Remove some usage of RtpRtcp::SetSSRC()
Bug: webrtc:10774
Change-Id: Ib8fa84f5d70ceb7e715405eae2901bcd7bdfebfe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146984
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28895}
diff --git a/call/flexfec_receive_stream_impl.cc b/call/flexfec_receive_stream_impl.cc
index 39b53af..f466cad 100644
--- a/call/flexfec_receive_stream_impl.cc
+++ b/call/flexfec_receive_stream_impl.cc
@@ -122,15 +122,16 @@
std::unique_ptr<RtpRtcp> CreateRtpRtcpModule(
Clock* clock,
ReceiveStatistics* receive_statistics,
- Transport* rtcp_send_transport,
+ const FlexfecReceiveStreamImpl::Config& config,
RtcpRttStats* rtt_stats) {
RtpRtcp::Configuration configuration;
configuration.audio = false;
configuration.receiver_only = true;
configuration.clock = clock;
configuration.receive_statistics = receive_statistics;
- configuration.outgoing_transport = rtcp_send_transport;
+ configuration.outgoing_transport = config.rtcp_send_transport;
configuration.rtt_stats = rtt_stats;
+ configuration.media_send_ssrc = config.local_ssrc;
return RtpRtcp::Create(configuration);
}
@@ -150,14 +151,13 @@
rtp_receive_statistics_(ReceiveStatistics::Create(clock)),
rtp_rtcp_(CreateRtpRtcpModule(clock,
rtp_receive_statistics_.get(),
- config_.rtcp_send_transport,
+ config_,
rtt_stats)),
process_thread_(process_thread) {
RTC_LOG(LS_INFO) << "FlexfecReceiveStreamImpl: " << config_.ToString();
// RTCP reporting.
rtp_rtcp_->SetRTCPStatus(config_.rtcp_mode);
- rtp_rtcp_->SetSSRC(config_.local_ssrc);
process_thread_->RegisterModule(rtp_rtcp_.get(), RTC_FROM_HERE);
// Register with transport.
diff --git a/call/rtp_video_sender.cc b/call/rtp_video_sender.cc
index 50cafb4..77f2ba9 100644
--- a/call/rtp_video_sender.cc
+++ b/call/rtp_video_sender.cc
@@ -110,15 +110,23 @@
std::vector<RtpStreamSender> rtp_streams;
const std::vector<uint32_t>& flexfec_protected_ssrcs =
rtp_config.flexfec.protected_media_ssrcs;
- for (uint32_t ssrc : rtp_config.ssrcs) {
+ RTC_DCHECK(rtp_config.rtx.ssrcs.empty() ||
+ rtp_config.rtx.ssrcs.size() == rtp_config.rtx.ssrcs.size());
+ for (size_t i = 0; i < rtp_config.ssrcs.size(); ++i) {
+ configuration.media_send_ssrc = rtp_config.ssrcs[i];
bool enable_flexfec = flexfec_sender != nullptr &&
std::find(flexfec_protected_ssrcs.begin(),
flexfec_protected_ssrcs.end(),
- ssrc) != flexfec_protected_ssrcs.end();
+ *configuration.media_send_ssrc) !=
+ flexfec_protected_ssrcs.end();
configuration.flexfec_sender = enable_flexfec ? flexfec_sender : nullptr;
auto playout_delay_oracle = absl::make_unique<PlayoutDelayOracle>();
configuration.ack_observer = playout_delay_oracle.get();
+ if (rtp_config.rtx.ssrcs.size() > i) {
+ configuration.rtx_send_ssrc = rtp_config.rtx.ssrcs[i];
+ }
+
auto rtp_rtcp = RtpRtcp::Create(configuration);
rtp_rtcp->SetSendingStatus(false);
rtp_rtcp->SetSendingMediaStatus(false);
@@ -571,7 +579,6 @@
for (size_t i = 0; i < rtp_config_.ssrcs.size(); ++i) {
uint32_t ssrc = rtp_config_.ssrcs[i];
RtpRtcp* const rtp_rtcp = rtp_streams_[i].rtp_rtcp.get();
- rtp_rtcp->SetSSRC(ssrc);
// Restore RTP state if previous existed.
auto it = suspended_ssrcs_.find(ssrc);
@@ -587,12 +594,10 @@
if (rtp_config_.rtx.ssrcs.empty())
return;
- // Configure RTX SSRCs.
RTC_DCHECK_EQ(rtp_config_.rtx.ssrcs.size(), rtp_config_.ssrcs.size());
for (size_t i = 0; i < rtp_config_.rtx.ssrcs.size(); ++i) {
uint32_t ssrc = rtp_config_.rtx.ssrcs[i];
RtpRtcp* const rtp_rtcp = rtp_streams_[i].rtp_rtcp.get();
- rtp_rtcp->SetRtxSsrc(ssrc);
auto it = suspended_ssrcs_.find(ssrc);
if (it != suspended_ssrcs_.end())
rtp_rtcp->SetRtxState(it->second);
diff --git a/modules/rtp_rtcp/source/nack_rtx_unittest.cc b/modules/rtp_rtcp/source/nack_rtx_unittest.cc
index 5c87f67..1f5d64a 100644
--- a/modules/rtp_rtcp/source/nack_rtx_unittest.cc
+++ b/modules/rtp_rtcp/source/nack_rtx_unittest.cc
@@ -135,11 +135,11 @@
configuration.receive_statistics = receive_statistics_.get();
configuration.outgoing_transport = &transport_;
configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
+ configuration.media_send_ssrc = kTestSsrc;
rtp_rtcp_module_ = RtpRtcp::Create(configuration);
rtp_sender_video_ = absl::make_unique<RTPSenderVideo>(
&fake_clock, rtp_rtcp_module_->RtpSender(), nullptr,
&playout_delay_oracle_, nullptr, false, false, FieldTrialBasedConfig());
- rtp_rtcp_module_->SetSSRC(kTestSsrc);
rtp_rtcp_module_->SetRTCPStatus(RtcpMode::kCompound);
rtp_rtcp_module_->SetStorePacketsStatus(true, 600);
EXPECT_EQ(0, rtp_rtcp_module_->SetSendingStatus(true));
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
index a0690c1..f4553e1 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
@@ -180,7 +180,6 @@
void SetUp() override {
// Send module.
- sender_.impl_->SetSSRC(kSenderSsrc);
EXPECT_EQ(0, sender_.impl_->SetSendingStatus(true));
sender_.impl_->SetSendingMediaStatus(true);
sender_.SetRemoteSsrc(kReceiverSsrc);
