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Add simulcast test coverage for H264.

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The test only exists #if defined(WEBRTC_USE_H264) because H264 is not
available in all testing environments (e.g. Android bots fail without
these guards).

Bug: webrtc:14884
Change-Id: Ic1ff6b16f49f6666df042304ee98d826778da122
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295508
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39424}
This commit is contained in:
Henrik Boström 2023-02-28 15:38:57 +01:00 committed by WebRTC LUCI CQ
parent 80663cd0da
commit 3d6e7f8a3a
1 changed files with 50 additions and 0 deletions
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50
pc/peer_connection_simulcast_unittest.cc
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@ -1112,6 +1112,56 @@ TEST_F(PeerConnectionSimulcastWithMediaFlowTests,
EXPECT_THAT(*outbound_rtps[2]->scalability_mode, StartsWith("L1T"));
}
#if defined(WEBRTC_USE_H264)
TEST_F(PeerConnectionSimulcastWithMediaFlowTests,
SendingThreeEncodings_H264_Simulcast) {
rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
std::vector<SimulcastLayer> layers =
CreateLayers({"f", "h", "q"}, /*active=*/true);
rtc::scoped_refptr<RtpTransceiverInterface> transceiver =
AddTransceiverWithSimulcastLayers(local_pc_wrapper, remote_pc_wrapper,
layers);
std::vector<RtpCodecCapability> codecs =
GetCapabilitiesAndRestrictToCodec(local_pc_wrapper, "H264");
transceiver->SetCodecPreferences(codecs);
NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper, layers);
local_pc_wrapper->WaitForConnection();
remote_pc_wrapper->WaitForConnection();
// Wait until media is flowing on all three layers.
// Ramp up time is needed before all three layers are sending.
EXPECT_TRUE_WAIT(HasOutboundRtpBytesSent(local_pc_wrapper, 3u),
kLongTimeoutForRampingUp.ms());
// No significant additional ramp up time should be needed so we use
// `kDefaultTimeout` and `log_during_ramp_up`.
EXPECT_TRUE_WAIT(HasOutboundRtpExpectedResolutions(
local_pc_wrapper,
{{"f", 320, 180}, {"h", 640, 360}, {"q", 1280, 720}},
/*log_during_ramp_up=*/true),
kDefaultTimeout.ms());
// Verify codec and scalability mode.
rtc::scoped_refptr<const RTCStatsReport> report = GetStats(local_pc_wrapper);
std::vector<const RTCOutboundRTPStreamStats*> outbound_rtps =
report->GetStatsOfType<RTCOutboundRTPStreamStats>();
ASSERT_THAT(outbound_rtps, SizeIs(3u));
EXPECT_THAT(GetCurrentCodecMimeType(report, *outbound_rtps[0]),
StrCaseEq("video/H264"));
EXPECT_THAT(GetCurrentCodecMimeType(report, *outbound_rtps[1]),
StrCaseEq("video/H264"));
EXPECT_THAT(GetCurrentCodecMimeType(report, *outbound_rtps[2]),
StrCaseEq("video/H264"));
EXPECT_THAT(*outbound_rtps[0]->scalability_mode, StartsWith("L1T"));
EXPECT_THAT(*outbound_rtps[1]->scalability_mode, StartsWith("L1T"));
EXPECT_THAT(*outbound_rtps[2]->scalability_mode, StartsWith("L1T"));
}
#endif // defined(WEBRTC_USE_H264)
// The legacy SVC path is triggered when VP9 us used, but `scalability_mode` has
// not been specified.
// TODO(https://crbug.com/webrtc/14889): When legacy VP9 SVC path has been