }
+
+ private:
+  scoped_ptr<CriticalSectionWrapper> crit_;
+  OveruseCallback* overuse_callback_;
+};
+
 class Call : public webrtc::Call, public PacketReceiver {
  public:
   Call(webrtc::VideoEngine* video_engine, const Call::Config& config);
@@ -78,6 +104,8 @@ class Call : public webrtc::Call, public PacketReceiver {
 
   scoped_ptr<RtpHeaderParser> rtp_header_parser_;
 
+  scoped_ptr<CpuOveruseObserverProxy> overuse_observer_proxy_;
+
   VideoEngine* video_engine_;
   ViERTP_RTCP* rtp_rtcp_;
   ViECodec* codec_;
@@ -185,6 +213,11 @@ Call::Call(webrtc::VideoEngine* video_engine, const Call::Config& config)
   assert(video_engine != NULL);
   assert(config.send_transport != NULL);
 
+  if (config.overuse_callback) {
+    overuse_observer_proxy_.reset(
+        new CpuOveruseObserverProxy(config.overuse_callback));
+  }
+
   global_trace_dispatcher->RegisterCallback(this, &config_);
 
   rtp_rtcp_ = ViERTP_RTCP::GetInterface(video_engine_);
@@ -236,8 +269,9 @@ VideoSendStream* Call::CreateVideoSendStream(
   assert(config.rtp.ssrcs.size() > 0);
   assert(config.rtp.ssrcs.size() >= config.codec.numberOfSimulcastStreams);
 
-  VideoSendStream* send_stream = new VideoSendStream(config_.send_transport,
-                                                     config_.overuse_detection,
+  VideoSendStream* send_stream = new VideoSendStream(
+      config_.send_transport,
+      overuse_observer_proxy_.get(),
       video_engine_,
       config,
       base_channel_id_);

webrtc/video/call_perf_tests.cc

@@ -45,6 +45,39 @@ static const uint32_t kReceiverLocalSsrc = 0x123456;
 static const uint8_t kSendPayloadType = 125;
 
 class CallPerfTest : public ::testing::Test {
+ public:
+  CallPerfTest()
+      : send_stream_(NULL), fake_encoder_(Clock::GetRealTimeClock()) {}
+ protected:
+  VideoSendStream::Config GetSendTestConfig(Call* call) {
+    VideoSendStream::Config config = call->GetDefaultSendConfig();
+    config.encoder = &fake_encoder_;
+    config.internal_source = false;
+    config.rtp.ssrcs.push_back(kSendSsrc);
+    test::FakeEncoder::SetCodecSettings(&config.codec, 1);
+    config.codec.plType = kSendPayloadType;
+    return config;
+  }
+  void RunVideoSendTest(Call* call,
+                        const VideoSendStream::Config& config,
+                        test::RtpRtcpObserver* observer) {
+    send_stream_ = call->CreateVideoSendStream(config);
+    scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer(
+        test::FrameGeneratorCapturer::Create(
+            send_stream_->Input(), 320, 240, 30, Clock::GetRealTimeClock()));
+    send_stream_->StartSending();
+    frame_generator_capturer->Start();
+
+    EXPECT_EQ(kEventSignaled, observer->Wait());
+
+    observer->StopSending();
+    frame_generator_capturer->Stop();
+    send_stream_->StopSending();
+    call->DestroyVideoSendStream(send_stream_);
+  }
+
+  VideoSendStream* send_stream_;
+  test::FakeEncoder fake_encoder_;
 };
 
 class SyncRtcpObserver : public test::RtpRtcpObserver {
@@ -236,15 +269,9 @@ TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSync) {
 
   observer.SetReceivers(receiver_call->Receiver(), sender_call->Receiver());
 
-  test::FakeEncoder fake_encoder(Clock::GetRealTimeClock());
   test::FakeDecoder fake_decoder;
 
-  VideoSendStream::Config send_config = sender_call->GetDefaultSendConfig();
-  send_config.rtp.ssrcs.push_back(kSendSsrc);
-  send_config.encoder = &fake_encoder;
-  send_config.internal_source = false;
-  test::FakeEncoder::SetCodecSettings(&send_config.codec, 1);
-  send_config.codec.plType = kSendPayloadType;
+  VideoSendStream::Config send_config = GetSendTestConfig(sender_call.get());
 
   VideoReceiveStream::Config receive_config =
       receiver_call->GetDefaultReceiveConfig();
@@ -301,4 +328,28 @@ TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSync) {
   receiver_call->DestroyVideoReceiveStream(receive_stream);
   VoiceEngine::Delete(voice_engine);
 }
+
+TEST_F(CallPerfTest, RegisterCpuOveruseObserver) {
+  // Verifies that either a normal or overuse callback is triggered.
+  class OveruseCallbackObserver : public test::RtpRtcpObserver,
+                                  public webrtc::OveruseCallback {
+   public:
+    OveruseCallbackObserver() : RtpRtcpObserver(kLongTimeoutMs) {}
+
+    virtual void OnOveruse() OVERRIDE {
+      observation_complete_->Set();
+    }
+    virtual void OnNormalUse() OVERRIDE {
+      observation_complete_->Set();
+    }
+  };
+
+  OveruseCallbackObserver observer;
+  Call::Config call_config(observer.SendTransport());
+  call_config.overuse_callback = &observer;
+  scoped_ptr<Call> call(Call::Create(call_config));
+
+  VideoSendStream::Config send_config = GetSendTestConfig(call.get());
+  RunVideoSendTest(call.get(), send_config, &observer);
+}
 }  // namespace webrtc

webrtc/video/loopback.cc

@@ -43,7 +43,6 @@ TEST_F(LoopbackTest, Test) {
 
   test::DirectTransport transport;
   Call::Config call_config(&transport);
-  call_config.overuse_detection = true;
   scoped_ptr<Call> call(Call::Create(call_config));
 
   // Loopback, call sends to itself.

webrtc/video/video_send_stream.cc

@@ -26,58 +26,8 @@
 namespace webrtc {
 namespace internal {
 
-// Super simple and temporary overuse logic. This will move to the application
-// as soon as the new API allows changing send codec on the fly.
-class ResolutionAdaptor : public webrtc::CpuOveruseObserver {
- public:
-  ResolutionAdaptor(ViECodec* codec, int channel, size_t width, size_t height)
-      : codec_(codec),
-        channel_(channel),
-        max_width_(width),
-        max_height_(height) {}
-
-  virtual ~ResolutionAdaptor() {}
-
-  virtual void OveruseDetected() OVERRIDE {
-    VideoCodec codec;
-    if (codec_->GetSendCodec(channel_, codec) != 0)
-      return;
-
-    if (codec.width / 2 < min_width || codec.height / 2 < min_height)
-      return;
-
-    codec.width /= 2;
-    codec.height /= 2;
-    codec_->SetSendCodec(channel_, codec);
-  }
-
-  virtual void NormalUsage() OVERRIDE {
-    VideoCodec codec;
-    if (codec_->GetSendCodec(channel_, codec) != 0)
-      return;
-
-    if (codec.width * 2u > max_width_ || codec.height * 2u > max_height_)
-      return;
-
-    codec.width *= 2;
-    codec.height *= 2;
-    codec_->SetSendCodec(channel_, codec);
-  }
-
- private:
-  // Temporary and arbitrary chosen minimum resolution.
-  static const size_t min_width = 160;
-  static const size_t min_height = 120;
-
-  ViECodec* codec_;
-  const int channel_;
-
-  const size_t max_width_;
-  const size_t max_height_;
-};
-
 VideoSendStream::VideoSendStream(newapi::Transport* transport,
-                                 bool overuse_detection,
+                                 CpuOveruseObserver* overuse_observer,
                                  webrtc::VideoEngine* video_engine,
                                  const VideoSendStream::Config& config,
                                  int base_channel)
@@ -169,11 +119,8 @@ VideoSendStream::VideoSendStream(newapi::Transport* transport,
   if (!SetCodec(config_.codec))
     abort();
 
-  if (overuse_detection) {
-    overuse_observer_.reset(new ResolutionAdaptor(
-        codec_, channel_, config_.codec.width, config_.codec.height));
-    video_engine_base_->RegisterCpuOveruseObserver(channel_,
-                                                   overuse_observer_.get());
+  if (overuse_observer) {
+    video_engine_base_->RegisterCpuOveruseObserver(channel_, overuse_observer);
   }
 
   image_process_ = ViEImageProcess::GetInterface(video_engine);

webrtc/video/video_send_stream.h

@@ -21,6 +21,7 @@
 
 namespace webrtc {
 
+class CpuOveruseObserver;
 class VideoEngine;
 class ViEBase;
 class ViECapture;
@@ -33,14 +34,12 @@ class ViERTP_RTCP;
 
 namespace internal {
 
-class ResolutionAdaptor;
-
 class VideoSendStream : public webrtc::VideoSendStream,
                         public VideoSendStreamInput,
                         public SendStatisticsProxy::StreamStatsProvider {
  public:
   VideoSendStream(newapi::Transport* transport,
-                  bool overuse_detection,
+                  CpuOveruseObserver* overuse_observer,
                   webrtc::VideoEngine* video_engine,
                   const VideoSendStream::Config& config,
                   int base_channel);
@@ -88,7 +87,6 @@ class VideoSendStream : public webrtc::VideoSendStream,
 
   int channel_;
   int capture_id_;
-  scoped_ptr<ResolutionAdaptor> overuse_observer_;
 
   scoped_ptr<SendStatisticsProxy> stats_proxy_;
 };