RtpPacketReceiver base class and OnRtpPacket, with a pre-parsed RTP packet.

webrtc/audio/audio_receive_stream.cc

Index: webrtc/audio/audio_receive_stream.cc
diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc
index 05d6edfa4c1fccd33b953ec2e38c3c456eefcb83..fbbff69bb9e9dbf487053e262e2b9a8c00821f03 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -67,6 +67,7 @@ AudioReceiveStream::AudioReceiveStream(
     const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
     webrtc::RtcEventLog* event_log)
     : config_(config),
+      rtp_config_(config.rtp.extensions, config.rtp.transport_cc),
       audio_state_(audio_state) {
   LOG(LS_INFO) << "AudioReceiveStream: " << config_.ToString();
   RTC_DCHECK_NE(config_.voe_channel_id, -1);
@@ -302,14 +303,16 @@ bool AudioReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) {
   return channel_proxy_->ReceivedRTCPPacket(packet, length);
 }
 
-bool AudioReceiveStream::DeliverRtp(const uint8_t* packet,
-                                    size_t length,
-                                    const PacketTime& packet_time) {
+const RtpPacketReceiver::RtpConfig& AudioReceiveStream::rtp_config() const {
+  return rtp_config_;
+}
+
+bool AudioReceiveStream::OnRtpPacket(const RtpPacketReceived& packet) {

[Taylor Brandstetter, 2017/02/09 20:20:11]

Personally, I prefer the "DeliverRtpPacket" style of naming. The
"OnReceivedPacket" style of naming is mostly used for private slots (for
example,
https://cs.chromium.org/chromium/src/third_party/webrtc/pc/channel.h?type=cs&...).

[nisse-webrtc, 2017/02/10 08:09:23]

I picked up the On* style from the VideoSinkInterface::OnFrame which we added a
while ago, and I think about it as an analogous RTP packet sink. I'd also
consider dropping the return value, making it a fire-and-forget method, but I'm
not sure if there's some code that really handles errors (besides logging and
stats). One exception is the UNKNWO_SSRC error from Call::DeliverRtp, but I hope
to reorganize the handling of unsignalled ssrcs so that case disappears.

   // TODO(solenberg): Tests call this function on a network thread, libjingle
   // calls on the worker thread. We should move towards always using a network
   // thread. Then this check can be enabled.
   // RTC_DCHECK(!thread_checker_.CalledOnValidThread());
-  return channel_proxy_->ReceivedRTPPacket(packet, length, packet_time);
+  return channel_proxy_->OnRTPPacket(packet);

[stefan-webrtc, 2017/02/09 13:40:06]

OnRtpPacket

[nisse-webrtc, 2017/02/10 08:09:23]

Done.

 }
 
 const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const {