Adding integration test for unsignaled inbound RTP stream stats.
The test isn't complete, since "track_id" ends up unset. But it's
better than having no test at all.
BUG=None
Review-Url: https://codereview.webrtc.org/2827643003
Cr-Commit-Position: refs/heads/master@{#17753}
diff --git a/webrtc/pc/peerconnection_integrationtest.cc b/webrtc/pc/peerconnection_integrationtest.cc
index 49771f6..3823f8b 100644
--- a/webrtc/pc/peerconnection_integrationtest.cc
+++ b/webrtc/pc/peerconnection_integrationtest.cc
@@ -106,6 +106,17 @@
return options;
}
+// Remove all stream information (SSRCs, track IDs, etc.) and "msid-semantic"
+// attribute from received SDP, simulating a legacy endpoint.
+void RemoveSsrcsAndMsids(cricket::SessionDescription* desc) {
+ for (ContentInfo& content : desc->contents()) {
+ MediaContentDescription* media_desc =
+ static_cast<MediaContentDescription*>(content.description);
+ media_desc->mutable_streams().clear();
+ }
+ desc->set_msid_supported(false);
+}
+
class SignalingMessageReceiver {
public:
virtual void ReceiveSdpMessage(const std::string& type,
@@ -408,7 +419,7 @@
// Returns a MockStatsObserver in a state after stats gathering finished,
// which can be used to access the gathered stats.
- rtc::scoped_refptr<MockStatsObserver> GetStatsForTrack(
+ rtc::scoped_refptr<MockStatsObserver> OldGetStatsForTrack(
webrtc::MediaStreamTrackInterface* track) {
rtc::scoped_refptr<MockStatsObserver> observer(
new rtc::RefCountedObject<MockStatsObserver>());
@@ -419,8 +430,18 @@
}
// Version that doesn't take a track "filter", and gathers all stats.
- rtc::scoped_refptr<MockStatsObserver> GetStats() {
- return GetStatsForTrack(nullptr);
+ rtc::scoped_refptr<MockStatsObserver> OldGetStats() {
+ return OldGetStatsForTrack(nullptr);
+ }
+
+ // Synchronously gets stats and returns them. If it times out, fails the test
+ // and returns null.
+ rtc::scoped_refptr<const webrtc::RTCStatsReport> NewGetStats() {
+ rtc::scoped_refptr<webrtc::MockRTCStatsCollectorCallback> callback(
+ new rtc::RefCountedObject<webrtc::MockRTCStatsCollectorCallback>());
+ peer_connection_->GetStats(callback);
+ EXPECT_TRUE_WAIT(callback->called(), kDefaultTimeout);
+ return callback->report();
}
int rendered_width() {
@@ -1099,7 +1120,7 @@
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(expected_cipher_suite),
- caller()->GetStats()->SrtpCipher(), kDefaultTimeout);
+ caller()->OldGetStats()->SrtpCipher(), kDefaultTimeout);
EXPECT_EQ(
1, caller_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
expected_cipher_suite));
@@ -1701,16 +1722,8 @@
// Add audio and video, testing that packets can be demuxed on payload type.
caller()->AddAudioVideoMediaStream();
callee()->AddAudioVideoMediaStream();
- // Remove all stream information (SSRCs, track IDs, etc.) and "msid-semantic"
- // attribute from received SDP, simulating a legacy endpoint.
- callee()->SetReceivedSdpMunger([](cricket::SessionDescription* desc) {
- for (ContentInfo& content : desc->contents()) {
- MediaContentDescription* media_desc =
- static_cast<MediaContentDescription*>(content.description);
- media_desc->mutable_streams().clear();
- }
- desc->set_msid_supported(false);
- });
+ // Remove SSRCs and MSIDs from the received offer SDP.
+ callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids);
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
ExpectNewFramesReceivedWithWait(
@@ -1787,7 +1800,7 @@
// Test that we can receive the audio output level from a remote audio track.
// TODO(deadbeef): Use a fake audio source and verify that the output level is
// exactly what the source on the other side was configured with.
-TEST_F(PeerConnectionIntegrationTest, GetAudioOutputLevelStats) {
+TEST_F(PeerConnectionIntegrationTest, GetAudioOutputLevelStatsWithOldStatsApi) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Just add an audio track.
@@ -1798,14 +1811,14 @@
// Get the audio output level stats. Note that the level is not available
// until an RTCP packet has been received.
- EXPECT_TRUE_WAIT(callee()->GetStats()->AudioOutputLevel() > 0,
+ EXPECT_TRUE_WAIT(callee()->OldGetStats()->AudioOutputLevel() > 0,
kMaxWaitForFramesMs);
}
// Test that an audio input level is reported.
// TODO(deadbeef): Use a fake audio source and verify that the input level is
// exactly what the source was configured with.
-TEST_F(PeerConnectionIntegrationTest, GetAudioInputLevelStats) {
+TEST_F(PeerConnectionIntegrationTest, GetAudioInputLevelStatsWithOldStatsApi) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Just add an audio track.
@@ -1816,12 +1829,12 @@
// Get the audio input level stats. The level should be available very
// soon after the test starts.
- EXPECT_TRUE_WAIT(caller()->GetStats()->AudioInputLevel() > 0,
+ EXPECT_TRUE_WAIT(caller()->OldGetStats()->AudioInputLevel() > 0,
kMaxWaitForStatsMs);
}
// Test that we can get incoming byte counts from both audio and video tracks.
-TEST_F(PeerConnectionIntegrationTest, GetBytesReceivedStats) {
+TEST_F(PeerConnectionIntegrationTest, GetBytesReceivedStatsWithOldStatsApi) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioVideoMediaStream();
@@ -1847,12 +1860,14 @@
// We received frames, so we definitely should have nonzero "received bytes"
// stats at this point.
- EXPECT_GT(callee()->GetStatsForTrack(remote_audio_track)->BytesReceived(), 0);
- EXPECT_GT(callee()->GetStatsForTrack(remote_video_track)->BytesReceived(), 0);
+ EXPECT_GT(callee()->OldGetStatsForTrack(remote_audio_track)->BytesReceived(),
+ 0);
+ EXPECT_GT(callee()->OldGetStatsForTrack(remote_video_track)->BytesReceived(),
+ 0);
}
// Test that we can get outgoing byte counts from both audio and video tracks.
-TEST_F(PeerConnectionIntegrationTest, GetBytesSentStats) {
+TEST_F(PeerConnectionIntegrationTest, GetBytesSentStatsWithOldStatsApi) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
auto audio_track = caller()->CreateLocalAudioTrack();
@@ -1869,8 +1884,38 @@
// The callee received frames, so we definitely should have nonzero "sent
// bytes" stats at this point.
- EXPECT_GT(caller()->GetStatsForTrack(audio_track)->BytesSent(), 0);
- EXPECT_GT(caller()->GetStatsForTrack(video_track)->BytesSent(), 0);
+ EXPECT_GT(caller()->OldGetStatsForTrack(audio_track)->BytesSent(), 0);
+ EXPECT_GT(caller()->OldGetStatsForTrack(video_track)->BytesSent(), 0);
+}
+
+// Test that we can get stats (using the new stats implemnetation) for
+// unsignaled streams. Meaning when SSRCs/MSIDs aren't signaled explicitly in
+// SDP.
+TEST_F(PeerConnectionIntegrationTest,
+ GetStatsForUnsignaledStreamWithNewStatsApi) {
+ ASSERT_TRUE(CreatePeerConnectionWrappers());
+ ConnectFakeSignaling();
+ caller()->AddAudioOnlyMediaStream();
+ // Remove SSRCs and MSIDs from the received offer SDP.
+ callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids);
+ caller()->CreateAndSetAndSignalOffer();
+ ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+ // Wait for one audio frame to be received by the callee.
+ ExpectNewFramesReceivedWithWait(0, 0, 1, 0, kMaxWaitForFramesMs);
+
+ // We received a frame, so we should have nonzero "bytes received" stats for
+ // the unsignaled stream, if stats are working for it.
+ rtc::scoped_refptr<const webrtc::RTCStatsReport> report =
+ callee()->NewGetStats();
+ ASSERT_NE(nullptr, report);
+ auto inbound_stream_stats =
+ report->GetStatsOfType<webrtc::RTCInboundRTPStreamStats>();
+ ASSERT_EQ(1U, inbound_stream_stats.size());
+ ASSERT_TRUE(inbound_stream_stats[0]->bytes_received.is_defined());
+ ASSERT_GT(*inbound_stream_stats[0]->bytes_received, 0U);
+ // TODO(deadbeef): Test that track_id is defined. This is not currently
+ // working since SSRCs are used to match RtpReceivers (and their tracks) with
+ // received stream stats in TrackMediaInfoMap.
}
// Test that DTLS 1.0 is used if both sides only support DTLS 1.0.
@@ -1908,10 +1953,10 @@
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
EXPECT_TRUE_WAIT(rtc::SSLStreamAdapter::IsAcceptableCipher(
- caller()->GetStats()->DtlsCipher(), rtc::KT_DEFAULT),
+ caller()->OldGetStats()->DtlsCipher(), rtc::KT_DEFAULT),
kDefaultTimeout);
EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
- caller()->GetStats()->SrtpCipher(), kDefaultTimeout);
+ caller()->OldGetStats()->SrtpCipher(), kDefaultTimeout);
EXPECT_EQ(1,
caller_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
kDefaultSrtpCryptoSuite));
@@ -1933,10 +1978,10 @@
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
EXPECT_TRUE_WAIT(rtc::SSLStreamAdapter::IsAcceptableCipher(
- caller()->GetStats()->DtlsCipher(), rtc::KT_DEFAULT),
+ caller()->OldGetStats()->DtlsCipher(), rtc::KT_DEFAULT),
kDefaultTimeout);
EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
- caller()->GetStats()->SrtpCipher(), kDefaultTimeout);
+ caller()->OldGetStats()->SrtpCipher(), kDefaultTimeout);
EXPECT_EQ(1,
caller_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
kDefaultSrtpCryptoSuite));
@@ -2765,7 +2810,7 @@
caller()->AddAudioOnlyMediaStream();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
- // Wait for one audio frame received by callee.
+ // Wait for one audio frame to be received by the callee.
ExpectNewFramesReceivedWithWait(0, 0, 1, 0, kMaxWaitForFramesMs);
ASSERT_GT(callee()->pc()->GetReceivers().size(), 0u);
auto receiver = callee()->pc()->GetReceivers()[0];
