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}diff --git a/api/stats/rtcstats_objects.h b/api/stats/rtcstats_objects.h
index 158ea92..8a1fe56 100644
--- a/api/stats/rtcstats_objects.h
+++ b/api/stats/rtcstats_objects.h
@@ -74,6 +74,12 @@
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 @@
// 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 @@
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*
diff --git a/pc/rtc_stats_collector.cc b/pc/rtc_stats_collector.cc
index df2f09c..7427356 100644
--- a/pc/rtc_stats_collector.cc
+++ b/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 @@
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 @@
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(
diff --git a/pc/rtc_stats_collector_unittest.cc b/pc/rtc_stats_collector_unittest.cc
index ec2a9eb..5b0d164 100644
--- a/pc/rtc_stats_collector_unittest.cc
+++ b/pc/rtc_stats_collector_unittest.cc
@@ -1684,6 +1684,7 @@
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 @@
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 @@
// 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 @@
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 @@
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 @@
// 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();
diff --git a/pc/rtc_stats_integrationtest.cc b/pc/rtc_stats_integrationtest.cc
index 13eb213..174e78e 100644
--- a/pc/rtc_stats_integrationtest.cc
+++ b/pc/rtc_stats_integrationtest.cc
@@ -740,8 +740,12 @@
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 @@
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();
}
diff --git a/stats/rtcstats_objects.cc b/stats/rtcstats_objects.cc
index de317aa..7c023d1 100644
--- a/stats/rtcstats_objects.cc
+++ b/stats/rtcstats_objects.cc
@@ -53,6 +53,10 @@
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 @@
&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 @@
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 @@
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 @@
&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 @@
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 @@
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() {}
