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[ModernStats] Mark obsolete stats as [[deprecated]].

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This includes the stats dictionaries that have been made obsolete in
the spec and whose IDs are prefixed "DEPRECATED_":
- RTCMediaStreamTrackStats
- RTCMediaStreamStats

There is an ongoing experiment to unship these stats dictionaries in
Chrome (https://crbug.com/1374215). Marking then as [[deprecated]] helps
alert other dependencies that these classes are deprecated.

In the meantime, the "DEPRECATED_RTCMediaStreamTrackStats" prefix makes
it possible to use the deprecated classes.

# Unrelated infra failures
NOTRY=True

Bug: webrtc:14175, webrtc:14419
Change-Id: I498d370294058a628278e1e5b027cd12e24ad31a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279700
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38439}
This commit is contained in:
Henrik Boström
2022-10-19 11:06:58 +02:00
committed by WebRTC LUCI CQ
parent 666c333625
commit 15166b2fa4
10 changed files with 199 additions and 165 deletions
Download Patch File Download Diff File Expand all files Collapse all files

30
pc/peer_connection_integrationtest.cc
View File

@ -1364,7 +1364,8 @@ TEST_P(PeerConnectionIntegrationTest, NewGetStatsManyAudioAndManyVideoStreams) {
}
ASSERT_TRUE(stat->track_id.is_defined());
const auto* track_stat =
caller_report->GetAs<webrtc::RTCMediaStreamTrackStats>(*stat->track_id);
caller_report->GetAs<webrtc::DEPRECATED_RTCMediaStreamTrackStats>(
*stat->track_id);
ASSERT_TRUE(track_stat);
outbound_track_ids.push_back(*track_stat->track_identifier);
}
@ -1388,7 +1389,8 @@ TEST_P(PeerConnectionIntegrationTest, NewGetStatsManyAudioAndManyVideoStreams) {
}
ASSERT_TRUE(stat->track_id.is_defined());
const auto* track_stat =
callee_report->GetAs<webrtc::RTCMediaStreamTrackStats>(*stat->track_id);
callee_report->GetAs<webrtc::DEPRECATED_RTCMediaStreamTrackStats>(
*stat->track_id);
ASSERT_TRUE(track_stat);
inbound_track_ids.push_back(*track_stat->track_identifier);
}
@ -1465,7 +1467,8 @@ TEST_P(PeerConnectionIntegrationTest,
callee()->NewGetStats();
ASSERT_NE(nullptr, report);
auto media_stats = report->GetStatsOfType<webrtc::RTCMediaStreamTrackStats>();
auto media_stats =
report->GetStatsOfType<webrtc::DEPRECATED_RTCMediaStreamTrackStats>();
auto audio_index = FindFirstMediaStatsIndexByKind("audio", media_stats);
ASSERT_GE(audio_index, 0);
EXPECT_TRUE(media_stats[audio_index]->audio_level.is_defined());
@ -1483,11 +1486,11 @@ void ModifySsrcs(cricket::SessionDescription* desc) {
}
}
// Test that the "RTCMediaSteamTrackStats" object is updated correctly when
// SSRCs are unsignaled, and the SSRC of the received (audio) stream changes.
// This should result in two "RTCInboundRTPStreamStats", but only one
// "RTCMediaStreamTrackStats", whose counters go up continuously rather than
// being reset to 0 once the SSRC change occurs.
// Test that the "DEPRECATED_RTCMediaStreamTrackStats" object is updated
// correctly when SSRCs are unsignaled, and the SSRC of the received (audio)
// stream changes. This should result in two "RTCInboundRTPStreamStats", but
// only one "DEPRECATED_RTCMediaStreamTrackStats", whose counters go up
// continuously rather than being reset to 0 once the SSRC change occurs.
//
// Regression test for this bug:
// https://bugs.chromium.org/p/webrtc/issues/detail?id=8158
@ -1519,7 +1522,8 @@ TEST_P(PeerConnectionIntegrationTest,
rtc::scoped_refptr<const webrtc::RTCStatsReport> report =
callee()->NewGetStats();
ASSERT_NE(nullptr, report);
auto track_stats = report->GetStatsOfType<webrtc::RTCMediaStreamTrackStats>();
auto track_stats =
report->GetStatsOfType<webrtc::DEPRECATED_RTCMediaStreamTrackStats>();
ASSERT_EQ(1U, track_stats.size());
ASSERT_TRUE(track_stats[0]->total_samples_received.is_defined());
ASSERT_GT(*track_stats[0]->total_samples_received, 0U);
@ -1539,7 +1543,8 @@ TEST_P(PeerConnectionIntegrationTest,
report = callee()->NewGetStats();
ASSERT_NE(nullptr, report);
track_stats = report->GetStatsOfType<webrtc::RTCMediaStreamTrackStats>();
track_stats =
report->GetStatsOfType<webrtc::DEPRECATED_RTCMediaStreamTrackStats>();
ASSERT_EQ(1U, track_stats.size());
ASSERT_TRUE(track_stats[0]->total_samples_received.is_defined());
// The "total samples received" stat should only be greater than it was
@ -2878,8 +2883,9 @@ TEST_P(PeerConnectionIntegrationTest, DisableAndEnableAudioPlayout) {
double GetAudioEnergyStat(PeerConnectionIntegrationWrapper* pc) {
auto report = pc->NewGetStats();
auto track_stats_list =
report->GetStatsOfType<webrtc::RTCMediaStreamTrackStats>();
const webrtc::RTCMediaStreamTrackStats* remote_track_stats = nullptr;
report->GetStatsOfType<webrtc::DEPRECATED_RTCMediaStreamTrackStats>();
const webrtc::DEPRECATED_RTCMediaStreamTrackStats* remote_track_stats =
nullptr;
for (const auto* track_stats : track_stats_list) {
if (track_stats->remote_source.is_defined() &&
*track_stats->remote_source) {