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Reland "Don't create channel_manager++ when media_engine is not set"

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This reverts commit c6c02efb56b24df04ed9ab61252c14c7bddcca93.

Reason for revert: Test now passes (and channel manager is gone)

Original change's description:
> Revert "Don't create channel_manager when media_engine is not set"
>
> This reverts commit c48ad732d6eb69f14dd6d44f801d62997cef2c2f.
>
> Reason for revert: breaks downstream project
>
> Original change's description:
> > Don't create channel_manager when media_engine is not set
> >
> > Also remove a bunch of functions in ChannelManager that were just
> > forwarding to MediaEngineInterface.
> >
> > Bug: webrtc:13931
> > Change-Id: Ia38591fd22c665cace16d032f5c1e384e413cded
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261304
> > Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#36801}
>
> Bug: webrtc:13931
> Change-Id: I1e260a2489547bd9483b50e043c28d2805b0fa5a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261660
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Owners-Override: Artem Titov <titovartem@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Cr-Commit-Position: refs/heads/main@{#36811}

Bug: webrtc:13931
Change-Id: I7b5b45b46095c18d489b6a9fe4c625971d6b3da6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261661
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36976}
This commit is contained in:
Harald Alvestrand 2022-05-23 14:57:47 +00:00 committed by WebRTC LUCI CQ
parent 2377226851
commit 8101e7b79b
10 changed files with 411 additions and 295 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

1
pc/BUILD.gn
View File

@ -329,6 +329,7 @@ rtc_source_set("media_session") {
"../rtc_base:checks",
"../rtc_base:logging",
"../rtc_base:stringutils",
"../rtc_base/memory:always_valid_pointer",
"../rtc_base/third_party/base64",
]
absl_deps = [

73
pc/data_channel_integrationtest.cc
View File

@ -58,10 +58,24 @@ namespace {
class DataChannelIntegrationTest
: public PeerConnectionIntegrationBaseTest,
public ::testing::WithParamInterface<SdpSemantics> {
public ::testing::WithParamInterface<std::tuple<SdpSemantics, bool>> {
protected:
DataChannelIntegrationTest()
: PeerConnectionIntegrationBaseTest(GetParam()) {}
: PeerConnectionIntegrationBaseTest(std::get<0>(GetParam())),
allow_media_(std::get<1>(GetParam())) {}
bool allow_media() { return allow_media_; }
bool CreatePeerConnectionWrappers() {
if (allow_media_) {
return PeerConnectionIntegrationBaseTest::CreatePeerConnectionWrappers();
}
return PeerConnectionIntegrationBaseTest::
CreatePeerConnectionWrappersWithoutMediaEngine();
}
private:
// True if media is allowed to be added
const bool allow_media_;
};
// Fake clock must be set before threads are started to prevent race on
@ -173,14 +187,18 @@ TEST_P(DataChannelIntegrationTest, EndToEndCallWithSctpDataChannel) {
// Expect that data channel created on caller side will show up for callee as
// well.
caller()->CreateDataChannel();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
if (allow_media()) {
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
}
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Ensure the existence of the SCTP data channel didn't impede audio/video.
MediaExpectations media_expectations;
media_expectations.ExpectBidirectionalAudioAndVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
if (allow_media()) {
// Ensure the existence of the SCTP data channel didn't impede audio/video.
MediaExpectations media_expectations;
media_expectations.ExpectBidirectionalAudioAndVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
}
// Caller data channel should already exist (it created one). Callee data
// channel may not exist yet, since negotiation happens in-band, not in SDP.
ASSERT_NE(nullptr, caller()->data_channel());
@ -202,7 +220,7 @@ TEST_P(DataChannelIntegrationTest, EndToEndCallWithSctpDataChannel) {
// data channel only, and sends messages of various sizes.
TEST_P(DataChannelIntegrationTest,
EndToEndCallWithSctpDataChannelVariousSizes) {
ASSERT_TRUE(CreatePeerConnectionWrappersWithoutMediaEngine());
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Expect that data channel created on caller side will show up for callee as
// well.
@ -241,7 +259,7 @@ TEST_P(DataChannelIntegrationTest,
// data channel only, and sends empty messages
TEST_P(DataChannelIntegrationTest,
EndToEndCallWithSctpDataChannelEmptyMessages) {
ASSERT_TRUE(CreatePeerConnectionWrappersWithoutMediaEngine());
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Expect that data channel created on caller side will show up for callee as
// well.
@ -291,7 +309,7 @@ TEST_P(DataChannelIntegrationTest,
// this test does not use TURN.
const size_t kLowestSafePayloadSizeLimit = 1225;
ASSERT_TRUE(CreatePeerConnectionWrappersWithoutMediaEngine());
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Expect that data channel created on caller side will show up for callee as
// well.
@ -328,7 +346,7 @@ TEST_P(DataChannelIntegrationTest, EndToEndCallWithSctpDataChannelHarmfulMtu) {
// The size of the smallest message that fails to be delivered.
const size_t kMessageSizeThatIsNotDelivered = 1157;
ASSERT_TRUE(CreatePeerConnectionWrappersWithoutMediaEngine());
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->CreateDataChannel();
caller()->CreateAndSetAndSignalOffer();
@ -369,8 +387,10 @@ TEST_P(DataChannelIntegrationTest, CalleeClosesSctpDataChannel) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->CreateDataChannel();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
if (allow_media()) {
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
}
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
ASSERT_NE(nullptr, caller()->data_channel());
@ -406,8 +426,10 @@ TEST_P(DataChannelIntegrationTest, SctpDataChannelConfigSentToOtherSide) {
init.id = 53;
init.maxRetransmits = 52;
caller()->CreateDataChannel("data-channel", &init);
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
if (allow_media()) {
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
}
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
@ -429,7 +451,7 @@ TEST_P(DataChannelIntegrationTest, StressTestUnorderedSctpDataChannel) {
virtual_socket_server()->set_delay_stddev(5);
virtual_socket_server()->UpdateDelayDistribution();
// Normal procedure, but with unordered data channel config.
ASSERT_TRUE(CreatePeerConnectionWrappersWithoutMediaEngine());
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
webrtc::DataChannelInit init;
init.ordered = false;
@ -633,6 +655,10 @@ TEST_P(DataChannelIntegrationTest, StressTestOpenCloseChannelWithDelay) {
// This test sets up a call between two parties with audio, and video. When
// audio and video are setup and flowing, an SCTP data channel is negotiated.
TEST_P(DataChannelIntegrationTest, AddSctpDataChannelInSubsequentOffer) {
// This test can't be performed without media.
if (!allow_media()) {
return;
}
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Do initial offer/answer with audio/video.
@ -665,6 +691,10 @@ TEST_P(DataChannelIntegrationTest, AddSctpDataChannelInSubsequentOffer) {
// Effectively the inverse of the test above. This was broken in M57; see
// https://crbug.com/711243
TEST_P(DataChannelIntegrationTest, SctpDataChannelToAudioVideoUpgrade) {
// This test can't be performed without media.
if (!allow_media()) {
return;
}
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Do initial offer/answer with just data channel.
@ -724,6 +754,10 @@ TEST_P(DataChannelIntegrationTest,
// Test that after closing PeerConnections, they stop sending any packets
// (ICE, DTLS, RTP...).
TEST_P(DataChannelIntegrationTest, ClosingConnectionStopsPacketFlow) {
// This test can't be performed without media.
if (!allow_media()) {
return;
}
// Set up audio/video/data, wait for some frames to be received.
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
@ -1055,8 +1089,9 @@ TEST_P(DataChannelIntegrationTest,
INSTANTIATE_TEST_SUITE_P(DataChannelIntegrationTest,
DataChannelIntegrationTest,
Values(SdpSemantics::kPlanB_DEPRECATED,
SdpSemantics::kUnifiedPlan));
Combine(Values(SdpSemantics::kPlanB_DEPRECATED,
SdpSemantics::kUnifiedPlan),
testing::Bool()));
TEST_F(DataChannelIntegrationTestUnifiedPlan,
EndToEndCallWithBundledSctpDataChannel) {

22
pc/media_session.cc
View File

@ -1566,9 +1566,7 @@ MediaSessionDescriptionFactory::MediaSessionDescriptionFactory(
const TransportDescriptionFactory* transport_desc_factory,
rtc::UniqueRandomIdGenerator* ssrc_generator)
: ssrc_generator_(ssrc_generator),
transport_desc_factory_(transport_desc_factory) {
RTC_DCHECK(ssrc_generator_);
}
transport_desc_factory_(transport_desc_factory) {}
MediaSessionDescriptionFactory::MediaSessionDescriptionFactory(
cricket::MediaEngineInterface* media_engine,
@ -2366,7 +2364,7 @@ bool MediaSessionDescriptionFactory::AddAudioContentForOffer(
if (!CreateMediaContentOffer(
media_description_options, session_options, filtered_codecs,
sdes_policy, GetCryptos(current_content), crypto_suites,
audio_rtp_extensions, ssrc_generator_, current_streams, audio.get(),
audio_rtp_extensions, ssrc_generator(), current_streams, audio.get(),
transport_desc_factory_->trials())) {
return false;
}
@ -2478,7 +2476,7 @@ bool MediaSessionDescriptionFactory::AddVideoContentForOffer(
if (!CreateMediaContentOffer(
media_description_options, session_options, filtered_codecs,
sdes_policy, GetCryptos(current_content), crypto_suites,
video_rtp_extensions, ssrc_generator_, current_streams, video.get(),
video_rtp_extensions, ssrc_generator(), current_streams, video.get(),
transport_desc_factory_->trials())) {
return false;
}
@ -2531,8 +2529,8 @@ bool MediaSessionDescriptionFactory::AddDataContentForOffer(
if (!CreateContentOffer(media_description_options, session_options,
sdes_policy, GetCryptos(current_content),
crypto_suites, RtpHeaderExtensions(), ssrc_generator_,
current_streams, data.get())) {
crypto_suites, RtpHeaderExtensions(),
ssrc_generator(), current_streams, data.get())) {
return false;
}
@ -2673,7 +2671,7 @@ bool MediaSessionDescriptionFactory::AddAudioContentForAnswer(
audio_transport->secure() ? cricket::SEC_DISABLED : secure();
if (!SetCodecsInAnswer(offer_audio_description, filtered_codecs,
media_description_options, session_options,
ssrc_generator_, current_streams, audio_answer.get(),
ssrc_generator(), current_streams, audio_answer.get(),
transport_desc_factory_->trials())) {
return false;
}
@ -2681,7 +2679,7 @@ bool MediaSessionDescriptionFactory::AddAudioContentForAnswer(
offer_audio_description, media_description_options, session_options,
sdes_policy, GetCryptos(current_content),
filtered_rtp_header_extensions(default_audio_rtp_header_extensions),
ssrc_generator_, enable_encrypted_rtp_header_extensions_,
ssrc_generator(), enable_encrypted_rtp_header_extensions_,
current_streams, bundle_enabled, audio_answer.get())) {
return false; // Fails the session setup.
}
@ -2809,7 +2807,7 @@ bool MediaSessionDescriptionFactory::AddVideoContentForAnswer(
video_transport->secure() ? cricket::SEC_DISABLED : secure();
if (!SetCodecsInAnswer(offer_video_description, filtered_codecs,
media_description_options, session_options,
ssrc_generator_, current_streams, video_answer.get(),
ssrc_generator(), current_streams, video_answer.get(),
transport_desc_factory_->trials())) {
return false;
}
@ -2817,7 +2815,7 @@ bool MediaSessionDescriptionFactory::AddVideoContentForAnswer(
offer_video_description, media_description_options, session_options,
sdes_policy, GetCryptos(current_content),
filtered_rtp_header_extensions(default_video_rtp_header_extensions),
ssrc_generator_, enable_encrypted_rtp_header_extensions_,
ssrc_generator(), enable_encrypted_rtp_header_extensions_,
current_streams, bundle_enabled, video_answer.get())) {
return false; // Failed the session setup.
}
@ -2890,7 +2888,7 @@ bool MediaSessionDescriptionFactory::AddDataContentForAnswer(
if (!CreateMediaContentAnswer(
offer_data_description, media_description_options, session_options,
sdes_policy, GetCryptos(current_content), RtpHeaderExtensions(),
ssrc_generator_, enable_encrypted_rtp_header_extensions_,
ssrc_generator(), enable_encrypted_rtp_header_extensions_,
current_streams, bundle_enabled, data_answer.get())) {
return false; // Fails the session setup.
}

10
pc/media_session.h
View File

@ -34,6 +34,7 @@
#include "pc/media_protocol_names.h"
#include "pc/session_description.h"
#include "pc/simulcast_description.h"
#include "rtc_base/memory/always_valid_pointer.h"
#include "rtc_base/unique_id_generator.h"
namespace webrtc {
@ -331,6 +332,10 @@ class MediaSessionDescriptionFactory {
void ComputeVideoCodecsIntersectionAndUnion();
rtc::UniqueRandomIdGenerator* ssrc_generator() const {
return ssrc_generator_.get();
}
bool is_unified_plan_ = false;
AudioCodecs audio_send_codecs_;
AudioCodecs audio_recv_codecs_;
@ -344,8 +349,9 @@ class MediaSessionDescriptionFactory {
VideoCodecs video_sendrecv_codecs_;
// Union of send and recv.
VideoCodecs all_video_codecs_;
// This object is not owned by the channel so it must outlive it.
rtc::UniqueRandomIdGenerator* const ssrc_generator_;
// This object may or may not be owned by this class.
webrtc::AlwaysValidPointer<rtc::UniqueRandomIdGenerator> const
ssrc_generator_;
bool enable_encrypted_rtp_header_extensions_ = false;
// TODO(zhihuang): Rename secure_ to sdec_policy_; rename the related getter
// and setter.

59
pc/peer_connection.cc
View File

@ -1172,8 +1172,10 @@ std::vector<rtc::scoped_refptr<RtpSenderInterface>> PeerConnection::GetSenders()
const {
RTC_DCHECK_RUN_ON(signaling_thread());
std::vector<rtc::scoped_refptr<RtpSenderInterface>> ret;
for (const auto& sender : rtp_manager()->GetSendersInternal()) {
ret.push_back(sender);
if (ConfiguredForMedia()) {
for (const auto& sender : rtp_manager()->GetSendersInternal()) {
ret.push_back(sender);
}
}
return ret;
}
@ -1182,8 +1184,10 @@ std::vector<rtc::scoped_refptr<RtpReceiverInterface>>
PeerConnection::GetReceivers() const {
RTC_DCHECK_RUN_ON(signaling_thread());
std::vector<rtc::scoped_refptr<RtpReceiverInterface>> ret;
for (const auto& receiver : rtp_manager()->GetReceiversInternal()) {
ret.push_back(receiver);
if (ConfiguredForMedia()) {
for (const auto& receiver : rtp_manager()->GetReceiversInternal()) {
ret.push_back(receiver);
}
}
return ret;
}
@ -1194,8 +1198,10 @@ PeerConnection::GetTransceivers() const {
RTC_CHECK(IsUnifiedPlan())
<< "GetTransceivers is only supported with Unified Plan SdpSemantics.";
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>> all_transceivers;
for (const auto& transceiver : rtp_manager()->transceivers()->List()) {
all_transceivers.push_back(transceiver);
if (ConfiguredForMedia()) {
for (const auto& transceiver : rtp_manager()->transceivers()->List()) {
all_transceivers.push_back(transceiver);
}
}
return all_transceivers;
}
@ -1814,12 +1820,13 @@ void PeerConnection::Close() {
NoteUsageEvent(UsageEvent::CLOSE_CALLED);
for (const auto& transceiver : rtp_manager()->transceivers()->List()) {
transceiver->internal()->SetPeerConnectionClosed();
if (!transceiver->stopped())
transceiver->StopInternal();
if (ConfiguredForMedia()) {
for (const auto& transceiver : rtp_manager()->transceivers()->List()) {
transceiver->internal()->SetPeerConnectionClosed();
if (!transceiver->stopped())
transceiver->StopInternal();
}
}
// Ensure that all asynchronous stats requests are completed before destroying
// the transport controller below.
if (stats_collector_) {
@ -1836,7 +1843,9 @@ void PeerConnection::Close() {
// WebRTC session description factory, the session description factory would
// call the transport controller.
sdp_handler_->ResetSessionDescFactory();
rtp_manager_->Close();
if (ConfiguredForMedia()) {
rtp_manager_->Close();
}
network_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
// Data channels will already have been unset via the DestroyAllChannels()
@ -2727,12 +2736,15 @@ void PeerConnection::ReportTransportStats() {
rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
std::map<std::string, std::set<cricket::MediaType>>
media_types_by_transport_name;
for (const auto& transceiver : rtp_manager()->transceivers()->UnsafeList()) {
if (transceiver->internal()->channel()) {
std::string transport_name(
transceiver->internal()->channel()->transport_name());
media_types_by_transport_name[transport_name].insert(
transceiver->media_type());
if (ConfiguredForMedia()) {
for (const auto& transceiver :
rtp_manager()->transceivers()->UnsafeList()) {
if (transceiver->internal()->channel()) {
std::string transport_name(
transceiver->internal()->channel()->transport_name());
media_types_by_transport_name[transport_name].insert(
transceiver->media_type());
}
}
}
@ -2880,10 +2892,13 @@ bool PeerConnection::OnTransportChanged(
DataChannelTransportInterface* data_channel_transport) {
RTC_DCHECK_RUN_ON(network_thread());
bool ret = true;
for (const auto& transceiver : rtp_manager()->transceivers()->UnsafeList()) {
cricket::ChannelInterface* channel = transceiver->internal()->channel();
if (channel && channel->mid() == mid) {
ret = channel->SetRtpTransport(rtp_transport);
if (ConfiguredForMedia()) {
for (const auto& transceiver :
rtp_manager()->transceivers()->UnsafeList()) {
cricket::ChannelInterface* channel = transceiver->internal()->channel();
if (channel && channel->mid() == mid) {
ret = channel->SetRtpTransport(rtp_transport);
}
}
}

3
pc/peer_connection.h
View File

@ -273,6 +273,9 @@ class PeerConnection : public PeerConnectionInternal,
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
GetTransceiversInternal() const override {
RTC_DCHECK_RUN_ON(signaling_thread());
if (!ConfiguredForMedia()) {
return {};
}
return rtp_manager()->transceivers()->List();
}

2
pc/rtp_transceiver.cc
View File

@ -23,6 +23,7 @@
#include "api/sequence_checker.h"
#include "media/base/codec.h"
#include "media/base/media_constants.h"
#include "media/base/media_engine.h"
#include "pc/channel.h"
#include "pc/rtp_media_utils.h"
#include "pc/session_description.h"
@ -602,7 +603,6 @@ void RtpTransceiver::StopTransceiverProcedure() {
RTCError RtpTransceiver::SetCodecPreferences(
rtc::ArrayView<RtpCodecCapability> codec_capabilities) {
RTC_DCHECK(unified_plan_);
// 3. If codecs is an empty list, set transceiver's [[PreferredCodecs]] slot
// to codecs and abort these steps.
if (codec_capabilities.empty()) {

523
pc/sdp_offer_answer.cc
View File

@ -1549,56 +1549,60 @@ RTCError SdpOfferAnswerHandler::ApplyLocalDescription(
<< ")";
return error;
}
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>> remove_list;
std::vector<rtc::scoped_refptr<MediaStreamInterface>> removed_streams;
for (const auto& transceiver_ext : transceivers()->List()) {
auto transceiver = transceiver_ext->internal();
if (transceiver->stopped()) {
continue;
}
// 2.2.7.1.1.(6-9): Set sender and receiver's transport slots.
// Note that code paths that don't set MID won't be able to use
// information about DTLS transports.
if (transceiver->mid()) {
auto dtls_transport = LookupDtlsTransportByMid(
context_->network_thread(), transport_controller_s(),
*transceiver->mid());
transceiver->sender_internal()->set_transport(dtls_transport);
transceiver->receiver_internal()->set_transport(dtls_transport);
}
const ContentInfo* content =
FindMediaSectionForTransceiver(transceiver, local_description());
if (!content) {
continue;
}
const MediaContentDescription* media_desc = content->media_description();
// 2.2.7.1.6: If description is of type "answer" or "pranswer", then run
// the following steps:
if (type == SdpType::kPrAnswer || type == SdpType::kAnswer) {
// 2.2.7.1.6.1: If direction is "sendonly" or "inactive", and
// transceiver's [[FiredDirection]] slot is either "sendrecv" or
// "recvonly", process the removal of a remote track for the media
// description, given transceiver, removeList, and muteTracks.
if (!RtpTransceiverDirectionHasRecv(media_desc->direction()) &&
(transceiver->fired_direction() &&
RtpTransceiverDirectionHasRecv(*transceiver->fired_direction()))) {
ProcessRemovalOfRemoteTrack(transceiver_ext, &remove_list,
&removed_streams);
if (ConfiguredForMedia()) {
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>> remove_list;
std::vector<rtc::scoped_refptr<MediaStreamInterface>> removed_streams;
for (const auto& transceiver_ext : transceivers()->List()) {
auto transceiver = transceiver_ext->internal();
if (transceiver->stopped()) {
continue;
}
// 2.2.7.1.1.(6-9): Set sender and receiver's transport slots.
// Note that code paths that don't set MID won't be able to use
// information about DTLS transports.
if (transceiver->mid()) {
auto dtls_transport = LookupDtlsTransportByMid(
context_->network_thread(), transport_controller_s(),
*transceiver->mid());
transceiver->sender_internal()->set_transport(dtls_transport);
transceiver->receiver_internal()->set_transport(dtls_transport);
}
const ContentInfo* content =
FindMediaSectionForTransceiver(transceiver, local_description());
if (!content) {
continue;
}
const MediaContentDescription* media_desc =
content->media_description();
// 2.2.7.1.6: If description is of type "answer" or "pranswer", then run
// the following steps:
if (type == SdpType::kPrAnswer || type == SdpType::kAnswer) {
// 2.2.7.1.6.1: If direction is "sendonly" or "inactive", and
// transceiver's [[FiredDirection]] slot is either "sendrecv" or
// "recvonly", process the removal of a remote track for the media
// description, given transceiver, removeList, and muteTracks.
if (!RtpTransceiverDirectionHasRecv(media_desc->direction()) &&
(transceiver->fired_direction() &&
RtpTransceiverDirectionHasRecv(
*transceiver->fired_direction()))) {
ProcessRemovalOfRemoteTrack(transceiver_ext, &remove_list,
&removed_streams);
}
// 2.2.7.1.6.2: Set transceiver's [[CurrentDirection]] and
// [[FiredDirection]] slots to direction.
transceiver->set_current_direction(media_desc->direction());
transceiver->set_fired_direction(media_desc->direction());
}
// 2.2.7.1.6.2: Set transceiver's [[CurrentDirection]] and
// [[FiredDirection]] slots to direction.
transceiver->set_current_direction(media_desc->direction());
transceiver->set_fired_direction(media_desc->direction());
}
}
auto observer = pc_->Observer();
for (const auto& transceiver : remove_list) {
observer->OnRemoveTrack(transceiver->receiver());
}
for (const auto& stream : removed_streams) {
observer->OnRemoveStream(stream);
auto observer = pc_->Observer();
for (const auto& transceiver : remove_list) {
observer->OnRemoveTrack(transceiver->receiver());
}
for (const auto& stream : removed_streams) {
observer->OnRemoveStream(stream);
}
}
} else {
// Media channels will be created only when offer is set. These may use new
@ -1642,35 +1646,39 @@ RTCError SdpOfferAnswerHandler::ApplyLocalDescription(
}
if (IsUnifiedPlan()) {
// We must use List and not ListInternal here because
// transceivers()->StableState() is indexed by the non-internal refptr.
for (const auto& transceiver_ext : transceivers()->List()) {
auto transceiver = transceiver_ext->internal();
if (transceiver->stopped()) {
continue;
}
const ContentInfo* content =
FindMediaSectionForTransceiver(transceiver, local_description());
if (!content) {
continue;
}
cricket::ChannelInterface* channel = transceiver->channel();
if (content->rejected || !channel || channel->local_streams().empty()) {
// 0 is a special value meaning "this sender has no associated send
// stream". Need to call this so the sender won't attempt to configure
// a no longer existing stream and run into DCHECKs in the lower
// layers.
transceiver->sender_internal()->SetSsrc(0);
} else {
// Get the StreamParams from the channel which could generate SSRCs.
const std::vector<StreamParams>& streams = channel->local_streams();
transceiver->sender_internal()->set_stream_ids(streams[0].stream_ids());
auto encodings = transceiver->sender_internal()->init_send_encodings();
transceiver->sender_internal()->SetSsrc(streams[0].first_ssrc());
if (!encodings.empty()) {
transceivers()
->StableState(transceiver_ext)
->SetInitSendEncodings(encodings);
if (ConfiguredForMedia()) {
// We must use List and not ListInternal here because
// transceivers()->StableState() is indexed by the non-internal refptr.
for (const auto& transceiver_ext : transceivers()->List()) {
auto transceiver = transceiver_ext->internal();
if (transceiver->stopped()) {
continue;
}
const ContentInfo* content =
FindMediaSectionForTransceiver(transceiver, local_description());
if (!content) {
continue;
}
cricket::ChannelInterface* channel = transceiver->channel();
if (content->rejected || !channel || channel->local_streams().empty()) {
// 0 is a special value meaning "this sender has no associated send
// stream". Need to call this so the sender won't attempt to configure
// a no longer existing stream and run into DCHECKs in the lower
// layers.
transceiver->sender_internal()->SetSsrc(0);
} else {
// Get the StreamParams from the channel which could generate SSRCs.
const std::vector<StreamParams>& streams = channel->local_streams();
transceiver->sender_internal()->set_stream_ids(
streams[0].stream_ids());
auto encodings =
transceiver->sender_internal()->init_send_encodings();
transceiver->sender_internal()->SetSsrc(streams[0].first_ssrc());
if (!encodings.empty()) {
transceivers()
->StableState(transceiver_ext)
->SetInitSendEncodings(encodings);
}
}
}
}
@ -1930,6 +1938,9 @@ void SdpOfferAnswerHandler::ApplyRemoteDescriptionUpdateTransceiverState(
SdpType sdp_type) {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(IsUnifiedPlan());
if (!ConfiguredForMedia()) {
return;
}
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>
now_receiving_transceivers;
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>> remove_list;
@ -2717,7 +2728,9 @@ RTCError SdpOfferAnswerHandler::UpdateSessionState(
} else {
RTC_DCHECK(type == SdpType::kAnswer);
ChangeSignalingState(PeerConnectionInterface::kStable);
transceivers()->DiscardStableStates();
if (ConfiguredForMedia()) {
transceivers()->DiscardStableStates();
}
}
// Update internal objects according to the session description's media
@ -3143,6 +3156,9 @@ bool SdpOfferAnswerHandler::CheckIfNegotiationIsNeeded() {
if (!cricket::GetFirstDataContent(description->description()->contents()))
return true;
}
if (!ConfiguredForMedia()) {
return false;
}
// 5. For each transceiver in connection's set of transceivers, perform the
// following checks:
@ -3254,7 +3270,6 @@ bool SdpOfferAnswerHandler::CheckIfNegotiationIsNeeded() {
}
}
}
// If all the preceding checks were performed and true was not returned,
// nothing remains to be negotiated; return false.
return false;
@ -3833,38 +3848,43 @@ void SdpOfferAnswerHandler::GetOptionsForOffer(
void SdpOfferAnswerHandler::GetOptionsForPlanBOffer(
const PeerConnectionInterface::RTCOfferAnswerOptions& offer_answer_options,
cricket::MediaSessionOptions* session_options) {
// Figure out transceiver directional preferences.
bool send_audio =
!rtp_manager()->GetAudioTransceiver()->internal()->senders().empty();
bool send_video =
!rtp_manager()->GetVideoTransceiver()->internal()->senders().empty();
// By default, generate sendrecv/recvonly m= sections.
bool recv_audio = true;
bool recv_video = true;
bool offer_new_data_description =
data_channel_controller()->HasDataChannels();
bool send_audio = false;
bool send_video = false;
bool recv_audio = false;
bool recv_video = false;
if (ConfiguredForMedia()) {
// Figure out transceiver directional preferences.
send_audio =
!rtp_manager()->GetAudioTransceiver()->internal()->senders().empty();
send_video =
!rtp_manager()->GetVideoTransceiver()->internal()->senders().empty();
// By default, generate sendrecv/recvonly m= sections.
recv_audio = true;
recv_video = true;
}
// By default, only offer a new m= section if we have media to send with it.
bool offer_new_audio_description = send_audio;
bool offer_new_video_description = send_video;
bool offer_new_data_description =
data_channel_controller()->HasDataChannels();
// The "offer_to_receive_X" options allow those defaults to be overridden.
if (offer_answer_options.offer_to_receive_audio !=
PeerConnectionInterface::RTCOfferAnswerOptions::kUndefined) {
recv_audio = (offer_answer_options.offer_to_receive_audio > 0);
offer_new_audio_description =
offer_new_audio_description ||
(offer_answer_options.offer_to_receive_audio > 0);
if (ConfiguredForMedia()) {
// The "offer_to_receive_X" options allow those defaults to be overridden.
if (offer_answer_options.offer_to_receive_audio !=
PeerConnectionInterface::RTCOfferAnswerOptions::kUndefined) {
recv_audio = (offer_answer_options.offer_to_receive_audio > 0);
offer_new_audio_description =
offer_new_audio_description ||
(offer_answer_options.offer_to_receive_audio > 0);
}
if (offer_answer_options.offer_to_receive_video !=
RTCOfferAnswerOptions::kUndefined) {
recv_video = (offer_answer_options.offer_to_receive_video > 0);
offer_new_video_description =
offer_new_video_description ||
(offer_answer_options.offer_to_receive_video > 0);
}
}
if (offer_answer_options.offer_to_receive_video !=
RTCOfferAnswerOptions::kUndefined) {
recv_video = (offer_answer_options.offer_to_receive_video > 0);
offer_new_video_description =
offer_new_video_description ||
(offer_answer_options.offer_to_receive_video > 0);
}
absl::optional<size_t> audio_index;
absl::optional<size_t> video_index;
absl::optional<size_t> data_index;
@ -3879,42 +3899,44 @@ void SdpOfferAnswerHandler::GetOptionsForPlanBOffer(
&audio_index, &video_index, &data_index, session_options);
}
// Add audio/video/data m= sections to the end if needed.
if (!audio_index && offer_new_audio_description) {
cricket::MediaDescriptionOptions options(
cricket::MEDIA_TYPE_AUDIO, cricket::CN_AUDIO,
RtpTransceiverDirectionFromSendRecv(send_audio, recv_audio), false);
options.header_extensions =
media_engine()->voice().GetRtpHeaderExtensions();
session_options->media_description_options.push_back(options);
audio_index = session_options->media_description_options.size() - 1;
}
if (!video_index && offer_new_video_description) {
cricket::MediaDescriptionOptions options(
cricket::MEDIA_TYPE_VIDEO, cricket::CN_VIDEO,
RtpTransceiverDirectionFromSendRecv(send_video, recv_video), false);
options.header_extensions =
media_engine()->video().GetRtpHeaderExtensions();
session_options->media_description_options.push_back(options);
video_index = session_options->media_description_options.size() - 1;
if (ConfiguredForMedia()) {
// Add audio/video/data m= sections to the end if needed.
if (!audio_index && offer_new_audio_description) {
cricket::MediaDescriptionOptions options(
cricket::MEDIA_TYPE_AUDIO, cricket::CN_AUDIO,
RtpTransceiverDirectionFromSendRecv(send_audio, recv_audio), false);
options.header_extensions =
media_engine()->voice().GetRtpHeaderExtensions();
session_options->media_description_options.push_back(options);
audio_index = session_options->media_description_options.size() - 1;
}
if (!video_index && offer_new_video_description) {
cricket::MediaDescriptionOptions options(
cricket::MEDIA_TYPE_VIDEO, cricket::CN_VIDEO,
RtpTransceiverDirectionFromSendRecv(send_video, recv_video), false);
options.header_extensions =
media_engine()->video().GetRtpHeaderExtensions();
session_options->media_description_options.push_back(options);
video_index = session_options->media_description_options.size() - 1;
}
cricket::MediaDescriptionOptions* audio_media_description_options =
!audio_index
? nullptr
: &session_options->media_description_options[*audio_index];
cricket::MediaDescriptionOptions* video_media_description_options =
!video_index
? nullptr
: &session_options->media_description_options[*video_index];
AddPlanBRtpSenderOptions(rtp_manager()->GetSendersInternal(),
audio_media_description_options,
video_media_description_options,
offer_answer_options.num_simulcast_layers);
}
if (!data_index && offer_new_data_description) {
session_options->media_description_options.push_back(
GetMediaDescriptionOptionsForActiveData(cricket::CN_DATA));
data_index = session_options->media_description_options.size() - 1;
}
cricket::MediaDescriptionOptions* audio_media_description_options =
!audio_index ? nullptr
: &session_options->media_description_options[*audio_index];
cricket::MediaDescriptionOptions* video_media_description_options =
!video_index ? nullptr
: &session_options->media_description_options[*video_index];
AddPlanBRtpSenderOptions(rtp_manager()->GetSendersInternal(),
audio_media_description_options,
video_media_description_options,
offer_answer_options.num_simulcast_layers);
}
void SdpOfferAnswerHandler::GetOptionsForUnifiedPlanOffer(
@ -4020,27 +4042,29 @@ void SdpOfferAnswerHandler::GetOptionsForUnifiedPlanOffer(
// and not associated). Reuse media sections marked as recyclable first,
// otherwise append to the end of the offer. New media sections should be
// added in the order they were added to the PeerConnection.
for (const auto& transceiver : transceivers()->ListInternal()) {
if (transceiver->mid() || transceiver->stopping()) {
continue;
if (ConfiguredForMedia()) {
for (const auto& transceiver : transceivers()->ListInternal()) {
if (transceiver->mid() || transceiver->stopping()) {
continue;
}
size_t mline_index;
if (!recycleable_mline_indices.empty()) {
mline_index = recycleable_mline_indices.front();
recycleable_mline_indices.pop();
session_options->media_description_options[mline_index] =
GetMediaDescriptionOptionsForTransceiver(
transceiver, mid_generator_.GenerateString(),
/*is_create_offer=*/true);
} else {
mline_index = session_options->media_description_options.size();
session_options->media_description_options.push_back(
GetMediaDescriptionOptionsForTransceiver(
transceiver, mid_generator_.GenerateString(),
/*is_create_offer=*/true));
}
// See comment above for why CreateOffer changes the transceiver's state.
transceiver->set_mline_index(mline_index);
}
size_t mline_index;
if (!recycleable_mline_indices.empty()) {
mline_index = recycleable_mline_indices.front();
recycleable_mline_indices.pop();
session_options->media_description_options[mline_index] =
GetMediaDescriptionOptionsForTransceiver(
transceiver, mid_generator_.GenerateString(),
/*is_create_offer=*/true);
} else {
mline_index = session_options->media_description_options.size();
session_options->media_description_options.push_back(
GetMediaDescriptionOptionsForTransceiver(
transceiver, mid_generator_.GenerateString(),
/*is_create_offer=*/true));
}
// See comment above for why CreateOffer changes the transceiver's state.
transceiver->set_mline_index(mline_index);
}
// Lastly, add a m-section if we have local data channels and an m section
// does not already exist.
@ -4080,25 +4104,32 @@ void SdpOfferAnswerHandler::GetOptionsForAnswer(
void SdpOfferAnswerHandler::GetOptionsForPlanBAnswer(
const PeerConnectionInterface::RTCOfferAnswerOptions& offer_answer_options,
cricket::MediaSessionOptions* session_options) {
// Figure out transceiver directional preferences.
bool send_audio =
!rtp_manager()->GetAudioTransceiver()->internal()->senders().empty();
bool send_video =
!rtp_manager()->GetVideoTransceiver()->internal()->senders().empty();
bool send_audio = false;
bool recv_audio = false;
bool send_video = false;
bool recv_video = false;
// By default, generate sendrecv/recvonly m= sections. The direction is also
// restricted by the direction in the offer.
bool recv_audio = true;
bool recv_video = true;
if (ConfiguredForMedia()) {
// Figure out transceiver directional preferences.
send_audio =
!rtp_manager()->GetAudioTransceiver()->internal()->senders().empty();
send_video =
!rtp_manager()->GetVideoTransceiver()->internal()->senders().empty();
// The "offer_to_receive_X" options allow those defaults to be overridden.
if (offer_answer_options.offer_to_receive_audio !=
RTCOfferAnswerOptions::kUndefined) {
recv_audio = (offer_answer_options.offer_to_receive_audio > 0);
}
if (offer_answer_options.offer_to_receive_video !=
RTCOfferAnswerOptions::kUndefined) {
recv_video = (offer_answer_options.offer_to_receive_video > 0);
// By default, generate sendrecv/recvonly m= sections. The direction is also
// restricted by the direction in the offer.
recv_audio = true;
recv_video = true;
// The "offer_to_receive_X" options allow those defaults to be overridden.
if (offer_answer_options.offer_to_receive_audio !=
RTCOfferAnswerOptions::kUndefined) {
recv_audio = (offer_answer_options.offer_to_receive_audio > 0);
}
if (offer_answer_options.offer_to_receive_video !=
RTCOfferAnswerOptions::kUndefined) {
recv_video = (offer_answer_options.offer_to_receive_video > 0);
}
}
absl::optional<size_t> audio_index;
@ -4121,10 +4152,12 @@ void SdpOfferAnswerHandler::GetOptionsForPlanBAnswer(
!video_index ? nullptr
: &session_options->media_description_options[*video_index];
AddPlanBRtpSenderOptions(rtp_manager()->GetSendersInternal(),
audio_media_description_options,
video_media_description_options,
offer_answer_options.num_simulcast_layers);
if (ConfiguredForMedia()) {
AddPlanBRtpSenderOptions(rtp_manager()->GetSendersInternal(),
audio_media_description_options,
video_media_description_options,
offer_answer_options.num_simulcast_layers);
}
}
void SdpOfferAnswerHandler::GetOptionsForUnifiedPlanAnswer(
@ -4469,6 +4502,9 @@ void SdpOfferAnswerHandler::UpdateRemoteSendersList(
void SdpOfferAnswerHandler::EnableSending() {
TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::EnableSending");
RTC_DCHECK_RUN_ON(signaling_thread());
if (!ConfiguredForMedia()) {
return;
}
for (const auto& transceiver : transceivers()->ListInternal()) {
cricket::ChannelInterface* channel = transceiver->channel();
if (channel) {
@ -4489,60 +4525,63 @@ RTCError SdpOfferAnswerHandler::PushdownMediaDescription(
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(sdesc);
// Note: This will perform an Invoke over to the worker thread, which we'll
// also do in a loop below.
if (!UpdatePayloadTypeDemuxingState(source, bundle_groups_by_mid)) {
// Note that this is never expected to fail, since RtpDemuxer doesn't return
// an error when changing payload type demux criteria, which is all this
// does.
return RTCError(RTCErrorType::INTERNAL_ERROR,
"Failed to update payload type demuxing state.");
}
// Push down the new SDP media section for each audio/video transceiver.
auto rtp_transceivers = transceivers()->ListInternal();
std::vector<
std::pair<cricket::ChannelInterface*, const MediaContentDescription*>>
channels;
for (const auto& transceiver : rtp_transceivers) {
const ContentInfo* content_info =
FindMediaSectionForTransceiver(transceiver, sdesc);
cricket::ChannelInterface* channel = transceiver->channel();
if (!channel || !content_info || content_info->rejected) {
continue;
}
const MediaContentDescription* content_desc =
content_info->media_description();
if (!content_desc) {
continue;
if (ConfiguredForMedia()) {
// Note: This will perform an Invoke over to the worker thread, which we'll
// also do in a loop below.
if (!UpdatePayloadTypeDemuxingState(source, bundle_groups_by_mid)) {
// Note that this is never expected to fail, since RtpDemuxer doesn't
// return an error when changing payload type demux criteria, which is all
// this does.
return RTCError(RTCErrorType::INTERNAL_ERROR,
"Failed to update payload type demuxing state.");
}
transceiver->OnNegotiationUpdate(type, content_desc);
channels.push_back(std::make_pair(channel, content_desc));
}
// Push down the new SDP media section for each audio/video transceiver.
auto rtp_transceivers = transceivers()->ListInternal();
std::vector<
std::pair<cricket::ChannelInterface*, const MediaContentDescription*>>
channels;
for (const auto& transceiver : rtp_transceivers) {
const ContentInfo* content_info =
FindMediaSectionForTransceiver(transceiver, sdesc);
cricket::ChannelInterface* channel = transceiver->channel();
if (!channel || !content_info || content_info->rejected) {
continue;
}
const MediaContentDescription* content_desc =
content_info->media_description();
if (!content_desc) {
continue;
}
// This for-loop of invokes helps audio impairment during re-negotiations.
// One of the causes is that downstairs decoder creation is synchronous at the
// moment, and that a decoder is created for each codec listed in the SDP.
//
// TODO(bugs.webrtc.org/12840): consider merging the invokes again after
// these projects have shipped:
// - bugs.webrtc.org/12462
// - crbug.com/1157227
// - crbug.com/1187289
for (const auto& entry : channels) {
std::string error;
bool success =
context_->worker_thread()->Invoke<bool>(RTC_FROM_HERE, [&]() {
return (source == cricket::CS_LOCAL)
? entry.first->SetLocalContent(entry.second, type, error)
: entry.first->SetRemoteContent(entry.second, type, error);
});
if (!success) {
return RTCError(RTCErrorType::INVALID_PARAMETER, error);
transceiver->OnNegotiationUpdate(type, content_desc);
channels.push_back(std::make_pair(channel, content_desc));
}
// This for-loop of invokes helps audio impairment during re-negotiations.
// One of the causes is that downstairs decoder creation is synchronous at
// the moment, and that a decoder is created for each codec listed in the
// SDP.
//
// TODO(bugs.webrtc.org/12840): consider merging the invokes again after
// these projects have shipped:
// - bugs.webrtc.org/12462
// - crbug.com/1157227
// - crbug.com/1187289
for (const auto& entry : channels) {
std::string error;
bool success =
context_->worker_thread()->Invoke<bool>(RTC_FROM_HERE, [&]() {
return (source == cricket::CS_LOCAL)
? entry.first->SetLocalContent(entry.second, type, error)
: entry.first->SetRemoteContent(entry.second, type,
error);
});
if (!success) {
return RTCError(RTCErrorType::INVALID_PARAMETER, error);
}
}
}
// Need complete offer/answer with an SCTP m= section before starting SCTP,
// according to https://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp-19
if (pc_->sctp_mid() && local_description() && remote_description()) {
@ -4596,6 +4635,9 @@ void SdpOfferAnswerHandler::RemoveStoppedTransceivers() {
// run the following steps:
if (!IsUnifiedPlan())
return;
if (!ConfiguredForMedia()) {
return;
}
// Traverse a copy of the transceiver list.
auto transceiver_list = transceivers()->List();
for (auto transceiver : transceiver_list) {
@ -4630,18 +4672,21 @@ void SdpOfferAnswerHandler::RemoveStoppedTransceivers() {
void SdpOfferAnswerHandler::RemoveUnusedChannels(
const SessionDescription* desc) {
RTC_DCHECK_RUN_ON(signaling_thread());
// Destroy video channel first since it may have a pointer to the
// voice channel.
const cricket::ContentInfo* video_info = cricket::GetFirstVideoContent(desc);
if (!video_info || video_info->rejected) {
rtp_manager()->GetVideoTransceiver()->internal()->ClearChannel();
}
if (ConfiguredForMedia()) {
// Destroy video channel first since it may have a pointer to the
// voice channel.
const cricket::ContentInfo* video_info =
cricket::GetFirstVideoContent(desc);
if (!video_info || video_info->rejected) {
rtp_manager()->GetVideoTransceiver()->internal()->ClearChannel();
}
const cricket::ContentInfo* audio_info = cricket::GetFirstAudioContent(desc);
if (!audio_info || audio_info->rejected) {
rtp_manager()->GetAudioTransceiver()->internal()->ClearChannel();
const cricket::ContentInfo* audio_info =
cricket::GetFirstAudioContent(desc);
if (!audio_info || audio_info->rejected) {
rtp_manager()->GetAudioTransceiver()->internal()->ClearChannel();
}
}
const cricket::ContentInfo* data_info = cricket::GetFirstDataContent(desc);
if (!data_info) {
RTCError error(RTCErrorType::OPERATION_ERROR_WITH_DATA,
@ -5200,4 +5245,8 @@ bool SdpOfferAnswerHandler::UpdatePayloadTypeDemuxingState(
});
}
bool SdpOfferAnswerHandler::ConfiguredForMedia() const {
return context_->media_engine();
}
} // namespace webrtc

3
pc/sdp_offer_answer.h
View File

@ -478,7 +478,7 @@ class SdpOfferAnswerHandler : public SdpStateProvider,
// This enables media to flow on all configured audio/video channels.
void EnableSending();
// Push the media parts of the local or remote session description
// down to all of the channels.
// down to all of the channels, and start SCTP if needed.
RTCError PushdownMediaDescription(
SdpType type,
cricket::ContentSource source,
@ -596,6 +596,7 @@ class SdpOfferAnswerHandler : public SdpStateProvider,
// ===================================================================
const cricket::AudioOptions& audio_options() { return audio_options_; }
const cricket::VideoOptions& video_options() { return video_options_; }
bool ConfiguredForMedia() const;
PeerConnectionSdpMethods* const pc_;
ConnectionContext* const context_;

10
pc/test/integration_test_helpers.h
View File

@ -373,9 +373,17 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
rtc::scoped_refptr<RtpSenderInterface> AddTrack(
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const std::vector<std::string>& stream_ids = {}) {
EXPECT_TRUE(track);
if (!track) {
return nullptr;
}
auto result = pc()->AddTrack(track, stream_ids);
EXPECT_EQ(RTCErrorType::NONE, result.error().type());
return result.MoveValue();
if (result.ok()) {
return result.MoveValue();
} else {
return nullptr;
}
}
std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceiversOfType(