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Revert "Exercise AV1 simulcast paths in tests."

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This reverts commit c8ab6c449c84f17fabf8da58456d396bdb5da762.

Reason for revert: new test fails to run upstream

Original change's description:
> Exercise AV1 simulcast paths in tests.
>
> This is something we get "for free" with the
> "WebRTC-AllowDisablingLegacyScalability" field trial that has been
> wired up to support VP9 simulcast.
>
> This test works and passes, however the ramp-up time is pretty bad.
> - VP9 simulcast takes approximately 4 seconds to ramp up.
> - VP9 SVC takes approximately 16 seconds to ramp up.
> - AV1 simulcast takes approximately 22 seconds to ramp up.
>
> A TODO is added (webrtc:15006) and the test is given extra timeout,
> a full minute to get bytes flowing on all layers.
>
> Despite ramp-up being bad, it's important to test that AV1 simulcast
> is in fact working to avoid regressions due to obsolete assumptions
> about which codec do or do not support simulcast. AV1 simulcast is an
> opt-in feature so there is no harm in the API not being perfect yet.
>
> Bug: webrtc:15005, webrtc:15006
> Change-Id: If0158d172647f0462bd6db802406249d93e01871
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/297982
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39586}

Bug: webrtc:15005, webrtc:15006
Change-Id: I7da6df8bb51219e7d0acfd3b62b4ec08e25bfdc7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298049
Owners-Override: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#39595}
This commit is contained in:
Jeremy Leconte 2023-03-17 19:40:35 +00:00 committed by WebRTC LUCI CQ
parent d9faa73cbb
commit 6a1ded02d9
1 changed files with 0 additions and 77 deletions
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77
pc/peer_connection_simulcast_unittest.cc
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@ -1506,81 +1506,4 @@ TEST_F(PeerConnectionSimulcastWithMediaFlowTests,
EXPECT_THAT(*outbound_rtps[2]->scalability_mode, StrEq("L1T3"));
}
// TODO(https://crbug.com/webrtc/15005): A field trial shouldn't be needed to
// get spec-compliant behavior! The same field trial is also used for VP9
// simulcast (https://crbug.com/webrtc/14884).
TEST_F(PeerConnectionSimulcastWithMediaFlowTests,
SendingThreeEncodings_AV1_Simulcast) {
test::ScopedFieldTrials field_trials(
"WebRTC-AllowDisablingLegacyScalability/Enabled/");
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, "AV1");
transceiver->SetCodecPreferences(codecs);
// Opt-in to spec-compliant simulcast by explicitly setting the
// `scalability_mode`.
rtc::scoped_refptr<RtpSenderInterface> sender = transceiver->sender();
RtpParameters parameters = sender->GetParameters();
ASSERT_EQ(parameters.encodings.size(), 3u);
parameters.encodings[0].scalability_mode = "L1T3";
parameters.encodings[1].scalability_mode = "L1T3";
parameters.encodings[2].scalability_mode = "L1T3";
sender->SetParameters(parameters);
NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper, layers);
local_pc_wrapper->WaitForConnection();
remote_pc_wrapper->WaitForConnection();
// GetParameters() does not report any fallback.
parameters = sender->GetParameters();
ASSERT_EQ(parameters.encodings.size(), 3u);
EXPECT_THAT(parameters.encodings[0].scalability_mode,
Optional(std::string("L1T3")));
EXPECT_THAT(parameters.encodings[1].scalability_mode,
Optional(std::string("L1T3")));
EXPECT_THAT(parameters.encodings[2].scalability_mode,
Optional(std::string("L1T3")));
// Wait until media is flowing on all three layers.
// Ramp up time is needed before all three layers are sending.
//
// This test is given 2X timeout because AV1 simulcast ramp-up time is
// terrible compared to other codecs.
// TODO(https://crbug.com/webrtc/15006): Improve the ramp-up time and stop
// giving this test extra long timeout.
EXPECT_TRUE_WAIT(HasOutboundRtpBytesSent(local_pc_wrapper, 3u),
(2 * kLongTimeoutForRampingUp).ms());
// Sometimes additional ramp up is needed to get the expected resolutions. If
// that has not happened yet we log (`log_during_ramp_up=true`).
EXPECT_TRUE_WAIT(HasOutboundRtpExpectedResolutions(
local_pc_wrapper,
{{"f", 320, 180}, {"h", 640, 360}, {"q", 1280, 720}},
/*log_during_ramp_up=*/true),
kLongTimeoutForRampingUp.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/AV1"));
EXPECT_THAT(GetCurrentCodecMimeType(report, *outbound_rtps[1]),
StrCaseEq("video/AV1"));
EXPECT_THAT(GetCurrentCodecMimeType(report, *outbound_rtps[2]),
StrCaseEq("video/AV1"));
EXPECT_THAT(*outbound_rtps[0]->scalability_mode, StrEq("L1T3"));
EXPECT_THAT(*outbound_rtps[1]->scalability_mode, StrEq("L1T3"));
EXPECT_THAT(*outbound_rtps[2]->scalability_mode, StrEq("L1T3"));
}
} // namespace webrtc