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Revert "Improve structuring of test for audio glitches."

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This reverts commit fdbaeda00362a385de85b4c08aa0b536062a8415.

Reason for revert: Breaks downstream project, see https://bugs.chromium.org/p/webrtc/issues/detail?id=12371

Original change's description:
> Improve structuring of test for audio glitches.
>
> Bug: webrtc:12361
> Change-Id: Ieddc3dafbb638b3bd73dd79bcafa499290fa4340
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201723
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32973}

TBR=hbos@webrtc.org,hta@webrtc.org

Change-Id: Ie337de79a80113958607a7508d136c05fe6d9167
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:12361
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202024
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32993}
This commit is contained in:
Alex Loiko
2021-01-15 09:58:50 +00:00
committed by Commit Bot
parent 98db5d1a11
commit 3ae09f5419
1 changed files with 32 additions and 54 deletions
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86
pc/peer_connection_integrationtest.cc
View File

@ -600,46 +600,6 @@ class PeerConnectionWrapper : public webrtc::PeerConnectionObserver,
webrtc::CreateSessionDescription(SdpType::kRollback, ""));
}
// Functions for querying stats.
void StartWatchingDelayStats() {
// Get the baseline numbers for audio_packets and audio_delay.
auto received_stats = 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());
audio_track_stats_id_ = track_stats->id();
ASSERT_TRUE(received_stats->Get(audio_track_stats_id_));
rtp_stats_id_ = rtp_stats->id();
ASSERT_EQ(audio_track_stats_id_, *rtp_stats->track_id);
audio_packets_stat_ = *rtp_stats->packets_received;
audio_delay_stat_ = *track_stats->relative_packet_arrival_delay;
}
void UpdateDelayStats(std::string tag, int desc_size) {
auto report = NewGetStats();
auto track_stats =
report->GetAs<webrtc::RTCMediaStreamTrackStats>(audio_track_stats_id_);
ASSERT_TRUE(track_stats);
auto rtp_stats =
report->GetAs<webrtc::RTCInboundRTPStreamStats>(rtp_stats_id_);
ASSERT_TRUE(rtp_stats);
auto delta_packets = *rtp_stats->packets_received - audio_packets_stat_;
auto delta_rpad =
*track_stats->relative_packet_arrival_delay - audio_delay_stat_;
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) << tag << " size " << desc_size;
// Increment trailing counters
audio_packets_stat_ = *rtp_stats->packets_received;
audio_delay_stat_ = *track_stats->relative_packet_arrival_delay;
}
private:
explicit PeerConnectionWrapper(const std::string& debug_name)
: debug_name_(debug_name) {}
@ -1108,12 +1068,6 @@ class PeerConnectionWrapper : public webrtc::PeerConnectionObserver,
peer_connection_signaling_state_history_;
webrtc::FakeRtcEventLogFactory* event_log_factory_;
// Variables for tracking delay stats on an audio track
int audio_packets_stat_ = 0;
double audio_delay_stat_ = 0.0;
std::string rtp_stats_id_;
std::string audio_track_stats_id_;
rtc::AsyncInvoker invoker_;
friend class PeerConnectionIntegrationBaseTest;
@ -1279,7 +1233,7 @@ class PeerConnectionIntegrationBaseTest : public ::testing::Test {
}
~PeerConnectionIntegrationBaseTest() {
// The PeerConnections should be deleted before the TurnCustomizers.
// The PeerConnections should deleted before the TurnCustomizers.
// A TurnPort is created with a raw pointer to a TurnCustomizer. The
// TurnPort has the same lifetime as the PeerConnection, so it's expected
// that the TurnCustomizer outlives the life of the PeerConnection or else
@ -5581,7 +5535,6 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, config));
ConnectFakeSignaling();
caller()->AddAudioTrack();
callee()->AddAudioTrack();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Wait until we can see the audio flowing.
@ -5589,10 +5542,21 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
media_expectations.CalleeExpectsSomeAudio();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
// Get the baseline numbers for audio_packets and audio_delay
// in both directions.
caller()->StartWatchingDelayStats();
callee()->StartWatchingDelayStats();
// 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.
@ -5614,8 +5578,22 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
ASSERT_GT(5000, elapsed_time_ms)
<< "Video transceivers: Negotiation took too long after "
<< current_size << " tracks added";
caller()->UpdateDelayStats("caller reception", current_size);
callee()->UpdateDelayStats("callee reception", current_size);
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;
}
}