commit 179a3923b9e402f427728d52b3024a3de1696a66
author Piotr (Peter) Slatala <psla@webrtc.org> Fri Nov 16 09:57:58 2018 -0800
committer Commit Bot <commit-bot@chromium.org> Fri Nov 16 19:31:11 2018 +0000
tree 493243790bfaaf67e5e555928dc099fa37782179
parent 8c1e73b024dae7d6fbecc85e890a7af97f8468db

Implement TargetBitrate, NetworkRoute and overhead features of media transport interface.

So far ANA was not available for media transport interface. With recent changes to media transport, we can now account for packet overhead, network route (ip/tcp/udp/turn overheads) and we can also use bandwidth estimate from the media transport.


Bug: webrtc:9719
Change-Id: I98c9a09dd418b763c339ee2ee05592e164cf9199
Reviewed-on: https://webrtc-review.googlesource.com/c/110367
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25677}

audio/channel_send.cc

diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index a8b93cc..e96ef4a 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -479,15 +479,23 @@
   audio_coding_.reset(AudioCodingModule::Create(AudioCodingModule::Config()));
 
   RtpRtcp::Configuration configuration;
+
+  // We gradually remove codepaths that depend on RTP when using media
+  // transport. All of this logic should be moved to the future
+  // RTPMediaTransport. In this case it means that overhead and bandwidth
+  // observers should not be called when using media transport.
+  if (!media_transport_) {
+    configuration.overhead_observer = this;
+    configuration.bandwidth_callback = rtcp_observer_.get();
+    configuration.transport_feedback_callback = feedback_observer_proxy_.get();
+  }
+
   configuration.audio = true;
   configuration.outgoing_transport = this;
-  configuration.overhead_observer = this;
-  configuration.bandwidth_callback = rtcp_observer_.get();
 
   configuration.paced_sender = rtp_packet_sender_proxy_.get();
   configuration.transport_sequence_number_allocator =
       seq_num_allocator_proxy_.get();
-  configuration.transport_feedback_callback = feedback_observer_proxy_.get();
 
   configuration.event_log = event_log_;
   configuration.rtt_stats = rtcp_rtt_stats;
@@ -500,6 +508,16 @@
   _rtpRtcpModule.reset(RtpRtcp::CreateRtpRtcp(configuration));
   _rtpRtcpModule->SetSendingMediaStatus(false);
 
+  // We want to invoke the 'TargetRateObserver' and |OnOverheadChanged|
+  // callbacks after the audio_coding_ is fully initialized.
+  if (media_transport_) {
+    RTC_DLOG(LS_INFO) << "Setting media_transport_ rate observers.";
+    media_transport_->AddTargetTransferRateObserver(this);
+    OnOverheadChanged(media_transport_->GetAudioPacketOverhead());
+  } else {
+    RTC_DLOG(LS_INFO) << "Not setting media_transport_ rate observers.";
+  }
+
   channel_state_.Reset();
 
   _moduleProcessThreadPtr->RegisterModule(_rtpRtcpModule.get(), RTC_FROM_HERE);
@@ -522,6 +540,10 @@
 ChannelSend::~ChannelSend() {
   RTC_DCHECK(construction_thread_.CalledOnValidThread());
 
+  if (media_transport_) {
+    media_transport_->RemoveTargetTransferRateObserver(this);
+  }
+
   StopSend();
 
   int error = audio_coding_->RegisterTransportCallback(NULL);
@@ -693,6 +715,12 @@
 
 // TODO(nisse): Delete always-true return value.
 bool ChannelSend::ReceivedRTCPPacket(const uint8_t* data, size_t length) {
+  if (media_transport_) {
+    // Ignore RTCP packets while media transport is used.
+    // Those packets should not arrive, but we are seeing occasional packets.
+    return 0;
+  }
+
   // Deliver RTCP packet to RTP/RTCP module for parsing
   _rtpRtcpModule->IncomingRtcpPacket(data, length);
 
@@ -710,11 +738,7 @@
   }
   retransmission_rate_limiter_->SetWindowSize(nack_window_ms);
 
-  // Invoke audio encoders OnReceivedRtt().
-  audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
-    if (*encoder)
-      (*encoder)->OnReceivedRtt(rtt);
-  });
+  OnReceivedRtt(rtt);
   return true;
 }
 
@@ -1010,6 +1034,19 @@
 }
 
 int64_t ChannelSend::GetRTT() const {
+  if (media_transport_) {
+    // GetRTT is generally used in the RTCP codepath, where media transport is
+    // not present and so it shouldn't be needed. But it's also invoked in
+    // 'GetStats' method, and for now returning media transport RTT here gives
+    // us "free" rtt stats for media transport.
+    auto target_rate = media_transport_->GetLatestTargetTransferRate();
+    if (target_rate.has_value()) {
+      return target_rate.value().network_estimate.round_trip_time.ms();
+    }
+
+    return 0;
+  }
+
   RtcpMode method = _rtpRtcpModule->RTCP();
   if (method == RtcpMode::kOff) {
     return 0;
@@ -1046,5 +1083,20 @@
   }
 }
 
+void ChannelSend::OnTargetTransferRate(TargetTransferRate rate) {
+  RTC_DCHECK(media_transport_);
+  OnReceivedRtt(rate.network_estimate.round_trip_time.ms());
+}
+
+void ChannelSend::OnReceivedRtt(int64_t rtt_ms) {
+  // Invoke audio encoders OnReceivedRtt().
+  audio_coding_->ModifyEncoder(
+      [rtt_ms](std::unique_ptr<AudioEncoder>* encoder) {
+        if (*encoder) {
+          (*encoder)->OnReceivedRtt(rtt_ms);
+        }
+      });
+}
+
 }  // namespace voe
 }  // namespace webrtc