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Prevent RTCP SR to be sent with bogus timestamp.

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This CL makes sure no RTCP SR is sent before there is a valid timestamp
to set in the SR, based on the first sent media packet.

BUG=webrtc:1600
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1506103006 .

Cr-Commit-Position: refs/heads/master@{#10964}
This commit is contained in:
mflodman 2015-12-10 10:10:44 +01:00
parent 48bf2382d9
commit 0b3d7eec07
2 changed files with 29 additions and 3 deletions
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10
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
View File

@ -183,8 +183,13 @@ int32_t ModuleRtpRtcpImpl::Process() {
set_rtt_ms(rtt_stats_->LastProcessedRtt());
}
if (rtcp_sender_.TimeToSendRTCPReport())
rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport);
// For sending streams, make sure to not send a SR before media has been sent.
if (rtcp_sender_.TimeToSendRTCPReport()) {
RTCPSender::FeedbackState state = GetFeedbackState();
// Prevent sending streams to send SR before any media has been sent.
if (!rtcp_sender_.Sending() || state.packets_sent > 0)
rtcp_sender_.SendRTCP(state, kRtcpReport);
}
if (UpdateRTCPReceiveInformationTimers()) {
// A receiver has timed out
@ -402,6 +407,7 @@ int32_t ModuleRtpRtcpImpl::SendOutgoingData(
const RTPFragmentationHeader* fragmentation,
const RTPVideoHeader* rtp_video_hdr) {
rtcp_sender_.SetLastRtpTime(time_stamp, capture_time_ms);
// Make sure an RTCP report isn't queued behind a key frame.
if (rtcp_sender_.TimeToSendRTCPReport(kVideoFrameKey == frame_type)) {
rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport);
}

22
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
View File

@ -346,6 +346,27 @@ TEST_F(RtpRtcpImplTest, RttForReceiverOnly) {
EXPECT_EQ(2 * kOneWayNetworkDelayMs, receiver_.impl_->rtt_ms());
}
TEST_F(RtpRtcpImplTest, NoSrBeforeMedia) {
// Ignore fake transport delays in this test.
sender_.transport_.SimulateNetworkDelay(0, &clock_);
receiver_.transport_.SimulateNetworkDelay(0, &clock_);
sender_.impl_->Process();
EXPECT_EQ(-1, sender_.RtcpSent().first_packet_time_ms);
// Verify no SR is sent before media has been sent, RR should still be sent
// from the receiving module though.
clock_.AdvanceTimeMilliseconds(2000);
int64_t current_time = clock_.TimeInMilliseconds();
sender_.impl_->Process();
receiver_.impl_->Process();
EXPECT_EQ(-1, sender_.RtcpSent().first_packet_time_ms);
EXPECT_EQ(receiver_.RtcpSent().first_packet_time_ms, current_time);
SendFrame(&sender_, kBaseLayerTid);
EXPECT_EQ(sender_.RtcpSent().first_packet_time_ms, current_time);
}
TEST_F(RtpRtcpImplTest, RtcpPacketTypeCounter_Nack) {
EXPECT_EQ(-1, receiver_.RtcpSent().first_packet_time_ms);
EXPECT_EQ(-1, sender_.RtcpReceived().first_packet_time_ms);
@ -522,5 +543,4 @@ TEST_F(RtpRtcpImplTest, UniqueNackRequests) {
EXPECT_EQ(6U, sender_.RtcpReceived().unique_nack_requests);
EXPECT_EQ(75, sender_.RtcpReceived().UniqueNackRequestsInPercent());
}
} // namespace webrtc