commit 24722b3c8487f92e0e7994ddf2f605cc5009a55f
author Seth Hampson <shampson@webrtc.org> Fri Dec 22 17:36:42 2017
committer Commit Bot <commit-bot@chromium.org> Mon Jan 08 18:57:19 2018
tree 997d035be842ea0c3d2abeb6ee393459007fe711
parent a9329dbae21cabd4b860e729ecd7769f5bec9ac7

Reland "Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator."

This is a reland of d2b912aed132c751919ed286439fb39bbd714dda
Original change's description:
> Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator.
>
> I followed the wiring path for the max bitrate.
> Doc:
> https://docs.google.com/a/google.com/document/d/1sGT6y00prOIErFuGD44zWZacDpR6Rkjg_HXA_Z3Vw4Q/edit?usp=sharing
>
> Bug: webrtc:8630
> Change-Id: I6b861816670442656721c20f81d035ee5eb6218c
> Reviewed-on: https://webrtc-review.googlesource.com/30380
> Commit-Queue: Seth Hampson <shampson@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Elad Alon <eladalon@webrtc.org>
> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21397}

TBR=solenberg@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org,stefan@webrtc.org

Bug: webrtc:8630
Change-Id: I7429d9e270c9ecb4dfaf6aef85d3055c47658631
Reviewed-on: https://webrtc-review.googlesource.com/35600
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21524}

audio/audio_send_stream.cc

diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index ff4ae97..2e5654a 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -243,7 +243,8 @@
        !webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe"))) {
     // Audio BWE is enabled.
     transport_->packet_sender()->SetAccountForAudioPackets(true);
-    ConfigureBitrateObserver(config_.min_bitrate_bps, config_.max_bitrate_bps);
+    ConfigureBitrateObserver(config_.min_bitrate_bps, config_.max_bitrate_bps,
+                             config_.bitrate_priority);
   }
   channel_proxy_->StartSend();
   sending_ = true;
@@ -631,6 +632,7 @@
       FindExtensionIds(new_config.rtp.extensions).transport_sequence_number;
   if (stream->config_.min_bitrate_bps == new_config.min_bitrate_bps &&
       stream->config_.max_bitrate_bps == new_config.max_bitrate_bps &&
+      stream->config_.bitrate_priority == new_config.bitrate_priority &&
       (FindExtensionIds(stream->config_.rtp.extensions)
                .transport_sequence_number == new_transport_seq_num_id ||
        !webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe"))) {
@@ -641,14 +643,16 @@
       (new_transport_seq_num_id != 0 ||
        !webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe"))) {
     stream->ConfigureBitrateObserver(new_config.min_bitrate_bps,
-                                     new_config.max_bitrate_bps);
+                                     new_config.max_bitrate_bps,
+                                     new_config.bitrate_priority);
   } else {
     stream->RemoveBitrateObserver();
   }
 }
 
 void AudioSendStream::ConfigureBitrateObserver(int min_bitrate_bps,
-                                               int max_bitrate_bps) {
+                                               int max_bitrate_bps,
+                                               double bitrate_priority) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   RTC_DCHECK_GE(max_bitrate_bps, min_bitrate_bps);
   rtc::Event thread_sync_event(false /* manual_reset */, false);
@@ -657,8 +661,10 @@
     // sure the bitrate limits in config_ are up-to-date.
     config_.min_bitrate_bps = min_bitrate_bps;
     config_.max_bitrate_bps = max_bitrate_bps;
+    config_.bitrate_priority = bitrate_priority;
+    // This either updates the current observer or adds a new observer.
     bitrate_allocator_->AddObserver(this, min_bitrate_bps, max_bitrate_bps, 0,
-                                    true, config_.track_id);
+                                    true, config_.track_id, bitrate_priority);
     thread_sync_event.Set();
   });
   thread_sync_event.Wait(rtc::Event::kForever);