Reland "Add AddTransceiver and GetTransceivers to PeerConnection"
This reverts commit 8b13f96e2d4b0449e54a3665121a4302ceb56e80.
Original change's description:
> Revert "Add AddTransceiver and GetTransceivers to PeerConnection"
>
> This reverts commit f93d2800d9b0d5818a5a383def0aaef3d441df3a.
>
> Reason for revert: https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2Fios-device%2F5804%2F%2B%2Frecipes%2Fsteps%2Fcompile%2F0%2Fstdout
>
> Original change's description:
> > Add AddTransceiver and GetTransceivers to PeerConnection
> >
> > WebRTC 1.0 has added the transceiver API to PeerConnection. This
> > is the first step towards exposing this to WebRTC consumers. For
> > now, transceivers can be added and fetched but there is not yet
> > support for creating offers/answers or setting local/remote
> > descriptions. That support ("Unified Plan") will be added in
> > follow-up CLs.
> >
> > The transceiver API is currently only available if the application
> > opts in by specifying the kUnifiedPlan SDP semantics when creating
> > the PeerConnection.
> >
> > Bug: webrtc:7600
> > Change-Id: I0b8ee24b489b45bb4c5f60b699bd20c61af01d8e
> > Reviewed-on: https://webrtc-review.googlesource.com/23880
> > Commit-Queue: Steve Anton <steveanton@webrtc.org>
> > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#20896}
>
> TBR=steveanton@webrtc.org,zhihuang@webrtc.org,hbos@webrtc.org,pthatcher@webrtc.org
>
> Change-Id: Ie91ea4988dba25c20e2532114d3a9d859a932d4c
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:7600
> Reviewed-on: https://webrtc-review.googlesource.com/26400
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20897}
TBR=steveanton@webrtc.org,zhihuang@webrtc.org,hbos@webrtc.org,pthatcher@webrtc.org
Change-Id: I19fdf08c54f09302794e998a0ffddb82ae0d7b41
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7600
Reviewed-on: https://webrtc-review.googlesource.com/26401
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20898}diff --git a/pc/peerconnection_rtp_unittest.cc b/pc/peerconnection_rtp_unittest.cc
index 0421124..e7ac8c7 100644
--- a/pc/peerconnection_rtp_unittest.cc
+++ b/pc/peerconnection_rtp_unittest.cc
@@ -27,11 +27,16 @@
#include "rtc_base/refcountedobject.h"
#include "rtc_base/scoped_ref_ptr.h"
#include "rtc_base/thread.h"
+#include "test/gmock.h"
// This file contains tests for RTP Media API-related behavior of
// |webrtc::PeerConnection|, see https://w3c.github.io/webrtc-pc/#rtp-media-api.
-namespace {
+namespace webrtc {
+
+using RTCConfiguration = PeerConnectionInterface::RTCConfiguration;
+using ::testing::ElementsAre;
+using ::testing::UnorderedElementsAre;
const uint32_t kDefaultTimeout = 10000u;
@@ -55,28 +60,37 @@
class PeerConnectionRtpTest : public testing::Test {
public:
PeerConnectionRtpTest()
- : pc_factory_(webrtc::CreatePeerConnectionFactory(
- rtc::Thread::Current(),
- rtc::Thread::Current(),
- rtc::Thread::Current(),
- FakeAudioCaptureModule::Create(),
- webrtc::CreateBuiltinAudioEncoderFactory(),
- webrtc::CreateBuiltinAudioDecoderFactory(),
- nullptr,
- nullptr)) {}
+ : pc_factory_(
+ CreatePeerConnectionFactory(rtc::Thread::Current(),
+ rtc::Thread::Current(),
+ rtc::Thread::Current(),
+ FakeAudioCaptureModule::Create(),
+ CreateBuiltinAudioEncoderFactory(),
+ CreateBuiltinAudioDecoderFactory(),
+ nullptr,
+ nullptr)) {}
- std::unique_ptr<webrtc::PeerConnectionWrapper> CreatePeerConnection() {
- webrtc::PeerConnectionInterface::RTCConfiguration config;
- auto observer = rtc::MakeUnique<webrtc::MockPeerConnectionObserver>();
+ std::unique_ptr<PeerConnectionWrapper> CreatePeerConnection() {
+ return CreatePeerConnection(RTCConfiguration());
+ }
+
+ std::unique_ptr<PeerConnectionWrapper> CreatePeerConnectionWithUnifiedPlan() {
+ RTCConfiguration config;
+ config.sdp_semantics = SdpSemantics::kUnifiedPlan;
+ return CreatePeerConnection(config);
+ }
+
+ std::unique_ptr<PeerConnectionWrapper> CreatePeerConnection(
+ const RTCConfiguration& config) {
+ auto observer = rtc::MakeUnique<MockPeerConnectionObserver>();
auto pc = pc_factory_->CreatePeerConnection(config, nullptr, nullptr,
observer.get());
- return std::unique_ptr<webrtc::PeerConnectionWrapper>(
- new webrtc::PeerConnectionWrapper(pc_factory_, pc,
- std::move(observer)));
+ return rtc::MakeUnique<PeerConnectionWrapper>(pc_factory_, pc,
+ std::move(observer));
}
protected:
- rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pc_factory_;
+ rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory_;
};
// These tests cover |webrtc::PeerConnectionObserver| callbacks firing upon
@@ -87,7 +101,7 @@
auto caller = CreatePeerConnection();
auto callee = CreatePeerConnection();
- rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
+ rtc::scoped_refptr<AudioTrackInterface> audio_track(
pc_factory_->CreateAudioTrack("audio_track", nullptr));
EXPECT_TRUE(caller->pc()->AddTrack(audio_track.get(), {}));
ASSERT_TRUE(
@@ -107,9 +121,9 @@
auto caller = CreatePeerConnection();
auto callee = CreatePeerConnection();
- rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
+ rtc::scoped_refptr<AudioTrackInterface> audio_track(
pc_factory_->CreateAudioTrack("audio_track", nullptr));
- auto stream = webrtc::MediaStream::Create("audio_stream");
+ auto stream = MediaStream::Create("audio_stream");
EXPECT_TRUE(caller->pc()->AddTrack(audio_track.get(), {stream.get()}));
ASSERT_TRUE(
callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
@@ -128,7 +142,7 @@
auto caller = CreatePeerConnection();
auto callee = CreatePeerConnection();
- rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
+ rtc::scoped_refptr<AudioTrackInterface> audio_track(
pc_factory_->CreateAudioTrack("audio_track", nullptr));
auto sender = caller->pc()->AddTrack(audio_track.get(), {});
ASSERT_TRUE(
@@ -150,9 +164,9 @@
auto caller = CreatePeerConnection();
auto callee = CreatePeerConnection();
- rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
+ rtc::scoped_refptr<AudioTrackInterface> audio_track(
pc_factory_->CreateAudioTrack("audio_track", nullptr));
- auto stream = webrtc::MediaStream::Create("audio_stream");
+ auto stream = MediaStream::Create("audio_stream");
auto sender = caller->pc()->AddTrack(audio_track.get(), {stream.get()});
ASSERT_TRUE(
callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
@@ -173,12 +187,12 @@
auto caller = CreatePeerConnection();
auto callee = CreatePeerConnection();
- rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track1(
+ rtc::scoped_refptr<AudioTrackInterface> audio_track1(
pc_factory_->CreateAudioTrack("audio_track1", nullptr));
- rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track2(
+ rtc::scoped_refptr<AudioTrackInterface> audio_track2(
pc_factory_->CreateAudioTrack("audio_track2", nullptr));
- auto stream = webrtc::MediaStream::Create("shared_audio_stream");
- std::vector<webrtc::MediaStreamInterface*> streams{stream.get()};
+ auto stream = MediaStream::Create("shared_audio_stream");
+ std::vector<MediaStreamInterface*> streams{stream.get()};
auto sender1 = caller->pc()->AddTrack(audio_track1.get(), streams);
auto sender2 = caller->pc()->AddTrack(audio_track2.get(), streams);
ASSERT_TRUE(
@@ -194,7 +208,7 @@
static_cast<webrtc::RTCError*>(nullptr)));
ASSERT_EQ(callee->observer()->add_track_events_.size(), 2u);
EXPECT_EQ(
- std::vector<rtc::scoped_refptr<webrtc::RtpReceiverInterface>>{
+ std::vector<rtc::scoped_refptr<RtpReceiverInterface>>{
callee->observer()->add_track_events_[0].receiver},
callee->observer()->remove_track_events_);
@@ -438,4 +452,146 @@
EXPECT_FALSE(observer->called());
}
-} // namespace
+// 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);
+
+ 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);
+
+ 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_F(PeerConnectionRtpTest, AddTransceiverWithInvalidKindReturnsError) {
+ auto caller = CreatePeerConnectionWithUnifiedPlan();
+
+ auto result = caller->pc()->AddTransceiver(cricket::MEDIA_TYPE_DATA);
+ EXPECT_EQ(RTCErrorType::INVALID_PARAMETER, result.error().type());
+}
+
+// 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));
+}
+
+} // namespace webrtc
