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Add documentation+DCHECK for the packet path from transport to call.

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An upcoming change will remove PeerConnection's involvement in
the RTCP packet path and keep the flow consistent with the RTP packet
path.

Bug: webrtc:11993
Change-Id: I7ce1bbe8bbc352a49310e2e55b55ca5d8d927935
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217389
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33927}
This commit is contained in:
Tommi 2021-05-05 11:04:30 +02:00 committed by WebRTC LUCI CQ
parent bb7ee95c71
commit 3f418ccf99
1 changed files with 14 additions and 0 deletions
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14
call/call.cc
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@ -1363,6 +1363,19 @@ PacketReceiver::DeliveryStatus Call::DeliverRtcp(MediaType media_type,
const uint8_t* packet,
size_t length) {
TRACE_EVENT0("webrtc", "Call::DeliverRtcp");
// TODO(bugs.webrtc.org/11993): This DCHECK is here just to maintain the
// invariant that currently the only call path to this function is via
// `PeerConnection::InitializeRtcpCallback()`. DeliverRtp on the other hand
// gets called via the channel classes and
// WebRtc[Audio|Video]Channel's `OnPacketReceived`. We'll remove the
// PeerConnection involvement as well as
// `JsepTransportController::OnRtcpPacketReceived_n` and `rtcp_handler`
// and make sure that the flow of packets is consistent from the
// `RtpTransport` class, via the *Channel and *Engine classes and into Call.
// This way we'll also know more about the context of the packet.
RTC_DCHECK_EQ(media_type, MediaType::ANY);
// TODO(pbos): Make sure it's a valid packet.
// Return DELIVERY_UNKNOWN_SSRC if it can be determined that
// there's no receiver of the packet.
@ -1409,6 +1422,7 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
rtc::CopyOnWriteBuffer packet,
int64_t packet_time_us) {
TRACE_EVENT0("webrtc", "Call::DeliverRtp");
RTC_DCHECK_NE(media_type, MediaType::ANY);
RtpPacketReceived parsed_packet;
if (!parsed_packet.Parse(std::move(packet)))