Set local ssrc at construction of Rtp module
The SetSSRC() method is slated for removal, make sure we set the local
SSRC at construction time.
Bug: webrtc:10774
Change-Id: I431e828caf60c5e0134adbe82d1d3345745cc6ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149827
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28926}
diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index 876ee69..f57858c 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -142,7 +142,6 @@
int payload_frequency) override;
// RTP+RTCP
- void SetLocalSSRC(uint32_t ssrc) override;
void SetRid(const std::string& rid,
int extension_id,
int repaired_extension_id) override;
@@ -279,7 +278,7 @@
int media_transport_sequence_number_ RTC_GUARDED_BY(encoder_queue_) = 0;
rtc::CriticalSection media_transport_lock_;
- // Currently set by SetLocalSSRC.
+ // Currently set to local SSRC at construction.
uint64_t media_transport_channel_id_ RTC_GUARDED_BY(&media_transport_lock_) =
0;
// Cache payload type and sampling frequency from most recent call to
@@ -702,6 +701,10 @@
configuration.rtcp_report_interval_ms = rtcp_report_interval_ms;
configuration.local_media_ssrc = ssrc;
+ if (media_transport_config_.media_transport) {
+ rtc::CritScope cs(&media_transport_lock_);
+ media_transport_channel_id_ = ssrc;
+ }
_rtpRtcpModule = RtpRtcp::Create(configuration);
_rtpRtcpModule->SetSendingMediaStatus(false);
@@ -951,17 +954,6 @@
payload_frequency, 0, 0);
}
-void ChannelSend::SetLocalSSRC(uint32_t ssrc) {
- RTC_DCHECK_RUN_ON(&worker_thread_checker_);
- RTC_DCHECK(!sending_);
-
- if (media_transport_config_.media_transport) {
- rtc::CritScope cs(&media_transport_lock_);
- media_transport_channel_id_ = ssrc;
- }
- _rtpRtcpModule->SetSSRC(ssrc);
-}
-
void ChannelSend::SetRid(const std::string& rid,
int extension_id,
int repaired_extension_id) {