Allow a received audio codec's payload type to change.

webrtc/media/engine/webrtcvoiceengine_unittest.cc

 /*
  *  Copyright (c) 2008 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 898 matching lines @@
 // Test that we can apply the same set of codecs again while playing.
 TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsWhilePlaying) {
   EXPECT_TRUE(SetupRecvStream());
   cricket::AudioRecvParameters parameters;
   parameters.codecs.push_back(kIsacCodec);
   parameters.codecs.push_back(kCn16000Codec);
   EXPECT_TRUE(channel_->SetRecvParameters(parameters));
   channel_->SetPlayout(true);
   EXPECT_TRUE(channel_->SetRecvParameters(parameters));
 
-  // Changing the payload type of a codec should fail.
-  parameters.codecs[0].id = 127;
+  // Remapping a payload type to a different codec should fail.
+  parameters.codecs[0] = kOpusCodec;
+  parameters.codecs[0].id = kIsacCodec.id;
   EXPECT_FALSE(channel_->SetRecvParameters(parameters));
   EXPECT_TRUE(GetRecvStream(kSsrcX).started());
 }
 
 // Test that we can add a codec while playing.
 TEST_F(WebRtcVoiceEngineTestFake, AddRecvCodecsWhilePlaying) {
   EXPECT_TRUE(SetupRecvStream());
   cricket::AudioRecvParameters parameters;
   parameters.codecs.push_back(kIsacCodec);
   parameters.codecs.push_back(kCn16000Codec);
   EXPECT_TRUE(channel_->SetRecvParameters(parameters));
   channel_->SetPlayout(true);
 
   parameters.codecs.push_back(kOpusCodec);
   EXPECT_TRUE(channel_->SetRecvParameters(parameters));
   EXPECT_TRUE(GetRecvStream(kSsrcX).started());
   webrtc::CodecInst gcodec;
   EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(kOpusCodec, &gcodec));
   EXPECT_EQ(kOpusCodec.id, gcodec.pltype);
 }
 
+// Test that we accept adding the same codec with a different payload type.
+// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=5847
+TEST_F(WebRtcVoiceEngineTestFake, ChangeRecvCodecPayloadType) {
+  EXPECT_TRUE(SetupRecvStream());
+  cricket::AudioRecvParameters parameters;
+  parameters.codecs.push_back(kIsacCodec);
+  EXPECT_TRUE(channel_->SetRecvParameters(parameters));
+
+  ++parameters.codecs[0].id;
+  EXPECT_TRUE(channel_->SetRecvParameters(parameters));
+}
+
 TEST_F(WebRtcVoiceEngineTestFake, SetSendBandwidthAuto) {
   EXPECT_TRUE(SetupSendStream());
 
   // Test that when autobw is enabled, bitrate is kept as the default
   // value. autobw is enabled for the following tests because the target
   // bitrate is <= 0.
 
   // ISAC, default bitrate == 32000.
   TestMaxSendBandwidth(kIsacCodec, 0, true, 32000);
 
@@ skipping 2929 matching lines @@
   // Without this cast, the comparison turned unsigned and, thus, failed for -1.
   const int num_specs = static_cast<int>(specs.size());
   EXPECT_GE(find_codec({"cn", 8000, 1}), num_specs);
   EXPECT_GE(find_codec({"cn", 16000, 1}), num_specs);
   EXPECT_EQ(find_codec({"cn", 32000, 1}), -1);
   EXPECT_GE(find_codec({"telephone-event", 8000, 1}), num_specs);
   EXPECT_GE(find_codec({"telephone-event", 16000, 1}), num_specs);
   EXPECT_GE(find_codec({"telephone-event", 32000, 1}), num_specs);
   EXPECT_GE(find_codec({"telephone-event", 48000, 1}), num_specs);
 }