Reland of Don't hardcode MediaType::ANY in FakeNetworkPipe.

webrtc/test/call_test.cc

Index: webrtc/test/call_test.cc
diff --git a/webrtc/test/call_test.cc b/webrtc/test/call_test.cc
index ad55421f667bc1f73fb195ecf2cb5dd8f3ecd49b..9ffe47497118d9fcdccb8c930e7be8dccc6e0891 100644
--- a/webrtc/test/call_test.cc
+++ b/webrtc/test/call_test.cc
@@ -26,6 +26,28 @@ namespace {
 const int kVideoRotationRtpExtensionId = 4;
 }
 
+void CallTest::PayloadDemuxer::SetReceiver(PacketReceiver* receiver) {
+  receiver_ = receiver;
+}
+
+PacketReceiver::DeliveryStatus CallTest::PayloadDemuxer::DeliverPacket(
+    MediaType media_type,
+    const uint8_t* packet,
+    size_t length,
+    const PacketTime& packet_time) {
+  if (media_type == MediaType::ANY) {
+    // This simplistic demux logic will not make much sense for RTCP
+    // packets, but it seems that doesn't matter.
+    RTC_CHECK_GE(length, 2);
+    uint8_t pt = packet[1] & 0x7f;
+    if (pt == kFakeVideoSendPayloadType || pt == kFlexfecPayloadType)
+      media_type = MediaType::VIDEO;
+    else
+      media_type = MediaType::AUDIO;

[sprang_webrtc, 2017/03/29 12:24:30]

nit: {} for if/else

[nisse-webrtc, 2017/03/29 12:37:55]

Done.

+  }
+  return receiver_->DeliverPacket(media_type, packet, length, packet_time);
+}
+
 CallTest::CallTest()
     : clock_(Clock::GetRealTimeClock()),
       video_send_config_(nullptr),
@@ -75,8 +97,15 @@ void CallTest::RunBaseTest(BaseTest* test) {
   send_transport_.reset(test->CreateSendTransport(sender_call_.get()));
 
   if (test->ShouldCreateReceivers()) {
-    send_transport_->SetReceiver(receiver_call_->Receiver());
-    receive_transport_->SetReceiver(sender_call_->Receiver());
+    if (num_video_streams_ > 0 && num_audio_streams_ > 0) {
+      receive_demuxer_.SetReceiver(receiver_call_->Receiver());
+      send_transport_->SetReceiver(&receive_demuxer_);
+      send_demuxer_.SetReceiver(sender_call_->Receiver());
+      receive_transport_->SetReceiver(&send_demuxer_);
+    } else {

[stefan-webrtc, 2017/03/29 12:27:25]

Is there a reason why we don't always use the demuxer?

[nisse-webrtc, 2017/03/29 12:37:55]

I'm afraid it is a bit brittle, so I don't want to depend on payload type it's
not really needed.

Hmm, but since the demuxer class now does the payload type demuxing only when it
gets MediaType::ANY as input, it would probably work fine to use it always.

[stefan-webrtc, 2017/03/30 06:47:11]

I'm ok with both options. Maybe good to have one less if statement. :)

[nisse-webrtc, 2017/03/30 06:56:45]

I don't have a strong opinion either. I think I'll leave this unchanged. Then
the tests which are audio-only or video-only (which are the majority) are forced
to configure the FakeNetworkPipe with the correct media type, and won't
accidentally start relying on the somewhat brittle demux logic.

+      send_transport_->SetReceiver(receiver_call_->Receiver());
+      receive_transport_->SetReceiver(sender_call_->Receiver());
+    }
     if (num_video_streams_ > 0)
       receiver_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
     if (num_audio_streams_ > 0)
@@ -458,11 +487,13 @@ void BaseTest::OnCallsCreated(Call* sender_call, Call* receiver_call) {
 
 test::PacketTransport* BaseTest::CreateSendTransport(Call* sender_call) {
   return new PacketTransport(sender_call, this, test::PacketTransport::kSender,
+                             MediaType::VIDEO,
                              FakeNetworkPipe::Config());
 }
 
 test::PacketTransport* BaseTest::CreateReceiveTransport() {
   return new PacketTransport(nullptr, this, test::PacketTransport::kReceiver,
+                             MediaType::VIDEO,
                              FakeNetworkPipe::Config());
 }