Revert "Improve structuring of test for audio glitches."
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}
diff --git a/pc/peer_connection_integrationtest.cc b/pc/peer_connection_integrationtest.cc
index 32bfd1a..6cf3b05 100644
--- a/pc/peer_connection_integrationtest.cc
+++ b/pc/peer_connection_integrationtest.cc
@@ -600,46 +600,6 @@
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 @@
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 @@
}
~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 @@
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 @@
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 @@
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;
}
}
