commit 6d6e7c5e1ac4b033e919cf562cc34ccc3b37f201
author solenberg <solenberg@webrtc.org> Wed Apr 13 09:07:30 2016 -0700
committer Commit bot <commit-bot@chromium.org> Wed Apr 13 16:07:38 2016 +0000
tree 30a9dfa318880df775c2bbb58472e03c4b35d7a7
parent 21a395ddf7136a02f251af3d3198eaaa4f688638

Fix bug causing audio to stop being sent when AudioSendStreams are recreated.

BUG=webrtc:5772

Review URL: https://codereview.webrtc.org/1881793006

Cr-Commit-Position: refs/heads/master@{#12347}

webrtc/media/engine/webrtcvoiceengine.cc

diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index 21f1210..cef7cdf 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -1121,6 +1121,7 @@
     RTC_DCHECK(!stream_);
     stream_ = call_->CreateAudioSendStream(config_);
     RTC_CHECK(stream_);
+    UpdateSendState();
   }
 
   bool SendTelephoneEvent(int payload_type, int event, int duration_ms) {

webrtc/media/engine/webrtcvoiceengine_unittest.cc

diff --git a/webrtc/media/engine/webrtcvoiceengine_unittest.cc b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
index f88b99b..1ed493b 100644
--- a/webrtc/media/engine/webrtcvoiceengine_unittest.cc
+++ b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
@@ -2176,6 +2176,31 @@
   EXPECT_FALSE(GetSendStream(kSsrc1).IsSending());
 }
 
+// Test that SetSendParameters() does not alter a stream's send state.
+TEST_F(WebRtcVoiceEngineTestFake, SendStateWhenStreamsAreRecreated) {
+  EXPECT_TRUE(SetupSendStream());
+  EXPECT_FALSE(GetSendStream(kSsrc1).IsSending());
+
+  // Turn on sending.
+  channel_->SetSend(true);
+  EXPECT_TRUE(GetSendStream(kSsrc1).IsSending());
+
+  // Changing RTP header extensions will recreate the AudioSendStream.
+  send_parameters_.extensions.push_back(
+      cricket::RtpHeaderExtension(kRtpAudioLevelHeaderExtension, 12));
+  EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
+  EXPECT_TRUE(GetSendStream(kSsrc1).IsSending());
+
+  // Turn off sending.
+  channel_->SetSend(false);
+  EXPECT_FALSE(GetSendStream(kSsrc1).IsSending());
+
+  // Changing RTP header extensions will recreate the AudioSendStream.
+  send_parameters_.extensions.clear();
+  EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
+  EXPECT_FALSE(GetSendStream(kSsrc1).IsSending());
+}
+
 // Test that we can create a channel and start playing out on it.
 TEST_F(WebRtcVoiceEngineTestFake, Playout) {
   EXPECT_TRUE(SetupRecvStream());