Implement RTC[In/Out]boundRtpStreamStats.contentType.

Spec: https://henbos.github.io/webrtc-provisional-stats/#dom-rtcinboundrtpstreamstats-contenttype

This already exists as a goog-stat. This CL only plumbs the value to the
new stats collector.

Note: There is currently no distinction between the extension being
missing and it being present but the value being "unspecified". Until
https://crbug.com/webrtc/10529 is fixed, this metric is only exposed if
SCREENSHARE was present.

Bug: webrtc:10452
Change-Id: Ic8723f4d0efb43ab72a560e954676facd3b90659
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131946
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27520}

api/stats/rtcstats_objects.h

@@ -74,6 +74,12 @@ struct RTCNetworkType {
   static const char* const kUnknown;
 };
 
+// https://webrtc.org/experiments/rtp-hdrext/video-content-type/
+struct RTCContentType {
+  static const char* const kUnspecified;
+  static const char* const kScreenshare;
+};
+
 // https://w3c.github.io/webrtc-stats/#certificatestats-dict*
 class RTC_EXPORT RTCCertificateStats final : public RTCStats {
  public:
@@ -418,6 +424,8 @@ class RTC_EXPORT RTCInboundRTPStreamStats final : public RTCRTPStreamStats {
   // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
   RTCStatsMember<double> gap_discard_rate;
   RTCStatsMember<uint32_t> frames_decoded;
+  // https://henbos.github.io/webrtc-provisional-stats/#dom-rtcinboundrtpstreamstats-contenttype
+  RTCStatsMember<std::string> content_type;
 };
 
 // https://w3c.github.io/webrtc-stats/#outboundrtpstats-dict*
@@ -438,6 +446,8 @@ class RTC_EXPORT RTCOutboundRTPStreamStats final : public RTCRTPStreamStats {
   RTCStatsMember<double> target_bitrate;
   RTCStatsMember<uint32_t> frames_encoded;
   RTCStatsMember<double> total_encode_time;
+  // https://henbos.github.io/webrtc-provisional-stats/#dom-rtcoutboundrtpstreamstats-contenttype
+  RTCStatsMember<std::string> content_type;
 };
 
 // https://w3c.github.io/webrtc-stats/#transportstats-dict*

pc/rtc_stats_collector.cc

@@ -19,6 +19,7 @@
 #include "api/candidate.h"
 #include "api/media_stream_interface.h"
 #include "api/peer_connection_interface.h"
+#include "api/video/video_content_type.h"
 #include "media/base/media_channel.h"
 #include "p2p/base/p2p_constants.h"
 #include "p2p/base/port.h"
@@ -266,6 +267,10 @@ void SetInboundRTPStreamStatsFromVideoReceiverInfo(
   inbound_video->frames_decoded = video_receiver_info.frames_decoded;
   if (video_receiver_info.qp_sum)
     inbound_video->qp_sum = *video_receiver_info.qp_sum;
+  // TODO(https://crbug.com/webrtc/10529): When info's |content_info| is
+  // optional, support the "unspecified" value.
+  if (video_receiver_info.content_type == VideoContentType::SCREENSHARE)
+    inbound_video->content_type = RTCContentType::kScreenshare;
 }
 
 // Provides the media independent counters (both audio and video).
@@ -322,6 +327,10 @@ void SetOutboundRTPStreamStatsFromVideoSenderInfo(
   outbound_video->total_encode_time =
       static_cast<double>(video_sender_info.total_encode_time_ms) /
       rtc::kNumMillisecsPerSec;
+  // TODO(https://crbug.com/webrtc/10529): When info's |content_info| is
+  // optional, support the "unspecified" value.
+  if (video_sender_info.content_type == VideoContentType::SCREENSHARE)
+    outbound_video->content_type = RTCContentType::kScreenshare;
 }
 
 void ProduceCertificateStatsFromSSLCertificateStats(

pc/rtc_stats_collector_unittest.cc

@@ -1684,6 +1684,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Video) {
   video_media_info.receivers[0].nacks_sent = 7;
   video_media_info.receivers[0].frames_decoded = 8;
   video_media_info.receivers[0].qp_sum = absl::nullopt;
+  video_media_info.receivers[0].content_type = VideoContentType::UNSPECIFIED;
 
   RtpCodecParameters codec_parameters;
   codec_parameters.payload_type = 42;
@@ -1718,6 +1719,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Video) {
   expected_video.fraction_lost = 4.5;
   expected_video.frames_decoded = 8;
   // |expected_video.qp_sum| should be undefined.
+  // |expected_video.content_type| should be undefined.
 
   ASSERT_TRUE(report->Get(expected_video.id()));
   EXPECT_EQ(
@@ -1727,6 +1729,8 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Video) {
   // Set previously undefined values and "GetStats" again.
   video_media_info.receivers[0].qp_sum = 9;
   expected_video.qp_sum = 9;
+  video_media_info.receivers[0].content_type = VideoContentType::SCREENSHARE;
+  expected_video.content_type = "screenshare";
   video_media_channel->SetStats(video_media_info);
 
   report = stats_->GetFreshStatsReport();
@@ -1806,6 +1810,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Video) {
   video_media_info.senders[0].frames_encoded = 8;
   video_media_info.senders[0].total_encode_time_ms = 9000;
   video_media_info.senders[0].qp_sum = absl::nullopt;
+  video_media_info.senders[0].content_type = VideoContentType::UNSPECIFIED;
 
   RtpCodecParameters codec_parameters;
   codec_parameters.payload_type = 42;
@@ -1843,6 +1848,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Video) {
   expected_video.bytes_sent = 6;
   expected_video.frames_encoded = 8;
   expected_video.total_encode_time = 9.0;
+  // |expected_video.content_type| should be undefined.
   // |expected_video.qp_sum| should be undefined.
   ASSERT_TRUE(report->Get(expected_video.id()));
 
@@ -1853,6 +1859,8 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Video) {
   // Set previously undefined values and "GetStats" again.
   video_media_info.senders[0].qp_sum = 9;
   expected_video.qp_sum = 9;
+  video_media_info.senders[0].content_type = VideoContentType::SCREENSHARE;
+  expected_video.content_type = "screenshare";
   video_media_channel->SetStats(video_media_info);
 
   report = stats_->GetFreshStatsReport();

pc/rtc_stats_integrationtest.cc

@@ -740,8 +740,12 @@ class RTCStatsReportVerifier {
     if (inbound_stream.media_type.is_defined() &&
         *inbound_stream.media_type == "video") {
       verifier.TestMemberIsDefined(inbound_stream.frames_decoded);
+      // The integration test is not set up to test screen share; don't require
+      // this to be present.
+      verifier.MarkMemberTested(inbound_stream.content_type, true);
     } else {
       verifier.TestMemberIsUndefined(inbound_stream.frames_decoded);
+      verifier.TestMemberIsUndefined(inbound_stream.content_type);
     }
     return verifier.ExpectAllMembersSuccessfullyTested();
   }
@@ -764,9 +768,13 @@ class RTCStatsReportVerifier {
       verifier.TestMemberIsDefined(outbound_stream.frames_encoded);
       verifier.TestMemberIsNonNegative<double>(
           outbound_stream.total_encode_time);
+      // The integration test is not set up to test screen share; don't require
+      // this to be present.
+      verifier.MarkMemberTested(outbound_stream.content_type, true);
     } else {
       verifier.TestMemberIsUndefined(outbound_stream.frames_encoded);
       verifier.TestMemberIsUndefined(outbound_stream.total_encode_time);
+      verifier.TestMemberIsUndefined(outbound_stream.content_type);
     }
     return verifier.ExpectAllMembersSuccessfullyTested();
   }

stats/rtcstats_objects.cc

@@ -53,6 +53,10 @@ const char* const RTCNetworkType::kWimax = "wimax";
 const char* const RTCNetworkType::kVpn = "vpn";
 const char* const RTCNetworkType::kUnknown = "unknown";
 
+// https://webrtc.org/experiments/rtp-hdrext/video-content-type/
+const char* const RTCContentType::kUnspecified = "unspecified";
+const char* const RTCContentType::kScreenshare = "screenshare";
+
 // clang-format off
 WEBRTC_RTCSTATS_IMPL(RTCCertificateStats, RTCStats, "certificate",
     &fingerprint,
@@ -587,7 +591,8 @@ WEBRTC_RTCSTATS_IMPL(
     &burst_discard_rate,
     &gap_loss_rate,
     &gap_discard_rate,
-    &frames_decoded)
+    &frames_decoded,
+    &content_type)
 // clang-format on
 
 RTCInboundRTPStreamStats::RTCInboundRTPStreamStats(const std::string& id,
@@ -613,7 +618,8 @@ RTCInboundRTPStreamStats::RTCInboundRTPStreamStats(std::string&& id,
       burst_discard_rate("burstDiscardRate"),
       gap_loss_rate("gapLossRate"),
       gap_discard_rate("gapDiscardRate"),
-      frames_decoded("framesDecoded") {}
+      frames_decoded("framesDecoded"),
+      content_type("contentType") {}
 
 RTCInboundRTPStreamStats::RTCInboundRTPStreamStats(
     const RTCInboundRTPStreamStats& other)
@@ -634,7 +640,8 @@ RTCInboundRTPStreamStats::RTCInboundRTPStreamStats(
       burst_discard_rate(other.burst_discard_rate),
       gap_loss_rate(other.gap_loss_rate),
       gap_discard_rate(other.gap_discard_rate),
-      frames_decoded(other.frames_decoded) {}
+      frames_decoded(other.frames_decoded),
+      content_type(other.content_type) {}
 
 RTCInboundRTPStreamStats::~RTCInboundRTPStreamStats() {}
 
@@ -645,7 +652,8 @@ WEBRTC_RTCSTATS_IMPL(
     &bytes_sent,
     &target_bitrate,
     &frames_encoded,
-    &total_encode_time)
+    &total_encode_time,
+    &content_type)
 // clang-format on
 
 RTCOutboundRTPStreamStats::RTCOutboundRTPStreamStats(const std::string& id,
@@ -659,7 +667,8 @@ RTCOutboundRTPStreamStats::RTCOutboundRTPStreamStats(std::string&& id,
       bytes_sent("bytesSent"),
       target_bitrate("targetBitrate"),
       frames_encoded("framesEncoded"),
-      total_encode_time("totalEncodeTime") {}
+      total_encode_time("totalEncodeTime"),
+      content_type("contentType") {}
 
 RTCOutboundRTPStreamStats::RTCOutboundRTPStreamStats(
     const RTCOutboundRTPStreamStats& other)
@@ -668,7 +677,8 @@ RTCOutboundRTPStreamStats::RTCOutboundRTPStreamStats(
       bytes_sent(other.bytes_sent),
       target_bitrate(other.target_bitrate),
       frames_encoded(other.frames_encoded),
-      total_encode_time(other.total_encode_time) {}
+      total_encode_time(other.total_encode_time),
+      content_type(other.content_type) {}
 
 RTCOutboundRTPStreamStats::~RTCOutboundRTPStreamStats() {}