Minor fixes and refactoring for RtpTransport until the Demux.
This change fixes some inefficiencies and quirks in the code that
originates in RtpTransport leading up to the demux.
This work is in preparation for more refactoring of the Demux stage
onwards.
Bug: webrtc:10297
Change-Id: I7b8f00134657d62c722939618a55a91a2b6040bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128220
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27185}
diff --git a/pc/srtp_transport.h b/pc/srtp_transport.h
index 7512711..e725733 100644
--- a/pc/srtp_transport.h
+++ b/pc/srtp_transport.h
@@ -116,9 +116,9 @@
void ConnectToRtpTransport();
void CreateSrtpSessions();
- void OnRtpPacketReceived(rtc::CopyOnWriteBuffer* packet,
+ void OnRtpPacketReceived(rtc::CopyOnWriteBuffer packet,
int64_t packet_time_us) override;
- void OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet,
+ void OnRtcpPacketReceived(rtc::CopyOnWriteBuffer packet,
int64_t packet_time_us) override;
void OnNetworkRouteChanged(
absl::optional<rtc::NetworkRoute> network_route) override;