Add RtpSenderInterface.SetStreams
This is a reland of df5731e44d510e9f23a35b77e9e102eb41919bf4 with fixes
to avoid existing chromium tests to fail.
Instead of replacing the existing RtpSender::set_stream_ids() to
also fire OnRenegotiationNeeded(), this CL keeps the old
set_stream_ids() and adds the new RtpSender::SetStreams() which sets
the stream IDs and fires the callback.
This allows existing callsites to maintain behavior, and reserve
SetStreams() for the cases when we want OnRenegotiationNeeded() to fire.
Using the SetStreams() name instead of SetStreamIDs() to match the W3C
spec and to make it more different that the existing set_stream_ids().
Original change's description:
> Improve spec compliance of SetStreamIDs in RtpSenderInterface
>
> This CL makes RtpSender::SetStreamIDs fire fire negotiationneeded
> event if needed and exposes the method on RtpSenderInterface.
>
> This is a spec-compliance change.
>
> Bug: webrtc:10129
> Change-Id: I2b98b92665c847102838b094516a79b24de0e47e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135121
> Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27974}
Bug: webrtc:10129
Change-Id: Ic0b322bfa25c157e3a39465ef8b486f898eaf6bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137439
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27992}
diff --git a/api/rtp_sender_interface.h b/api/rtp_sender_interface.h
index e0a29a3..e6140db 100644
--- a/api/rtp_sender_interface.h
+++ b/api/rtp_sender_interface.h
@@ -61,6 +61,11 @@
// tracks.
virtual std::vector<std::string> stream_ids() const = 0;
+ // Sets the IDs of the media streams associated with this sender's track.
+ // These are signalled in the SDP so that the remote side can associate
+ // tracks.
+ virtual void SetStreams(const std::vector<std::string>& stream_ids) {}
+
// Returns the list of encoding parameters that will be applied when the SDP
// local description is set. These initial encoding parameters can be set by
// PeerConnection::AddTransceiver, and later updated with Get/SetParameters.
@@ -112,6 +117,7 @@
rtc::scoped_refptr<FrameEncryptorInterface>)
PROXY_CONSTMETHOD0(rtc::scoped_refptr<FrameEncryptorInterface>,
GetFrameEncryptor)
+PROXY_METHOD1(void, SetStreams, const std::vector<std::string>&)
END_PROXY_MAP()
} // namespace webrtc
diff --git a/pc/peer_connection.cc b/pc/peer_connection.cc
index 065247d..9902d84 100644
--- a/pc/peer_connection.cc
+++ b/pc/peer_connection.cc
@@ -1632,14 +1632,14 @@
(track->kind() == MediaStreamTrackInterface::kAudioKind));
sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
signaling_thread(),
- AudioRtpSender::Create(worker_thread(), id, stats_.get()));
+ AudioRtpSender::Create(worker_thread(), id, stats_.get(), this));
NoteUsageEvent(UsageEvent::AUDIO_ADDED);
} else {
RTC_DCHECK_EQ(media_type, cricket::MEDIA_TYPE_VIDEO);
RTC_DCHECK(!track ||
(track->kind() == MediaStreamTrackInterface::kVideoKind));
sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
- signaling_thread(), VideoRtpSender::Create(worker_thread(), id));
+ signaling_thread(), VideoRtpSender::Create(worker_thread(), id, this));
NoteUsageEvent(UsageEvent::VIDEO_ADDED);
}
bool set_track_succeeded = sender->SetTrack(track);
@@ -1722,14 +1722,14 @@
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> new_sender;
if (kind == MediaStreamTrackInterface::kAudioKind) {
auto audio_sender = AudioRtpSender::Create(
- worker_thread(), rtc::CreateRandomUuid(), stats_.get());
+ worker_thread(), rtc::CreateRandomUuid(), stats_.get(), this);
audio_sender->SetMediaChannel(voice_media_channel());
new_sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
signaling_thread(), audio_sender);
GetAudioTransceiver()->internal()->AddSender(new_sender);
} else if (kind == MediaStreamTrackInterface::kVideoKind) {
auto video_sender =
- VideoRtpSender::Create(worker_thread(), rtc::CreateRandomUuid());
+ VideoRtpSender::Create(worker_thread(), rtc::CreateRandomUuid(), this);
video_sender->SetMediaChannel(video_media_channel());
new_sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
signaling_thread(), video_sender);
@@ -7130,6 +7130,12 @@
return ret;
}
+void PeerConnection::OnSetStreams() {
+ RTC_DCHECK_RUN_ON(signaling_thread());
+ if (IsUnifiedPlan())
+ UpdateNegotiationNeeded();
+}
+
PeerConnectionObserver* PeerConnection::Observer() const {
// In earlier production code, the pointer was not cleared on close,
// which might have led to undefined behavior if the observer was not
diff --git a/pc/peer_connection.h b/pc/peer_connection.h
index 3009185..86fc1a8 100644
--- a/pc/peer_connection.h
+++ b/pc/peer_connection.h
@@ -25,6 +25,7 @@
#include "pc/peer_connection_factory.h"
#include "pc/peer_connection_internal.h"
#include "pc/rtc_stats_collector.h"
+#include "pc/rtp_sender.h"
#include "pc/rtp_transceiver.h"
#include "pc/sctp_transport.h"
#include "pc/stats_collector.h"
@@ -57,6 +58,7 @@
public DataChannelProviderInterface,
public DataChannelSink,
public JsepTransportController::Observer,
+ public RtpSenderBase::SetStreamsObserver,
public rtc::MessageHandler,
public sigslot::has_slots<> {
public:
@@ -1055,6 +1057,9 @@
rtc::scoped_refptr<DtlsTransport> dtls_transport,
MediaTransportInterface* media_transport) override;
+ // RtpSenderBase::SetStreamsObserver override.
+ void OnSetStreams() override;
+
// Returns the observer. Will crash on CHECK if the observer is removed.
PeerConnectionObserver* Observer() const RTC_RUN_ON(signaling_thread());
diff --git a/pc/peer_connection_rtp_unittest.cc b/pc/peer_connection_rtp_unittest.cc
index 6925300..6336f1f 100644
--- a/pc/peer_connection_rtp_unittest.cc
+++ b/pc/peer_connection_rtp_unittest.cc
@@ -491,15 +491,8 @@
ASSERT_TRUE(callee->CreateAnswerAndSetAsLocal());
// Change the stream ID in the offer.
- // TODO(https://crbug.com/webrtc/10129): When RtpSenderInterface::SetStreams
- // is supported, this can use that instead of munging the SDP.
- auto offer = caller->CreateOffer();
- auto contents = offer->description()->contents();
- ASSERT_EQ(1u, contents.size());
- auto& stream_params = contents[0].media_description()->mutable_streams();
- ASSERT_EQ(1u, stream_params.size());
- stream_params[0].set_stream_ids({"stream2"});
- ASSERT_TRUE(callee->SetRemoteDescription(std::move(offer)));
+ caller->pc()->GetSenders()[0]->SetStreams({"stream2"});
+ ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
ASSERT_EQ(1u, transceiver->receiver()->streams().size());
EXPECT_EQ("stream2", transceiver->receiver()->streams()[0]->id());
}
@@ -1797,12 +1790,7 @@
// This test exercises the code path that fires a NegotiationNeeded
// notification when the stream IDs of the local description differ from
-// the ones in the transceiver. Since SetStreams() is not yet available
-// on RtpSenderInterface, adding a track is used to trigger the check for
-// the NegotiationNeeded notification.
-// TODO(https://crbug.com/webrtc/10129): Replace this test with a test that
-// checks that calling SetStreams() on a sender fires the notification once
-// the method becomes available in RtpSenderInterface.
+// the ones in the transceiver.
TEST_F(PeerConnectionRtpTestUnifiedPlan,
ChangeAssociatedStreamsTriggersRenegotiation) {
auto caller = CreatePeerConnection();
@@ -1817,18 +1805,15 @@
ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
caller->observer()->clear_negotiation_needed();
- SessionDescriptionInterface* cld = const_cast<SessionDescriptionInterface*>(
- caller->pc()->current_local_description());
- ASSERT_EQ(cld->description()->contents().size(), 1u);
-
- cricket::SessionDescription* description = cld->description();
- cricket::ContentInfo& content_info = description->contents()[0];
- ASSERT_EQ(content_info.media_description()->mutable_streams().size(), 1u);
- content_info.media_description()->mutable_streams()[0].set_stream_ids(
- {"stream3", "stream4", "stream5"});
-
- ASSERT_TRUE(caller->AddTrack(caller->CreateAudioTrack("a2")));
+ transceiver->sender()->SetStreams({"stream3", "stream4", "stream5"});
EXPECT_TRUE(caller->observer()->negotiation_needed());
+
+ ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
+ auto callee_streams = callee->pc()->GetReceivers()[0]->streams();
+ ASSERT_EQ(3u, callee_streams.size());
+ EXPECT_EQ("stream3", callee_streams[0]->id());
+ EXPECT_EQ("stream4", callee_streams[1]->id());
+ EXPECT_EQ("stream5", callee_streams[2]->id());
}
INSTANTIATE_TEST_SUITE_P(PeerConnectionRtpTest,
diff --git a/pc/rtp_sender.cc b/pc/rtp_sender.cc
index 22957d3..6ef7f9f 100644
--- a/pc/rtp_sender.cc
+++ b/pc/rtp_sender.cc
@@ -16,6 +16,7 @@
#include "api/audio_options.h"
#include "api/media_stream_interface.h"
#include "media/base/media_engine.h"
+#include "pc/peer_connection.h"
#include "pc/stats_collector.h"
#include "rtc_base/checks.h"
#include "rtc_base/helpers.h"
@@ -118,8 +119,12 @@
return false;
}
-RtpSenderBase::RtpSenderBase(rtc::Thread* worker_thread, const std::string& id)
- : worker_thread_(worker_thread), id_(id) {
+RtpSenderBase::RtpSenderBase(rtc::Thread* worker_thread,
+ const std::string& id,
+ SetStreamsObserver* set_streams_observer)
+ : worker_thread_(worker_thread),
+ id_(id),
+ set_streams_observer_(set_streams_observer) {
RTC_DCHECK(worker_thread);
init_parameters_.encodings.emplace_back();
}
@@ -215,6 +220,12 @@
return result;
}
+void RtpSenderBase::SetStreams(const std::vector<std::string>& stream_ids) {
+ set_stream_ids(stream_ids);
+ if (set_streams_observer_)
+ set_streams_observer_->OnSetStreams();
+}
+
bool RtpSenderBase::SetTrack(MediaStreamTrackInterface* track) {
TRACE_EVENT0("webrtc", "RtpSenderBase::SetTrack");
if (stopped_) {
@@ -317,6 +328,7 @@
RemoveTrackFromStats();
}
media_channel_ = nullptr;
+ set_streams_observer_ = nullptr;
stopped_ = true;
}
@@ -395,15 +407,18 @@
rtc::scoped_refptr<AudioRtpSender> AudioRtpSender::Create(
rtc::Thread* worker_thread,
const std::string& id,
- StatsCollector* stats) {
+ StatsCollector* stats,
+ SetStreamsObserver* set_streams_observer) {
return rtc::scoped_refptr<AudioRtpSender>(
- new rtc::RefCountedObject<AudioRtpSender>(worker_thread, id, stats));
+ new rtc::RefCountedObject<AudioRtpSender>(worker_thread, id, stats,
+ set_streams_observer));
}
AudioRtpSender::AudioRtpSender(rtc::Thread* worker_thread,
const std::string& id,
- StatsCollector* stats)
- : RtpSenderBase(worker_thread, id),
+ StatsCollector* stats,
+ SetStreamsObserver* set_streams_observer)
+ : RtpSenderBase(worker_thread, id, set_streams_observer),
stats_(stats),
dtmf_sender_proxy_(DtmfSenderProxy::Create(
rtc::Thread::Current(),
@@ -539,14 +554,17 @@
rtc::scoped_refptr<VideoRtpSender> VideoRtpSender::Create(
rtc::Thread* worker_thread,
- const std::string& id) {
+ const std::string& id,
+ SetStreamsObserver* set_streams_observer) {
return rtc::scoped_refptr<VideoRtpSender>(
- new rtc::RefCountedObject<VideoRtpSender>(worker_thread, id));
+ new rtc::RefCountedObject<VideoRtpSender>(worker_thread, id,
+ set_streams_observer));
}
VideoRtpSender::VideoRtpSender(rtc::Thread* worker_thread,
- const std::string& id)
- : RtpSenderBase(worker_thread, id) {}
+ const std::string& id,
+ SetStreamsObserver* set_streams_observer)
+ : RtpSenderBase(worker_thread, id, set_streams_observer) {}
VideoRtpSender::~VideoRtpSender() {
Stop();
diff --git a/pc/rtp_sender.h b/pc/rtp_sender.h
index c786d90..82ef711 100644
--- a/pc/rtp_sender.h
+++ b/pc/rtp_sender.h
@@ -74,6 +74,12 @@
// Shared implementation for RtpSenderInternal interface.
class RtpSenderBase : public RtpSenderInternal, public ObserverInterface {
public:
+ class SetStreamsObserver {
+ public:
+ virtual ~SetStreamsObserver() = default;
+ virtual void OnSetStreams() = 0;
+ };
+
// Sets the underlying MediaEngine channel associated with this RtpSender.
// A VoiceMediaChannel should be used for audio RtpSenders and
// a VideoMediaChannel should be used for video RtpSenders.
@@ -104,6 +110,7 @@
void set_stream_ids(const std::vector<std::string>& stream_ids) override {
stream_ids_ = stream_ids;
}
+ void SetStreams(const std::vector<std::string>& stream_ids) override;
std::string id() const override { return id_; }
@@ -143,7 +150,12 @@
RTCError DisableEncodingLayers(const std::vector<std::string>& rid) override;
protected:
- RtpSenderBase(rtc::Thread* worker_thread, const std::string& id);
+ // If |set_streams_observer| is not null, it is invoked when SetStreams()
+ // is called. |set_streams_observer| is not owned by this object. If not
+ // null, it must be valid at least until this sender becomes stopped.
+ RtpSenderBase(rtc::Thread* worker_thread,
+ const std::string& id,
+ SetStreamsObserver* set_streams_observer);
// TODO(nisse): Since SSRC == 0 is technically valid, figure out
// some other way to test if we have a valid SSRC.
bool can_send_track() const { return track_ && ssrc_; }
@@ -183,6 +195,8 @@
// const method.
mutable absl::optional<std::string> last_transaction_id_;
std::vector<std::string> disabled_rids_;
+
+ SetStreamsObserver* set_streams_observer_ = nullptr;
};
// LocalAudioSinkAdapter receives data callback as a sink to the local
@@ -215,9 +229,14 @@
// The sender is initialized with no track to send and no associated streams.
// StatsCollector provided so that Add/RemoveLocalAudioTrack can be called
// at the appropriate times.
- static rtc::scoped_refptr<AudioRtpSender> Create(rtc::Thread* worker_thread,
- const std::string& id,
- StatsCollector* stats);
+ // If |set_streams_observer| is not null, it is invoked when SetStreams()
+ // is called. |set_streams_observer| is not owned by this object. If not
+ // null, it must be valid at least until this sender becomes stopped.
+ static rtc::scoped_refptr<AudioRtpSender> Create(
+ rtc::Thread* worker_thread,
+ const std::string& id,
+ StatsCollector* stats,
+ SetStreamsObserver* set_streams_observer);
virtual ~AudioRtpSender();
// DtmfSenderProvider implementation.
@@ -240,7 +259,8 @@
protected:
AudioRtpSender(rtc::Thread* worker_thread,
const std::string& id,
- StatsCollector* stats);
+ StatsCollector* stats,
+ SetStreamsObserver* set_streams_observer);
void SetSend() override;
void ClearSend() override;
@@ -274,8 +294,13 @@
public:
// Construct an RtpSender for video with the given sender ID.
// The sender is initialized with no track to send and no associated streams.
- static rtc::scoped_refptr<VideoRtpSender> Create(rtc::Thread* worker_thread,
- const std::string& id);
+ // If |set_streams_observer| is not null, it is invoked when SetStreams()
+ // is called. |set_streams_observer| is not owned by this object. If not
+ // null, it must be valid at least until this sender becomes stopped.
+ static rtc::scoped_refptr<VideoRtpSender> Create(
+ rtc::Thread* worker_thread,
+ const std::string& id,
+ SetStreamsObserver* set_streams_observer);
virtual ~VideoRtpSender();
// ObserverInterface implementation
@@ -291,7 +316,9 @@
rtc::scoped_refptr<DtmfSenderInterface> GetDtmfSender() const override;
protected:
- VideoRtpSender(rtc::Thread* worker_thread, const std::string& id);
+ VideoRtpSender(rtc::Thread* worker_thread,
+ const std::string& id,
+ SetStreamsObserver* set_streams_observer);
void SetSend() override;
void ClearSend() override;
diff --git a/pc/rtp_sender_receiver_unittest.cc b/pc/rtp_sender_receiver_unittest.cc
index 90ca03f..d602a1d 100644
--- a/pc/rtp_sender_receiver_unittest.cc
+++ b/pc/rtp_sender_receiver_unittest.cc
@@ -83,6 +83,13 @@
static const uint32_t kVideoSsrcSimulcast = 102;
static const uint32_t kVideoSimulcastLayerCount = 2;
static const int kDefaultTimeout = 10000; // 10 seconds.
+
+class MockSetStreamsObserver
+ : public webrtc::RtpSenderBase::SetStreamsObserver {
+ public:
+ MOCK_METHOD0(OnSetStreams, void());
+};
+
} // namespace
namespace webrtc {
@@ -187,10 +194,14 @@
const rtc::scoped_refptr<LocalAudioSource>& source) {
audio_track_ = AudioTrack::Create(kAudioTrackId, source);
EXPECT_TRUE(local_stream_->AddTrack(audio_track_));
+ std::unique_ptr<MockSetStreamsObserver> set_streams_observer =
+ absl::make_unique<MockSetStreamsObserver>();
audio_rtp_sender_ =
- AudioRtpSender::Create(worker_thread_, audio_track_->id(), nullptr);
+ AudioRtpSender::Create(worker_thread_, audio_track_->id(), nullptr,
+ set_streams_observer.get());
ASSERT_TRUE(audio_rtp_sender_->SetTrack(audio_track_));
- audio_rtp_sender_->set_stream_ids({local_stream_->id()});
+ EXPECT_CALL(*set_streams_observer, OnSetStreams());
+ audio_rtp_sender_->SetStreams({local_stream_->id()});
audio_rtp_sender_->SetMediaChannel(voice_media_channel_);
audio_rtp_sender_->SetSsrc(kAudioSsrc);
audio_rtp_sender_->GetOnDestroyedSignal()->connect(
@@ -200,7 +211,7 @@
void CreateAudioRtpSenderWithNoTrack() {
audio_rtp_sender_ =
- AudioRtpSender::Create(worker_thread_, /*id=*/"", nullptr);
+ AudioRtpSender::Create(worker_thread_, /*id=*/"", nullptr, nullptr);
audio_rtp_sender_->SetMediaChannel(voice_media_channel_);
}
@@ -248,16 +259,20 @@
void CreateVideoRtpSender(bool is_screencast, uint32_t ssrc = kVideoSsrc) {
AddVideoTrack(is_screencast);
- video_rtp_sender_ =
- VideoRtpSender::Create(worker_thread_, video_track_->id());
+ std::unique_ptr<MockSetStreamsObserver> set_streams_observer =
+ absl::make_unique<MockSetStreamsObserver>();
+ video_rtp_sender_ = VideoRtpSender::Create(
+ worker_thread_, video_track_->id(), set_streams_observer.get());
ASSERT_TRUE(video_rtp_sender_->SetTrack(video_track_));
- video_rtp_sender_->set_stream_ids({local_stream_->id()});
+ EXPECT_CALL(*set_streams_observer, OnSetStreams());
+ video_rtp_sender_->SetStreams({local_stream_->id()});
video_rtp_sender_->SetMediaChannel(video_media_channel_);
video_rtp_sender_->SetSsrc(ssrc);
VerifyVideoChannelInput(ssrc);
}
void CreateVideoRtpSenderWithNoTrack() {
- video_rtp_sender_ = VideoRtpSender::Create(worker_thread_, /*id=*/"");
+ video_rtp_sender_ =
+ VideoRtpSender::Create(worker_thread_, /*id=*/"", nullptr);
video_rtp_sender_->SetMediaChannel(video_media_channel_);
}
@@ -428,7 +443,7 @@
void RunDisableSimulcastLayersWithoutMediaEngineTest(
const std::vector<std::string>& all_layers,
const std::vector<std::string>& disabled_layers) {
- auto sender = VideoRtpSender::Create(rtc::Thread::Current(), "1");
+ auto sender = VideoRtpSender::Create(rtc::Thread::Current(), "1", nullptr);
RtpParameters parameters;
parameters.encodings.resize(all_layers.size());
for (size_t i = 0; i < all_layers.size(); ++i) {
@@ -820,7 +835,7 @@
TEST_F(RtpSenderReceiverTest, AudioSenderCanSetParametersBeforeNegotiation) {
audio_rtp_sender_ =
- AudioRtpSender::Create(worker_thread_, /*id=*/"", nullptr);
+ AudioRtpSender::Create(worker_thread_, /*id=*/"", nullptr, nullptr);
RtpParameters params = audio_rtp_sender_->GetParameters();
ASSERT_EQ(1u, params.encodings.size());
@@ -838,10 +853,13 @@
audio_track_ = AudioTrack::Create(kAudioTrackId, nullptr);
EXPECT_TRUE(local_stream_->AddTrack(audio_track_));
- audio_rtp_sender_ =
- AudioRtpSender::Create(worker_thread_, audio_track_->id(), nullptr);
+ std::unique_ptr<MockSetStreamsObserver> set_streams_observer =
+ absl::make_unique<MockSetStreamsObserver>();
+ audio_rtp_sender_ = AudioRtpSender::Create(
+ worker_thread_, audio_track_->id(), nullptr, set_streams_observer.get());
ASSERT_TRUE(audio_rtp_sender_->SetTrack(audio_track_));
- audio_rtp_sender_->set_stream_ids({local_stream_->id()});
+ EXPECT_CALL(*set_streams_observer, OnSetStreams());
+ audio_rtp_sender_->SetStreams({local_stream_->id()});
std::vector<RtpEncodingParameters> init_encodings(1);
init_encodings[0].max_bitrate_bps = 60000;
@@ -869,7 +887,7 @@
TEST_F(RtpSenderReceiverTest,
AudioSenderMustCallGetParametersBeforeSetParametersBeforeNegotiation) {
audio_rtp_sender_ =
- AudioRtpSender::Create(worker_thread_, /*id=*/"", nullptr);
+ AudioRtpSender::Create(worker_thread_, /*id=*/"", nullptr, nullptr);
RtpParameters params;
RTCError result = audio_rtp_sender_->SetParameters(params);
@@ -1048,7 +1066,8 @@
}
TEST_F(RtpSenderReceiverTest, VideoSenderCanSetParametersBeforeNegotiation) {
- video_rtp_sender_ = VideoRtpSender::Create(worker_thread_, /*id=*/"");
+ video_rtp_sender_ =
+ VideoRtpSender::Create(worker_thread_, /*id=*/"", nullptr);
RtpParameters params = video_rtp_sender_->GetParameters();
ASSERT_EQ(1u, params.encodings.size());
@@ -1065,10 +1084,13 @@
TEST_F(RtpSenderReceiverTest, VideoSenderInitParametersMovedAfterNegotiation) {
AddVideoTrack(false);
- video_rtp_sender_ =
- VideoRtpSender::Create(worker_thread_, video_track_->id());
+ std::unique_ptr<MockSetStreamsObserver> set_streams_observer =
+ absl::make_unique<MockSetStreamsObserver>();
+ video_rtp_sender_ = VideoRtpSender::Create(worker_thread_, video_track_->id(),
+ set_streams_observer.get());
ASSERT_TRUE(video_rtp_sender_->SetTrack(video_track_));
- video_rtp_sender_->set_stream_ids({local_stream_->id()});
+ EXPECT_CALL(*set_streams_observer, OnSetStreams());
+ video_rtp_sender_->SetStreams({local_stream_->id()});
std::vector<RtpEncodingParameters> init_encodings(2);
init_encodings[0].max_bitrate_bps = 60000;
@@ -1103,10 +1125,13 @@
VideoSenderInitParametersMovedAfterManualSimulcastAndNegotiation) {
AddVideoTrack(false);
- video_rtp_sender_ =
- VideoRtpSender::Create(worker_thread_, video_track_->id());
+ std::unique_ptr<MockSetStreamsObserver> set_streams_observer =
+ absl::make_unique<MockSetStreamsObserver>();
+ video_rtp_sender_ = VideoRtpSender::Create(worker_thread_, video_track_->id(),
+ set_streams_observer.get());
ASSERT_TRUE(video_rtp_sender_->SetTrack(video_track_));
- video_rtp_sender_->set_stream_ids({local_stream_->id()});
+ EXPECT_CALL(*set_streams_observer, OnSetStreams());
+ video_rtp_sender_->SetStreams({local_stream_->id()});
std::vector<RtpEncodingParameters> init_encodings(1);
init_encodings[0].max_bitrate_bps = 60000;
@@ -1137,7 +1162,8 @@
TEST_F(RtpSenderReceiverTest,
VideoSenderMustCallGetParametersBeforeSetParametersBeforeNegotiation) {
- video_rtp_sender_ = VideoRtpSender::Create(worker_thread_, /*id=*/"");
+ video_rtp_sender_ =
+ VideoRtpSender::Create(worker_thread_, /*id=*/"", nullptr);
RtpParameters params;
RTCError result = video_rtp_sender_->SetParameters(params);
@@ -1527,13 +1553,16 @@
TEST_F(RtpSenderReceiverTest,
PropagatesVideoTrackContentHintSetBeforeEnabling) {
AddVideoTrack();
+ std::unique_ptr<MockSetStreamsObserver> set_streams_observer =
+ absl::make_unique<MockSetStreamsObserver>();
// Setting detailed overrides the default non-screencast mode. This should be
// applied even if the track is set on construction.
video_track_->set_content_hint(VideoTrackInterface::ContentHint::kDetailed);
- video_rtp_sender_ =
- VideoRtpSender::Create(worker_thread_, video_track_->id());
+ video_rtp_sender_ = VideoRtpSender::Create(worker_thread_, video_track_->id(),
+ set_streams_observer.get());
ASSERT_TRUE(video_rtp_sender_->SetTrack(video_track_));
- video_rtp_sender_->set_stream_ids({local_stream_->id()});
+ EXPECT_CALL(*set_streams_observer, OnSetStreams());
+ video_rtp_sender_->SetStreams({local_stream_->id()});
video_rtp_sender_->SetMediaChannel(video_media_channel_);
video_track_->set_enabled(true);
diff --git a/pc/test/mock_rtp_sender_internal.h b/pc/test/mock_rtp_sender_internal.h
index fa16220..2cf0173 100644
--- a/pc/test/mock_rtp_sender_internal.h
+++ b/pc/test/mock_rtp_sender_internal.h
@@ -47,6 +47,7 @@
MOCK_METHOD1(SetMediaChannel, void(cricket::MediaChannel*));
MOCK_METHOD1(SetSsrc, void(uint32_t));
MOCK_METHOD1(set_stream_ids, void(const std::vector<std::string>&));
+ MOCK_METHOD1(SetStreams, void(const std::vector<std::string>&));
MOCK_METHOD1(set_init_send_encodings,
void(const std::vector<RtpEncodingParameters>&));
MOCK_METHOD0(Stop, void());
