From 55dd70842c3d2741b08da8e2807cc233ee797ccc Mon Sep 17 00:00:00 2001
From: Taylor Brandstetter <deadbeef@webrtc.org>
Date: Tue, 3 May 2016 13:50:11 -0700
Subject: [PATCH] Support RtpEncodingParameters::active in voice engine.

R=solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/1943073003 .

Cr-Commit-Position: refs/heads/master@{#12615}
---
 webrtc/media/engine/webrtcvoiceengine.cc      |  9 ++++++---
 .../engine/webrtcvoiceengine_unittest.cc      | 20 +++++++++++++++++++
 2 files changed, 26 insertions(+), 3 deletions(-)

diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index ed67ce84ba..464d6133d0 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -1216,16 +1216,19 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
     return rtp_parameters_;
   }
 
-  void set_rtp_parameters(const webrtc::RtpParameters& parameters) {
+  void SetRtpParameters(const webrtc::RtpParameters& parameters) {
     RTC_CHECK_EQ(1UL, parameters.encodings.size());
     rtp_parameters_ = parameters;
+    // parameters.encodings[0].active could have changed.
+    UpdateSendState();
   }
 
  private:
   void UpdateSendState() {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
     RTC_DCHECK(stream_);
-    if (send_ && source_ != nullptr) {
+    RTC_DCHECK_EQ(1UL, rtp_parameters_.encodings.size());
+    if (send_ && source_ != nullptr && rtp_parameters_.encodings[0].active) {
       stream_->Start();
     } else {  // !send || source_ = nullptr
       stream_->Stop();
@@ -1457,7 +1460,7 @@ bool WebRtcVoiceMediaChannel::SetRtpParameters(
   // Codecs are handled at the WebRtcVoiceMediaChannel level.
   webrtc::RtpParameters reduced_params = parameters;
   reduced_params.codecs.clear();
-  it->second->set_rtp_parameters(reduced_params);
+  it->second->SetRtpParameters(reduced_params);
   return true;
 }
 
diff --git a/webrtc/media/engine/webrtcvoiceengine_unittest.cc b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
index c72775a985..e5af99fca7 100644
--- a/webrtc/media/engine/webrtcvoiceengine_unittest.cc
+++ b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
@@ -892,6 +892,26 @@ TEST_F(WebRtcVoiceEngineTestFake,
   EXPECT_FALSE(channel_->SetRtpParameters(kSsrc1, parameters));
 }
 
+// Test that a stream will not be sending if its encoding is made
+// inactive through SetRtpParameters.
+TEST_F(WebRtcVoiceEngineTestFake, SetRtpParametersEncodingsActive) {
+  EXPECT_TRUE(SetupSendStream());
+  SetSend(channel_, true);
+  EXPECT_TRUE(GetSendStream(kSsrc1).IsSending());
+  // Get current parameters and change "active" to false.
+  webrtc::RtpParameters parameters = channel_->GetRtpParameters(kSsrc1);
+  ASSERT_EQ(1u, parameters.encodings.size());
+  ASSERT_TRUE(parameters.encodings[0].active);
+  parameters.encodings[0].active = false;
+  EXPECT_TRUE(channel_->SetRtpParameters(kSsrc1, parameters));
+  EXPECT_FALSE(GetSendStream(kSsrc1).IsSending());
+
+  // Now change it back to active and verify we resume sending.
+  parameters.encodings[0].active = true;
+  EXPECT_TRUE(channel_->SetRtpParameters(kSsrc1, parameters));
+  EXPECT_TRUE(GetSendStream(kSsrc1).IsSending());
+}
+
 // Test that SetRtpParameters configures the correct encoding channel for each
 // SSRC.
 TEST_F(WebRtcVoiceEngineTestFake, RtpParametersArePerStream) {