This is a reland of f351c3408a0c7f695447a2a9f4e6a1719a0d6a26 Reland history: The original CL broke tests in chromium which were manually tested in the first reland. Another small fix was added to the reland to fix a downstream bug, which caused separate tests to fail in chromium. These were not caught because the chromium trybot was down. These are temporarily disabled in chrome to allow this change to roll in. Original change's description: > Reland "Adds support for multiple or no media stream ids." > > This is a reland of 1550292efe680ac79a18004705c908b1cdca54cb > > Original change's description: > > Adds support for multiple or no media stream ids. > > > > With Unified Plan SDP semantics, this adds support for specifying > > either no media stream ids or multiple media stream ids for a > > transceiver/sender/receiver. This includes serializing/deserializing > > SDPs with multiple a=msid lines in a m section, or an "a=msid:- > > <appdata>" line to indicate the no stream case. Note that this does > > not synchronize between multiple streams, this is still just supported > > based upon the first media stream id. > > > > Bug: webrtc:7932, webrtc:7933 > > Change-Id: Ib7433929af7b2925abe2824b485b360cec12f275 > > Reviewed-on: https://webrtc-review.googlesource.com/61341 > > Commit-Queue: Seth Hampson <shampson@webrtc.org> > > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#22611} > > Bug: webrtc:7932, webrtc:7933 > Change-Id: Ica272ac18088103e65cccf6b96a6d3ecccb178ed > Reviewed-on: https://webrtc-review.googlesource.com/65560 > Commit-Queue: Seth Hampson <shampson@webrtc.org> > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22687} TBR=deadbeef@webrtc.org Bug: webrtc:7932, webrtc:7933 Change-Id: Ideb30219b2f952dd51428cd4e8bd43ef49df5b17 Reviewed-on: https://webrtc-review.googlesource.com/66280 Commit-Queue: Seth Hampson <shampson@webrtc.org> Reviewed-by: Seth Hampson <shampson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22699}
1335 lines
54 KiB
C++
1335 lines
54 KiB
C++
/*
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* Copyright 2017 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 <vector>
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#include "api/audio_codecs/builtin_audio_decoder_factory.h"
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#include "api/audio_codecs/builtin_audio_encoder_factory.h"
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#include "api/jsep.h"
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#include "api/mediastreaminterface.h"
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#include "api/peerconnectioninterface.h"
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#include "api/umametrics.h"
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#include "pc/mediasession.h"
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#include "pc/mediastream.h"
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#include "pc/mediastreamtrack.h"
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#include "pc/peerconnectionwrapper.h"
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#include "pc/sdputils.h"
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#include "pc/test/fakeaudiocapturemodule.h"
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#include "pc/test/mockpeerconnectionobservers.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/gunit.h"
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#include "rtc_base/ptr_util.h"
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#include "rtc_base/refcountedobject.h"
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#include "rtc_base/scoped_ref_ptr.h"
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#include "rtc_base/thread.h"
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#include "test/gmock.h"
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// This file contains tests for RTP Media API-related behavior of
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// |webrtc::PeerConnection|, see https://w3c.github.io/webrtc-pc/#rtp-media-api.
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namespace webrtc {
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using RTCConfiguration = PeerConnectionInterface::RTCConfiguration;
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using ::testing::ElementsAre;
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using ::testing::UnorderedElementsAre;
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const uint32_t kDefaultTimeout = 10000u;
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template <typename MethodFunctor>
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class OnSuccessObserver : public rtc::RefCountedObject<
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webrtc::SetRemoteDescriptionObserverInterface> {
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public:
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explicit OnSuccessObserver(MethodFunctor on_success)
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: on_success_(std::move(on_success)) {}
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// webrtc::SetRemoteDescriptionObserverInterface implementation.
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void OnSetRemoteDescriptionComplete(webrtc::RTCError error) override {
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RTC_CHECK(error.ok());
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on_success_();
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}
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private:
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MethodFunctor on_success_;
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};
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class PeerConnectionRtpTest : public testing::Test {
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public:
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PeerConnectionRtpTest()
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: pc_factory_(
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CreatePeerConnectionFactory(rtc::Thread::Current(),
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rtc::Thread::Current(),
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rtc::Thread::Current(),
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FakeAudioCaptureModule::Create(),
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CreateBuiltinAudioEncoderFactory(),
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CreateBuiltinAudioDecoderFactory(),
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nullptr,
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nullptr)) {}
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std::unique_ptr<PeerConnectionWrapper> CreatePeerConnection() {
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return CreatePeerConnection(RTCConfiguration());
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}
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std::unique_ptr<PeerConnectionWrapper> CreatePeerConnectionWithPlanB() {
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RTCConfiguration config;
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config.sdp_semantics = SdpSemantics::kPlanB;
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return CreatePeerConnection(config);
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}
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std::unique_ptr<PeerConnectionWrapper> CreatePeerConnectionWithUnifiedPlan() {
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RTCConfiguration config;
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config.sdp_semantics = SdpSemantics::kUnifiedPlan;
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return CreatePeerConnection(config);
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}
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std::unique_ptr<PeerConnectionWrapper> CreatePeerConnection(
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const RTCConfiguration& config) {
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auto observer = rtc::MakeUnique<MockPeerConnectionObserver>();
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auto pc = pc_factory_->CreatePeerConnection(config, nullptr, nullptr,
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observer.get());
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return rtc::MakeUnique<PeerConnectionWrapper>(pc_factory_, pc,
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std::move(observer));
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}
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protected:
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rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory_;
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};
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// These tests cover |webrtc::PeerConnectionObserver| callbacks firing upon
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// setting the remote description.
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class PeerConnectionRtpCallbacksTest : public PeerConnectionRtpTest {};
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TEST_F(PeerConnectionRtpCallbacksTest, AddTrackWithoutStreamFiresOnAddTrack) {
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auto caller = CreatePeerConnection();
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auto callee = CreatePeerConnection();
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ASSERT_TRUE(caller->AddTrack(caller->CreateAudioTrack("audio_track")));
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ASSERT_TRUE(
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callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
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static_cast<webrtc::RTCError*>(nullptr)));
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ASSERT_EQ(callee->observer()->add_track_events_.size(), 1u);
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// Since we are not supporting the no stream case with Plan B, there should be
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// a generated stream, even though we didn't set one with AddTrack.
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auto& add_track_event = callee->observer()->add_track_events_[0];
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ASSERT_EQ(add_track_event.streams.size(), 1u);
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EXPECT_TRUE(add_track_event.streams[0]->FindAudioTrack("audio_track"));
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EXPECT_EQ(add_track_event.streams, add_track_event.receiver->streams());
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}
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TEST_F(PeerConnectionRtpCallbacksTest, AddTrackWithStreamFiresOnAddTrack) {
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auto caller = CreatePeerConnection();
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auto callee = CreatePeerConnection();
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ASSERT_TRUE(caller->AddTrack(caller->CreateAudioTrack("audio_track"),
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{"audio_stream"}));
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ASSERT_TRUE(
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callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
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static_cast<webrtc::RTCError*>(nullptr)));
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ASSERT_EQ(callee->observer()->add_track_events_.size(), 1u);
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auto& add_track_event = callee->observer()->add_track_events_[0];
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ASSERT_EQ(add_track_event.streams.size(), 1u);
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EXPECT_EQ("audio_stream", add_track_event.streams[0]->id());
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EXPECT_TRUE(add_track_event.streams[0]->FindAudioTrack("audio_track"));
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EXPECT_EQ(add_track_event.streams, add_track_event.receiver->streams());
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}
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TEST_F(PeerConnectionRtpCallbacksTest,
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RemoveTrackWithoutStreamFiresOnRemoveTrack) {
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auto caller = CreatePeerConnection();
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auto callee = CreatePeerConnection();
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auto sender = caller->AddTrack(caller->CreateAudioTrack("audio_track"), {});
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ASSERT_TRUE(
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callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
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static_cast<webrtc::RTCError*>(nullptr)));
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ASSERT_EQ(callee->observer()->add_track_events_.size(), 1u);
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EXPECT_TRUE(caller->pc()->RemoveTrack(sender));
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ASSERT_TRUE(
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callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
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static_cast<webrtc::RTCError*>(nullptr)));
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ASSERT_EQ(callee->observer()->add_track_events_.size(), 1u);
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EXPECT_EQ(callee->observer()->GetAddTrackReceivers(),
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callee->observer()->remove_track_events_);
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}
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TEST_F(PeerConnectionRtpCallbacksTest,
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RemoveTrackWithStreamFiresOnRemoveTrack) {
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auto caller = CreatePeerConnection();
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auto callee = CreatePeerConnection();
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auto sender = caller->AddTrack(caller->CreateAudioTrack("audio_track"),
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{"audio_stream"});
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ASSERT_TRUE(
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callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
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static_cast<webrtc::RTCError*>(nullptr)));
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ASSERT_EQ(callee->observer()->add_track_events_.size(), 1u);
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EXPECT_TRUE(caller->pc()->RemoveTrack(sender));
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ASSERT_TRUE(
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callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
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static_cast<webrtc::RTCError*>(nullptr)));
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ASSERT_EQ(callee->observer()->add_track_events_.size(), 1u);
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EXPECT_EQ(callee->observer()->GetAddTrackReceivers(),
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callee->observer()->remove_track_events_);
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}
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TEST_F(PeerConnectionRtpCallbacksTest,
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RemoveTrackWithSharedStreamFiresOnRemoveTrack) {
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auto caller = CreatePeerConnection();
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auto callee = CreatePeerConnection();
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const char kSharedStreamId[] = "shared_audio_stream";
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auto sender1 = caller->AddTrack(caller->CreateAudioTrack("audio_track1"),
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{kSharedStreamId});
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auto sender2 = caller->AddTrack(caller->CreateAudioTrack("audio_track2"),
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{kSharedStreamId});
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ASSERT_TRUE(
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callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
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static_cast<webrtc::RTCError*>(nullptr)));
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ASSERT_EQ(callee->observer()->add_track_events_.size(), 2u);
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// Remove "audio_track1".
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EXPECT_TRUE(caller->pc()->RemoveTrack(sender1));
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ASSERT_TRUE(
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callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
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static_cast<webrtc::RTCError*>(nullptr)));
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ASSERT_EQ(callee->observer()->add_track_events_.size(), 2u);
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EXPECT_EQ(
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std::vector<rtc::scoped_refptr<RtpReceiverInterface>>{
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callee->observer()->add_track_events_[0].receiver},
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callee->observer()->remove_track_events_);
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// Remove "audio_track2".
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EXPECT_TRUE(caller->pc()->RemoveTrack(sender2));
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ASSERT_TRUE(
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callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
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static_cast<webrtc::RTCError*>(nullptr)));
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ASSERT_EQ(callee->observer()->add_track_events_.size(), 2u);
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EXPECT_EQ(callee->observer()->GetAddTrackReceivers(),
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callee->observer()->remove_track_events_);
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}
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// Tests the edge case that if a stream ID changes for a given track that both
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// OnRemoveTrack and OnAddTrack is fired.
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TEST_F(PeerConnectionRtpCallbacksTest,
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RemoteStreamIdChangesFiresOnRemoveAndOnAddTrack) {
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auto caller = CreatePeerConnection();
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auto callee = CreatePeerConnection();
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const char kStreamId1[] = "stream1";
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const char kStreamId2[] = "stream2";
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caller->AddTrack(caller->CreateAudioTrack("audio_track1"), {kStreamId1});
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ASSERT_TRUE(
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callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
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static_cast<webrtc::RTCError*>(nullptr)));
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EXPECT_EQ(callee->observer()->add_track_events_.size(), 1u);
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// Change the stream ID of the sender in the session description.
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auto offer = caller->CreateOfferAndSetAsLocal();
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auto audio_desc = offer->description()->GetContentDescriptionByName("audio");
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ASSERT_EQ(audio_desc->mutable_streams().size(), 1u);
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audio_desc->mutable_streams()[0].set_stream_ids({kStreamId2});
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ASSERT_TRUE(
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callee->SetRemoteDescription(CloneSessionDescription(offer.get()),
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static_cast<webrtc::RTCError*>(nullptr)));
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ASSERT_EQ(callee->observer()->add_track_events_.size(), 2u);
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EXPECT_EQ(callee->observer()->add_track_events_[1].streams[0]->id(),
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kStreamId2);
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ASSERT_EQ(callee->observer()->remove_track_events_.size(), 1u);
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EXPECT_EQ(callee->observer()->remove_track_events_[0]->streams()[0]->id(),
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kStreamId1);
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}
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// Tests that setting a remote description with sending transceivers will fire
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// the OnTrack callback for each transceiver and setting a remote description
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// with receive only transceivers will not call OnTrack. One transceiver is
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// created without any stream_ids, while the other is created with multiple
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// stream_ids.
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TEST_F(PeerConnectionRtpCallbacksTest, UnifiedPlanAddTransceiverCallsOnTrack) {
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const std::string kStreamId1 = "video_stream1";
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const std::string kStreamId2 = "video_stream2";
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auto caller = CreatePeerConnectionWithUnifiedPlan();
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auto callee = CreatePeerConnectionWithUnifiedPlan();
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auto audio_transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
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RtpTransceiverInit video_transceiver_init;
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video_transceiver_init.stream_ids = {kStreamId1, kStreamId2};
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auto video_transceiver =
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caller->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, video_transceiver_init);
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ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
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ASSERT_EQ(0u, caller->observer()->on_track_transceivers_.size());
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ASSERT_EQ(2u, callee->observer()->on_track_transceivers_.size());
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EXPECT_EQ(audio_transceiver->mid(),
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callee->pc()->GetTransceivers()[0]->mid());
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EXPECT_EQ(video_transceiver->mid(),
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callee->pc()->GetTransceivers()[1]->mid());
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std::vector<rtc::scoped_refptr<MediaStreamInterface>> audio_streams =
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callee->pc()->GetTransceivers()[0]->receiver()->streams();
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std::vector<rtc::scoped_refptr<MediaStreamInterface>> video_streams =
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callee->pc()->GetTransceivers()[1]->receiver()->streams();
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ASSERT_EQ(0u, audio_streams.size());
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ASSERT_EQ(2u, video_streams.size());
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EXPECT_EQ(kStreamId1, video_streams[0]->id());
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EXPECT_EQ(kStreamId2, video_streams[1]->id());
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}
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// Test that doing additional offer/answer exchanges with no changes to tracks
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// will cause no additional OnTrack calls after the tracks have been negotiated.
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TEST_F(PeerConnectionRtpCallbacksTest, UnifiedPlanReofferDoesNotCallOnTrack) {
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auto caller = CreatePeerConnectionWithUnifiedPlan();
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auto callee = CreatePeerConnectionWithUnifiedPlan();
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caller->AddAudioTrack("audio");
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callee->AddAudioTrack("audio");
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ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
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EXPECT_EQ(1u, caller->observer()->on_track_transceivers_.size());
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EXPECT_EQ(1u, callee->observer()->on_track_transceivers_.size());
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// If caller reoffers with no changes expect no additional OnTrack calls.
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ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
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EXPECT_EQ(1u, caller->observer()->on_track_transceivers_.size());
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EXPECT_EQ(1u, callee->observer()->on_track_transceivers_.size());
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// Also if callee reoffers with no changes expect no additional OnTrack calls.
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ASSERT_TRUE(callee->ExchangeOfferAnswerWith(caller.get()));
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EXPECT_EQ(1u, caller->observer()->on_track_transceivers_.size());
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EXPECT_EQ(1u, callee->observer()->on_track_transceivers_.size());
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}
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// Test that OnTrack is called when the transceiver direction changes to send
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// the track.
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TEST_F(PeerConnectionRtpCallbacksTest, UnifiedPlanSetDirectionCallsOnTrack) {
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auto caller = CreatePeerConnectionWithUnifiedPlan();
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auto callee = CreatePeerConnectionWithUnifiedPlan();
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auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
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transceiver->SetDirection(RtpTransceiverDirection::kInactive);
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ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
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EXPECT_EQ(0u, caller->observer()->on_track_transceivers_.size());
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EXPECT_EQ(0u, callee->observer()->on_track_transceivers_.size());
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transceiver->SetDirection(RtpTransceiverDirection::kSendOnly);
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ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
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EXPECT_EQ(0u, caller->observer()->on_track_transceivers_.size());
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EXPECT_EQ(1u, callee->observer()->on_track_transceivers_.size());
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// If the direction changes but it is still receiving on the remote side, then
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// OnTrack should not be fired again.
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transceiver->SetDirection(RtpTransceiverDirection::kSendRecv);
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ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
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EXPECT_EQ(0u, caller->observer()->on_track_transceivers_.size());
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EXPECT_EQ(1u, callee->observer()->on_track_transceivers_.size());
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}
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// Test that OnTrack is called twice when a sendrecv call is started, the callee
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// changes the direction to inactive, then changes it back to sendrecv.
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TEST_F(PeerConnectionRtpCallbacksTest,
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UnifiedPlanSetDirectionHoldCallsOnTrackTwice) {
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auto caller = CreatePeerConnectionWithUnifiedPlan();
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auto callee = CreatePeerConnectionWithUnifiedPlan();
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auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
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ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
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EXPECT_EQ(0u, caller->observer()->on_track_transceivers_.size());
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EXPECT_EQ(1u, callee->observer()->on_track_transceivers_.size());
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// Put the call on hold by no longer receiving the track.
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callee->pc()->GetTransceivers()[0]->SetDirection(
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RtpTransceiverDirection::kInactive);
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ASSERT_TRUE(callee->ExchangeOfferAnswerWith(caller.get()));
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EXPECT_EQ(0u, caller->observer()->on_track_transceivers_.size());
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EXPECT_EQ(1u, callee->observer()->on_track_transceivers_.size());
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// Resume the call by changing the direction to recvonly. This should call
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// OnTrack again on the callee side.
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callee->pc()->GetTransceivers()[0]->SetDirection(
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RtpTransceiverDirection::kRecvOnly);
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ASSERT_TRUE(callee->ExchangeOfferAnswerWith(caller.get()));
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EXPECT_EQ(0u, caller->observer()->on_track_transceivers_.size());
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EXPECT_EQ(2u, callee->observer()->on_track_transceivers_.size());
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}
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// These tests examine the state of the peer connection as a result of
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// performing SetRemoteDescription().
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class PeerConnectionRtpObserverTest : public PeerConnectionRtpTest {};
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TEST_F(PeerConnectionRtpObserverTest, AddSenderWithoutStreamAddsReceiver) {
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auto caller = CreatePeerConnection();
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auto callee = CreatePeerConnection();
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ASSERT_TRUE(caller->AddTrack(caller->CreateAudioTrack("audio_track"), {}));
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ASSERT_TRUE(
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callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
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static_cast<webrtc::RTCError*>(nullptr)));
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EXPECT_EQ(callee->pc()->GetReceivers().size(), 1u);
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auto receiver_added = callee->pc()->GetReceivers()[0];
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EXPECT_EQ("audio_track", receiver_added->track()->id());
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// Since we are not supporting the no stream case with Plan B, there should be
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// a generated stream, even though we didn't set one with AddTrack.
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EXPECT_EQ(receiver_added->streams().size(), 1u);
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EXPECT_TRUE(receiver_added->streams()[0]->FindAudioTrack("audio_track"));
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}
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TEST_F(PeerConnectionRtpObserverTest, AddSenderWithStreamAddsReceiver) {
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auto caller = CreatePeerConnection();
|
|
auto callee = CreatePeerConnection();
|
|
|
|
ASSERT_TRUE(caller->AddTrack(caller->CreateAudioTrack("audio_track"),
|
|
{"audio_stream"}));
|
|
ASSERT_TRUE(
|
|
callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
|
|
static_cast<webrtc::RTCError*>(nullptr)));
|
|
|
|
EXPECT_EQ(callee->pc()->GetReceivers().size(), 1u);
|
|
auto receiver_added = callee->pc()->GetReceivers()[0];
|
|
EXPECT_EQ("audio_track", receiver_added->track()->id());
|
|
EXPECT_EQ(receiver_added->streams().size(), 1u);
|
|
EXPECT_EQ("audio_stream", receiver_added->streams()[0]->id());
|
|
EXPECT_TRUE(receiver_added->streams()[0]->FindAudioTrack("audio_track"));
|
|
}
|
|
|
|
TEST_F(PeerConnectionRtpObserverTest,
|
|
RemoveSenderWithoutStreamRemovesReceiver) {
|
|
auto caller = CreatePeerConnection();
|
|
auto callee = CreatePeerConnection();
|
|
|
|
auto sender = caller->AddTrack(caller->CreateAudioTrack("audio_track"), {});
|
|
ASSERT_TRUE(sender);
|
|
ASSERT_TRUE(
|
|
callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
|
|
static_cast<webrtc::RTCError*>(nullptr)));
|
|
ASSERT_EQ(callee->pc()->GetReceivers().size(), 1u);
|
|
auto receiver = callee->pc()->GetReceivers()[0];
|
|
ASSERT_TRUE(caller->pc()->RemoveTrack(sender));
|
|
ASSERT_TRUE(
|
|
callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
|
|
static_cast<webrtc::RTCError*>(nullptr)));
|
|
|
|
// TODO(hbos): When we implement Unified Plan, receivers will not be removed.
|
|
// Instead, the transceiver owning the receiver will become inactive.
|
|
EXPECT_EQ(callee->pc()->GetReceivers().size(), 0u);
|
|
}
|
|
|
|
TEST_F(PeerConnectionRtpObserverTest, RemoveSenderWithStreamRemovesReceiver) {
|
|
auto caller = CreatePeerConnection();
|
|
auto callee = CreatePeerConnection();
|
|
|
|
auto sender = caller->AddTrack(caller->CreateAudioTrack("audio_track"),
|
|
{"audio_stream"});
|
|
ASSERT_TRUE(sender);
|
|
ASSERT_TRUE(
|
|
callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
|
|
static_cast<webrtc::RTCError*>(nullptr)));
|
|
ASSERT_EQ(callee->pc()->GetReceivers().size(), 1u);
|
|
auto receiver = callee->pc()->GetReceivers()[0];
|
|
ASSERT_TRUE(caller->pc()->RemoveTrack(sender));
|
|
ASSERT_TRUE(
|
|
callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
|
|
static_cast<webrtc::RTCError*>(nullptr)));
|
|
|
|
// TODO(hbos): When we implement Unified Plan, receivers will not be removed.
|
|
// Instead, the transceiver owning the receiver will become inactive.
|
|
EXPECT_EQ(callee->pc()->GetReceivers().size(), 0u);
|
|
}
|
|
|
|
TEST_F(PeerConnectionRtpObserverTest,
|
|
RemoveSenderWithSharedStreamRemovesReceiver) {
|
|
auto caller = CreatePeerConnection();
|
|
auto callee = CreatePeerConnection();
|
|
|
|
const char kSharedStreamId[] = "shared_audio_stream";
|
|
auto sender1 = caller->AddTrack(caller->CreateAudioTrack("audio_track1"),
|
|
{kSharedStreamId});
|
|
auto sender2 = caller->AddTrack(caller->CreateAudioTrack("audio_track2"),
|
|
{kSharedStreamId});
|
|
ASSERT_TRUE(
|
|
callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
|
|
static_cast<webrtc::RTCError*>(nullptr)));
|
|
|
|
ASSERT_EQ(callee->pc()->GetReceivers().size(), 2u);
|
|
rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver1;
|
|
rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver2;
|
|
if (callee->pc()->GetReceivers()[0]->track()->id() == "audio_track1") {
|
|
receiver1 = callee->pc()->GetReceivers()[0];
|
|
receiver2 = callee->pc()->GetReceivers()[1];
|
|
} else {
|
|
receiver1 = callee->pc()->GetReceivers()[1];
|
|
receiver2 = callee->pc()->GetReceivers()[0];
|
|
}
|
|
EXPECT_EQ("audio_track1", receiver1->track()->id());
|
|
EXPECT_EQ("audio_track2", receiver2->track()->id());
|
|
|
|
// Remove "audio_track1".
|
|
EXPECT_TRUE(caller->pc()->RemoveTrack(sender1));
|
|
ASSERT_TRUE(
|
|
callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
|
|
static_cast<webrtc::RTCError*>(nullptr)));
|
|
// Only |receiver2| should remain.
|
|
// TODO(hbos): When we implement Unified Plan, receivers will not be removed.
|
|
// Instead, the transceiver owning the receiver will become inactive.
|
|
EXPECT_EQ(
|
|
std::vector<rtc::scoped_refptr<webrtc::RtpReceiverInterface>>{receiver2},
|
|
callee->pc()->GetReceivers());
|
|
|
|
// Remove "audio_track2".
|
|
EXPECT_TRUE(caller->pc()->RemoveTrack(sender2));
|
|
ASSERT_TRUE(
|
|
callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
|
|
static_cast<webrtc::RTCError*>(nullptr)));
|
|
// TODO(hbos): When we implement Unified Plan, receivers will not be removed.
|
|
// Instead, the transceiver owning the receiver will become inactive.
|
|
EXPECT_EQ(callee->pc()->GetReceivers().size(), 0u);
|
|
}
|
|
|
|
// Invokes SetRemoteDescription() twice in a row without synchronizing the two
|
|
// calls and examine the state of the peer connection inside the callbacks to
|
|
// ensure that the second call does not occur prematurely, contaminating the
|
|
// state of the peer connection of the first callback.
|
|
TEST_F(PeerConnectionRtpObserverTest,
|
|
StatesCorrelateWithSetRemoteDescriptionCall) {
|
|
auto caller = CreatePeerConnection();
|
|
auto callee = CreatePeerConnection();
|
|
|
|
// Create SDP for adding a track and for removing it. This will be used in the
|
|
// first and second SetRemoteDescription() calls.
|
|
auto sender = caller->AddTrack(caller->CreateAudioTrack("audio_track"), {});
|
|
auto srd1_sdp = caller->CreateOfferAndSetAsLocal();
|
|
EXPECT_TRUE(caller->pc()->RemoveTrack(sender));
|
|
auto srd2_sdp = caller->CreateOfferAndSetAsLocal();
|
|
|
|
// In the first SetRemoteDescription() callback, check that we have a
|
|
// receiver for the track.
|
|
auto pc = callee->pc();
|
|
bool srd1_callback_called = false;
|
|
auto srd1_callback = [&srd1_callback_called, &pc]() {
|
|
EXPECT_EQ(pc->GetReceivers().size(), 1u);
|
|
srd1_callback_called = true;
|
|
};
|
|
|
|
// In the second SetRemoteDescription() callback, check that the receiver has
|
|
// been removed.
|
|
// TODO(hbos): When we implement Unified Plan, receivers will not be removed.
|
|
// Instead, the transceiver owning the receiver will become inactive.
|
|
// https://crbug.com/webrtc/7600
|
|
bool srd2_callback_called = false;
|
|
auto srd2_callback = [&srd2_callback_called, &pc]() {
|
|
EXPECT_TRUE(pc->GetReceivers().empty());
|
|
srd2_callback_called = true;
|
|
};
|
|
|
|
// Invoke SetRemoteDescription() twice in a row without synchronizing the two
|
|
// calls. The callbacks verify that the two calls are synchronized, as in, the
|
|
// effects of the second SetRemoteDescription() call must not have happened by
|
|
// the time the first callback is invoked. If it has then the receiver that is
|
|
// added as a result of the first SetRemoteDescription() call will already
|
|
// have been removed as a result of the second SetRemoteDescription() call
|
|
// when the first callback is invoked.
|
|
callee->pc()->SetRemoteDescription(
|
|
std::move(srd1_sdp),
|
|
new OnSuccessObserver<decltype(srd1_callback)>(srd1_callback));
|
|
callee->pc()->SetRemoteDescription(
|
|
std::move(srd2_sdp),
|
|
new OnSuccessObserver<decltype(srd2_callback)>(srd2_callback));
|
|
EXPECT_TRUE_WAIT(srd1_callback_called, kDefaultTimeout);
|
|
EXPECT_TRUE_WAIT(srd2_callback_called, kDefaultTimeout);
|
|
}
|
|
|
|
// Tests that with Unified Plan if the the stream id changes for a track when
|
|
// when setting a new remote description, that the media stream is updated
|
|
// appropriately for the receiver.
|
|
TEST_F(PeerConnectionRtpObserverTest, RemoteStreamIdChangesUpdatesReceiver) {
|
|
auto caller = CreatePeerConnectionWithUnifiedPlan();
|
|
auto callee = CreatePeerConnectionWithUnifiedPlan();
|
|
|
|
const char kStreamId1[] = "stream1";
|
|
const char kStreamId2[] = "stream2";
|
|
caller->AddTrack(caller->CreateAudioTrack("audio_track1"), {kStreamId1});
|
|
ASSERT_TRUE(
|
|
callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
|
|
static_cast<webrtc::RTCError*>(nullptr)));
|
|
EXPECT_EQ(callee->observer()->add_track_events_.size(), 1u);
|
|
|
|
// Change the stream id of the sender in the session description.
|
|
auto offer = caller->CreateOfferAndSetAsLocal();
|
|
auto contents = offer->description()->contents();
|
|
ASSERT_EQ(contents.size(), 1u);
|
|
ASSERT_EQ(contents[0].media_description()->mutable_streams().size(), 1u);
|
|
contents[0].media_description()->mutable_streams()[0].set_stream_ids(
|
|
{kStreamId2});
|
|
|
|
// Set the remote description and verify that the stream was updated properly.
|
|
ASSERT_TRUE(
|
|
callee->SetRemoteDescription(CloneSessionDescription(offer.get()),
|
|
static_cast<webrtc::RTCError*>(nullptr)));
|
|
auto receivers = callee->pc()->GetReceivers();
|
|
ASSERT_EQ(receivers.size(), 1u);
|
|
ASSERT_EQ(receivers[0]->streams().size(), 1u);
|
|
EXPECT_EQ(receivers[0]->streams()[0]->id(), kStreamId2);
|
|
}
|
|
|
|
// This tests a regression caught by a downstream client, that occured when
|
|
// applying a remote description with a SessionDescription object that
|
|
// contained StreamParams that didn't have ids. Although there were multiple
|
|
// remote audio senders, FindSenderInfo didn't find them as unique, because
|
|
// it looked up by StreamParam.id, which none had. This meant only one
|
|
// AudioRtpReceiver was created, as opposed to one for each remote sender.
|
|
TEST_F(PeerConnectionRtpObserverTest,
|
|
MultipleRemoteSendersWithoutStreamParamIdAddsMultipleReceivers) {
|
|
auto caller = CreatePeerConnection();
|
|
auto callee = CreatePeerConnection();
|
|
|
|
const char kStreamId1[] = "stream1";
|
|
const char kStreamId2[] = "stream2";
|
|
caller->AddAudioTrack("audio_track1", {kStreamId1});
|
|
caller->AddAudioTrack("audio_track2", {kStreamId2});
|
|
|
|
auto offer = caller->CreateOfferAndSetAsLocal();
|
|
auto mutable_streams =
|
|
cricket::GetFirstAudioContentDescription(offer->description())
|
|
->mutable_streams();
|
|
ASSERT_EQ(mutable_streams.size(), 2u);
|
|
// Clear the IDs in the StreamParams.
|
|
mutable_streams[0].id.clear();
|
|
mutable_streams[1].id.clear();
|
|
ASSERT_TRUE(
|
|
callee->SetRemoteDescription(CloneSessionDescription(offer.get())));
|
|
|
|
auto receivers = callee->pc()->GetReceivers();
|
|
ASSERT_EQ(receivers.size(), 2u);
|
|
ASSERT_EQ(receivers[0]->streams().size(), 1u);
|
|
EXPECT_EQ(kStreamId1, receivers[0]->streams()[0]->id());
|
|
ASSERT_EQ(receivers[1]->streams().size(), 1u);
|
|
EXPECT_EQ(kStreamId2, receivers[1]->streams()[0]->id());
|
|
}
|
|
|
|
// Tests for the legacy SetRemoteDescription() function signature.
|
|
class PeerConnectionRtpLegacyObserverTest : public PeerConnectionRtpTest {};
|
|
|
|
// Sanity test making sure the callback is invoked.
|
|
TEST_F(PeerConnectionRtpLegacyObserverTest, OnSuccess) {
|
|
auto caller = CreatePeerConnection();
|
|
auto callee = CreatePeerConnection();
|
|
|
|
std::string error;
|
|
ASSERT_TRUE(
|
|
callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(), &error));
|
|
}
|
|
|
|
// Verifies legacy behavior: The observer is not called if if the peer
|
|
// connection is destroyed because the asynchronous callback is executed in the
|
|
// peer connection's message handler.
|
|
TEST_F(PeerConnectionRtpLegacyObserverTest,
|
|
ObserverNotCalledIfPeerConnectionDereferenced) {
|
|
auto caller = CreatePeerConnection();
|
|
auto callee = CreatePeerConnection();
|
|
|
|
rtc::scoped_refptr<webrtc::MockSetSessionDescriptionObserver> observer =
|
|
new rtc::RefCountedObject<webrtc::MockSetSessionDescriptionObserver>();
|
|
|
|
auto offer = caller->CreateOfferAndSetAsLocal();
|
|
callee->pc()->SetRemoteDescription(observer, offer.release());
|
|
callee = nullptr;
|
|
rtc::Thread::Current()->ProcessMessages(0);
|
|
EXPECT_FALSE(observer->called());
|
|
}
|
|
|
|
// RtpTransceiver Tests.
|
|
|
|
// Test that by default there are no transceivers with Unified Plan.
|
|
TEST_F(PeerConnectionRtpTest, PeerConnectionHasNoTransceivers) {
|
|
auto caller = CreatePeerConnectionWithUnifiedPlan();
|
|
EXPECT_THAT(caller->pc()->GetTransceivers(), ElementsAre());
|
|
}
|
|
|
|
// Test that a transceiver created with the audio kind has the correct initial
|
|
// properties.
|
|
TEST_F(PeerConnectionRtpTest, AddTransceiverHasCorrectInitProperties) {
|
|
auto caller = CreatePeerConnectionWithUnifiedPlan();
|
|
|
|
auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
|
|
EXPECT_EQ(rtc::nullopt, transceiver->mid());
|
|
EXPECT_FALSE(transceiver->stopped());
|
|
EXPECT_EQ(RtpTransceiverDirection::kSendRecv, transceiver->direction());
|
|
EXPECT_EQ(rtc::nullopt, transceiver->current_direction());
|
|
}
|
|
|
|
// Test that adding a transceiver with the audio kind creates an audio sender
|
|
// and audio receiver with the receiver having a live audio track.
|
|
TEST_F(PeerConnectionRtpTest,
|
|
AddAudioTransceiverCreatesAudioSenderAndReceiver) {
|
|
auto caller = CreatePeerConnectionWithUnifiedPlan();
|
|
|
|
auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
|
|
EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, transceiver->media_type());
|
|
|
|
ASSERT_TRUE(transceiver->sender());
|
|
EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, transceiver->sender()->media_type());
|
|
|
|
ASSERT_TRUE(transceiver->receiver());
|
|
EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, transceiver->receiver()->media_type());
|
|
|
|
auto track = transceiver->receiver()->track();
|
|
ASSERT_TRUE(track);
|
|
EXPECT_EQ(MediaStreamTrackInterface::kAudioKind, track->kind());
|
|
EXPECT_EQ(MediaStreamTrackInterface::TrackState::kLive, track->state());
|
|
}
|
|
|
|
// Test that adding a transceiver with the video kind creates an video sender
|
|
// and video receiver with the receiver having a live video track.
|
|
TEST_F(PeerConnectionRtpTest,
|
|
AddAudioTransceiverCreatesVideoSenderAndReceiver) {
|
|
auto caller = CreatePeerConnectionWithUnifiedPlan();
|
|
|
|
auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
|
|
EXPECT_EQ(cricket::MEDIA_TYPE_VIDEO, transceiver->media_type());
|
|
|
|
ASSERT_TRUE(transceiver->sender());
|
|
EXPECT_EQ(cricket::MEDIA_TYPE_VIDEO, transceiver->sender()->media_type());
|
|
|
|
ASSERT_TRUE(transceiver->receiver());
|
|
EXPECT_EQ(cricket::MEDIA_TYPE_VIDEO, transceiver->receiver()->media_type());
|
|
|
|
auto track = transceiver->receiver()->track();
|
|
ASSERT_TRUE(track);
|
|
EXPECT_EQ(MediaStreamTrackInterface::kVideoKind, track->kind());
|
|
EXPECT_EQ(MediaStreamTrackInterface::TrackState::kLive, track->state());
|
|
}
|
|
|
|
// Test that after a call to AddTransceiver, the transceiver shows in
|
|
// GetTransceivers(), the transceiver's sender shows in GetSenders(), and the
|
|
// transceiver's receiver shows in GetReceivers().
|
|
TEST_F(PeerConnectionRtpTest, AddTransceiverShowsInLists) {
|
|
auto caller = CreatePeerConnectionWithUnifiedPlan();
|
|
|
|
auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
|
|
EXPECT_EQ(
|
|
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>{transceiver},
|
|
caller->pc()->GetTransceivers());
|
|
EXPECT_EQ(
|
|
std::vector<rtc::scoped_refptr<RtpSenderInterface>>{
|
|
transceiver->sender()},
|
|
caller->pc()->GetSenders());
|
|
EXPECT_EQ(
|
|
std::vector<rtc::scoped_refptr<RtpReceiverInterface>>{
|
|
transceiver->receiver()},
|
|
caller->pc()->GetReceivers());
|
|
}
|
|
|
|
// Test that the direction passed in through the AddTransceiver init parameter
|
|
// is set in the returned transceiver.
|
|
TEST_F(PeerConnectionRtpTest, AddTransceiverWithDirectionIsReflected) {
|
|
auto caller = CreatePeerConnectionWithUnifiedPlan();
|
|
|
|
RtpTransceiverInit init;
|
|
init.direction = RtpTransceiverDirection::kSendOnly;
|
|
auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO, init);
|
|
EXPECT_EQ(RtpTransceiverDirection::kSendOnly, transceiver->direction());
|
|
}
|
|
|
|
// Test that calling AddTransceiver with a track creates a transceiver which has
|
|
// its sender's track set to the passed-in track.
|
|
TEST_F(PeerConnectionRtpTest, AddTransceiverWithTrackCreatesSenderWithTrack) {
|
|
auto caller = CreatePeerConnectionWithUnifiedPlan();
|
|
|
|
auto audio_track = caller->CreateAudioTrack("audio track");
|
|
auto transceiver = caller->AddTransceiver(audio_track);
|
|
|
|
auto sender = transceiver->sender();
|
|
ASSERT_TRUE(sender->track());
|
|
EXPECT_EQ(audio_track, sender->track());
|
|
|
|
auto receiver = transceiver->receiver();
|
|
ASSERT_TRUE(receiver->track());
|
|
EXPECT_EQ(MediaStreamTrackInterface::kAudioKind, receiver->track()->kind());
|
|
EXPECT_EQ(MediaStreamTrackInterface::TrackState::kLive,
|
|
receiver->track()->state());
|
|
}
|
|
|
|
// Test that calling AddTransceiver twice with the same track creates distinct
|
|
// transceivers, senders with the same track.
|
|
TEST_F(PeerConnectionRtpTest,
|
|
AddTransceiverTwiceWithSameTrackCreatesMultipleTransceivers) {
|
|
auto caller = CreatePeerConnectionWithUnifiedPlan();
|
|
|
|
auto audio_track = caller->CreateAudioTrack("audio track");
|
|
|
|
auto transceiver1 = caller->AddTransceiver(audio_track);
|
|
auto transceiver2 = caller->AddTransceiver(audio_track);
|
|
|
|
EXPECT_NE(transceiver1, transceiver2);
|
|
|
|
auto sender1 = transceiver1->sender();
|
|
auto sender2 = transceiver2->sender();
|
|
EXPECT_NE(sender1, sender2);
|
|
EXPECT_EQ(audio_track, sender1->track());
|
|
EXPECT_EQ(audio_track, sender2->track());
|
|
|
|
EXPECT_THAT(caller->pc()->GetTransceivers(),
|
|
UnorderedElementsAre(transceiver1, transceiver2));
|
|
EXPECT_THAT(caller->pc()->GetSenders(),
|
|
UnorderedElementsAre(sender1, sender2));
|
|
}
|
|
|
|
// RtpTransceiver error handling tests.
|
|
|
|
TEST_F(PeerConnectionRtpTest, AddTransceiverWithInvalidKindReturnsError) {
|
|
auto caller = CreatePeerConnectionWithUnifiedPlan();
|
|
|
|
auto result = caller->pc()->AddTransceiver(cricket::MEDIA_TYPE_DATA);
|
|
EXPECT_EQ(RTCErrorType::INVALID_PARAMETER, result.error().type());
|
|
}
|
|
|
|
TEST_F(PeerConnectionRtpTest, UnifiedPlanCanClosePeerConnection) {
|
|
auto caller = CreatePeerConnectionWithUnifiedPlan();
|
|
|
|
caller->pc()->Close();
|
|
}
|
|
|
|
// Unified Plan AddTrack tests.
|
|
|
|
class PeerConnectionRtpUnifiedPlanTest : public PeerConnectionRtpTest {};
|
|
|
|
// Test that adding an audio track creates a new audio RtpSender with the given
|
|
// track.
|
|
TEST_F(PeerConnectionRtpUnifiedPlanTest, AddAudioTrackCreatesAudioSender) {
|
|
auto caller = CreatePeerConnectionWithUnifiedPlan();
|
|
|
|
auto audio_track = caller->CreateAudioTrack("a");
|
|
auto sender = caller->AddTrack(audio_track);
|
|
ASSERT_TRUE(sender);
|
|
|
|
EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, sender->media_type());
|
|
EXPECT_EQ(audio_track, sender->track());
|
|
}
|
|
|
|
// Test that adding a video track creates a new video RtpSender with the given
|
|
// track.
|
|
TEST_F(PeerConnectionRtpUnifiedPlanTest, AddVideoTrackCreatesVideoSender) {
|
|
auto caller = CreatePeerConnectionWithUnifiedPlan();
|
|
|
|
auto video_track = caller->CreateVideoTrack("a");
|
|
auto sender = caller->AddTrack(video_track);
|
|
ASSERT_TRUE(sender);
|
|
|
|
EXPECT_EQ(cricket::MEDIA_TYPE_VIDEO, sender->media_type());
|
|
EXPECT_EQ(video_track, sender->track());
|
|
}
|
|
|
|
// Test that adding a track to a new PeerConnection creates an RtpTransceiver
|
|
// with the sender that AddTrack returns and in the sendrecv direction.
|
|
TEST_F(PeerConnectionRtpUnifiedPlanTest, AddFirstTrackCreatesTransceiver) {
|
|
auto caller = CreatePeerConnectionWithUnifiedPlan();
|
|
|
|
auto sender = caller->AddAudioTrack("a");
|
|
ASSERT_TRUE(sender);
|
|
|
|
auto transceivers = caller->pc()->GetTransceivers();
|
|
ASSERT_EQ(1u, transceivers.size());
|
|
EXPECT_EQ(sender, transceivers[0]->sender());
|
|
EXPECT_EQ(RtpTransceiverDirection::kSendRecv, transceivers[0]->direction());
|
|
}
|
|
|
|
// Test that if a transceiver of the same type but no track had been added to
|
|
// the PeerConnection and later a call to AddTrack is made, the resulting sender
|
|
// is the transceiver's sender and the sender's track is the newly-added track.
|
|
TEST_F(PeerConnectionRtpUnifiedPlanTest, AddTrackReusesTransceiver) {
|
|
auto caller = CreatePeerConnectionWithUnifiedPlan();
|
|
|
|
auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
|
|
auto audio_track = caller->CreateAudioTrack("a");
|
|
auto sender = caller->AddTrack(audio_track);
|
|
ASSERT_TRUE(sender);
|
|
|
|
auto transceivers = caller->pc()->GetTransceivers();
|
|
ASSERT_EQ(1u, transceivers.size());
|
|
EXPECT_EQ(transceiver, transceivers[0]);
|
|
EXPECT_EQ(sender, transceiver->sender());
|
|
EXPECT_EQ(audio_track, sender->track());
|
|
}
|
|
|
|
// Test that adding two tracks to a new PeerConnection creates two
|
|
// RtpTransceivers in the same order.
|
|
TEST_F(PeerConnectionRtpUnifiedPlanTest, TwoAddTrackCreatesTwoTransceivers) {
|
|
auto caller = CreatePeerConnectionWithUnifiedPlan();
|
|
|
|
auto sender1 = caller->AddAudioTrack("a");
|
|
auto sender2 = caller->AddVideoTrack("v");
|
|
ASSERT_TRUE(sender2);
|
|
|
|
auto transceivers = caller->pc()->GetTransceivers();
|
|
ASSERT_EQ(2u, transceivers.size());
|
|
EXPECT_EQ(sender1, transceivers[0]->sender());
|
|
EXPECT_EQ(sender2, transceivers[1]->sender());
|
|
}
|
|
|
|
// Test that if there are multiple transceivers with no sending track then a
|
|
// later call to AddTrack will use the one of the same type as the newly-added
|
|
// track.
|
|
TEST_F(PeerConnectionRtpUnifiedPlanTest, AddTrackReusesTransceiverOfType) {
|
|
auto caller = CreatePeerConnectionWithUnifiedPlan();
|
|
|
|
auto audio_transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
|
|
auto video_transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
|
|
auto sender = caller->AddVideoTrack("v");
|
|
|
|
ASSERT_EQ(2u, caller->pc()->GetTransceivers().size());
|
|
EXPECT_NE(sender, audio_transceiver->sender());
|
|
EXPECT_EQ(sender, video_transceiver->sender());
|
|
}
|
|
|
|
// Test that if the only transceivers that do not have a sending track have a
|
|
// different type from the added track, then AddTrack will create a new
|
|
// transceiver for the track.
|
|
TEST_F(PeerConnectionRtpUnifiedPlanTest,
|
|
AddTrackDoesNotReuseTransceiverOfWrongType) {
|
|
auto caller = CreatePeerConnectionWithUnifiedPlan();
|
|
|
|
caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
|
|
auto sender = caller->AddVideoTrack("v");
|
|
|
|
auto transceivers = caller->pc()->GetTransceivers();
|
|
ASSERT_EQ(2u, transceivers.size());
|
|
EXPECT_NE(sender, transceivers[0]->sender());
|
|
EXPECT_EQ(sender, transceivers[1]->sender());
|
|
}
|
|
|
|
// Test that the first available transceiver is reused by AddTrack when multiple
|
|
// are available.
|
|
TEST_F(PeerConnectionRtpUnifiedPlanTest,
|
|
AddTrackReusesFirstMatchingTransceiver) {
|
|
auto caller = CreatePeerConnectionWithUnifiedPlan();
|
|
|
|
caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
|
|
caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
|
|
auto sender = caller->AddAudioTrack("a");
|
|
|
|
auto transceivers = caller->pc()->GetTransceivers();
|
|
ASSERT_EQ(2u, transceivers.size());
|
|
EXPECT_EQ(sender, transceivers[0]->sender());
|
|
EXPECT_NE(sender, transceivers[1]->sender());
|
|
}
|
|
|
|
// Test that a call to AddTrack that reuses a transceiver will change the
|
|
// direction from inactive to sendonly.
|
|
TEST_F(PeerConnectionRtpUnifiedPlanTest,
|
|
AddTrackChangesDirectionFromInactiveToSendOnly) {
|
|
auto caller = CreatePeerConnectionWithUnifiedPlan();
|
|
|
|
RtpTransceiverInit init;
|
|
init.direction = RtpTransceiverDirection::kInactive;
|
|
auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO, init);
|
|
|
|
caller->observer()->clear_negotiation_needed();
|
|
ASSERT_TRUE(caller->AddAudioTrack("a"));
|
|
EXPECT_TRUE(caller->observer()->negotiation_needed());
|
|
|
|
EXPECT_EQ(RtpTransceiverDirection::kSendOnly, transceiver->direction());
|
|
}
|
|
|
|
// Test that a call to AddTrack that reuses a transceiver will change the
|
|
// direction from recvonly to sendrecv.
|
|
TEST_F(PeerConnectionRtpUnifiedPlanTest,
|
|
AddTrackChangesDirectionFromRecvOnlyToSendRecv) {
|
|
auto caller = CreatePeerConnectionWithUnifiedPlan();
|
|
|
|
RtpTransceiverInit init;
|
|
init.direction = RtpTransceiverDirection::kRecvOnly;
|
|
auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO, init);
|
|
|
|
caller->observer()->clear_negotiation_needed();
|
|
ASSERT_TRUE(caller->AddAudioTrack("a"));
|
|
EXPECT_TRUE(caller->observer()->negotiation_needed());
|
|
|
|
EXPECT_EQ(RtpTransceiverDirection::kSendRecv, transceiver->direction());
|
|
}
|
|
|
|
TEST_F(PeerConnectionRtpUnifiedPlanTest, AddTrackCreatesSenderWithTrackId) {
|
|
const std::string kTrackId = "audio_track";
|
|
|
|
auto caller = CreatePeerConnectionWithUnifiedPlan();
|
|
|
|
auto audio_track = caller->CreateAudioTrack(kTrackId);
|
|
auto sender = caller->AddTrack(audio_track);
|
|
|
|
EXPECT_EQ(kTrackId, sender->id());
|
|
}
|
|
|
|
// Unified Plan AddTrack error handling.
|
|
|
|
TEST_F(PeerConnectionRtpUnifiedPlanTest, AddTrackErrorIfClosed) {
|
|
auto caller = CreatePeerConnectionWithUnifiedPlan();
|
|
|
|
auto audio_track = caller->CreateAudioTrack("a");
|
|
caller->pc()->Close();
|
|
|
|
caller->observer()->clear_negotiation_needed();
|
|
auto result = caller->pc()
|
|
->AddTrack(audio_track, std::vector<std::string>());
|
|
EXPECT_EQ(RTCErrorType::INVALID_STATE, result.error().type());
|
|
EXPECT_FALSE(caller->observer()->negotiation_needed());
|
|
}
|
|
|
|
TEST_F(PeerConnectionRtpUnifiedPlanTest, AddTrackErrorIfTrackAlreadyHasSender) {
|
|
auto caller = CreatePeerConnectionWithUnifiedPlan();
|
|
|
|
auto audio_track = caller->CreateAudioTrack("a");
|
|
ASSERT_TRUE(caller->AddTrack(audio_track));
|
|
|
|
caller->observer()->clear_negotiation_needed();
|
|
auto result = caller->pc()
|
|
->AddTrack(audio_track, std::vector<std::string>());
|
|
EXPECT_EQ(RTCErrorType::INVALID_PARAMETER, result.error().type());
|
|
EXPECT_FALSE(caller->observer()->negotiation_needed());
|
|
}
|
|
|
|
// Unified Plan RemoveTrack tests.
|
|
|
|
// Test that calling RemoveTrack on a sender with a previously-added track
|
|
// clears the sender's track.
|
|
TEST_F(PeerConnectionRtpUnifiedPlanTest, RemoveTrackClearsSenderTrack) {
|
|
auto caller = CreatePeerConnectionWithUnifiedPlan();
|
|
|
|
auto sender = caller->AddAudioTrack("a");
|
|
ASSERT_TRUE(caller->pc()->RemoveTrack(sender));
|
|
|
|
EXPECT_FALSE(sender->track());
|
|
}
|
|
|
|
// Test that calling RemoveTrack on a sender where the transceiver is configured
|
|
// in the sendrecv direction changes the transceiver's direction to recvonly.
|
|
TEST_F(PeerConnectionRtpUnifiedPlanTest,
|
|
RemoveTrackChangesDirectionFromSendRecvToRecvOnly) {
|
|
auto caller = CreatePeerConnectionWithUnifiedPlan();
|
|
|
|
RtpTransceiverInit init;
|
|
init.direction = RtpTransceiverDirection::kSendRecv;
|
|
auto transceiver =
|
|
caller->AddTransceiver(caller->CreateAudioTrack("a"), init);
|
|
|
|
caller->observer()->clear_negotiation_needed();
|
|
ASSERT_TRUE(caller->pc()->RemoveTrack(transceiver->sender()));
|
|
EXPECT_TRUE(caller->observer()->negotiation_needed());
|
|
|
|
EXPECT_EQ(RtpTransceiverDirection::kRecvOnly, transceiver->direction());
|
|
EXPECT_TRUE(caller->observer()->renegotiation_needed_);
|
|
}
|
|
|
|
// Test that calling RemoveTrack on a sender where the transceiver is configured
|
|
// in the sendonly direction changes the transceiver's direction to inactive.
|
|
TEST_F(PeerConnectionRtpUnifiedPlanTest,
|
|
RemoveTrackChangesDirectionFromSendOnlyToInactive) {
|
|
auto caller = CreatePeerConnectionWithUnifiedPlan();
|
|
|
|
RtpTransceiverInit init;
|
|
init.direction = RtpTransceiverDirection::kSendOnly;
|
|
auto transceiver =
|
|
caller->AddTransceiver(caller->CreateAudioTrack("a"), init);
|
|
|
|
caller->observer()->clear_negotiation_needed();
|
|
ASSERT_TRUE(caller->pc()->RemoveTrack(transceiver->sender()));
|
|
EXPECT_TRUE(caller->observer()->negotiation_needed());
|
|
|
|
EXPECT_EQ(RtpTransceiverDirection::kInactive, transceiver->direction());
|
|
}
|
|
|
|
// Test that calling RemoveTrack with a sender that has a null track results in
|
|
// no change in state.
|
|
TEST_F(PeerConnectionRtpUnifiedPlanTest, RemoveTrackWithNullSenderTrackIsNoOp) {
|
|
auto caller = CreatePeerConnectionWithUnifiedPlan();
|
|
|
|
auto sender = caller->AddAudioTrack("a");
|
|
auto transceiver = caller->pc()->GetTransceivers()[0];
|
|
ASSERT_TRUE(sender->SetTrack(nullptr));
|
|
|
|
caller->observer()->clear_negotiation_needed();
|
|
ASSERT_TRUE(caller->pc()->RemoveTrack(sender));
|
|
EXPECT_FALSE(caller->observer()->negotiation_needed());
|
|
|
|
EXPECT_EQ(RtpTransceiverDirection::kSendRecv, transceiver->direction());
|
|
}
|
|
|
|
// Unified Plan RemoveTrack error handling.
|
|
|
|
TEST_F(PeerConnectionRtpUnifiedPlanTest, RemoveTrackErrorIfClosed) {
|
|
auto caller = CreatePeerConnectionWithUnifiedPlan();
|
|
|
|
auto sender = caller->AddAudioTrack("a");
|
|
caller->pc()->Close();
|
|
|
|
caller->observer()->clear_negotiation_needed();
|
|
EXPECT_FALSE(caller->pc()->RemoveTrack(sender));
|
|
EXPECT_FALSE(caller->observer()->negotiation_needed());
|
|
}
|
|
|
|
TEST_F(PeerConnectionRtpUnifiedPlanTest,
|
|
RemoveTrackNoErrorIfTrackAlreadyRemoved) {
|
|
auto caller = CreatePeerConnectionWithUnifiedPlan();
|
|
|
|
auto sender = caller->AddAudioTrack("a");
|
|
ASSERT_TRUE(caller->pc()->RemoveTrack(sender));
|
|
|
|
caller->observer()->clear_negotiation_needed();
|
|
EXPECT_TRUE(caller->pc()->RemoveTrack(sender));
|
|
EXPECT_FALSE(caller->observer()->negotiation_needed());
|
|
}
|
|
|
|
// Test that OnRenegotiationNeeded is fired if SetDirection is called on an
|
|
// active RtpTransceiver with a new direction.
|
|
TEST_F(PeerConnectionRtpUnifiedPlanTest,
|
|
RenegotiationNeededAfterTransceiverSetDirection) {
|
|
auto caller = CreatePeerConnectionWithUnifiedPlan();
|
|
|
|
auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
|
|
|
|
caller->observer()->clear_negotiation_needed();
|
|
transceiver->SetDirection(RtpTransceiverDirection::kInactive);
|
|
EXPECT_TRUE(caller->observer()->negotiation_needed());
|
|
}
|
|
|
|
// Test that OnRenegotiationNeeded is not fired if SetDirection is called on an
|
|
// active RtpTransceiver with current direction.
|
|
TEST_F(PeerConnectionRtpUnifiedPlanTest,
|
|
NoRenegotiationNeededAfterTransceiverSetSameDirection) {
|
|
auto caller = CreatePeerConnectionWithUnifiedPlan();
|
|
|
|
auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
|
|
|
|
caller->observer()->clear_negotiation_needed();
|
|
transceiver->SetDirection(transceiver->direction());
|
|
EXPECT_FALSE(caller->observer()->negotiation_needed());
|
|
}
|
|
|
|
// Test that OnRenegotiationNeeded is not fired if SetDirection is called on a
|
|
// stopped RtpTransceiver.
|
|
TEST_F(PeerConnectionRtpUnifiedPlanTest,
|
|
NoRenegotiationNeededAfterSetDirectionOnStoppedTransceiver) {
|
|
auto caller = CreatePeerConnectionWithUnifiedPlan();
|
|
|
|
auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
|
|
transceiver->Stop();
|
|
|
|
caller->observer()->clear_negotiation_needed();
|
|
transceiver->SetDirection(RtpTransceiverDirection::kInactive);
|
|
EXPECT_FALSE(caller->observer()->negotiation_needed());
|
|
}
|
|
|
|
// Test MSID signaling between Unified Plan and Plan B endpoints. There are two
|
|
// options for this kind of signaling: media section based (a=msid) and ssrc
|
|
// based (a=ssrc MSID). While JSEP only specifies media section MSID signaling,
|
|
// we want to ensure compatibility with older Plan B endpoints that might expect
|
|
// ssrc based MSID signaling. Thus we test here that Unified Plan offers both
|
|
// types but answers with the same type as the offer.
|
|
|
|
class PeerConnectionMsidSignalingTest : public PeerConnectionRtpTest {};
|
|
|
|
TEST_F(PeerConnectionMsidSignalingTest, UnifiedPlanTalkingToOurself) {
|
|
auto caller = CreatePeerConnectionWithUnifiedPlan();
|
|
caller->AddAudioTrack("caller_audio");
|
|
auto callee = CreatePeerConnectionWithUnifiedPlan();
|
|
callee->AddAudioTrack("callee_audio");
|
|
auto caller_observer = caller->RegisterFakeMetricsObserver();
|
|
|
|
ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
|
|
|
|
// Offer should have had both a=msid and a=ssrc MSID lines.
|
|
auto* offer = callee->pc()->remote_description();
|
|
EXPECT_EQ((cricket::kMsidSignalingMediaSection |
|
|
cricket::kMsidSignalingSsrcAttribute),
|
|
offer->description()->msid_signaling());
|
|
|
|
// Answer should have had only a=msid lines.
|
|
auto* answer = caller->pc()->remote_description();
|
|
EXPECT_EQ(cricket::kMsidSignalingMediaSection,
|
|
answer->description()->msid_signaling());
|
|
// Check that this is counted correctly
|
|
EXPECT_TRUE(caller_observer->ExpectOnlySingleEnumCount(
|
|
kEnumCounterSdpSemanticNegotiated, kSdpSemanticNegotiatedUnifiedPlan));
|
|
}
|
|
|
|
TEST_F(PeerConnectionMsidSignalingTest, PlanBOfferToUnifiedPlanAnswer) {
|
|
auto caller = CreatePeerConnectionWithPlanB();
|
|
caller->AddAudioTrack("caller_audio");
|
|
auto callee = CreatePeerConnectionWithUnifiedPlan();
|
|
callee->AddAudioTrack("callee_audio");
|
|
|
|
ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
|
|
|
|
// Offer should have only a=ssrc MSID lines.
|
|
auto* offer = callee->pc()->remote_description();
|
|
EXPECT_EQ(cricket::kMsidSignalingSsrcAttribute,
|
|
offer->description()->msid_signaling());
|
|
|
|
// Answer should have only a=ssrc MSID lines to match the offer.
|
|
auto* answer = caller->pc()->remote_description();
|
|
EXPECT_EQ(cricket::kMsidSignalingSsrcAttribute,
|
|
answer->description()->msid_signaling());
|
|
}
|
|
|
|
// This tests that a Plan B endpoint appropriately sets the remote description
|
|
// from a Unified Plan offer. When the Unified Plan offer contains a=msid lines
|
|
// that signal no stream ids or multiple stream ids we expect that the Plan B
|
|
// endpoint always has exactly one media stream per track.
|
|
TEST_F(PeerConnectionMsidSignalingTest, UnifiedPlanToPlanBAnswer) {
|
|
const std::string kStreamId1 = "audio_stream_1";
|
|
const std::string kStreamId2 = "audio_stream_2";
|
|
|
|
auto caller = CreatePeerConnectionWithUnifiedPlan();
|
|
caller->AddAudioTrack("caller_audio", {kStreamId1, kStreamId2});
|
|
caller->AddVideoTrack("caller_video", {});
|
|
auto callee = CreatePeerConnectionWithPlanB();
|
|
callee->AddAudioTrack("callee_audio");
|
|
caller->AddVideoTrack("callee_video");
|
|
|
|
ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
|
|
|
|
// Offer should have had both a=msid and a=ssrc MSID lines.
|
|
auto* offer = callee->pc()->remote_description();
|
|
EXPECT_EQ((cricket::kMsidSignalingMediaSection |
|
|
cricket::kMsidSignalingSsrcAttribute),
|
|
offer->description()->msid_signaling());
|
|
|
|
// Callee should always have 1 stream for all of it's receivers.
|
|
const auto& track_events = callee->observer()->add_track_events_;
|
|
ASSERT_EQ(2u, track_events.size());
|
|
ASSERT_EQ(1u, track_events[0].streams.size());
|
|
EXPECT_EQ(kStreamId1, track_events[0].streams[0]->id());
|
|
ASSERT_EQ(1u, track_events[1].streams.size());
|
|
// This autogenerated a stream id for the empty one signalled.
|
|
EXPECT_FALSE(track_events[1].streams[0]->id().empty());
|
|
}
|
|
|
|
TEST_F(PeerConnectionMsidSignalingTest, PureUnifiedPlanToUs) {
|
|
auto caller = CreatePeerConnectionWithUnifiedPlan();
|
|
caller->AddAudioTrack("caller_audio");
|
|
auto callee = CreatePeerConnectionWithUnifiedPlan();
|
|
callee->AddAudioTrack("callee_audio");
|
|
|
|
auto offer = caller->CreateOffer();
|
|
// Simulate a pure Unified Plan offerer by setting the MSID signaling to media
|
|
// section only.
|
|
offer->description()->set_msid_signaling(cricket::kMsidSignalingMediaSection);
|
|
|
|
ASSERT_TRUE(
|
|
caller->SetLocalDescription(CloneSessionDescription(offer.get())));
|
|
ASSERT_TRUE(callee->SetRemoteDescription(std::move(offer)));
|
|
|
|
// Answer should have only a=msid to match the offer.
|
|
auto answer = callee->CreateAnswer();
|
|
EXPECT_EQ(cricket::kMsidSignalingMediaSection,
|
|
answer->description()->msid_signaling());
|
|
}
|
|
|
|
// Test that the correct UMA metrics are reported for simple/complex SDP.
|
|
|
|
class SdpFormatReceivedTest : public PeerConnectionRtpTest {};
|
|
|
|
#ifdef HAVE_SCTP
|
|
TEST_F(SdpFormatReceivedTest, DataChannelOnlyIsReportedAsNoTracks) {
|
|
auto caller = CreatePeerConnectionWithUnifiedPlan();
|
|
caller->CreateDataChannel("dc");
|
|
auto callee = CreatePeerConnectionWithUnifiedPlan();
|
|
auto callee_metrics = callee->RegisterFakeMetricsObserver();
|
|
|
|
ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOffer()));
|
|
|
|
EXPECT_TRUE(callee_metrics->ExpectOnlySingleEnumCount(
|
|
kEnumCounterSdpFormatReceived, kSdpFormatReceivedNoTracks));
|
|
}
|
|
#endif // HAVE_SCTP
|
|
|
|
TEST_F(SdpFormatReceivedTest, SimpleUnifiedPlanIsReportedAsSimple) {
|
|
auto caller = CreatePeerConnectionWithUnifiedPlan();
|
|
caller->AddAudioTrack("audio");
|
|
caller->AddVideoTrack("video");
|
|
auto callee = CreatePeerConnectionWithPlanB();
|
|
auto callee_metrics = callee->RegisterFakeMetricsObserver();
|
|
|
|
ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOffer()));
|
|
|
|
EXPECT_TRUE(callee_metrics->ExpectOnlySingleEnumCount(
|
|
kEnumCounterSdpFormatReceived, kSdpFormatReceivedSimple));
|
|
}
|
|
|
|
TEST_F(SdpFormatReceivedTest, SimplePlanBIsReportedAsSimple) {
|
|
auto caller = CreatePeerConnectionWithPlanB();
|
|
caller->AddVideoTrack("video"); // Video only.
|
|
auto callee = CreatePeerConnectionWithUnifiedPlan();
|
|
auto callee_metrics = callee->RegisterFakeMetricsObserver();
|
|
|
|
ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOffer()));
|
|
|
|
EXPECT_TRUE(callee_metrics->ExpectOnlySingleEnumCount(
|
|
kEnumCounterSdpFormatReceived, kSdpFormatReceivedSimple));
|
|
}
|
|
|
|
TEST_F(SdpFormatReceivedTest, ComplexUnifiedIsReportedAsComplexUnifiedPlan) {
|
|
auto caller = CreatePeerConnectionWithUnifiedPlan();
|
|
caller->AddAudioTrack("audio1");
|
|
caller->AddAudioTrack("audio2");
|
|
caller->AddVideoTrack("video");
|
|
auto callee = CreatePeerConnectionWithPlanB();
|
|
auto callee_metrics = callee->RegisterFakeMetricsObserver();
|
|
|
|
ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOffer()));
|
|
|
|
EXPECT_TRUE(callee_metrics->ExpectOnlySingleEnumCount(
|
|
kEnumCounterSdpFormatReceived, kSdpFormatReceivedComplexUnifiedPlan));
|
|
}
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|
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TEST_F(SdpFormatReceivedTest, ComplexPlanBIsReportedAsComplexPlanB) {
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auto caller = CreatePeerConnectionWithPlanB();
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|
caller->AddVideoTrack("video1");
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caller->AddVideoTrack("video2");
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|
auto callee = CreatePeerConnectionWithUnifiedPlan();
|
|
auto callee_metrics = callee->RegisterFakeMetricsObserver();
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|
|
|
ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOffer()));
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|
|
|
EXPECT_TRUE(callee_metrics->ExpectOnlySingleEnumCount(
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|
kEnumCounterSdpFormatReceived, kSdpFormatReceivedComplexPlanB));
|
|
}
|
|
|
|
// Sender setups in a call.
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|
|
|
class PeerConnectionSenderTest : public PeerConnectionRtpTest {};
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|
|
|
TEST_F(PeerConnectionSenderTest, CreateTwoSendersWithSameTrack) {
|
|
auto caller = CreatePeerConnection();
|
|
auto callee = CreatePeerConnection();
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|
|
|
auto track = caller->CreateAudioTrack("audio_track");
|
|
auto sender1 = caller->AddTrack(track);
|
|
ASSERT_TRUE(sender1);
|
|
// We need to temporarily reset the track for the subsequent AddTrack() to
|
|
// succeed.
|
|
EXPECT_TRUE(sender1->SetTrack(nullptr));
|
|
auto sender2 = caller->AddTrack(track);
|
|
EXPECT_TRUE(sender2);
|
|
EXPECT_TRUE(sender1->SetTrack(track));
|
|
|
|
// TODO(hbos): When https://crbug.com/webrtc/8734 is resolved, this should
|
|
// return true, and doing |callee->SetRemoteDescription()| should work.
|
|
EXPECT_FALSE(caller->CreateOfferAndSetAsLocal());
|
|
}
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|
|
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} // namespace webrtc
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