Adding REMB to receive stream configuration, the send side will always
react to incoming REMB for now.
Adding a test to verify the receive side is generating RTCP REMB and
will follow up with a send side test as soon as the bitrate stats are
wired up for the new API.
TEST=See above.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5286 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/video/call.cc b/video/call.cc
index 69e8f84..12daa8e 100644
--- a/video/call.cc
+++ b/video/call.cc
@@ -277,7 +277,9 @@
}
VideoReceiveStream::Config Call::GetDefaultReceiveConfig() {
- return VideoReceiveStream::Config();
+ VideoReceiveStream::Config config;
+ config.rtp.remb = true;
+ return config;
}
VideoReceiveStream* Call::CreateVideoReceiveStream(
diff --git a/video/call_tests.cc b/video/call_tests.cc
index 34931d1..96ad832 100644
--- a/video/call_tests.cc
+++ b/video/call_tests.cc
@@ -936,4 +936,53 @@
DestroyStreams();
}
+
+TEST_F(CallTest, ReceiveStreamSendsRemb) {
+ class RembObserver : public test::RtpRtcpObserver {
+ public:
+ RembObserver() : test::RtpRtcpObserver(kDefaultTimeoutMs) {}
+
+ virtual Action OnReceiveRtcp(const uint8_t* packet,
+ size_t length) OVERRIDE {
+ RTCPUtility::RTCPParserV2 parser(packet, length, true);
+ EXPECT_TRUE(parser.IsValid());
+
+ bool received_psfb = false;
+ bool received_remb = false;
+ RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
+ while (packet_type != RTCPUtility::kRtcpNotValidCode) {
+ if (packet_type == RTCPUtility::kRtcpPsfbRembCode) {
+ const RTCPUtility::RTCPPacket& packet = parser.Packet();
+ EXPECT_EQ(packet.PSFBAPP.SenderSSRC, kReceiverLocalSsrc);
+ received_psfb = true;
+ } else if (packet_type == RTCPUtility::kRtcpPsfbRembItemCode) {
+ const RTCPUtility::RTCPPacket& packet = parser.Packet();
+ EXPECT_GT(packet.REMBItem.BitRate, 0u);
+ EXPECT_EQ(packet.REMBItem.NumberOfSSRCs, 1u);
+ EXPECT_EQ(packet.REMBItem.SSRCs[0], kSendSsrc);
+ received_remb = true;
+ }
+ packet_type = parser.Iterate();
+ }
+ if (received_psfb && received_remb)
+ observation_complete_->Set();
+ return SEND_PACKET;
+ }
+ } observer;
+
+ CreateCalls(Call::Config(observer.SendTransport()),
+ Call::Config(observer.ReceiveTransport()));
+ observer.SetReceivers(receiver_call_->Receiver(), sender_call_->Receiver());
+ CreateTestConfigs();
+ CreateStreams();
+ CreateFrameGenerator();
+ StartSending();
+
+ EXPECT_EQ(kEventSignaled, observer.Wait())
+ << "Timed out while waiting for a receiver RTCP REMB packet to be sent.";
+
+ StopSending();
+ observer.StopSending();
+ DestroyStreams();
+}
} // namespace webrtc
diff --git a/video/video_receive_stream.cc b/video/video_receive_stream.cc
index 27dafc9..31c8524 100644
--- a/video/video_receive_stream.cc
+++ b/video/video_receive_stream.cc
@@ -61,6 +61,7 @@
assert(config_.rtp.remote_ssrc != config_.rtp.local_ssrc);
rtp_rtcp_->SetLocalSSRC(channel_, config_.rtp.local_ssrc);
+ rtp_rtcp_->SetRembStatus(channel_, false, config_.rtp.remb);
network_ = ViENetwork::GetInterface(video_engine);
assert(network_ != NULL);
diff --git a/video/video_send_stream.cc b/video/video_send_stream.cc
index df03b13..713cdb9 100644
--- a/video/video_send_stream.cc
+++ b/video/video_send_stream.cc
@@ -115,6 +115,8 @@
}
}
+ rtp_rtcp_->SetRembStatus(channel_, true, false);
+
// Enable NACK, FEC or both.
if (config_.rtp.fec.red_payload_type != -1) {
assert(config_.rtp.fec.ulpfec_payload_type != -1);
diff --git a/video_receive_stream.h b/video_receive_stream.h
index e5a6829..e548728 100644
--- a/video_receive_stream.h
+++ b/video_receive_stream.h
@@ -108,7 +108,8 @@
Rtp()
: remote_ssrc(0),
local_ssrc(0),
- rtcp_mode(newapi::kRtcpReducedSize) {}
+ rtcp_mode(newapi::kRtcpReducedSize),
+ remb(false) {}
// Synchronization source (stream identifier) to be received.
uint32_t remote_ssrc;
@@ -118,6 +119,9 @@
// See RtcpMode for description.
newapi::RtcpMode rtcp_mode;
+ // See draft-alvestrand-rmcat-remb for information.
+ bool remb;
+
// See NackConfig for description.
NackConfig nack;
