commit cf439a04e5df8889f4be985c1b76aa09b7d3e721
author Per K <perkj@webrtc.org> Thu Jan 05 13:01:39 2023
committer WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> Thu Jan 05 13:54:02 2023
tree 3f38c893d196fa439fed6ff82724a17b8b173782
parent 7f8680cf6fb239a2118ebae56ad255b9e8ee5fc6

Introduce PacketReceiver::DeliverRtpPacket and PacketReceier::DeliverRtcpPacket

DeliverRtpPacket use a parsed RTP packet as argument where the RTP extensions are supposed to be known.
This method is implemented in webrt::Call and temporary used by the exising method  Call::DeliverRtp, but the idea is to instead avoid extra packet parsing by forwarding a RtpPacketReceived from RtpTransport::DemuxRtpPacket via  WebrtcVideoChannel::OnPacketReceived and WebrtcVoiceChannel.

DeliverRtcpPacket is also implemented in Call and is directly used in PeerConnection::InitializeRtcpCallback.

Bug: webrtc:14795, webrtc:7135
Change-Id: Ib6ffe8e1229ac07fa459ee2fc9a0af8455a23bac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290401
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39015}

call/packet_receiver.h

diff --git a/call/packet_receiver.h b/call/packet_receiver.h
index 13d3b84..a97bb96 100644
--- a/call/packet_receiver.h
+++ b/call/packet_receiver.h
@@ -10,7 +10,10 @@
 #ifndef CALL_PACKET_RECEIVER_H_
 #define CALL_PACKET_RECEIVER_H_
 
+#include "absl/functional/any_invocable.h"
 #include "api/media_types.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "rtc_base/checks.h"
 #include "rtc_base/copy_on_write_buffer.h"
 
 namespace webrtc {
@@ -27,6 +30,28 @@
                                        rtc::CopyOnWriteBuffer packet,
                                        int64_t packet_time_us) = 0;
 
+  // Demux RTCP packets. Must be called on the worker thread.
+  virtual void DeliverRtcpPacket(rtc::CopyOnWriteBuffer packet) {
+    // TODO(perkj, https://bugs.webrtc.org/7135): Implement in FakeCall and
+    // FakeNetworkPipe.
+    RTC_CHECK_NOTREACHED();
+  }
+
+  // Invoked once when a packet packet is received that can not be demuxed.
+  // If the method returns true, a new attempt is made to demux the packet.
+  using OnUndemuxablePacketHandler =
+      absl::AnyInvocable<bool(const RtpPacketReceived& parsed_packet)>;
+
+  // Demux RTP packets. Must be called on the worker thread.
+  virtual void DeliverRtpPacket(
+      MediaType media_type,
+      RtpPacketReceived packet,
+      OnUndemuxablePacketHandler undemuxable_packet_handler) {
+    // TODO(perkj, https://bugs.webrtc.org/7135): Implement in FakeCall and
+    // FakeNetworkPipe.
+    RTC_CHECK_NOTREACHED();
+  }
+
  protected:
   virtual ~PacketReceiver() {}
 };