mirror of
https://github.com/mollyim/webrtc.git
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6f727da62b
This reverts commit 71db9acc40.
Reason for revert: breaks downstream project.
Reason for force push: win bot broken.
Original change's description:
> RtpTransceiverInterface: introduce SetOfferedRtpHeaderExtensions.
>
> This change adds exposure of a new transceiver method for
> modifying the extensions offered in the next SDP negotiation,
> following spec details in https://w3c.github.io/webrtc-extensions/#rtcrtptransceiver-interface.
>
> Features:
> - The interface allows to control the negotiated direction as
> per https://tools.ietf.org/html/rfc5285#page-7.
> - The interface allows to remove an extension from SDP
> negotiation by modifying the direction to
> RtpTransceiverDirection::kStopped.
>
> Note: support for signalling directionality of header extensions
> in the SDP isn't implemented yet.
>
> https://chromestatus.com/feature/5680189201711104.
> Intent to prototype: https://groups.google.com/a/chromium.org/g/blink-dev/c/65YdUi02yZk
>
> Bug: chromium:1051821
> Change-Id: Iaabc34446f038c46d93c442e90c2a77f77d542d4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176408
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Markus Handell <handellm@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31487}
TBR=hta@webrtc.org,handellm@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
No-Try: true
Bug: chromium:1051821
Change-Id: I70e1a07225d7eeec7480fa5577d8ff647eba6902
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177103
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31516}
144 lines
6.1 KiB
C++
144 lines
6.1 KiB
C++
/*
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* Copyright 2020 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include <memory>
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#include <tuple>
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#include "api/rtc_event_log/rtc_event_log_factory.h"
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#include "api/task_queue/default_task_queue_factory.h"
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#include "media/base/fake_media_engine.h"
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#include "p2p/base/fake_port_allocator.h"
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#include "pc/media_session.h"
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#include "pc/peer_connection_wrapper.h"
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#include "rtc_base/gunit.h"
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#include "rtc_base/strings/string_builder.h"
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#include "test/gmock.h"
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namespace webrtc {
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using ::testing::Combine;
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using ::testing::ElementsAre;
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using ::testing::Field;
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using ::testing::Return;
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using ::testing::Values;
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class PeerConnectionHeaderExtensionTest
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: public ::testing::TestWithParam<
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std::tuple<cricket::MediaType, SdpSemantics>> {
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protected:
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std::unique_ptr<PeerConnectionWrapper> CreatePeerConnection(
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cricket::MediaType media_type,
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absl::optional<SdpSemantics> semantics,
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std::vector<RtpHeaderExtensionCapability> extensions) {
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auto voice = std::make_unique<cricket::FakeVoiceEngine>();
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auto video = std::make_unique<cricket::FakeVideoEngine>();
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if (media_type == cricket::MediaType::MEDIA_TYPE_AUDIO)
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voice->SetRtpHeaderExtensions(extensions);
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else
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video->SetRtpHeaderExtensions(extensions);
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auto media_engine = std::make_unique<cricket::CompositeMediaEngine>(
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std::move(voice), std::move(video));
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PeerConnectionFactoryDependencies factory_dependencies;
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factory_dependencies.network_thread = rtc::Thread::Current();
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factory_dependencies.worker_thread = rtc::Thread::Current();
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factory_dependencies.signaling_thread = rtc::Thread::Current();
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factory_dependencies.task_queue_factory = CreateDefaultTaskQueueFactory();
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factory_dependencies.media_engine = std::move(media_engine);
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factory_dependencies.call_factory = CreateCallFactory();
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factory_dependencies.event_log_factory =
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std::make_unique<RtcEventLogFactory>(
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factory_dependencies.task_queue_factory.get());
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auto pc_factory =
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CreateModularPeerConnectionFactory(std::move(factory_dependencies));
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auto fake_port_allocator = std::make_unique<cricket::FakePortAllocator>(
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rtc::Thread::Current(), nullptr);
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auto observer = std::make_unique<MockPeerConnectionObserver>();
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PeerConnectionInterface::RTCConfiguration config;
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if (semantics)
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config.sdp_semantics = *semantics;
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auto pc = pc_factory->CreatePeerConnection(
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config, std::move(fake_port_allocator), nullptr, observer.get());
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observer->SetPeerConnectionInterface(pc.get());
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return std::make_unique<PeerConnectionWrapper>(pc_factory, pc,
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std::move(observer));
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}
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};
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TEST_P(PeerConnectionHeaderExtensionTest, TransceiverOffersHeaderExtensions) {
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cricket::MediaType media_type;
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SdpSemantics semantics;
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std::tie(media_type, semantics) = GetParam();
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if (semantics != SdpSemantics::kUnifiedPlan)
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return;
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std::vector<RtpHeaderExtensionCapability> extensions(
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{RtpHeaderExtensionCapability("uri1", 1,
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RtpTransceiverDirection::kStopped),
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RtpHeaderExtensionCapability("uri2", 2,
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RtpTransceiverDirection::kSendOnly),
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RtpHeaderExtensionCapability("uri3", 3,
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RtpTransceiverDirection::kRecvOnly),
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RtpHeaderExtensionCapability("uri4", 4,
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RtpTransceiverDirection::kSendRecv)});
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std::unique_ptr<PeerConnectionWrapper> wrapper =
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CreatePeerConnection(media_type, semantics, extensions);
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auto transceiver = wrapper->AddTransceiver(media_type);
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EXPECT_EQ(transceiver->HeaderExtensionsToOffer(), extensions);
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}
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TEST_P(PeerConnectionHeaderExtensionTest,
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SenderReceiverCapabilitiesReturnNotStoppedExtensions) {
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cricket::MediaType media_type;
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SdpSemantics semantics;
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std::tie(media_type, semantics) = GetParam();
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std::unique_ptr<PeerConnectionWrapper> wrapper = CreatePeerConnection(
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media_type, semantics,
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std::vector<RtpHeaderExtensionCapability>(
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{RtpHeaderExtensionCapability("uri1", 1,
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RtpTransceiverDirection::kSendRecv),
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RtpHeaderExtensionCapability("uri2", 2,
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RtpTransceiverDirection::kStopped),
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RtpHeaderExtensionCapability("uri3", 3,
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RtpTransceiverDirection::kRecvOnly)}));
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EXPECT_THAT(wrapper->pc_factory()
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->GetRtpSenderCapabilities(media_type)
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.header_extensions,
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ElementsAre(Field(&RtpHeaderExtensionCapability::uri, "uri1"),
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Field(&RtpHeaderExtensionCapability::uri, "uri3")));
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EXPECT_EQ(wrapper->pc_factory()
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->GetRtpReceiverCapabilities(media_type)
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.header_extensions,
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wrapper->pc_factory()
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->GetRtpSenderCapabilities(media_type)
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.header_extensions);
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}
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INSTANTIATE_TEST_SUITE_P(
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,
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PeerConnectionHeaderExtensionTest,
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Combine(Values(SdpSemantics::kPlanB, SdpSemantics::kUnifiedPlan),
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Values(cricket::MediaType::MEDIA_TYPE_AUDIO,
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cricket::MediaType::MEDIA_TYPE_VIDEO)),
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[](const testing::TestParamInfo<
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PeerConnectionHeaderExtensionTest::ParamType>& info) {
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cricket::MediaType media_type;
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SdpSemantics semantics;
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std::tie(media_type, semantics) = info.param;
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return (rtc::StringBuilder("With")
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<< (semantics == SdpSemantics::kPlanB ? "PlanB" : "UnifiedPlan")
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<< "And"
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<< (media_type == cricket::MediaType::MEDIA_TYPE_AUDIO ? "Voice"
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: "Video")
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<< "Engine")
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.str();
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});
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} // namespace webrtc
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