Add a simulcast test that verifies media is flowing on all layers.

Previous tests only asserted that O/A succeeded and that the number of
encodings was as expected. This test goes further and also asserts that
bytesSent eventually becomes non-zero (after an initial ramp-up time).

Let's get testing straight before we add VP9 simulcast support.

Bug: webrtc:14885, webrtc:14884
Change-Id: Idccce66698a077264fa0df2c448c8474d2439aea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291960
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39271}

4 files changed, 210 insertions, 5 deletions

diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index 0be2fabab3..4411fce723 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -2753,6 +2753,7 @@ if (rtc_include_tests && !build_with_chromium) {
       "../api/task_queue:default_task_queue_factory",
       "../api/units:time_delta",
       "../api/video:builtin_video_bitrate_allocator_factory",
+      "../api/video:resolution",
       "../api/video:video_frame",
       "../api/video:video_rtp_headers",
       "../api/video_codecs:builtin_video_decoder_factory",
diff --git a/pc/peer_connection_simulcast_unittest.cc b/pc/peer_connection_simulcast_unittest.cc
index b9681dca34..3b9d7c89aa 100644
--- a/pc/peer_connection_simulcast_unittest.cc
+++ b/pc/peer_connection_simulcast_unittest.cc
@@ -22,6 +22,8 @@
 #include "api/audio/audio_mixer.h"
 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "api/audio_codecs/opus_audio_decoder_factory.h"
+#include "api/audio_codecs/opus_audio_encoder_factory.h"
 #include "api/create_peerconnection_factory.h"
 #include "api/jsep.h"
 #include "api/media_types.h"
@@ -32,6 +34,7 @@
 #include "api/rtp_transceiver_direction.h"
 #include "api/rtp_transceiver_interface.h"
 #include "api/scoped_refptr.h"
+#include "api/stats/rtcstats_objects.h"
 #include "api/uma_metrics.h"
 #include "api/video/video_codec_constants.h"
 #include "api/video_codecs/builtin_video_decoder_factory.h"
@@ -42,11 +45,15 @@
 #include "modules/audio_processing/include/audio_processing.h"
 #include "pc/channel_interface.h"
 #include "pc/peer_connection_wrapper.h"
+#include "pc/sdp_utils.h"
 #include "pc/session_description.h"
 #include "pc/simulcast_description.h"
 #include "pc/test/fake_audio_capture_module.h"
 #include "pc/test/mock_peer_connection_observers.h"
+#include "pc/test/peer_connection_test_wrapper.h"
 #include "rtc_base/checks.h"
+#include "rtc_base/gunit.h"
+#include "rtc_base/physical_socket_server.h"
 #include "rtc_base/strings/string_builder.h"
 #include "rtc_base/thread.h"
 #include "rtc_base/unique_id_generator.h"
@@ -117,6 +124,9 @@ std::vector<SimulcastLayer> CreateLayers(int num_layers, bool active) {
 
 namespace webrtc {
 
+constexpr TimeDelta kDefaultTimeout = TimeDelta::Seconds(5);
+constexpr TimeDelta kLongTimeoutForRampingUp = TimeDelta::Seconds(20);
+
 class PeerConnectionSimulcastTests : public ::testing::Test {
  public:
   PeerConnectionSimulcastTests()
@@ -794,4 +804,189 @@ INSTANTIATE_TEST_SUITE_P(NumberOfSendEncodings,
                          PeerConnectionSimulcastMetricsTests,
                          ::testing::Range(0, kMaxLayersInMetricsTest));
 #endif
+
+// Inherits some helper methods from PeerConnectionSimulcastTests but
+// uses real threads and PeerConnectionTestWrapper to create fake media streams
+// with flowing media and establish connections.
+class PeerConnectionSimulcastWithMediaFlowTests
+    : public PeerConnectionSimulcastTests {
+ public:
+  PeerConnectionSimulcastWithMediaFlowTests()
+      : background_thread_(std::make_unique<rtc::Thread>(&pss_)) {
+    RTC_CHECK(background_thread_->Start());
+  }
+
+  rtc::scoped_refptr<PeerConnectionTestWrapper> CreatePc() {
+    auto pc_wrapper = rtc::make_ref_counted<PeerConnectionTestWrapper>(
+        "pc", &pss_, background_thread_.get(), background_thread_.get());
+    pc_wrapper->CreatePc({}, webrtc::CreateOpusAudioEncoderFactory(),
+                         webrtc::CreateOpusAudioDecoderFactory());
+    return pc_wrapper;
+  }
+
+  rtc::scoped_refptr<RtpTransceiverInterface> AddTransceiverWithSimulcastLayers(
+      rtc::scoped_refptr<PeerConnectionTestWrapper> local,
+      rtc::scoped_refptr<PeerConnectionTestWrapper> remote,
+      std::vector<SimulcastLayer> init_layers) {
+    rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
+        local->GetUserMedia(
+            /*audio=*/false, cricket::AudioOptions(), /*video=*/true,
+            {.width = 1280, .height = 720});
+    rtc::scoped_refptr<VideoTrackInterface> track = stream->GetVideoTracks()[0];
+
+    RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
+        transceiver_or_error = local->pc()->AddTransceiver(
+            track, CreateTransceiverInit(init_layers));
+    EXPECT_TRUE(transceiver_or_error.ok());
+    return transceiver_or_error.value();
+  }
+
+  void ExchangeIceCandidates(
+      rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper,
+      rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper) {
+    local_pc_wrapper->SignalOnIceCandidateReady.connect(
+        remote_pc_wrapper.get(), &PeerConnectionTestWrapper::AddIceCandidate);
+    remote_pc_wrapper->SignalOnIceCandidateReady.connect(
+        local_pc_wrapper.get(), &PeerConnectionTestWrapper::AddIceCandidate);
+  }
+
+  void NegotiateWithSimulcastTweaks(
+      rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper,
+      rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper,
+      std::vector<SimulcastLayer> init_layers) {
+    // Create and set offer for `local_pc_wrapper`.
+    std::unique_ptr<SessionDescriptionInterface> offer =
+        CreateOffer(local_pc_wrapper);
+    rtc::scoped_refptr<MockSetSessionDescriptionObserver> p1 =
+        SetLocalDescription(local_pc_wrapper, offer.get());
+    // Modify the offer before handoff because `remote_pc_wrapper` only supports
+    // receiving singlecast.
+    SimulcastDescription simulcast_description = RemoveSimulcast(offer.get());
+    rtc::scoped_refptr<MockSetSessionDescriptionObserver> p2 =
+        SetRemoteDescription(remote_pc_wrapper, offer.get());
+    EXPECT_TRUE(Await({p1, p2}));
+
+    // Create and set answer for `remote_pc_wrapper`.
+    std::unique_ptr<SessionDescriptionInterface> answer =
+        CreateAnswer(remote_pc_wrapper);
+    p1 = SetLocalDescription(remote_pc_wrapper, answer.get());
+    // Modify the answer before handoff because `local_pc_wrapper` should still
+    // send simulcast.
+    cricket::MediaContentDescription* mcd_answer =
+        answer->description()->contents()[0].media_description();
+    mcd_answer->mutable_streams().clear();
+    std::vector<SimulcastLayer> simulcast_layers =
+        simulcast_description.send_layers().GetAllLayers();
+    cricket::SimulcastLayerList& receive_layers =
+        mcd_answer->simulcast_description().receive_layers();
+    for (const auto& layer : simulcast_layers) {
+      receive_layers.AddLayer(layer);
+    }
+    p2 = SetRemoteDescription(local_pc_wrapper, answer.get());
+    EXPECT_TRUE(Await({p1, p2}));
+  }
+
+  bool HasOutboundRtpBytesSent(
+      rtc::scoped_refptr<PeerConnectionTestWrapper> pc_wrapper,
+      size_t num_layers) {
+    rtc::scoped_refptr<const RTCStatsReport> report = GetStats(pc_wrapper);
+    std::vector<const RTCOutboundRTPStreamStats*> outbound_rtps =
+        report->GetStatsOfType<RTCOutboundRTPStreamStats>();
+    if (outbound_rtps.size() != num_layers) {
+      return false;
+    }
+    for (const auto* outbound_rtp : outbound_rtps) {
+      if (!outbound_rtp->bytes_sent.is_defined() ||
+          *outbound_rtp->bytes_sent == 0u) {
+        return false;
+      }
+    }
+    return true;
+  }
+
+ protected:
+  std::unique_ptr<SessionDescriptionInterface> CreateOffer(
+      rtc::scoped_refptr<PeerConnectionTestWrapper> pc_wrapper) {
+    auto observer =
+        rtc::make_ref_counted<MockCreateSessionDescriptionObserver>();
+    pc_wrapper->pc()->CreateOffer(observer.get(), {});
+    EXPECT_EQ_WAIT(true, observer->called(), kDefaultTimeout.ms());
+    return observer->MoveDescription();
+  }
+
+  std::unique_ptr<SessionDescriptionInterface> CreateAnswer(
+      rtc::scoped_refptr<PeerConnectionTestWrapper> pc_wrapper) {
+    auto observer =
+        rtc::make_ref_counted<MockCreateSessionDescriptionObserver>();
+    pc_wrapper->pc()->CreateAnswer(observer.get(), {});
+    EXPECT_EQ_WAIT(true, observer->called(), kDefaultTimeout.ms());
+    return observer->MoveDescription();
+  }
+
+  rtc::scoped_refptr<MockSetSessionDescriptionObserver> SetLocalDescription(
+      rtc::scoped_refptr<PeerConnectionTestWrapper> pc_wrapper,
+      SessionDescriptionInterface* sdp) {
+    auto observer = rtc::make_ref_counted<MockSetSessionDescriptionObserver>();
+    pc_wrapper->pc()->SetLocalDescription(
+        observer.get(), CloneSessionDescription(sdp).release());
+    return observer;
+  }
+
+  rtc::scoped_refptr<MockSetSessionDescriptionObserver> SetRemoteDescription(
+      rtc::scoped_refptr<PeerConnectionTestWrapper> pc_wrapper,
+      SessionDescriptionInterface* sdp) {
+    auto observer = rtc::make_ref_counted<MockSetSessionDescriptionObserver>();
+    pc_wrapper->pc()->SetRemoteDescription(
+        observer.get(), CloneSessionDescription(sdp).release());
+    return observer;
+  }
+
+  // To avoid ICE candidates arriving before the remote endpoint has received
+  // the offer it is important to SetLocalDescription() and
+  // SetRemoteDescription() are kicked off without awaiting in-between. This
+  // helper is used to await multiple observers.
+  bool Await(std::vector<rtc::scoped_refptr<MockSetSessionDescriptionObserver>>
+                 observers) {
+    for (auto& observer : observers) {
+      EXPECT_EQ_WAIT(true, observer->called(), kDefaultTimeout.ms());
+      if (!observer->result()) {
+        return false;
+      }
+    }
+    return true;
+  }
+
+  rtc::scoped_refptr<const RTCStatsReport> GetStats(
+      rtc::scoped_refptr<PeerConnectionTestWrapper> pc_wrapper) {
+    auto callback = rtc::make_ref_counted<MockRTCStatsCollectorCallback>();
+    pc_wrapper->pc()->GetStats(callback.get());
+    EXPECT_TRUE_WAIT(callback->called(), kDefaultTimeout.ms());
+    return callback->report();
+  }
+
+  rtc::PhysicalSocketServer pss_;
+  std::unique_ptr<rtc::Thread> background_thread_;
+};
+
+// TODO(https://crbug.com/webrtc/14884): When VP9 simulast is supported, use
+// SetCodecPreferences() and pass a test like this with VP9.
+TEST_F(PeerConnectionSimulcastWithMediaFlowTests,
+       SimulcastSendsAllLayersWithDefaultCodec) {
+  rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
+  rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
+  ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
+
+  std::vector<SimulcastLayer> layers = CreateLayers({"f", "h", "q"}, true);
+  AddTransceiverWithSimulcastLayers(local_pc_wrapper, remote_pc_wrapper,
+                                    layers);
+
+  NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper, layers);
+  local_pc_wrapper->WaitForConnection();
+  remote_pc_wrapper->WaitForConnection();
+
+  // Wait until media is flowing on all three layers.
+  EXPECT_TRUE_WAIT(HasOutboundRtpBytesSent(local_pc_wrapper, 3u),
+                   kLongTimeoutForRampingUp.ms());
+}
+
 }  // namespace webrtc
diff --git a/pc/test/peer_connection_test_wrapper.cc b/pc/test/peer_connection_test_wrapper.cc
index 8325e59510..c35628eae3 100644
--- a/pc/test/peer_connection_test_wrapper.cc
+++ b/pc/test/peer_connection_test_wrapper.cc
@@ -318,7 +318,8 @@ rtc::scoped_refptr<webrtc::MediaStreamInterface>
 PeerConnectionTestWrapper::GetUserMedia(
     bool audio,
     const cricket::AudioOptions& audio_options,
-    bool video) {
+    bool video,
+    webrtc::Resolution resolution) {
   std::string stream_id =
       kStreamIdBase + rtc::ToString(num_get_user_media_calls_++);
   rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
@@ -341,6 +342,8 @@ PeerConnectionTestWrapper::GetUserMedia(
     webrtc::FakePeriodicVideoSource::Config config;
     config.frame_interval_ms = 100;
     config.timestamp_offset_ms = rtc::TimeMillis();
+    config.width = resolution.width;
+    config.height = resolution.height;
 
     auto source = rtc::make_ref_counted<webrtc::FakePeriodicVideoTrackSource>(
         config, /* remote */ false);
diff --git a/pc/test/peer_connection_test_wrapper.h b/pc/test/peer_connection_test_wrapper.h
index cda3ecb73b..23a6996c30 100644
--- a/pc/test/peer_connection_test_wrapper.h
+++ b/pc/test/peer_connection_test_wrapper.h
@@ -26,7 +26,9 @@
 #include "api/rtp_receiver_interface.h"
 #include "api/scoped_refptr.h"
 #include "api/sequence_checker.h"
+#include "api/video/resolution.h"
 #include "pc/test/fake_audio_capture_module.h"
+#include "pc/test/fake_periodic_video_source.h"
 #include "pc/test/fake_video_track_renderer.h"
 #include "rtc_base/third_party/sigslot/sigslot.h"
 #include "rtc_base/thread.h"
@@ -106,16 +108,20 @@ class PeerConnectionTestWrapper
   sigslot::signal1<const std::string&> SignalOnSdpReady;
   sigslot::signal1<webrtc::DataChannelInterface*> SignalOnDataChannel;
 
+  rtc::scoped_refptr<webrtc::MediaStreamInterface> GetUserMedia(
+      bool audio,
+      const cricket::AudioOptions& audio_options,
+      bool video,
+      webrtc::Resolution resolution = {
+          .width = webrtc::FakePeriodicVideoSource::kDefaultWidth,
+          .height = webrtc::FakePeriodicVideoSource::kDefaultHeight});
+
  private:
   void SetLocalDescription(webrtc::SdpType type, const std::string& sdp);
   void SetRemoteDescription(webrtc::SdpType type, const std::string& sdp);
   bool CheckForConnection();
   bool CheckForAudio();
   bool CheckForVideo();
-  rtc::scoped_refptr<webrtc::MediaStreamInterface> GetUserMedia(
-      bool audio,
-      const cricket::AudioOptions& audio_options,
-      bool video);
 
   webrtc::test::ScopedKeyValueConfig field_trials_;
   std::string name_;