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Adding integration test for unsignaled inbound RTP stream stats.

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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}
This commit is contained in:
deadbeef
2017-04-18 16:01:17 -07:00
committed by Commit bot
parent 59edb9298e
commit d8ad788a2b
2 changed files with 93 additions and 29 deletions
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103
webrtc/pc/peerconnection_integrationtest.cc
View File

@ -106,6 +106,17 @@ PeerConnectionInterface::RTCOfferAnswerOptions IceRestartOfferAnswerOptions() {
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 @@ class PeerConnectionWrapper : public webrtc::PeerConnectionObserver,
// 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 @@ class PeerConnectionWrapper : public webrtc::PeerConnectionObserver,
}
// 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 @@ class PeerConnectionIntegrationTest : public testing::Test {
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 @@ TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithoutSsrcOrMsidSignaling) {
// 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_F(PeerConnectionIntegrationTest,
// 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 @@ TEST_F(PeerConnectionIntegrationTest, GetAudioOutputLevelStats) {
// 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 @@ TEST_F(PeerConnectionIntegrationTest, GetAudioInputLevelStats) {
// 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 @@ TEST_F(PeerConnectionIntegrationTest, GetBytesReceivedStats) {
// 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 @@ TEST_F(PeerConnectionIntegrationTest, GetBytesSentStats) {
// 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 @@ TEST_F(PeerConnectionIntegrationTest, Dtls10CipherStatsAndUmaMetrics) {
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 @@ TEST_F(PeerConnectionIntegrationTest, Dtls12CipherStatsAndUmaMetrics) {
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 @@ TEST_F(PeerConnectionIntegrationTest, GetSources) {
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];

19
webrtc/pc/test/mockpeerconnectionobservers.h
View File

@ -225,6 +225,25 @@ class MockStatsObserver : public webrtc::StatsObserver {
} stats_;
};
// Helper class that just stores the report from the callback.
class MockRTCStatsCollectorCallback : public webrtc::RTCStatsCollectorCallback {
public:
rtc::scoped_refptr<const RTCStatsReport> report() { return report_; }
bool called() const { return called_; }
protected:
void OnStatsDelivered(
const rtc::scoped_refptr<const RTCStatsReport>& report) override {
report_ = report;
called_ = true;
}
private:
bool called_ = false;
rtc::scoped_refptr<const RTCStatsReport> report_;
};
} // namespace webrtc
#endif // WEBRTC_PC_TEST_MOCKPEERCONNECTIONOBSERVERS_H_