Add send-side BWE to WebRtcVoiceEngine under a finch experiment.

talk/media/webrtc/webrtcvoiceengine_unittest.cc

Index: talk/media/webrtc/webrtcvoiceengine_unittest.cc
diff --git a/talk/media/webrtc/webrtcvoiceengine_unittest.cc b/talk/media/webrtc/webrtcvoiceengine_unittest.cc
index a62bcb225fc7c6111fed44c0a43507f1ecf7b353..19cceb4b99362298f5e281f6497c897f49996f0c 100644
--- a/talk/media/webrtc/webrtcvoiceengine_unittest.cc
+++ b/talk/media/webrtc/webrtcvoiceengine_unittest.cc
@@ -29,6 +29,8 @@
 #include "webrtc/base/byteorder.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/call.h"
+#include "webrtc/p2p/base/faketransportcontroller.h"
+#include "webrtc/test/field_trial.h"
 #include "talk/media/base/constants.h"
 #include "talk/media/base/fakemediaengine.h"
 #include "talk/media/base/fakenetworkinterface.h"
@@ -36,7 +38,6 @@
 #include "talk/media/webrtc/fakewebrtccall.h"
 #include "talk/media/webrtc/fakewebrtcvoiceengine.h"
 #include "talk/media/webrtc/webrtcvoiceengine.h"
-#include "webrtc/p2p/base/faketransportcontroller.h"
 #include "talk/session/media/channel.h"
 
 using cricket::kRtpAudioLevelHeaderExtension;
@@ -74,10 +75,13 @@ class FakeVoEWrapper : public cricket::VoEWrapper {
 
 class WebRtcVoiceEngineTestFake : public testing::Test {
  public:
-  WebRtcVoiceEngineTestFake()
+  explicit WebRtcVoiceEngineTestFake() : WebRtcVoiceEngineTestFake("") {}
+
+  explicit WebRtcVoiceEngineTestFake(const char* field_trials)
       : call_(webrtc::Call::Config()),
         engine_(new FakeVoEWrapper(&voe_)),
-        channel_(nullptr) {
+        channel_(nullptr),
+        override_field_trials_(field_trials) {
     send_parameters_.codecs.push_back(kPcmuCodec);
     recv_parameters_.codecs.push_back(kPcmuCodec);
   }
@@ -281,6 +285,29 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
     EXPECT_EQ(0u, GetRecvStreamConfig(kSsrc2).rtp.extensions.size());
   }
 
+  void TestExtensionFilter(const std::vector<std::string>& extensions,
+                           const std::string& expected_extension) {
+    EXPECT_TRUE(SetupEngineWithSendStream());
+    cricket::AudioSendParameters parameters = send_parameters_;
+    int expected_id = -1;
+    int id = 1;
+    for (const std::string& extension : extensions) {
+      if (extension == expected_extension)
+        expected_id = id;
+      parameters.extensions.push_back(
+          cricket::RtpHeaderExtension(extension, id++));
+    }
+    EXPECT_TRUE(channel_->SetSendParameters(parameters));
+    const cricket::FakeAudioSendStream send_stream = GetSendStream(kSsrc1);
+
+    // Verify that only one of them has been set, and that it is the one with
+    // highest priority (transport sequence number).
+    ASSERT_EQ(1u, send_stream.GetConfig().rtp.extensions.size());
+    EXPECT_EQ(expected_id, send_stream.GetConfig().rtp.extensions[0].id);
+    EXPECT_EQ(expected_extension,
+              send_stream.GetConfig().rtp.extensions[0].name);
+  }
+
   webrtc::AudioSendStream::Stats GetAudioSendStreamStats() const {
     webrtc::AudioSendStream::Stats stats;
     stats.local_ssrc = 12;
@@ -399,6 +426,9 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
   cricket::VoiceMediaChannel* channel_;
   cricket::AudioSendParameters send_parameters_;
   cricket::AudioRecvParameters recv_parameters_;
+
+ private:
+  webrtc::test::ScopedFieldTrials override_field_trials_;
 };
 
 // Tests that our stub library "works".
@@ -433,6 +463,18 @@ TEST_F(WebRtcVoiceEngineTestFake, CodecPreference) {
   }
 }
 
+TEST_F(WebRtcVoiceEngineTestFake, OpusSupportsTransportCc) {
+  const std::vector<cricket::AudioCodec>& codecs = engine_.codecs();
+  bool opus_found = false;
+  for (cricket::AudioCodec codec : codecs) {
+    if (codec.name == "opus") {
+      EXPECT_TRUE(HasTransportCc(codec));
+      opus_found = true;
+    }
+  }
+  EXPECT_TRUE(opus_found);
+}
+
 // Tests that we can find codecs by name or id, and that we interpret the
 // clockrate and bitrate fields properly.
 TEST_F(WebRtcVoiceEngineTestFake, FindCodec) {
@@ -1276,6 +1318,29 @@ TEST_F(WebRtcVoiceEngineTestFake, ChangeOpusFecStatus) {
   EXPECT_TRUE(voe_.GetCodecFEC(channel_num));
 }
 
+TEST_F(WebRtcVoiceEngineTestFake, TransportCcCanBeEnabledAndDisabled) {
+  EXPECT_TRUE(SetupEngine());
+  cricket::AudioSendParameters send_parameters;
+  send_parameters.codecs.push_back(kOpusCodec);
+  EXPECT_TRUE(send_parameters.codecs[0].feedback_params.params().empty());
+  EXPECT_TRUE(channel_->SetSendParameters(send_parameters));
+
+  cricket::AudioRecvParameters recv_parameters;
+  recv_parameters.codecs.push_back(kOpusCodec);
+  EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters));
+  EXPECT_TRUE(
+      channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrc1)));
+  ASSERT_TRUE(call_.GetAudioReceiveStream(kSsrc1) != nullptr);
+  EXPECT_FALSE(
+      call_.GetAudioReceiveStream(kSsrc1)->GetConfig().rtp.transport_cc);
+
+  send_parameters.codecs = engine_.codecs();
+  EXPECT_TRUE(channel_->SetSendParameters(send_parameters));
+  ASSERT_TRUE(call_.GetAudioReceiveStream(kSsrc1) != nullptr);
+  EXPECT_TRUE(
+      call_.GetAudioReceiveStream(kSsrc1)->GetConfig().rtp.transport_cc);
+}
+
 // Test maxplaybackrate <= 8000 triggers Opus narrow band mode.
 TEST_F(WebRtcVoiceEngineTestFake, SetOpusMaxPlaybackRateNb) {
   EXPECT_TRUE(SetupEngineWithSendStream());
@@ -1918,6 +1983,37 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsBadRED5) {
   EXPECT_FALSE(voe_.GetRED(channel_num));
 }
 
+class WebRtcVoiceEngineWithSendSideBweTest : public WebRtcVoiceEngineTestFake {
+ public:
+  WebRtcVoiceEngineWithSendSideBweTest()
+      : WebRtcVoiceEngineTestFake("WebRTC-Audio-SendSideBwe/Enabled/") {}
+};
+
+TEST_F(WebRtcVoiceEngineWithSendSideBweTest,
+       SupportsTransportSequenceNumberHeaderExtension) {
+  cricket::RtpCapabilities capabilities = engine_.GetCapabilities();
+  ASSERT_FALSE(capabilities.header_extensions.empty());
+  for (const cricket::RtpHeaderExtension& extension :
+       capabilities.header_extensions) {
+    if (extension.uri == cricket::kRtpTransportSequenceNumberHeaderExtension) {
+      EXPECT_EQ(cricket::kRtpTransportSequenceNumberHeaderExtensionDefaultId,
+                extension.id);
+      return;
+    }
+  }
+  FAIL() << "Transport sequence number extension not in header-extension list.";
+}
+
+TEST_F(WebRtcVoiceEngineWithSendSideBweTest,
+       FiltersExtensionsPicksTransportSeqNum) {
+  // Enable three redundant extensions.
+  std::vector<std::string> extensions;
+  extensions.push_back(cricket::kRtpAbsoluteSenderTimeHeaderExtension);
+  extensions.push_back(cricket::kRtpTransportSequenceNumberHeaderExtension);
+  TestExtensionFilter(extensions,
+                      cricket::kRtpTransportSequenceNumberHeaderExtension);
+}
+
 // Test support for audio level header extension.
 TEST_F(WebRtcVoiceEngineTestFake, SendAudioLevelHeaderExtensions) {
   TestSetSendRtpHeaderExtensions(kRtpAudioLevelHeaderExtension);