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Remove dependencies on 'track' stats from PC integration tests.

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This unblocks the deletion of this deprecated stats object.

Bug: webrtc:14175
Change-Id: I850c028fc9556a36191909afa3d635a7e6b65b69
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288582
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38983}
This commit is contained in:
Henrik Boström 2022-12-24 10:33:39 +01:00 committed by WebRTC LUCI CQ
parent c412a9c177
commit 323e9df6d1
3 changed files with 24 additions and 41 deletions
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53
pc/peer_connection_integrationtest.cc
View File

@ -1326,8 +1326,8 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
}
// Test that the new GetStats() returns stats for all outgoing/incoming streams
// with the correct track IDs if there are more than one audio and more than one
// video senders/receivers.
// with the correct track identifiers if there are more than one audio and more
// than one video senders/receivers.
TEST_P(PeerConnectionIntegrationTest, NewGetStatsManyAudioAndManyVideoStreams) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
@ -1362,12 +1362,12 @@ TEST_P(PeerConnectionIntegrationTest, NewGetStatsManyAudioAndManyVideoStreams) {
ASSERT_TRUE(stat->frames_encoded.is_defined());
EXPECT_GE(*stat->frames_encoded, *stat->key_frames_encoded);
}
ASSERT_TRUE(stat->track_id.is_defined());
const auto* track_stat =
caller_report->GetAs<webrtc::DEPRECATED_RTCMediaStreamTrackStats>(
*stat->track_id);
ASSERT_TRUE(track_stat);
outbound_track_ids.push_back(*track_stat->track_identifier);
ASSERT_TRUE(stat->media_source_id.is_defined());
const RTCMediaSourceStats* media_source =
static_cast<const RTCMediaSourceStats*>(
caller_report->Get(*stat->media_source_id));
ASSERT_TRUE(media_source);
outbound_track_ids.push_back(*media_source->track_identifier);
}
EXPECT_THAT(outbound_track_ids, UnorderedElementsAreArray(track_ids));
@ -1387,12 +1387,7 @@ TEST_P(PeerConnectionIntegrationTest, NewGetStatsManyAudioAndManyVideoStreams) {
ASSERT_TRUE(stat->frames_decoded.is_defined());
EXPECT_GE(*stat->frames_decoded, *stat->key_frames_decoded);
}
ASSERT_TRUE(stat->track_id.is_defined());
const auto* track_stat =
callee_report->GetAs<webrtc::DEPRECATED_RTCMediaStreamTrackStats>(
*stat->track_id);
ASSERT_TRUE(track_stat);
inbound_track_ids.push_back(*track_stat->track_identifier);
inbound_track_ids.push_back(*stat->track_identifier);
}
EXPECT_THAT(inbound_track_ids, UnorderedElementsAreArray(track_ids));
}
@ -1467,11 +1462,11 @@ TEST_P(PeerConnectionIntegrationTest,
callee()->NewGetStats();
ASSERT_NE(nullptr, report);
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());
auto inbound_rtps =
report->GetStatsOfType<webrtc::RTCInboundRTPStreamStats>();
auto index = FindFirstMediaStatsIndexByKind("audio", inbound_rtps);
ASSERT_GE(index, 0);
EXPECT_TRUE(inbound_rtps[index]->audio_level.is_defined());
}
// Helper for test below.
@ -2882,22 +2877,14 @@ TEST_P(PeerConnectionIntegrationTest, DisableAndEnableAudioPlayout) {
double GetAudioEnergyStat(PeerConnectionIntegrationWrapper* pc) {
auto report = pc->NewGetStats();
auto track_stats_list =
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) {
remote_track_stats = track_stats;
break;
}
}
if (!remote_track_stats->total_audio_energy.is_defined()) {
auto inbound_rtps =
report->GetStatsOfType<webrtc::RTCInboundRTPStreamStats>();
RTC_CHECK(!inbound_rtps.empty());
auto* inbound_rtp = inbound_rtps[0];
if (!inbound_rtp->total_audio_energy.is_defined()) {
return 0.0;
}
return *remote_track_stats->total_audio_energy;
return *inbound_rtp->total_audio_energy;
}
// Test that if audio playout is disabled via the SetAudioPlayout() method, then

7
pc/test/integration_test_helpers.cc
View File

@ -46,10 +46,9 @@ void RemoveSsrcsAndKeepMsids(cricket::SessionDescription* desc) {
int FindFirstMediaStatsIndexByKind(
const std::string& kind,
const std::vector<const webrtc::DEPRECATED_RTCMediaStreamTrackStats*>&
media_stats_vec) {
for (size_t i = 0; i < media_stats_vec.size(); i++) {
if (media_stats_vec[i]->kind.ValueToString() == kind) {
const std::vector<const webrtc::RTCInboundRTPStreamStats*>& inbound_rtps) {
for (size_t i = 0; i < inbound_rtps.size(); i++) {
if (*inbound_rtps[i]->kind == kind) {
return i;
}
}

5
pc/test/integration_test_helpers.h
View File

@ -171,12 +171,9 @@ void RemoveSsrcsAndMsids(cricket::SessionDescription* desc);
// endpoint that only signals a=msid lines to convey stream_ids.
void RemoveSsrcsAndKeepMsids(cricket::SessionDescription* desc);
// TODO(https://crbug.com/webrtc/14175): Stop depending on "track" stats, the
// metrics we're interested in are already available in "inbound-rtp".
int FindFirstMediaStatsIndexByKind(
const std::string& kind,
const std::vector<const webrtc::DEPRECATED_RTCMediaStreamTrackStats*>&
media_stats_vec);
const std::vector<const webrtc::RTCInboundRTPStreamStats*>& inbound_rtps);
class TaskQueueMetronome : public webrtc::Metronome {
public: