Implement Outbound/InboundRtpStreamStats.mid.
This is what allowed us to remove "transceiver" stats from the spec.
Bug: webrtc:14191
Change-Id: I687a2dd97de016832005cb4271f6e1a0e0560cd3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266022
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#37247}
diff --git a/api/stats/rtcstats_objects.h b/api/stats/rtcstats_objects.h
index 5465d15..b2ebdd6 100644
--- a/api/stats/rtcstats_objects.h
+++ b/api/stats/rtcstats_objects.h
@@ -452,6 +452,7 @@
// TODO(https://crbug.com/webrtc/14174): Implement trackIdentifier and kind.
RTCStatsMember<std::string> track_identifier;
+ RTCStatsMember<std::string> mid;
RTCStatsMember<std::string> remote_id;
RTCStatsMember<uint32_t> packets_received;
RTCStatsMember<uint64_t> fec_packets_received;
@@ -515,6 +516,7 @@
RTCStatsMember<std::string> media_source_id;
RTCStatsMember<std::string> remote_id;
+ RTCStatsMember<std::string> mid;
RTCStatsMember<std::string> rid;
RTCStatsMember<uint32_t> packets_sent;
RTCStatsMember<uint64_t> retransmitted_packets_sent;
diff --git a/pc/rtc_stats_collector.cc b/pc/rtc_stats_collector.cc
index efc90a3..04ea8b2 100644
--- a/pc/rtc_stats_collector.cc
+++ b/pc/rtc_stats_collector.cc
@@ -422,6 +422,7 @@
timestamp_us);
SetInboundRTPStreamStatsFromMediaReceiverInfo(voice_receiver_info,
inbound_audio.get());
+ inbound_audio->mid = mid;
inbound_audio->media_type = "audio";
inbound_audio->kind = "audio";
if (voice_receiver_info.codec_payload_type) {
@@ -524,6 +525,7 @@
RTCInboundRTPStreamStats* inbound_video) {
SetInboundRTPStreamStatsFromMediaReceiverInfo(video_receiver_info,
inbound_video);
+ inbound_video->mid = mid;
inbound_video->media_type = "video";
inbound_video->kind = "video";
if (video_receiver_info.codec_payload_type) {
@@ -615,6 +617,7 @@
RTCOutboundRTPStreamStats* outbound_audio) {
SetOutboundRTPStreamStatsFromMediaSenderInfo(voice_sender_info,
outbound_audio);
+ outbound_audio->mid = mid;
outbound_audio->media_type = "audio";
outbound_audio->kind = "audio";
if (voice_sender_info.target_bitrate > 0) {
@@ -634,6 +637,7 @@
RTCOutboundRTPStreamStats* outbound_video) {
SetOutboundRTPStreamStatsFromMediaSenderInfo(video_sender_info,
outbound_video);
+ outbound_video->mid = mid;
outbound_video->media_type = "video";
outbound_video->kind = "video";
if (video_sender_info.codec_payload_type) {
diff --git a/pc/rtc_stats_collector_unittest.cc b/pc/rtc_stats_collector_unittest.cc
index 89d0daa..c008c1b 100644
--- a/pc/rtc_stats_collector_unittest.cc
+++ b/pc/rtc_stats_collector_unittest.cc
@@ -2099,6 +2099,7 @@
expected_audio.media_type = "audio";
expected_audio.kind = "audio";
expected_audio.track_identifier = "RemoteAudioTrackID";
+ expected_audio.mid = "AudioMid";
expected_audio.track_id = stats_of_track_type[0]->id();
expected_audio.transport_id = "RTCTransport_TransportName_1";
expected_audio.codec_id = "RTCCodec_AudioMid_Inbound_42";
@@ -2208,6 +2209,7 @@
expected_video.media_type = "video";
expected_video.kind = "video";
expected_video.track_identifier = "RemoteVideoTrackID";
+ expected_video.mid = "VideoMid";
expected_video.track_id = IdForType<RTCMediaStreamTrackStats>(report.get());
expected_video.transport_id = "RTCTransport_TransportName_1";
expected_video.codec_id = "RTCCodec_VideoMid_Inbound_42";
@@ -2300,6 +2302,7 @@
report->timestamp_us());
expected_audio.media_source_id = "RTCAudioSource_50";
// `expected_audio.remote_id` should be undefined.
+ expected_audio.mid = "AudioMid";
expected_audio.ssrc = 1;
expected_audio.media_type = "audio";
expected_audio.kind = "audio";
@@ -2387,6 +2390,7 @@
report->timestamp_us());
expected_video.media_source_id = "RTCVideoSource_50";
// `expected_video.remote_id` should be undefined.
+ expected_video.mid = "VideoMid";
expected_video.ssrc = 1;
expected_video.media_type = "video";
expected_video.kind = "video";
@@ -2728,6 +2732,7 @@
RTCOutboundRTPStreamStats expected_audio("RTCOutboundRTPAudioStream_1",
report->timestamp_us());
expected_audio.media_source_id = "RTCAudioSource_50";
+ expected_audio.mid = "AudioMid";
expected_audio.ssrc = 1;
expected_audio.media_type = "audio";
expected_audio.kind = "audio";
diff --git a/pc/rtc_stats_integrationtest.cc b/pc/rtc_stats_integrationtest.cc
index 6125137..a071786 100644
--- a/pc/rtc_stats_integrationtest.cc
+++ b/pc/rtc_stats_integrationtest.cc
@@ -799,6 +799,7 @@
*inbound_stream.media_type == "audio");
verifier.TestMemberIsOptionalIDReference(
inbound_stream.remote_id, RTCRemoteOutboundRtpStreamStats::kType);
+ verifier.TestMemberIsDefined(inbound_stream.mid);
verifier.TestMemberIsDefined(inbound_stream.track_identifier);
if (inbound_stream.kind.is_defined() &&
*inbound_stream.media_type == "video") {
@@ -926,6 +927,7 @@
const RTCOutboundRTPStreamStats& outbound_stream) {
RTCStatsVerifier verifier(report_.get(), &outbound_stream);
VerifyRTCRTPStreamStats(outbound_stream, verifier);
+ verifier.TestMemberIsDefined(outbound_stream.mid);
if (outbound_stream.kind.is_defined() &&
*outbound_stream.media_type == "video") {
verifier.TestMemberIsIDReference(outbound_stream.media_source_id,
diff --git a/stats/rtcstats_objects.cc b/stats/rtcstats_objects.cc
index 0605f93..77a2414 100644
--- a/stats/rtcstats_objects.cc
+++ b/stats/rtcstats_objects.cc
@@ -641,6 +641,7 @@
WEBRTC_RTCSTATS_IMPL(
RTCInboundRTPStreamStats, RTCReceivedRtpStreamStats, "inbound-rtp",
&track_identifier,
+ &mid,
&remote_id,
&packets_received,
&fec_packets_received,
@@ -690,6 +691,7 @@
int64_t timestamp_us)
: RTCReceivedRtpStreamStats(std::move(id), timestamp_us),
track_identifier("trackIdentifier"),
+ mid("mid"),
remote_id("remoteId"),
packets_received("packetsReceived"),
fec_packets_received("fecPacketsReceived"),
@@ -735,6 +737,7 @@
const RTCInboundRTPStreamStats& other)
: RTCReceivedRtpStreamStats(other),
track_identifier(other.track_identifier),
+ mid(other.mid),
remote_id(other.remote_id),
packets_received(other.packets_received),
fec_packets_received(other.fec_packets_received),
@@ -784,6 +787,7 @@
RTCOutboundRTPStreamStats, RTCRTPStreamStats, "outbound-rtp",
&media_source_id,
&remote_id,
+ &mid,
&rid,
&packets_sent,
&retransmitted_packets_sent,
@@ -821,6 +825,7 @@
: RTCRTPStreamStats(std::move(id), timestamp_us),
media_source_id("mediaSourceId"),
remote_id("remoteId"),
+ mid("mid"),
rid("rid"),
packets_sent("packetsSent"),
retransmitted_packets_sent("retransmittedPacketsSent"),
@@ -854,6 +859,7 @@
: RTCRTPStreamStats(other),
media_source_id(other.media_source_id),
remote_id(other.remote_id),
+ mid(other.mid),
rid(other.rid),
packets_sent(other.packets_sent),
retransmitted_packets_sent(other.retransmitted_packets_sent),
