Disable reduced-size RTCP in default config.
Verifies that reduced-size isn't configured in WebRtcVideoEngine2
without explicit configuration (which doesn't exist). Also disables REMB
in the default config because it requires reconfiguration.
Adds default-config tests to make sure that they don't contain
parameters that need to be negotiated between clients.
BUG=chromium:497103, webrtc:4745
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1171533002
Cr-Commit-Position: refs/heads/master@{#9384}
diff --git a/talk/media/webrtc/webrtcvideoengine2_unittest.cc b/talk/media/webrtc/webrtcvideoengine2_unittest.cc
index 8ec7980..d83e8c4 100644
--- a/talk/media/webrtc/webrtcvideoengine2_unittest.cc
+++ b/talk/media/webrtc/webrtcvideoengine2_unittest.cc
@@ -1267,6 +1267,11 @@
EXPECT_EQ(1u, fake_call_->GetVideoReceiveStreams().size());
}
+TEST_F(WebRtcVideoChannel2Test, RtcpIsCompoundByDefault) {
+ FakeVideoReceiveStream* stream = AddRecvStream();
+ EXPECT_EQ(webrtc::newapi::kRtcpCompound, stream->GetConfig().rtp.rtcp_mode);
+}
+
TEST_F(WebRtcVideoChannel2Test, RembIsEnabledByDefault) {
FakeVideoReceiveStream* stream = AddRecvStream();
EXPECT_TRUE(stream->GetConfig().rtp.remb);
diff --git a/webrtc/test/call_test.cc b/webrtc/test/call_test.cc
index e912982..2548659 100644
--- a/webrtc/test/call_test.cc
+++ b/webrtc/test/call_test.cc
@@ -108,6 +108,7 @@
assert(receive_configs_.empty());
assert(allocated_decoders_.empty());
VideoReceiveStream::Config config;
+ config.rtp.remb = true;
config.rtp.local_ssrc = kReceiverLocalSsrc;
for (const RtpExtension& extension : send_config_.rtp.extensions)
config.rtp.extensions.push_back(extension);
diff --git a/webrtc/video/bitrate_estimator_tests.cc b/webrtc/video/bitrate_estimator_tests.cc
index 9209560..6e71ca2 100644
--- a/webrtc/video/bitrate_estimator_tests.cc
+++ b/webrtc/video/bitrate_estimator_tests.cc
@@ -151,6 +151,7 @@
// receive_config_.decoders will be set by every stream separately.
receive_config_.rtp.remote_ssrc = send_config_.rtp.ssrcs[0];
receive_config_.rtp.local_ssrc = kReceiverLocalSsrc;
+ receive_config_.rtp.remb = true;
receive_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
receive_config_.rtp.extensions.push_back(
diff --git a/webrtc/video/end_to_end_tests.cc b/webrtc/video/end_to_end_tests.cc
index 28504c4..ce5a0e5 100644
--- a/webrtc/video/end_to_end_tests.cc
+++ b/webrtc/video/end_to_end_tests.cc
@@ -2789,4 +2789,50 @@
call->DestroyVideoReceiveStream(receive_stream);
}
}
+
+void VerifyEmptyNackConfig(const NackConfig& config) {
+ EXPECT_EQ(0, config.rtp_history_ms)
+ << "Enabling NACK requires rtcp-fb: nack negotiation.";
+}
+
+void VerifyEmptyFecConfig(const FecConfig& config) {
+ EXPECT_EQ(-1, config.ulpfec_payload_type)
+ << "Enabling FEC requires rtpmap: ulpfec negotiation.";
+ EXPECT_EQ(-1, config.red_payload_type)
+ << "Enabling FEC requires rtpmap: red negotiation.";
+ EXPECT_EQ(-1, config.red_rtx_payload_type)
+ << "Enabling RTX in FEC requires rtpmap: rtx negotiation.";
+}
+
+TEST_F(EndToEndTest, VerifyDefaultSendConfigParameters) {
+ VideoSendStream::Config default_send_config;
+ EXPECT_EQ(0, default_send_config.rtp.nack.rtp_history_ms)
+ << "Enabling NACK require rtcp-fb: nack negotiation.";
+ EXPECT_TRUE(default_send_config.rtp.rtx.ssrcs.empty())
+ << "Enabling RTX requires rtpmap: rtx negotiation.";
+ EXPECT_TRUE(default_send_config.rtp.extensions.empty())
+ << "Enabling RTP extensions require negotiation.";
+
+ VerifyEmptyNackConfig(default_send_config.rtp.nack);
+ VerifyEmptyFecConfig(default_send_config.rtp.fec);
+}
+
+TEST_F(EndToEndTest, VerifyDefaultReceiveConfigParameters) {
+ VideoReceiveStream::Config default_receive_config;
+ EXPECT_EQ(newapi::kRtcpCompound, default_receive_config.rtp.rtcp_mode)
+ << "Reduced-size RTCP require rtcp-rsize to be negotiated.";
+ EXPECT_FALSE(default_receive_config.rtp.remb)
+ << "REMB require rtcp-fb: goog-remb to be negotiated.";
+ EXPECT_FALSE(
+ default_receive_config.rtp.rtcp_xr.receiver_reference_time_report)
+ << "RTCP XR settings require rtcp-xr to be negotiated.";
+ 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
diff --git a/webrtc/video/loopback.cc b/webrtc/video/loopback.cc
index 881fa49..7e5ca07 100644
--- a/webrtc/video/loopback.cc
+++ b/webrtc/video/loopback.cc
@@ -115,6 +115,7 @@
receive_config.rtp.remote_ssrc = send_config.rtp.ssrcs[0];
receive_config.rtp.local_ssrc = kReceiverLocalSsrc;
receive_config.rtp.nack.rtp_history_ms = 1000;
+ receive_config.rtp.remb = true;
receive_config.rtp.rtx[kVideoPayloadType].ssrc = kSendRtxSsrc;
receive_config.rtp.rtx[kVideoPayloadType].payload_type = kRtxVideoPayloadType;
receive_config.rtp.extensions.push_back(
diff --git a/webrtc/video_receive_stream.h b/webrtc/video_receive_stream.h
index b523b1e..5e5ece5 100644
--- a/webrtc/video_receive_stream.h
+++ b/webrtc/video_receive_stream.h
@@ -107,8 +107,8 @@
Rtp()
: remote_ssrc(0),
local_ssrc(0),
- rtcp_mode(newapi::kRtcpReducedSize),
- remb(true) {}
+ rtcp_mode(newapi::kRtcpCompound),
+ remb(false) {}
std::string ToString() const;
// Synchronization source (stream identifier) to be received.
