stats: remove media_type which was an alias for kind

The web compat requirement that was the reason for keeping
is now solved in Chromium and its stats bindings.

BUG=webrtc:9674

Change-Id: Ifb722769414b2bcc5f4d36d7dff87a875336e039
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303860
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40024}

api/stats/rtcstats_objects.h

@@ -368,9 +368,6 @@ class RTC_EXPORT RTCRtpStreamStats : public RTCStats {
   RTCStatsMember<std::string> transport_id;
   RTCStatsMember<std::string> codec_id;
 
-  // Obsolete
-  RTCStatsMember<std::string> media_type;  // renamed to kind.
-
  protected:
   RTCRtpStreamStats(std::string id, Timestamp timestamp);
 };

pc/rtc_stats_collector.cc

@@ -463,7 +463,6 @@ std::unique_ptr<RTCInboundRtpStreamStats> CreateInboundAudioStreamStats(
                                                 inbound_audio.get());
   inbound_audio->transport_id = transport_id;
   inbound_audio->mid = mid;
-  inbound_audio->media_type = "audio";
   inbound_audio->kind = "audio";
   if (voice_receiver_info.codec_payload_type.has_value()) {
     auto codec_param_it = voice_media_info.receive_codecs.find(
@@ -609,7 +608,6 @@ CreateInboundRTPStreamStatsFromVideoReceiverInfo(
                                                 inbound_video.get());
   inbound_video->transport_id = transport_id;
   inbound_video->mid = mid;
-  inbound_video->media_type = "video";
   inbound_video->kind = "video";
   if (video_receiver_info.codec_payload_type.has_value()) {
     auto codec_param_it = video_media_info.receive_codecs.find(
@@ -737,7 +735,6 @@ CreateOutboundRTPStreamStatsFromVoiceSenderInfo(
                                                outbound_audio.get());
   outbound_audio->transport_id = transport_id;
   outbound_audio->mid = mid;
-  outbound_audio->media_type = "audio";
   outbound_audio->kind = "audio";
   if (voice_sender_info.target_bitrate.has_value() &&
       *voice_sender_info.target_bitrate > 0) {
@@ -774,7 +771,6 @@ CreateOutboundRTPStreamStatsFromVideoSenderInfo(
                                                outbound_video.get());
   outbound_video->transport_id = transport_id;
   outbound_video->mid = mid;
-  outbound_video->media_type = "video";
   outbound_video->kind = "video";
   if (video_sender_info.codec_payload_type.has_value()) {
     auto codec_param_it = video_media_info.send_codecs.find(

pc/rtc_stats_collector_unittest.cc

@@ -2530,7 +2530,6 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRtpStreamStats_Audio) {
   RTCInboundRtpStreamStats expected_audio("ITTransportName1A1",
                                           report->timestamp());
   expected_audio.ssrc = 1;
-  expected_audio.media_type = "audio";
   expected_audio.kind = "audio";
   expected_audio.track_identifier = "RemoteAudioTrackID";
   expected_audio.mid = "AudioMid";
@@ -2701,7 +2700,6 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRtpStreamStats_Video) {
   RTCInboundRtpStreamStats expected_video("ITTransportName1V1",
                                           report->timestamp());
   expected_video.ssrc = 1;
-  expected_video.media_type = "video";
   expected_video.kind = "video";
   expected_video.track_identifier = "RemoteVideoTrackID";
   expected_video.mid = "VideoMid";
@@ -2877,7 +2875,6 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRtpStreamStats_Audio) {
   // `expected_audio.remote_id` should be undefined.
   expected_audio.mid = "AudioMid";
   expected_audio.ssrc = 1;
-  expected_audio.media_type = "audio";
   expected_audio.kind = "audio";
   expected_audio.track_id =
       IdForType<DEPRECATED_RTCMediaStreamTrackStats>(report.get());
@@ -2973,7 +2970,6 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRtpStreamStats_Video) {
   // `expected_video.remote_id` should be undefined.
   expected_video.mid = "VideoMid";
   expected_video.ssrc = 1;
-  expected_video.media_type = "video";
   expected_video.kind = "video";
   expected_video.track_id = stats_of_track_type[0]->id();
   expected_video.transport_id = "TTransportName1";
@@ -3319,7 +3315,6 @@ TEST_F(RTCStatsCollectorTest, CollectNoStreamRTCOutboundRtpStreamStats_Audio) {
   expected_audio.media_source_id = "SA50";
   expected_audio.mid = "AudioMid";
   expected_audio.ssrc = 1;
-  expected_audio.media_type = "audio";
   expected_audio.kind = "audio";
   expected_audio.track_id =
       IdForType<DEPRECATED_RTCMediaStreamTrackStats>(report.get());

pc/rtc_stats_integrationtest.cc

@@ -723,11 +723,9 @@ class RTCStatsReportVerifier {
     // hierarcy.
     if (stream.type() == RTCInboundRtpStreamStats::kType ||
         stream.type() == RTCOutboundRtpStreamStats::kType) {
-      verifier.TestMemberIsDefined(stream.media_type);
       verifier.TestMemberIsIDReference(
           stream.track_id, DEPRECATED_RTCMediaStreamTrackStats::kType);
     } else {
-      verifier.TestMemberIsUndefined(stream.media_type);
       verifier.TestMemberIsUndefined(stream.track_id);
     }
     verifier.TestMemberIsIDReference(stream.transport_id,

stats/rtcstats_objects.cc

@@ -401,8 +401,7 @@ WEBRTC_RTCSTATS_IMPL(RTCRtpStreamStats, RTCStats, "rtp",
     &kind,
     &track_id,
     &transport_id,
-    &codec_id,
+    &codec_id)
-    &media_type)
 // clang-format on
 
 RTCRtpStreamStats::RTCRtpStreamStats(std::string id, Timestamp timestamp)
@@ -411,8 +410,7 @@ RTCRtpStreamStats::RTCRtpStreamStats(std::string id, Timestamp timestamp)
       kind("kind"),
       track_id("trackId"),
       transport_id("transportId"),
-      codec_id("codecId"),
+      codec_id("codecId") {}
-      media_type("mediaType") {}
 
 RTCRtpStreamStats::RTCRtpStreamStats(const RTCRtpStreamStats& other) = default;