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_receive.h b/audio/channel_receive.h
index 1fe64b9..dadeab3 100644
--- a/audio/channel_receive.h
+++ b/audio/channel_receive.h
@@ -116,9 +116,6 @@
// Produces the transport-related timestamps; current_delay_ms is left unset.
virtual absl::optional<Syncable::Info> GetSyncInfo() const = 0;
- // RTP+RTCP
- virtual void SetLocalSSRC(uint32_t ssrc) = 0;
-
virtual void RegisterReceiverCongestionControlObjects(
PacketRouter* packet_router) = 0;
virtual void ResetReceiverCongestionControlObjects() = 0;
@@ -145,6 +142,7 @@
const MediaTransportConfig& media_transport_config,
Transport* rtcp_send_transport,
RtcEventLog* rtc_event_log,
+ uint32_t local_ssrc,
uint32_t remote_ssrc,
size_t jitter_buffer_max_packets,
bool jitter_buffer_fast_playout,