Reland "Don't create channel_manager++ when media_engine is not set"
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}
diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index 24ea680..d4acaed 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -329,6 +329,7 @@
"../rtc_base:checks",
"../rtc_base:logging",
"../rtc_base:stringutils",
+ "../rtc_base/memory:always_valid_pointer",
"../rtc_base/third_party/base64",
]
absl_deps = [
diff --git a/pc/data_channel_integrationtest.cc b/pc/data_channel_integrationtest.cc
index d184a81..faec76d 100644
--- a/pc/data_channel_integrationtest.cc
+++ b/pc/data_channel_integrationtest.cc
@@ -58,10 +58,24 @@
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 @@
// 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 @@
// 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 @@
// 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 @@
// 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 @@
// 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 @@
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 @@
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 @@
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 @@
// 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 @@
// 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 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 @@
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) {
diff --git a/pc/media_session.cc b/pc/media_session.cc
index ec4b6a2..3f7dbb5 100644
--- a/pc/media_session.cc
+++ b/pc/media_session.cc
@@ -1566,9 +1566,7 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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.
}
diff --git a/pc/media_session.h b/pc/media_session.h
index ef2893d..3711110 100644
--- a/pc/media_session.h
+++ b/pc/media_session.h
@@ -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 @@
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 @@
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.
diff --git a/pc/peer_connection.cc b/pc/peer_connection.cc
index c3b29d0..c277ac0 100644
--- a/pc/peer_connection.cc
+++ b/pc/peer_connection.cc
@@ -1172,8 +1172,10 @@
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 @@
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 @@
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 @@
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 @@
// 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 @@
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 @@
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);
+ }
}
}
diff --git a/pc/peer_connection.h b/pc/peer_connection.h
index 51dbea8..7dd6446 100644
--- a/pc/peer_connection.h
+++ b/pc/peer_connection.h
@@ -273,6 +273,9 @@
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
GetTransceiversInternal() const override {
RTC_DCHECK_RUN_ON(signaling_thread());
+ if (!ConfiguredForMedia()) {
+ return {};
+ }
return rtp_manager()->transceivers()->List();
}
diff --git a/pc/rtp_transceiver.cc b/pc/rtp_transceiver.cc
index 4a0fce7..5ca662c 100644
--- a/pc/rtp_transceiver.cc
+++ b/pc/rtp_transceiver.cc
@@ -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 @@
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()) {
diff --git a/pc/sdp_offer_answer.cc b/pc/sdp_offer_answer.cc
index f84d37c..f163851 100644
--- a/pc/sdp_offer_answer.cc
+++ b/pc/sdp_offer_answer.cc
@@ -1549,56 +1549,60 @@
<< ")";
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;
}
- // 2.2.7.1.6.2: Set transceiver's [[CurrentDirection]] and
- // [[FiredDirection]] slots to direction.
- transceiver->set_current_direction(media_desc->direction());
- transceiver->set_fired_direction(media_desc->direction());
+
+ // 2.2.7.1.1.(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());
+ }
}
- }
- 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 @@
}
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 @@
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 @@
} 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 @@
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 @@
}
}
}
-
// 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::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();
+ 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.
- bool recv_audio = true;
- bool recv_video = true;
-
+ // 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 @@
&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 @@
// 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::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 @@
!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::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 @@
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 @@
// 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::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::ConfiguredForMedia() const {
+ return context_->media_engine();
+}
+
} // namespace webrtc
diff --git a/pc/sdp_offer_answer.h b/pc/sdp_offer_answer.h
index ab3dc20..6a0aaaa 100644
--- a/pc/sdp_offer_answer.h
+++ b/pc/sdp_offer_answer.h
@@ -478,7 +478,7 @@
// 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 @@
// ===================================================================
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_;
diff --git a/pc/test/integration_test_helpers.h b/pc/test/integration_test_helpers.h
index 35aedbd..a1d3a54 100644
--- a/pc/test/integration_test_helpers.h
+++ b/pc/test/integration_test_helpers.h
@@ -373,9 +373,17 @@
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(
