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Reland "Enable DD and VLA header extensions by default for Simulcast/SVC"
This is a reland of commit 33c7edd58a
taking into account GFD which can be enabled by field trials and somewhat conflicts with DD
Original change's description:
> Enable DD and VLA header extensions by default for Simulcast/SVC
>
> When Simulcast (more than one encoding) or SVC (a scalability mode
> other than the default L1T1) is used, enable the AV1 Dependency
> Descriptor and the video-layer-allocations RTP header extensions by
> default.
>
> The RTP header extensions API can be used to disable them if needed.
>
> BUG=webrtc:15378
>
> Change-Id: I587ac32c9d681461496a136f6950b007e72da86d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326100
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Cr-Commit-Position: refs/heads/main@{#41332}
Bug: webrtc:15378
Change-Id: I190edc9435083c0a0a65a6959363f3c41e4a3d1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330563
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41615}
This commit is contained in:
parent
7f8470aeee
commit
6a3bbefd58
2 changed files with 104 additions and 0 deletions
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@ -171,6 +171,35 @@ RtpTransceiver::RtpTransceiver(
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: media_engine()->voice().send_codecs());
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senders_.push_back(sender);
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receivers_.push_back(receiver);
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// Set default header extensions depending on whether simulcast/SVC is used.
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RtpParameters parameters = sender->internal()->GetParametersInternal();
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bool uses_simulcast = parameters.encodings.size() > 1;
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bool uses_svc = !parameters.encodings.empty() &&
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parameters.encodings[0].scalability_mode.has_value() &&
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parameters.encodings[0].scalability_mode !=
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ScalabilityModeToString(ScalabilityMode::kL1T1);
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if (uses_simulcast || uses_svc) {
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// Enable DD and VLA extensions, can be deactivated by the API.
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// Skip this if the GFD extension was enabled via field trial
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// for backward compability reasons.
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bool uses_gfd =
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absl::c_find_if(
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header_extensions_to_negotiate_,
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[](const RtpHeaderExtensionCapability& ext) {
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return ext.uri == RtpExtension::kGenericFrameDescriptorUri00 &&
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ext.direction != webrtc::RtpTransceiverDirection::kStopped;
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}) != header_extensions_to_negotiate_.end();
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if (!uses_gfd) {
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for (RtpHeaderExtensionCapability& ext :
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header_extensions_to_negotiate_) {
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if (ext.uri == RtpExtension::kVideoLayersAllocationUri ||
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ext.uri == RtpExtension::kDependencyDescriptorUri) {
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ext.direction = RtpTransceiverDirection::kSendRecv;
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}
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}
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}
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}
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}
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RtpTransceiver::~RtpTransceiver() {
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@ -481,6 +481,81 @@ TEST_F(RtpTransceiverTestForHeaderExtensions,
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RtpTransceiverDirection::kStopped)));
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}
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TEST_F(RtpTransceiverTestForHeaderExtensions,
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SimulcastOrSvcEnablesExtensionsByDefault) {
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std::vector<RtpHeaderExtensionCapability> extensions = {
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{RtpExtension::kDependencyDescriptorUri, 1,
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RtpTransceiverDirection::kStopped},
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{RtpExtension::kVideoLayersAllocationUri, 2,
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RtpTransceiverDirection::kStopped},
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};
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// Default is stopped.
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auto sender = rtc::make_ref_counted<MockRtpSenderInternal>();
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auto transceiver = rtc::make_ref_counted<RtpTransceiver>(
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RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
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rtc::Thread::Current(), sender),
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RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
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rtc::Thread::Current(), rtc::Thread::Current(), receiver_),
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context(), extensions,
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/* on_negotiation_needed= */ [] {});
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std::vector<webrtc::RtpHeaderExtensionCapability> header_extensions =
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transceiver->GetHeaderExtensionsToNegotiate();
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ASSERT_EQ(header_extensions.size(), 2u);
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EXPECT_EQ(header_extensions[0].uri, RtpExtension::kDependencyDescriptorUri);
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EXPECT_EQ(header_extensions[0].direction, RtpTransceiverDirection::kStopped);
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EXPECT_EQ(header_extensions[1].uri, RtpExtension::kVideoLayersAllocationUri);
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EXPECT_EQ(header_extensions[1].direction, RtpTransceiverDirection::kStopped);
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// Simulcast, i.e. more than one encoding.
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RtpParameters simulcast_parameters;
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simulcast_parameters.encodings.resize(2);
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auto simulcast_sender = rtc::make_ref_counted<MockRtpSenderInternal>();
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EXPECT_CALL(*simulcast_sender, GetParametersInternal())
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.WillRepeatedly(Return(simulcast_parameters));
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auto simulcast_transceiver = rtc::make_ref_counted<RtpTransceiver>(
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RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
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rtc::Thread::Current(), simulcast_sender),
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RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
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rtc::Thread::Current(), rtc::Thread::Current(), receiver_),
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context(), extensions,
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/* on_negotiation_needed= */ [] {});
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auto simulcast_extensions =
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simulcast_transceiver->GetHeaderExtensionsToNegotiate();
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ASSERT_EQ(simulcast_extensions.size(), 2u);
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EXPECT_EQ(simulcast_extensions[0].uri,
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RtpExtension::kDependencyDescriptorUri);
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EXPECT_EQ(simulcast_extensions[0].direction,
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RtpTransceiverDirection::kSendRecv);
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EXPECT_EQ(simulcast_extensions[1].uri,
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RtpExtension::kVideoLayersAllocationUri);
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EXPECT_EQ(simulcast_extensions[1].direction,
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RtpTransceiverDirection::kSendRecv);
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// SVC, a single encoding with a scalabilityMode other than L1T1.
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webrtc::RtpParameters svc_parameters;
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svc_parameters.encodings.resize(1);
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svc_parameters.encodings[0].scalability_mode = "L3T3";
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auto svc_sender = rtc::make_ref_counted<MockRtpSenderInternal>();
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EXPECT_CALL(*svc_sender, GetParametersInternal())
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.WillRepeatedly(Return(svc_parameters));
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auto svc_transceiver = rtc::make_ref_counted<RtpTransceiver>(
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RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
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rtc::Thread::Current(), svc_sender),
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RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
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rtc::Thread::Current(), rtc::Thread::Current(), receiver_),
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context(), extensions,
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/* on_negotiation_needed= */ [] {});
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std::vector<webrtc::RtpHeaderExtensionCapability> svc_extensions =
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svc_transceiver->GetHeaderExtensionsToNegotiate();
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ASSERT_EQ(svc_extensions.size(), 2u);
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EXPECT_EQ(svc_extensions[0].uri, RtpExtension::kDependencyDescriptorUri);
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EXPECT_EQ(svc_extensions[0].direction, RtpTransceiverDirection::kSendRecv);
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EXPECT_EQ(svc_extensions[1].uri, RtpExtension::kVideoLayersAllocationUri);
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EXPECT_EQ(svc_extensions[1].direction, RtpTransceiverDirection::kSendRecv);
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}
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} // namespace
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} // namespace webrtc
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