From cb44343006c1fcf282ae3ce55640ba297be2ad74 Mon Sep 17 00:00:00 2001
From: deadbeef <deadbeef@webrtc.org>
Date: Mon, 12 Dec 2016 11:12:36 -0800
Subject: [PATCH] Add SSRC to RtpEncodingParameters for audio.

Was added for video initially, but not for audio.

BUG=webrtc:6862

Review-Url: https://codereview.webrtc.org/2568553002
Cr-Commit-Position: refs/heads/master@{#15552}
---
 webrtc/media/engine/webrtcvoiceengine.cc        |  2 ++
 .../media/engine/webrtcvoiceengine_unittest.cc  | 17 +++++++++++++++++
 2 files changed, 19 insertions(+)

diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index eec9340640..2a0583c909 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -1217,6 +1217,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
     config_.voe_channel_id = ch;
     config_.rtp.extensions = extensions;
     config_.audio_network_adaptor_config = audio_network_adaptor_config;
+    rtp_parameters_.encodings[0].ssrc = rtc::Optional<uint32_t>(ssrc);
     RecreateAudioSendStream(send_codec_spec);
   }
 
@@ -1742,6 +1743,7 @@ webrtc::RtpParameters WebRtcVoiceMediaChannel::GetRtpReceiveParameters(
   for (const AudioCodec& codec : recv_codecs_) {
     rtp_params.codecs.push_back(codec.ToCodecParameters());
   }
+  rtp_params.encodings[0].ssrc = rtc::Optional<uint32_t>(ssrc);
   return rtp_params;
 }
 
diff --git a/webrtc/media/engine/webrtcvoiceengine_unittest.cc b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
index d90695de26..bddefdd145 100644
--- a/webrtc/media/engine/webrtcvoiceengine_unittest.cc
+++ b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
@@ -1113,6 +1113,14 @@ TEST_F(WebRtcVoiceEngineTestFake, GetRtpSendParametersCodecs) {
   EXPECT_EQ(kPcmuCodec.ToCodecParameters(), rtp_parameters.codecs[1]);
 }
 
+// Test that GetRtpSendParameters returns an SSRC.
+TEST_F(WebRtcVoiceEngineTestFake, GetRtpSendParametersSsrc) {
+  EXPECT_TRUE(SetupSendStream());
+  webrtc::RtpParameters rtp_parameters = channel_->GetRtpSendParameters(kSsrc1);
+  ASSERT_EQ(1u, rtp_parameters.encodings.size());
+  EXPECT_EQ(kSsrc1, rtp_parameters.encodings[0].ssrc);
+}
+
 // Test that if we set/get parameters multiple times, we get the same results.
 TEST_F(WebRtcVoiceEngineTestFake, SetAndGetRtpSendParameters) {
   EXPECT_TRUE(SetupSendStream());
@@ -1146,6 +1154,15 @@ TEST_F(WebRtcVoiceEngineTestFake, GetRtpReceiveParametersCodecs) {
   EXPECT_EQ(kPcmuCodec.ToCodecParameters(), rtp_parameters.codecs[1]);
 }
 
+// Test that GetRtpReceiveParameters returns an SSRC.
+TEST_F(WebRtcVoiceEngineTestFake, GetRtpReceiveParametersSsrc) {
+  EXPECT_TRUE(SetupRecvStream());
+  webrtc::RtpParameters rtp_parameters =
+      channel_->GetRtpReceiveParameters(kSsrc1);
+  ASSERT_EQ(1u, rtp_parameters.encodings.size());
+  EXPECT_EQ(kSsrc1, rtp_parameters.encodings[0].ssrc);
+}
+
 // Test that if we set/get parameters multiple times, we get the same results.
 TEST_F(WebRtcVoiceEngineTestFake, SetAndGetRtpReceiveParameters) {
   EXPECT_TRUE(SetupRecvStream());