From 31fec4048248364070fca8a3ff554c9caa5b1b81 Mon Sep 17 00:00:00 2001
From: solenberg <solenberg@webrtc.org>
Date: Fri, 6 May 2016 02:13:12 -0700
Subject: [PATCH] Set rtcp_send_transport for AudioReceiveStreams. This was
 forgotten in https://codereview.webrtc.org/1909333002/.

BUG=webrtc:4690, webrtc:5079, webrtc:5762

Review-Url: https://codereview.webrtc.org/1951833002
Cr-Commit-Position: refs/heads/master@{#12640}
---
 webrtc/media/engine/webrtcvoiceengine.cc      |  6 ++--
 .../engine/webrtcvoiceengine_unittest.cc      | 29 +++++++++++++++++++
 2 files changed, 33 insertions(+), 2 deletions(-)

diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index 464d6133d0..d873d8b767 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -1262,12 +1262,14 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
                            bool use_transport_cc,
                            const std::string& sync_group,
                            const std::vector<webrtc::RtpExtension>& extensions,
-                           webrtc::Call* call)
+                           webrtc::Call* call,
+                           webrtc::Transport* rtcp_send_transport)
       : call_(call), config_() {
     RTC_DCHECK_GE(ch, 0);
     RTC_DCHECK(call);
     config_.rtp.remote_ssrc = remote_ssrc;
     config_.rtp.local_ssrc = local_ssrc;
+    config_.rtcp_send_transport = rtcp_send_transport;
     config_.voe_channel_id = ch;
     config_.sync_group = sync_group;
     RecreateAudioReceiveStream(use_transport_cc, extensions);
@@ -2099,7 +2101,7 @@ bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
       ssrc, new WebRtcAudioReceiveStream(channel, ssrc, receiver_reports_ssrc_,
                                          recv_transport_cc_enabled_,
                                          sp.sync_label, recv_rtp_extensions_,
-                                         call_)));
+                                         call_, this)));
 
   SetNack(channel, send_codec_spec_.nack_enabled);
   SetPlayout(channel, playout_);
diff --git a/webrtc/media/engine/webrtcvoiceengine_unittest.cc b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
index e5af99fca7..5d8dd90d02 100644
--- a/webrtc/media/engine/webrtcvoiceengine_unittest.cc
+++ b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
@@ -494,6 +494,35 @@ TEST_F(WebRtcVoiceEngineTestFake, CreateChannel) {
   EXPECT_TRUE(SetupChannel());
 }
 
+// Test that we can add a send stream and that it has the correct defaults.
+TEST_F(WebRtcVoiceEngineTestFake, CreateSendStream) {
+  EXPECT_TRUE(SetupChannel());
+  EXPECT_TRUE(
+      channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrc1)));
+  const webrtc::AudioSendStream::Config& config = GetSendStreamConfig(kSsrc1);
+  EXPECT_EQ(kSsrc1, config.rtp.ssrc);
+  EXPECT_EQ("", config.rtp.c_name);
+  EXPECT_EQ(0u, config.rtp.extensions.size());
+  EXPECT_EQ(static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_),
+            config.send_transport);
+}
+
+// Test that we can add a receive stream and that it has the correct defaults.
+TEST_F(WebRtcVoiceEngineTestFake, CreateRecvStream) {
+  EXPECT_TRUE(SetupChannel());
+  EXPECT_TRUE(
+      channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrc1)));
+  const webrtc::AudioReceiveStream::Config& config =
+      GetRecvStreamConfig(kSsrc1);
+  EXPECT_EQ(kSsrc1, config.rtp.remote_ssrc);
+  EXPECT_EQ(0xFA17FA17, config.rtp.local_ssrc);
+  EXPECT_FALSE(config.rtp.transport_cc);
+  EXPECT_EQ(0u, config.rtp.extensions.size());
+  EXPECT_EQ(static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_),
+            config.rtcp_send_transport);
+  EXPECT_EQ("", config.sync_group);
+}
+
 // Tests that the list of supported codecs is created properly and ordered
 // correctly (such that opus appears first).
 TEST_F(WebRtcVoiceEngineTestFake, CodecOrder) {