Move RTP/RTCP demuxing logic from BaseChannel to RtpTransport.

webrtc/pc/channel_unittest.cc

 /*
  *  Copyright 2009 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 1542 matching lines @@
     if (rtcp_mux) {
       flags |= RTCP_MUX;
     }
     CreateChannels(flags, flags);
     EXPECT_TRUE(SendInitiate());
     EXPECT_TRUE(channel1_->NeedsRtcpTransport());
     EXPECT_EQ(rtcp_mux, !channel2_->NeedsRtcpTransport());
     EXPECT_TRUE(SendAccept());
     EXPECT_EQ(rtcp_mux, !channel1_->NeedsRtcpTransport());
     EXPECT_EQ(rtcp_mux, !channel2_->NeedsRtcpTransport());
-    EXPECT_TRUE(channel1_->bundle_filter()->FindPayloadType(pl_type1));
-    EXPECT_TRUE(channel2_->bundle_filter()->FindPayloadType(pl_type1));
-    EXPECT_FALSE(channel1_->bundle_filter()->FindPayloadType(pl_type2));
-    EXPECT_FALSE(channel2_->bundle_filter()->FindPayloadType(pl_type2));
+    EXPECT_TRUE(channel1_->HandlesPayloadType(pl_type1));
+    EXPECT_TRUE(channel2_->HandlesPayloadType(pl_type1));
+    EXPECT_FALSE(channel1_->HandlesPayloadType(pl_type2));
+    EXPECT_FALSE(channel2_->HandlesPayloadType(pl_type2));
 
     // Both channels can receive pl_type1 only.
     SendCustomRtp1(kSsrc1, ++sequence_number1_1, pl_type1);
     SendCustomRtp2(kSsrc2, ++sequence_number2_2, pl_type1);
     WaitForThreads();
     EXPECT_TRUE(CheckCustomRtp2(kSsrc1, sequence_number1_1, pl_type1));
     EXPECT_TRUE(CheckCustomRtp1(kSsrc2, sequence_number2_2, pl_type1));
     EXPECT_TRUE(CheckNoRtp1());
     EXPECT_TRUE(CheckNoRtp2());
 
@@ skipping 2113 matching lines @@
   ASSERT_TRUE(voice_channel_.Init_w(nullptr, nullptr, &fake_rtp_dtls_transport_,
                                     &fake_rtcp_dtls_transport_));
   EXPECT_DEATH(voice_channel_.SetTransports(&fake_rtp_dtls_transport_,
                                             &fake_rtp_dtls_transport_),
                "");
 }
 
 #endif  // RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
 
 // TODO(pthatcher): TestSetReceiver?