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Add a test to detect excessive audio delay during renegotiation.

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This version uses relative_packet_arrival_delay as the target metric.

Bug: none
Change-Id: Ie6eb575ce4d13fd005f026862892b14bd4fb1135
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201620
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32962}
This commit is contained in:
Harald Alvestrand
2021-01-13 12:31:53 +00:00
committed by Commit Bot
parent 0e8dd039be
commit 94324f2774
1 changed files with 70 additions and 1 deletions
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71
pc/peer_connection_integrationtest.cc
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@ -5461,7 +5461,7 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
}
TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
ReegotiateManyAudioTransceivers) {
RenegotiateManyAudioTransceivers) {
PeerConnectionInterface::RTCConfiguration config;
config.sdp_semantics = SdpSemantics::kUnifiedPlan;
ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, config));
@ -5528,6 +5528,75 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
}
}
TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
RenegotiateManyVideoTransceiversAndWatchAudioDelay) {
PeerConnectionInterface::RTCConfiguration config;
config.sdp_semantics = SdpSemantics::kUnifiedPlan;
ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, config));
ConnectFakeSignaling();
caller()->AddAudioTrack();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Wait until we can see the audio flowing.
MediaExpectations media_expectations;
media_expectations.CalleeExpectsSomeAudio();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
// Get the baseline numbers for audio_packets and audio_delay.
auto received_stats = callee()->NewGetStats();
auto track_stats =
received_stats->GetStatsOfType<webrtc::RTCMediaStreamTrackStats>()[0];
ASSERT_TRUE(track_stats->relative_packet_arrival_delay.is_defined());
auto rtp_stats =
received_stats->GetStatsOfType<webrtc::RTCInboundRTPStreamStats>()[0];
ASSERT_TRUE(rtp_stats->packets_received.is_defined());
ASSERT_TRUE(rtp_stats->track_id.is_defined());
auto audio_track_stats_id = track_stats->id();
ASSERT_TRUE(received_stats->Get(audio_track_stats_id));
auto rtp_stats_id = rtp_stats->id();
ASSERT_EQ(audio_track_stats_id, *rtp_stats->track_id);
auto audio_packets = *rtp_stats->packets_received;
auto audio_delay = *track_stats->relative_packet_arrival_delay;
int current_size = caller()->pc()->GetTransceivers().size();
// Add more tracks until we get close to having issues.
// Making this number very large makes the test very slow.
while (current_size < 32) {
// Double the number of tracks
for (int i = 0; i < current_size; i++) {
caller()->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
}
current_size = caller()->pc()->GetTransceivers().size();
RTC_LOG(LS_INFO) << "Renegotiating with " << current_size << " tracks";
auto start_time_ms = rtc::TimeMillis();
caller()->CreateAndSetAndSignalOffer();
// We want to stop when the time exceeds one second.
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
auto elapsed_time_ms = rtc::TimeMillis() - start_time_ms;
RTC_LOG(LS_INFO) << "Renegotiating took " << elapsed_time_ms << " ms";
// This is a guard against the test using excessive amounts of time.
ASSERT_GT(5000, elapsed_time_ms)
<< "Video transceivers: Negotiation took too long after "
<< current_size << " tracks added";
auto report = callee()->NewGetStats();
track_stats =
report->GetAs<webrtc::RTCMediaStreamTrackStats>(audio_track_stats_id);
ASSERT_TRUE(track_stats);
rtp_stats = report->GetAs<webrtc::RTCInboundRTPStreamStats>(rtp_stats_id);
ASSERT_TRUE(rtp_stats);
auto delta_packets = *rtp_stats->packets_received - audio_packets;
auto delta_rpad = *track_stats->relative_packet_arrival_delay - audio_delay;
auto recent_delay = delta_packets > 0 ? delta_rpad / delta_packets : -1;
// An average relative packet arrival delay over the renegotiation of
// > 100 ms indicates that something is dramatically wrong, and will impact
// quality for sure.
ASSERT_GT(0.1, recent_delay);
// Increment trailing counters
audio_packets = *rtp_stats->packets_received;
audio_delay = *track_stats->relative_packet_arrival_delay;
}
}
INSTANTIATE_TEST_SUITE_P(PeerConnectionIntegrationTest,
PeerConnectionIntegrationTest,
Values(SdpSemantics::kPlanB,