Add send transports to individual webrtc::Call streams.

talk/media/webrtc/webrtcvoiceengine_unittest.cc

Index: talk/media/webrtc/webrtcvoiceengine_unittest.cc
diff --git a/talk/media/webrtc/webrtcvoiceengine_unittest.cc b/talk/media/webrtc/webrtcvoiceengine_unittest.cc
index c559e39f8392af12a4b7a275ad15c011a5da53b5..3575a65abcce1359aa1ac8eed09f336cf42c23b6 100644
--- a/talk/media/webrtc/webrtcvoiceengine_unittest.cc
+++ b/talk/media/webrtc/webrtcvoiceengine_unittest.cc
@@ -3354,7 +3354,7 @@ TEST(WebRtcVoiceEngineTest, CoInitialize) {
 #endif
 
 TEST_F(WebRtcVoiceEngineTestFake, SetsSyncGroupFromSyncLabel) {
-  cricket::FakeCall call(webrtc::Call::Config(nullptr));
+  cricket::FakeCall call(webrtc::Call::Config{});

[pbos-webrtc, 2015/08/26 12:21:21]

Is {} or () preferred? Or choose whatever you want? Haven't seen this before, so
I prefer () since it's familiar?

[Andrew MacDonald, 2015/08/26 16:32:35]

goog c-style arbiters prefer () in general:
https://g3doc.corp.google.com/devtools/library_club/g3doc/totw/totw-088.md?cl...

but at the end you can see they present either brace initialization or extra
parentheses as a solution to most vexing parse.

However, uniform initialization syntax is banned in Chromium anyway:
https://chromium-cpp.appspot.com/

So I suppose extra parentheses is the way to go?

[the sun, 2015/08/27 08:42:20]

Thanks for pointing this out! Using extra parenthesis.

   const uint32 kAudioSsrc = 123;
   const std::string kSyncLabel = "AvSyncLabel";
 
@@ -3384,7 +3384,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetsSyncGroupFromSyncLabel) {
 TEST_F(WebRtcVoiceEngineTestFake, CanChangeCombinedBweOption) {
   // Test that changing the combined_audio_video_bwe option results in the
   // expected state changes on an associated Call.
-  cricket::FakeCall call(webrtc::Call::Config(nullptr));
+  cricket::FakeCall call(webrtc::Call::Config{});
   const uint32 kAudioSsrc1 = 223;
   const uint32 kAudioSsrc2 = 224;
 
@@ -3437,8 +3437,8 @@ TEST_F(WebRtcVoiceEngineTestFake, CanChangeCombinedBweOption) {
 TEST_F(WebRtcVoiceEngineTestFake, SetCallConfiguresAudioReceiveChannels) {
   // Test that calling SetCall() on the voice media channel results in the
   // expected state changes in Call.
-  cricket::FakeCall call(webrtc::Call::Config(nullptr));
-  cricket::FakeCall call2(webrtc::Call::Config(nullptr));
+  cricket::FakeCall call(webrtc::Call::Config{});
+  cricket::FakeCall call2(webrtc::Call::Config{});
   const uint32 kAudioSsrc1 = 223;
   const uint32 kAudioSsrc2 = 224;
 
@@ -3474,7 +3474,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetCallConfiguresAudioReceiveChannels) {
 TEST_F(WebRtcVoiceEngineTestFake, ConfigureCombinedBweForNewRecvStreams) {
   // Test that adding receive streams after enabling combined bandwidth
   // estimation will correctly configure each channel.
-  cricket::FakeCall call(webrtc::Call::Config(nullptr));
+  cricket::FakeCall call(webrtc::Call::Config{});
 
   EXPECT_TRUE(SetupEngine());
   cricket::WebRtcVoiceMediaChannel* media_channel =
@@ -3502,7 +3502,7 @@ TEST_F(WebRtcVoiceEngineTestFake, ConfigureCombinedBweForNewRecvStreams) {
 TEST_F(WebRtcVoiceEngineTestFake, ConfiguresAudioReceiveStreamRtpExtensions) {
   // Test that setting the header extensions results in the expected state
   // changes on an associated Call.
-  cricket::FakeCall call(webrtc::Call::Config(nullptr));
+  cricket::FakeCall call(webrtc::Call::Config{});
   std::vector<uint32> ssrcs;
   ssrcs.push_back(223);
   ssrcs.push_back(224);
@@ -3556,7 +3556,7 @@ TEST_F(WebRtcVoiceEngineTestFake, ConfiguresAudioReceiveStreamRtpExtensions) {
 
 TEST_F(WebRtcVoiceEngineTestFake, DeliverAudioPacket_Call) {
   // Test that packets are forwarded to the Call when configured accordingly.
-  cricket::FakeCall call(webrtc::Call::Config(nullptr));
+  cricket::FakeCall call(webrtc::Call::Config{});
   const uint32 kAudioSsrc = 1;
   rtc::Buffer kPcmuPacket(kPcmuFrame, sizeof(kPcmuFrame));
   static const unsigned char kRtcp[] = {