Rename RTCCodecStats.codec -> mimeType, parameters -> sdpFmtpLine.

As per https://github.com/w3c/webrtc-stats/pull/168.

NOTRY due to broken linux_ubsan_vptr, all other tests passed.

BUG=webrtc:7061
NOTRY=True

Review-Url: https://codereview.webrtc.org/2718383002
Cr-Commit-Position: refs/heads/master@{#16907}
diff --git a/webrtc/api/stats/rtcstats_objects.h b/webrtc/api/stats/rtcstats_objects.h
index baf0d28..cc01b5e 100644
--- a/webrtc/api/stats/rtcstats_objects.h
+++ b/webrtc/api/stats/rtcstats_objects.h
@@ -87,12 +87,12 @@
   ~RTCCodecStats() override;
 
   RTCStatsMember<uint32_t> payload_type;
-  RTCStatsMember<std::string> codec;
+  RTCStatsMember<std::string> mime_type;
   RTCStatsMember<uint32_t> clock_rate;
   // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7061
   RTCStatsMember<uint32_t> channels;
   // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7061
-  RTCStatsMember<std::string> parameters;
+  RTCStatsMember<std::string> sdp_fmtp_line;
   // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7061
   RTCStatsMember<std::string> implementation;
 };
diff --git a/webrtc/pc/rtcstats_integrationtest.cc b/webrtc/pc/rtcstats_integrationtest.cc
index 34adb4c..1176b54 100644
--- a/webrtc/pc/rtcstats_integrationtest.cc
+++ b/webrtc/pc/rtcstats_integrationtest.cc
@@ -334,10 +334,10 @@
       const RTCCodecStats& codec) {
     RTCStatsVerifier verifier(report_, &codec);
     verifier.TestMemberIsDefined(codec.payload_type);
-    verifier.TestMemberIsDefined(codec.codec);
+    verifier.TestMemberIsDefined(codec.mime_type);
     verifier.TestMemberIsPositive<uint32_t>(codec.clock_rate);
     verifier.TestMemberIsUndefined(codec.channels);
-    verifier.TestMemberIsUndefined(codec.parameters);
+    verifier.TestMemberIsUndefined(codec.sdp_fmtp_line);
     verifier.TestMemberIsUndefined(codec.implementation);
     return verifier.ExpectAllMembersSuccessfullyTested();
   }
diff --git a/webrtc/pc/rtcstatscollector.cc b/webrtc/pc/rtcstatscollector.cc
index abb4ab9..36a4bde 100644
--- a/webrtc/pc/rtcstatscollector.cc
+++ b/webrtc/pc/rtcstatscollector.cc
@@ -177,7 +177,7 @@
       RTCCodecStatsIDFromDirectionMediaAndPayload(inbound, audio, payload_type),
       timestamp_us));
   codec_stats->payload_type = payload_type;
-  codec_stats->codec = codec_params.mime_type();
+  codec_stats->mime_type = codec_params.mime_type();
   if (codec_params.clock_rate) {
     codec_stats->clock_rate = static_cast<uint32_t>(*codec_params.clock_rate);
   }
diff --git a/webrtc/pc/rtcstatscollector_unittest.cc b/webrtc/pc/rtcstatscollector_unittest.cc
index eea33e0..db26096 100644
--- a/webrtc/pc/rtcstatscollector_unittest.cc
+++ b/webrtc/pc/rtcstatscollector_unittest.cc
@@ -826,25 +826,25 @@
   RTCCodecStats expected_inbound_audio_codec(
       "RTCCodec_InboundAudio_1", report->timestamp_us());
   expected_inbound_audio_codec.payload_type = 1;
-  expected_inbound_audio_codec.codec = "audio/opus";
+  expected_inbound_audio_codec.mime_type = "audio/opus";
   expected_inbound_audio_codec.clock_rate = 1337;
 
   RTCCodecStats expected_outbound_audio_codec(
       "RTCCodec_OutboundAudio_2", report->timestamp_us());
   expected_outbound_audio_codec.payload_type = 2;
-  expected_outbound_audio_codec.codec = "audio/isac";
+  expected_outbound_audio_codec.mime_type = "audio/isac";
   expected_outbound_audio_codec.clock_rate = 1338;
 
   RTCCodecStats expected_inbound_video_codec(
       "RTCCodec_InboundVideo_3", report->timestamp_us());
   expected_inbound_video_codec.payload_type = 3;
-  expected_inbound_video_codec.codec = "video/H264";
+  expected_inbound_video_codec.mime_type = "video/H264";
   expected_inbound_video_codec.clock_rate = 1339;
 
   RTCCodecStats expected_outbound_video_codec(
       "RTCCodec_OutboundVideo_4", report->timestamp_us());
   expected_outbound_video_codec.payload_type = 4;
-  expected_outbound_video_codec.codec = "video/VP8";
+  expected_outbound_video_codec.mime_type = "video/VP8";
   expected_outbound_video_codec.clock_rate = 1340;
 
   ASSERT_TRUE(report->Get(expected_inbound_audio_codec.id()));
diff --git a/webrtc/stats/rtcstats_objects.cc b/webrtc/stats/rtcstats_objects.cc
index 68b33f5..9a7ebed 100644
--- a/webrtc/stats/rtcstats_objects.cc
+++ b/webrtc/stats/rtcstats_objects.cc
@@ -72,10 +72,10 @@
 
 WEBRTC_RTCSTATS_IMPL(RTCCodecStats, RTCStats, "codec",
     &payload_type,
-    &codec,
+    &mime_type,
     &clock_rate,
     &channels,
-    &parameters,
+    &sdp_fmtp_line,
     &implementation);
 
 RTCCodecStats::RTCCodecStats(
@@ -87,10 +87,10 @@
     std::string&& id, int64_t timestamp_us)
     : RTCStats(std::move(id), timestamp_us),
       payload_type("payloadType"),
-      codec("codec"),
+      mime_type("mimeType"),
       clock_rate("clockRate"),
       channels("channels"),
-      parameters("parameters"),
+      sdp_fmtp_line("sdpFmtpLine"),
       implementation("implementation") {
 }
 
@@ -98,10 +98,10 @@
     const RTCCodecStats& other)
     : RTCStats(other.id(), other.timestamp_us()),
       payload_type(other.payload_type),
-      codec(other.codec),
+      mime_type(other.mime_type),
       clock_rate(other.clock_rate),
       channels(other.channels),
-      parameters(other.parameters),
+      sdp_fmtp_line(other.sdp_fmtp_line),
       implementation(other.implementation) {
 }