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}

pc/rtpsenderreceiver_unittest.cc

diff --git a/pc/rtpsenderreceiver_unittest.cc b/pc/rtpsenderreceiver_unittest.cc
index 03bb84b..6334f72 100644
--- a/pc/rtpsenderreceiver_unittest.cc
+++ b/pc/rtpsenderreceiver_unittest.cc
@@ -12,6 +12,7 @@
 #include <string>
 #include <utility>
 
+#include "api/rtpparameters.h"
 #include "media/base/fakemediaengine.h"
 #include "media/base/rtpdataengine.h"
 #include "media/engine/fakewebrtccall.h"
@@ -600,6 +601,28 @@
   DestroyAudioRtpSender();
 }
 
+TEST_F(RtpSenderReceiverTest, SetAudioBitratePriority) {
+  CreateAudioRtpSender();
+
+  webrtc::RtpParameters params = audio_rtp_sender_->GetParameters();
+  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(webrtc::kDefaultBitratePriority,
+            params.encodings[0].bitrate_priority);
+  double new_bitrate_priority = 2.0;
+  params.encodings[0].bitrate_priority = new_bitrate_priority;
+  EXPECT_TRUE(audio_rtp_sender_->SetParameters(params));
+
+  params = audio_rtp_sender_->GetParameters();
+  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(new_bitrate_priority, params.encodings[0].bitrate_priority);
+
+  params = voice_media_channel_->GetRtpSendParameters(kAudioSsrc);
+  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(new_bitrate_priority, params.encodings[0].bitrate_priority);
+
+  DestroyAudioRtpSender();
+}
+
 TEST_F(RtpSenderReceiverTest, VideoSenderCanSetParameters) {
   CreateVideoRtpSender();
 
@@ -636,6 +659,28 @@
   DestroyVideoRtpSender();
 }
 
+TEST_F(RtpSenderReceiverTest, SetVideoBitratePriority) {
+  CreateVideoRtpSender();
+
+  webrtc::RtpParameters params = video_rtp_sender_->GetParameters();
+  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(webrtc::kDefaultBitratePriority,
+            params.encodings[0].bitrate_priority);
+  double new_bitrate_priority = 2.0;
+  params.encodings[0].bitrate_priority = new_bitrate_priority;
+  EXPECT_TRUE(video_rtp_sender_->SetParameters(params));
+
+  params = video_rtp_sender_->GetParameters();
+  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(new_bitrate_priority, params.encodings[0].bitrate_priority);
+
+  params = video_media_channel_->GetRtpSendParameters(kVideoSsrc);
+  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(new_bitrate_priority, params.encodings[0].bitrate_priority);
+
+  DestroyVideoRtpSender();
+}
+
 TEST_F(RtpSenderReceiverTest, AudioReceiverCanSetParameters) {
   CreateAudioRtpReceiver();