Removing `preference` field from `cricket::Codec`.

webrtc/api/webrtcsdp_unittest.cc

 /*
  *  Copyright 2011 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 1012 matching lines @@
   // Creates an audio content description with no streams, and some default
   // configuration.
   AudioContentDescription* CreateAudioContentDescription() {
     AudioContentDescription* audio = new AudioContentDescription();
     audio->set_rtcp_mux(true);
     audio->set_rtcp_reduced_size(true);
     audio->AddCrypto(CryptoParams(1, "AES_CM_128_HMAC_SHA1_32",
         "inline:NzB4d1BINUAvLEw6UzF3WSJ+PSdFcGdUJShpX1Zj|2^20|1:32",
         "dummy_session_params"));
     audio->set_protocol(cricket::kMediaProtocolSavpf);
-    AudioCodec opus(111, "opus", 48000, 0, 2, 3);
+    AudioCodec opus(111, "opus", 48000, 0, 2);
     audio->AddCodec(opus);
-    audio->AddCodec(AudioCodec(103, "ISAC", 16000, 32000, 1, 2));
-    audio->AddCodec(AudioCodec(104, "ISAC", 32000, 56000, 1, 1));
+    audio->AddCodec(AudioCodec(103, "ISAC", 16000, 32000, 1));
+    audio->AddCodec(AudioCodec(104, "ISAC", 32000, 56000, 1));
     return audio;
   }
 
   // Creates a video content description with no streams, and some default
   // configuration.
   VideoContentDescription* CreateVideoContentDescription() {
     VideoContentDescription* video = new VideoContentDescription();
     video->AddCrypto(CryptoParams(
         1, "AES_CM_128_HMAC_SHA1_80",
         "inline:d0RmdmcmVCspeEc3QGZiNWpVLFJhQX1cfHAwJSoj|2^20|1:32", ""));
     video->set_protocol(cricket::kMediaProtocolSavpf);
     video->AddCodec(
         VideoCodec(120, JsepSessionDescription::kDefaultVideoCodecName,
                    JsepSessionDescription::kMaxVideoCodecWidth,
                    JsepSessionDescription::kMaxVideoCodecHeight,
-                   JsepSessionDescription::kDefaultVideoCodecFramerate,
-                   JsepSessionDescription::kDefaultVideoCodecPreference));
+                   JsepSessionDescription::kDefaultVideoCodecFramerate));
     return video;
   }
 
   template <class MCD>
   void CompareMediaContentDescription(const MCD* cd1,
                                       const MCD* cd2) {
     // type
     EXPECT_EQ(cd1->type(), cd1->type());
 
     // content direction
@@ skipping 329 matching lines @@
     EXPECT_EQ(new_sdp, message);
     return true;
   }
 
   void AddSctpDataChannel() {
     rtc::scoped_ptr<DataContentDescription> data(
         new DataContentDescription());
     data_desc_ = data.get();
     data_desc_->set_protocol(cricket::kMediaProtocolDtlsSctp);
     DataCodec codec(cricket::kGoogleSctpDataCodecId,
-                    cricket::kGoogleSctpDataCodecName, 0);
+                    cricket::kGoogleSctpDataCodecName);
     codec.SetParam(cricket::kCodecParamPort, kDefaultSctpPort);
     data_desc_->AddCodec(codec);
     desc_.AddContent(kDataContentName, NS_JINGLE_DRAFT_SCTP, data.release());
     EXPECT_TRUE(desc_.AddTransportInfo(TransportInfo(
         kDataContentName, TransportDescription(kUfragData, kPwdData))));
   }
 
   void AddRtpDataChannel() {
     rtc::scoped_ptr<DataContentDescription> data(
         new DataContentDescription());
     data_desc_ = data.get();
 
-    data_desc_->AddCodec(DataCodec(101, "google-data", 1));
+    data_desc_->AddCodec(DataCodec(101, "google-data"));
     StreamParams data_stream;
     data_stream.id = kDataChannelMsid;
     data_stream.cname = kDataChannelCname;
     data_stream.sync_label = kDataChannelLabel;
     data_stream.ssrcs.push_back(kDataChannelSsrc);
     data_desc_->AddStream(data_stream);
     data_desc_->AddCrypto(CryptoParams(
         1, "AES_CM_128_HMAC_SHA1_80",
         "inline:FvLcvU2P3ZWmQxgPAgcDu7Zl9vftYElFOjEzhWs5", ""));
     data_desc_->set_protocol(cricket::kMediaProtocolSavpf);
@@ skipping 561 matching lines @@
 
 TEST_F(WebRtcSdpTest, SerializeWithSctpDataChannelAndNewPort) {
   AddSctpDataChannel();
   JsepSessionDescription jsep_desc(kDummyString);
 
   ASSERT_TRUE(jsep_desc.Initialize(desc_.Copy(), kSessionId, kSessionVersion));
   DataContentDescription* dcdesc = static_cast<DataContentDescription*>(
       jsep_desc.description()->GetContentDescriptionByName(kDataContentName));
 
   const int kNewPort = 1234;
-  cricket::DataCodec codec(
-        cricket::kGoogleSctpDataCodecId, cricket::kGoogleSctpDataCodecName, 0);
+  cricket::DataCodec codec(cricket::kGoogleSctpDataCodecId,
+                           cricket::kGoogleSctpDataCodecName);
   codec.SetParam(cricket::kCodecParamPort, kNewPort);
   dcdesc->AddOrReplaceCodec(codec);
 
   std::string message = webrtc::SdpSerialize(jsep_desc, false);
 
   std::string expected_sdp = kSdpString;
   expected_sdp.append(kSdpSctpDataChannelString);
 
   char default_portstr[16];
   char new_portstr[16];
@@ skipping 160 matching lines @@
       // The rtpmap line for static payload codec is optional.
       "a=rtpmap:18 G729/16000\r\n"
       "a=rtpmap:103 ISAC/16000\r\n";
 
   JsepSessionDescription jdesc(kDummyString);
   EXPECT_TRUE(SdpDeserialize(kSdpNoRtpmapString, &jdesc));
   cricket::AudioContentDescription* audio =
     static_cast<AudioContentDescription*>(
         jdesc.description()->GetContentDescriptionByName(cricket::CN_AUDIO));
   AudioCodecs ref_codecs;
-  // The codecs in the AudioContentDescription will be sorted by preference.
-  ref_codecs.push_back(AudioCodec(0, "PCMU", 8000, 0, 1, 3));
-  ref_codecs.push_back(AudioCodec(18, "G729", 16000, 0, 1, 2));
-  ref_codecs.push_back(AudioCodec(103, "ISAC", 16000, 32000, 1, 1));
+  // The codecs in the AudioContentDescription should be in the same order as
+  // the payload types (<fmt>s) on the m= line.
+  ref_codecs.push_back(AudioCodec(0, "PCMU", 8000, 0, 1));
+  ref_codecs.push_back(AudioCodec(18, "G729", 16000, 0, 1));
+  ref_codecs.push_back(AudioCodec(103, "ISAC", 16000, 32000, 1));
   EXPECT_EQ(ref_codecs, audio->codecs());
 }
 
 TEST_F(WebRtcSdpTest, DeserializeSessionDescriptionWithoutRtpmapButWithFmtp) {
   static const char kSdpNoRtpmapString[] =
       "v=0\r\n"
       "o=- 11 22 IN IP4 127.0.0.1\r\n"
       "s=-\r\n"
       "t=0 0\r\n"
       "m=audio 49232 RTP/AVP 18 103\r\n"
@@ skipping 947 matching lines @@
   EXPECT_TRUE(
       SdpDeserialize(kUnifiedPlanSdpFullString, &deserialized_description));
 
   EXPECT_TRUE(CompareSessionDescription(jdesc_, deserialized_description));
 }
 
 TEST_F(WebRtcSdpTest, SerializeUnifiedPlanSessionDescription) {
   MakeUnifiedPlanDescription();
   TestSerialize(jdesc_, true);
 }