Signal target bitrate only for screenshare streams

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>
@@ skipping 2978 matching lines @@
 
   // TODO(philipel): Implement statistics for the new video jitter buffer.
   if (GetParam() == new_jb_enabled)
     return;
 
   RunBaseTest(&test);
 }
 
 class RtcpXrObserver : public test::EndToEndTest {
  public:
-  explicit RtcpXrObserver(bool enable_rrtr)
+  RtcpXrObserver(bool enable_rrtr, bool enable_target_bitrate)
       : EndToEndTest(test::CallTest::kDefaultTimeoutMs),
         enable_rrtr_(enable_rrtr),
+        enable_target_bitrate_(enable_target_bitrate),
         sent_rtcp_sr_(0),
         sent_rtcp_rr_(0),
         sent_rtcp_rrtr_(0),
         sent_rtcp_target_bitrate_(false),
         sent_rtcp_dlrr_(0) {}
 
  private:
   // Receive stream should send RR packets (and RRTR packets if enabled).
   Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
     rtc::CritScope lock(&crit_);
@@ skipping 22 matching lines @@
     if (parser.xr()->num_packets() > 0) {
       EXPECT_FALSE(parser.xr()->rrtr());
       if (parser.xr()->dlrr())
         ++sent_rtcp_dlrr_;
       if (parser.xr()->target_bitrate())
         sent_rtcp_target_bitrate_ = true;
     }
 
     if (sent_rtcp_sr_ > kNumRtcpReportPacketsToObserve &&
         sent_rtcp_rr_ > kNumRtcpReportPacketsToObserve &&
-        sent_rtcp_target_bitrate_) {
+        (sent_rtcp_target_bitrate_ || !enable_target_bitrate_)) {
       if (enable_rrtr_) {
         EXPECT_GT(sent_rtcp_rrtr_, 0);
         EXPECT_GT(sent_rtcp_dlrr_, 0);
       } else {
         EXPECT_EQ(sent_rtcp_rrtr_, 0);
         EXPECT_EQ(sent_rtcp_dlrr_, 0);
       }
+      EXPECT_EQ(enable_target_bitrate_, sent_rtcp_target_bitrate_);
       observation_complete_.Set();
     }
     return SEND_PACKET;
   }
 
   void ModifyVideoConfigs(
       VideoSendStream::Config* send_config,
       std::vector<VideoReceiveStream::Config>* receive_configs,
       VideoEncoderConfig* encoder_config) override {
+    if (enable_target_bitrate_) {
+      // TargetBitrate only signaled for screensharing.
+      encoder_config->content_type = VideoEncoderConfig::ContentType::kScreen;
+    }
     (*receive_configs)[0].rtp.rtcp_mode = RtcpMode::kReducedSize;
     (*receive_configs)[0].rtp.rtcp_xr.receiver_reference_time_report =
         enable_rrtr_;
   }
 
   void PerformTest() override {
     EXPECT_TRUE(Wait())
         << "Timed out while waiting for RTCP SR/RR packets to be sent.";
   }
 
   static const int kNumRtcpReportPacketsToObserve = 5;
 
   rtc::CriticalSection crit_;
-  bool enable_rrtr_;
+  const bool enable_rrtr_;
+  const bool enable_target_bitrate_;
   int sent_rtcp_sr_;
   int sent_rtcp_rr_ GUARDED_BY(&crit_);
   int sent_rtcp_rrtr_ GUARDED_BY(&crit_);
   bool sent_rtcp_target_bitrate_ GUARDED_BY(&crit_);
   int sent_rtcp_dlrr_;
 };
 
-TEST_P(EndToEndTest, TestExtendedReportsWithRrtr) {
-  RtcpXrObserver test(true);
+TEST_P(EndToEndTest, TestExtendedReportsWithRrtrWithoutTargetBitrate) {
+  RtcpXrObserver test(true, false);
   RunBaseTest(&test);
 }
 
-TEST_P(EndToEndTest, TestExtendedReportsWithoutRrtr) {
-  RtcpXrObserver test(false);
+TEST_P(EndToEndTest, TestExtendedReportsWithoutRrtrWithoutTargetBitrate) {
+  RtcpXrObserver test(false, false);
   RunBaseTest(&test);
 }
 
+TEST_P(EndToEndTest, TestExtendedReportsWithRrtrWithTargetBitrate) {
+  RtcpXrObserver test(true, true);
+  RunBaseTest(&test);
+}
+
+TEST_P(EndToEndTest, TestExtendedReportsWithoutRrtrWithTargetBitrate) {
+  RtcpXrObserver test(false, true);
+  RunBaseTest(&test);
+}
+
 TEST_P(EndToEndTest, TestReceivedRtpPacketStats) {
   static const size_t kNumRtpPacketsToSend = 5;
   class ReceivedRtpStatsObserver : public test::EndToEndTest {
    public:
     ReceivedRtpStatsObserver()
         : EndToEndTest(kDefaultTimeoutMs),
           receive_stream_(nullptr),
           sent_rtp_(0) {}
 
    private:
@@ skipping 963 matching lines @@
     std::unique_ptr<VideoEncoder> encoder_;
     std::unique_ptr<VideoDecoder> decoder_;
     rtc::CriticalSection crit_;
     int recorded_frames_ GUARDED_BY(crit_);
   } test(this);
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc