Wire up send-side bandwidth estimation.

webrtc/video/end_to_end_tests.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.
  */
 #include <algorithm>
 #include <map>
 #include <sstream>
 #include <string>
 
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/event.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/call.h"
 #include "webrtc/frame_callback.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
 #include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h"
 #include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/interface/event_wrapper.h"
 #include "webrtc/system_wrappers/interface/metrics.h"
@@ skipping 1398 matching lines @@
       return observer_;
     }
 
    private:
     RtpExtensionHeaderObserver* observer_;
   } tester;
 
   tester.RunTest();
 }
 
+TEST_F(EndToEndTest, ReceivesTransportFeedback) {
+  static const int kExtensionId = 5;
+
+  class TransportFeedbackObserver : public test::DirectTransport {
+   public:
+    TransportFeedbackObserver(rtc::Event* done_event) : done_(done_event) {}
+    virtual ~TransportFeedbackObserver() {}
+
+    bool SendRtcp(const uint8_t* data, size_t length) override {
+      RTCPUtility::RTCPParserV2 parser(data, length, true);
+      EXPECT_TRUE(parser.IsValid());
+
+      RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
+      while (packet_type != RTCPUtility::RTCPPacketTypes::kInvalid) {
+        if (packet_type == RTCPUtility::RTCPPacketTypes::kTransportFeedback) {
+          done_->Set();
+          break;
+        }
+        packet_type = parser.Iterate();
+      }
+
+      return test::DirectTransport::SendRtcp(data, length);
+    }
+
+    rtc::Event* done_;
+  };
+
+  class TransportFeedbackTester : public MultiStreamTest {
+   public:
+    TransportFeedbackTester() : done_(false, false) {}
+    virtual ~TransportFeedbackTester() {}
+
+   protected:
+    void Wait() override {
+      EXPECT_TRUE(done_.Wait(CallTest::kDefaultTimeoutMs));
+    }
+
+    void UpdateSendConfig(
+        size_t stream_index,
+        VideoSendStream::Config* send_config,
+        VideoEncoderConfig* encoder_config,
+        test::FrameGeneratorCapturer** frame_generator) override {
+      send_config->rtp.extensions.push_back(
+          RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId));
+    }
+
+    void UpdateReceiveConfig(
+        size_t stream_index,
+        VideoReceiveStream::Config* receive_config) override {
+      receive_config->rtp.extensions.push_back(
+          RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId));
+    }
+
+    virtual test::DirectTransport* CreateReceiveTransport() {
+      return new TransportFeedbackObserver(&done_);
+    }
+
+   private:
+    rtc::Event done_;
+  } tester;
+  tester.RunTest();
+}
 TEST_F(EndToEndTest, ObserversEncodedFrames) {
   class EncodedFrameTestObserver : public EncodedFrameObserver {
    public:
     EncodedFrameTestObserver()
         : length_(0),
           frame_type_(kFrameEmpty),
           called_(EventWrapper::Create()) {}
     virtual ~EncodedFrameTestObserver() {}
 
     virtual void EncodedFrameCallback(const EncodedFrame& encoded_frame) {
@@ skipping 1601 matching lines @@
   EXPECT_TRUE(default_receive_config.rtp.rtx.empty())
       << "Enabling RTX requires rtpmap: rtx negotiation.";
   EXPECT_TRUE(default_receive_config.rtp.extensions.empty())
       << "Enabling RTP extensions require negotiation.";
 
   VerifyEmptyNackConfig(default_receive_config.rtp.nack);
   VerifyEmptyFecConfig(default_receive_config.rtp.fec);
 }
 
 }  // namespace webrtc