Delete almost all use of MediaConstraintsInterface in the PeerConnection API
Bug: webrtc:9239
Change-Id: I04f4370f624346bf72c7e4e090b57987b558213b
Reviewed-on: https://webrtc-review.googlesource.com/74420
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24396}diff --git a/api/BUILD.gn b/api/BUILD.gn
index 7271ece..28b3535 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -469,7 +469,6 @@
deps = [
":array_view",
":libjingle_peerconnection_api",
- ":libjingle_peerconnection_test_api",
":ortc_api",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
diff --git a/api/mediaconstraintsinterface.cc b/api/mediaconstraintsinterface.cc
index fb4481f..c892606 100644
--- a/api/mediaconstraintsinterface.cc
+++ b/api/mediaconstraintsinterface.cc
@@ -260,4 +260,48 @@
}
}
+bool CopyConstraintsIntoOfferAnswerOptions(
+ const MediaConstraintsInterface* constraints,
+ PeerConnectionInterface::RTCOfferAnswerOptions* offer_answer_options) {
+ if (!constraints) {
+ return true;
+ }
+
+ bool value = false;
+ size_t mandatory_constraints_satisfied = 0;
+
+ if (FindConstraint(constraints,
+ MediaConstraintsInterface::kOfferToReceiveAudio, &value,
+ &mandatory_constraints_satisfied)) {
+ offer_answer_options->offer_to_receive_audio =
+ value ? PeerConnectionInterface::RTCOfferAnswerOptions::
+ kOfferToReceiveMediaTrue
+ : 0;
+ }
+
+ if (FindConstraint(constraints,
+ MediaConstraintsInterface::kOfferToReceiveVideo, &value,
+ &mandatory_constraints_satisfied)) {
+ offer_answer_options->offer_to_receive_video =
+ value ? PeerConnectionInterface::RTCOfferAnswerOptions::
+ kOfferToReceiveMediaTrue
+ : 0;
+ }
+ if (FindConstraint(constraints,
+ MediaConstraintsInterface::kVoiceActivityDetection, &value,
+ &mandatory_constraints_satisfied)) {
+ offer_answer_options->voice_activity_detection = value;
+ }
+ if (FindConstraint(constraints, MediaConstraintsInterface::kUseRtpMux, &value,
+ &mandatory_constraints_satisfied)) {
+ offer_answer_options->use_rtp_mux = value;
+ }
+ if (FindConstraint(constraints, MediaConstraintsInterface::kIceRestart,
+ &value, &mandatory_constraints_satisfied)) {
+ offer_answer_options->ice_restart = value;
+ }
+
+ return mandatory_constraints_satisfied == constraints->GetMandatory().size();
+}
+
} // namespace webrtc
diff --git a/api/mediaconstraintsinterface.h b/api/mediaconstraintsinterface.h
index 54ab706..3c85dfb 100644
--- a/api/mediaconstraintsinterface.h
+++ b/api/mediaconstraintsinterface.h
@@ -13,9 +13,10 @@
// http://www.w3.org/TR/mediacapture-streams/#mediastreamconstraints and also
// used in WebRTC: http://dev.w3.org/2011/webrtc/editor/webrtc.html#constraints.
-// This interface is being deprecated in Chrome, and may be removed
-// from WebRTC too.
-// https://bugs.chromium.org/p/webrtc/issues/detail?id=5617
+// Implementation of the w3c constraints spec is the responsibility of the
+// browser. Chrome no longer uses the constraints api declared here, and it will
+// be removed from WebRTC.
+// https://bugs.chromium.org/p/webrtc/issues/detail?id=9239
#ifndef API_MEDIACONSTRAINTSINTERFACE_H_
#define API_MEDIACONSTRAINTSINTERFACE_H_
@@ -144,6 +145,10 @@
const MediaConstraintsInterface* constraints,
cricket::AudioOptions* options);
+bool CopyConstraintsIntoOfferAnswerOptions(
+ const MediaConstraintsInterface* constraints,
+ PeerConnectionInterface::RTCOfferAnswerOptions* offer_answer_options);
+
} // namespace webrtc
#endif // API_MEDIACONSTRAINTSINTERFACE_H_
diff --git a/api/peerconnectionfactoryproxy.h b/api/peerconnectionfactoryproxy.h
index db52083..d885cc1 100644
--- a/api/peerconnectionfactoryproxy.h
+++ b/api/peerconnectionfactoryproxy.h
@@ -31,13 +31,6 @@
// removed.
using PeerConnectionFactoryInterface::CreateVideoSource;
PROXY_METHOD1(void, SetOptions, const Options&)
-PROXY_METHOD5(rtc::scoped_refptr<PeerConnectionInterface>,
- CreatePeerConnection,
- const PeerConnectionInterface::RTCConfiguration&,
- const MediaConstraintsInterface*,
- std::unique_ptr<cricket::PortAllocator>,
- std::unique_ptr<rtc::RTCCertificateGeneratorInterface>,
- PeerConnectionObserver*);
PROXY_METHOD4(rtc::scoped_refptr<PeerConnectionInterface>,
CreatePeerConnection,
const PeerConnectionInterface::RTCConfiguration&,
diff --git a/api/peerconnectioninterface.cc b/api/peerconnectioninterface.cc
index ddaeb36..05aa53f 100644
--- a/api/peerconnectioninterface.cc
+++ b/api/peerconnectioninterface.cc
@@ -247,12 +247,4 @@
return CreateVideoSource(std::unique_ptr<cricket::VideoCapturer>(capturer));
}
-rtc::scoped_refptr<VideoTrackSourceInterface>
-PeerConnectionFactoryInterface::CreateVideoSource(
- cricket::VideoCapturer* capturer,
- const MediaConstraintsInterface* constraints) {
- return CreateVideoSource(std::unique_ptr<cricket::VideoCapturer>(capturer),
- constraints);
-}
-
} // namespace webrtc
diff --git a/api/peerconnectioninterface.h b/api/peerconnectioninterface.h
index 593084e..84a0501 100644
--- a/api/peerconnectioninterface.h
+++ b/api/peerconnectioninterface.h
@@ -846,22 +846,12 @@
// Create a new offer.
// The CreateSessionDescriptionObserver callback will be called when done.
virtual void CreateOffer(CreateSessionDescriptionObserver* observer,
- const MediaConstraintsInterface* constraints) {}
-
- // TODO(jiayl): remove the default impl and the old interface when chromium
- // code is updated.
- virtual void CreateOffer(CreateSessionDescriptionObserver* observer,
- const RTCOfferAnswerOptions& options) {}
+ const RTCOfferAnswerOptions& options) = 0;
// Create an answer to an offer.
// The CreateSessionDescriptionObserver callback will be called when done.
virtual void CreateAnswer(CreateSessionDescriptionObserver* observer,
- const RTCOfferAnswerOptions& options) {}
- // Deprecated - use version above.
- // TODO(hta): Remove and remove default implementations when all callers
- // are updated.
- virtual void CreateAnswer(CreateSessionDescriptionObserver* observer,
- const MediaConstraintsInterface* constraints) {}
+ const RTCOfferAnswerOptions& options) = 0;
// Sets the local session description.
// The PeerConnection takes the ownership of |desc| even if it fails.
@@ -1293,10 +1283,6 @@
// TODO(deadbeef): Remove these once safe to do so.
virtual rtc::scoped_refptr<VideoTrackSourceInterface> CreateVideoSource(
cricket::VideoCapturer* capturer);
- virtual rtc::scoped_refptr<VideoTrackSourceInterface> CreateVideoSource(
- cricket::VideoCapturer* capturer,
- const MediaConstraintsInterface* constraints);
-
// Creates a new local VideoTrack. The same |source| can be used in several
// tracks.
virtual rtc::scoped_refptr<VideoTrackInterface> CreateVideoTrack(
diff --git a/api/peerconnectionproxy.h b/api/peerconnectionproxy.h
index 7fc5e17..6b66bcc 100644
--- a/api/peerconnectionproxy.h
+++ b/api/peerconnectionproxy.h
@@ -88,14 +88,6 @@
PROXY_METHOD2(void,
CreateOffer,
CreateSessionDescriptionObserver*,
- const MediaConstraintsInterface*)
-PROXY_METHOD2(void,
- CreateAnswer,
- CreateSessionDescriptionObserver*,
- const MediaConstraintsInterface*)
-PROXY_METHOD2(void,
- CreateOffer,
- CreateSessionDescriptionObserver*,
const RTCOfferAnswerOptions&)
PROXY_METHOD2(void,
CreateAnswer,
diff --git a/examples/BUILD.gn b/examples/BUILD.gn
index 5a19a37..76526eb 100644
--- a/examples/BUILD.gn
+++ b/examples/BUILD.gn
@@ -721,7 +721,6 @@
deps += [
"../api:libjingle_peerconnection_api",
- "../api:libjingle_peerconnection_test_api",
"../api/audio_codecs:builtin_audio_decoder_factory",
"../api/audio_codecs:builtin_audio_encoder_factory",
"../api/video:video_frame",
@@ -853,7 +852,6 @@
}
deps = [
"../api:libjingle_peerconnection_api",
- "../api:libjingle_peerconnection_test_api",
"../api/audio_codecs:builtin_audio_decoder_factory",
"../api/audio_codecs:builtin_audio_encoder_factory",
"../api/video:video_frame",
diff --git a/examples/peerconnection/client/conductor.cc b/examples/peerconnection/client/conductor.cc
index 89c8984..65d4aca 100644
--- a/examples/peerconnection/client/conductor.cc
+++ b/examples/peerconnection/client/conductor.cc
@@ -16,7 +16,6 @@
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
-#include "api/test/fakeconstraints.h"
#include "api/video_codecs/builtin_video_decoder_factory.h"
#include "api/video_codecs/builtin_video_encoder_factory.h"
#include "examples/peerconnection/client/defaults.h"
diff --git a/examples/unityplugin/simple_peer_connection.cc b/examples/unityplugin/simple_peer_connection.cc
index df03e37..36e1937 100644
--- a/examples/unityplugin/simple_peer_connection.cc
+++ b/examples/unityplugin/simple_peer_connection.cc
@@ -15,7 +15,6 @@
#include "absl/memory/memory.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
-#include "api/test/fakeconstraints.h"
#include "api/videosourceproxy.h"
#include "media/engine/internaldecoderfactory.h"
#include "media/engine/internalencoderfactory.h"
@@ -164,12 +163,11 @@
webrtc::PeerConnectionInterface::IceServer stun_server;
stun_server.uri = GetPeerConnectionString();
config_.servers.push_back(stun_server);
-
- webrtc::FakeConstraints constraints;
- constraints.SetAllowDtlsSctpDataChannels();
+ config_.enable_rtp_data_channel = true;
+ config_.enable_dtls_srtp = false;
peer_connection_ = g_peer_connection_factory->CreatePeerConnection(
- config_, &constraints, nullptr, nullptr, this);
+ config_, nullptr, nullptr, this);
return peer_connection_.get() != nullptr;
}
@@ -207,12 +205,12 @@
if (!peer_connection_.get())
return false;
- webrtc::FakeConstraints constraints;
+ webrtc::PeerConnectionInterface::RTCOfferAnswerOptions options;
if (mandatory_receive_) {
- constraints.SetMandatoryReceiveAudio(true);
- constraints.SetMandatoryReceiveVideo(true);
+ options.offer_to_receive_audio = true;
+ options.offer_to_receive_video = true;
}
- peer_connection_->CreateOffer(this, &constraints);
+ peer_connection_->CreateOffer(this, options);
return true;
}
@@ -220,12 +218,12 @@
if (!peer_connection_.get())
return false;
- webrtc::FakeConstraints constraints;
+ webrtc::PeerConnectionInterface::RTCOfferAnswerOptions options;
if (mandatory_receive_) {
- constraints.SetMandatoryReceiveAudio(true);
- constraints.SetMandatoryReceiveVideo(true);
+ options.offer_to_receive_audio = true;
+ options.offer_to_receive_video = true;
}
- peer_connection_->CreateAnswer(this, &constraints);
+ peer_connection_->CreateAnswer(this, options);
return true;
}
diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index 03b6d80..8e41ce0 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -338,7 +338,6 @@
deps = [
":pc_test_utils",
"../api:libjingle_peerconnection_api",
- "../api:libjingle_peerconnection_test_api",
"../api:rtc_stats_api",
"../api/audio_codecs:builtin_audio_decoder_factory",
"../api/audio_codecs:builtin_audio_encoder_factory",
diff --git a/pc/localaudiosource_unittest.cc b/pc/localaudiosource_unittest.cc
index 54766ca..fe3a0e2 100644
--- a/pc/localaudiosource_unittest.cc
+++ b/pc/localaudiosource_unittest.cc
@@ -13,14 +13,11 @@
#include <string>
#include <vector>
-#include "api/test/fakeconstraints.h"
#include "media/base/fakemediaengine.h"
#include "media/base/fakevideorenderer.h"
#include "rtc_base/gunit.h"
using webrtc::LocalAudioSource;
-using webrtc::MediaConstraintsInterface;
-using webrtc::MediaSourceInterface;
TEST(LocalAudioSourceTest, InitWithAudioOptions) {
cricket::AudioOptions audio_options;
diff --git a/pc/peerconnection.cc b/pc/peerconnection.cc
index e068f60..9155b71 100644
--- a/pc/peerconnection.cc
+++ b/pc/peerconnection.cc
@@ -773,50 +773,6 @@
session_options->bundle_enabled = rtc_options.use_rtp_mux;
}
-bool ConvertConstraintsToOfferAnswerOptions(
- const MediaConstraintsInterface* constraints,
- PeerConnectionInterface::RTCOfferAnswerOptions* offer_answer_options) {
- if (!constraints) {
- return true;
- }
-
- bool value = false;
- size_t mandatory_constraints_satisfied = 0;
-
- if (FindConstraint(constraints,
- MediaConstraintsInterface::kOfferToReceiveAudio, &value,
- &mandatory_constraints_satisfied)) {
- offer_answer_options->offer_to_receive_audio =
- value ? PeerConnectionInterface::RTCOfferAnswerOptions::
- kOfferToReceiveMediaTrue
- : 0;
- }
-
- if (FindConstraint(constraints,
- MediaConstraintsInterface::kOfferToReceiveVideo, &value,
- &mandatory_constraints_satisfied)) {
- offer_answer_options->offer_to_receive_video =
- value ? PeerConnectionInterface::RTCOfferAnswerOptions::
- kOfferToReceiveMediaTrue
- : 0;
- }
- if (FindConstraint(constraints,
- MediaConstraintsInterface::kVoiceActivityDetection, &value,
- &mandatory_constraints_satisfied)) {
- offer_answer_options->voice_activity_detection = value;
- }
- if (FindConstraint(constraints, MediaConstraintsInterface::kUseRtpMux, &value,
- &mandatory_constraints_satisfied)) {
- offer_answer_options->use_rtp_mux = value;
- }
- if (FindConstraint(constraints, MediaConstraintsInterface::kIceRestart,
- &value, &mandatory_constraints_satisfied)) {
- offer_answer_options->ice_restart = value;
- }
-
- return mandatory_constraints_satisfied == constraints->GetMandatory().size();
-}
-
PeerConnection::PeerConnection(PeerConnectionFactory* factory,
std::unique_ptr<RtcEventLog> event_log,
std::unique_ptr<Call> call)
@@ -1733,23 +1689,6 @@
}
void PeerConnection::CreateOffer(CreateSessionDescriptionObserver* observer,
- const MediaConstraintsInterface* constraints) {
- TRACE_EVENT0("webrtc", "PeerConnection::CreateOffer");
-
- PeerConnectionInterface::RTCOfferAnswerOptions offer_answer_options;
- // Always create an offer even if |ConvertConstraintsToOfferAnswerOptions|
- // returns false for now. Because |ConvertConstraintsToOfferAnswerOptions|
- // compares the mandatory fields parsed with the mandatory fields added in the
- // |constraints| and some downstream applications might create offers with
- // mandatory fields which would not be parsed in the helper method. For
- // example, in Chromium/remoting, |kEnableDtlsSrtp| is added to the
- // |constraints| as a mandatory field but it is not parsed.
- ConvertConstraintsToOfferAnswerOptions(constraints, &offer_answer_options);
-
- CreateOffer(observer, offer_answer_options);
-}
-
-void PeerConnection::CreateOffer(CreateSessionDescriptionObserver* observer,
const RTCOfferAnswerOptions& options) {
TRACE_EVENT0("webrtc", "PeerConnection::CreateOffer");
@@ -1861,29 +1800,6 @@
return receiving_transceivers;
}
-void PeerConnection::CreateAnswer(
- CreateSessionDescriptionObserver* observer,
- const MediaConstraintsInterface* constraints) {
- TRACE_EVENT0("webrtc", "PeerConnection::CreateAnswer");
-
- if (!observer) {
- RTC_LOG(LS_ERROR) << "CreateAnswer - observer is NULL.";
- return;
- }
-
- PeerConnectionInterface::RTCOfferAnswerOptions offer_answer_options;
- if (!ConvertConstraintsToOfferAnswerOptions(constraints,
- &offer_answer_options)) {
- std::string error = "CreateAnswer called with invalid constraints.";
- RTC_LOG(LS_ERROR) << error;
- PostCreateSessionDescriptionFailure(
- observer, RTCError(RTCErrorType::INVALID_PARAMETER, std::move(error)));
- return;
- }
-
- CreateAnswer(observer, offer_answer_options);
-}
-
void PeerConnection::CreateAnswer(CreateSessionDescriptionObserver* observer,
const RTCOfferAnswerOptions& options) {
TRACE_EVENT0("webrtc", "PeerConnection::CreateAnswer");
diff --git a/pc/peerconnection.h b/pc/peerconnection.h
index 25fbe1f..2942709 100644
--- a/pc/peerconnection.h
+++ b/pc/peerconnection.h
@@ -157,14 +157,8 @@
const override;
// JSEP01
- // Deprecated, use version without constraints.
- void CreateOffer(CreateSessionDescriptionObserver* observer,
- const MediaConstraintsInterface* constraints) override;
void CreateOffer(CreateSessionDescriptionObserver* observer,
const RTCOfferAnswerOptions& options) override;
- // Deprecated, use version without constraints.
- void CreateAnswer(CreateSessionDescriptionObserver* observer,
- const MediaConstraintsInterface* constraints) override;
void CreateAnswer(CreateSessionDescriptionObserver* observer,
const RTCOfferAnswerOptions& options) override;
void SetLocalDescription(SetSessionDescriptionObserver* observer,
diff --git a/pc/peerconnection_crypto_unittest.cc b/pc/peerconnection_crypto_unittest.cc
index 5a8b5b8..1ccfe55 100644
--- a/pc/peerconnection_crypto_unittest.cc
+++ b/pc/peerconnection_crypto_unittest.cc
@@ -598,9 +598,11 @@
new rtc::RefCountedObject<MockCreateSessionDescriptionObserver>();
observers.push_back(observer);
if (sdp_type_ == SdpType::kOffer) {
- pc->pc()->CreateOffer(observer, nullptr);
+ pc->pc()->CreateOffer(observer,
+ PeerConnectionInterface::RTCOfferAnswerOptions());
} else {
- pc->pc()->CreateAnswer(observer, nullptr);
+ pc->pc()->CreateAnswer(observer,
+ PeerConnectionInterface::RTCOfferAnswerOptions());
}
}
for (auto& observer : observers) {
diff --git a/pc/peerconnection_ice_unittest.cc b/pc/peerconnection_ice_unittest.cc
index 36e0a09..23a49e9 100644
--- a/pc/peerconnection_ice_unittest.cc
+++ b/pc/peerconnection_ice_unittest.cc
@@ -975,9 +975,9 @@
new cricket::FakePortAllocator(rtc::Thread::Current(), nullptr));
port_allocator_ = port_allocator.get();
rtc::scoped_refptr<PeerConnectionInterface> pc(
- pc_factory_->CreatePeerConnection(
- config, nullptr /* constraint */, std::move(port_allocator),
- nullptr /* cert_generator */, &observer_));
+ pc_factory_->CreatePeerConnection(config, std::move(port_allocator),
+ nullptr /* cert_generator */,
+ &observer_));
EXPECT_TRUE(pc.get());
pc_ = std::move(pc.get());
}
diff --git a/pc/peerconnection_integrationtest.cc b/pc/peerconnection_integrationtest.cc
index 6def198..1b8054f 100644
--- a/pc/peerconnection_integrationtest.cc
+++ b/pc/peerconnection_integrationtest.cc
@@ -28,7 +28,6 @@
#include "api/peerconnectioninterface.h"
#include "api/peerconnectionproxy.h"
#include "api/rtpreceiverinterface.h"
-#include "api/test/fakeconstraints.h"
#include "api/video_codecs/builtin_video_decoder_factory.h"
#include "api/video_codecs/builtin_video_encoder_factory.h"
#include "api/video_codecs/sdp_video_format.h"
@@ -81,7 +80,6 @@
using webrtc::DtmfSender;
using webrtc::DtmfSenderInterface;
using webrtc::DtmfSenderObserverInterface;
-using webrtc::FakeConstraints;
using webrtc::FakeVideoTrackRenderer;
using webrtc::MediaConstraintsInterface;
using webrtc::MediaStreamInterface;
@@ -236,8 +234,8 @@
PeerConnectionWrapper* client(new PeerConnectionWrapper(debug_name));
webrtc::PeerConnectionDependencies dependencies(nullptr);
dependencies.cert_generator = std::move(cert_generator);
- if (!client->Init(nullptr, nullptr, nullptr, std::move(dependencies),
- network_thread, worker_thread, nullptr)) {
+ if (!client->Init(nullptr, nullptr, std::move(dependencies), network_thread,
+ worker_thread, nullptr)) {
delete client;
return nullptr;
}
@@ -561,8 +559,7 @@
explicit PeerConnectionWrapper(const std::string& debug_name)
: debug_name_(debug_name) {}
- bool Init(const MediaConstraintsInterface* constraints,
- const PeerConnectionFactory::Options* options,
+ bool Init(const PeerConnectionFactory::Options* options,
const PeerConnectionInterface::RTCConfiguration* config,
webrtc::PeerConnectionDependencies dependencies,
rtc::Thread* network_thread,
@@ -619,13 +616,11 @@
}
dependencies.allocator = std::move(port_allocator);
- peer_connection_ =
- CreatePeerConnection(constraints, config, std::move(dependencies));
+ peer_connection_ = CreatePeerConnection(config, std::move(dependencies));
return peer_connection_.get() != nullptr;
}
rtc::scoped_refptr<webrtc::PeerConnectionInterface> CreatePeerConnection(
- const MediaConstraintsInterface* constraints,
const PeerConnectionInterface::RTCConfiguration* config,
webrtc::PeerConnectionDependencies dependencies) {
PeerConnectionInterface::RTCConfiguration modified_config;
@@ -640,12 +635,6 @@
// ratios and not specific resolutions, is this even necessary?
modified_config.set_cpu_adaptation(false);
- // Use the legacy interface.
- if (constraints != nullptr) {
- return peer_connection_factory_->CreatePeerConnection(
- modified_config, constraints, std::move(dependencies.allocator),
- std::move(dependencies.cert_generator), this);
- }
dependencies.observer = this;
return peer_connection_factory_->CreatePeerConnection(
modified_config, std::move(dependencies));
@@ -1154,7 +1143,6 @@
// log factory will be used.
std::unique_ptr<PeerConnectionWrapper> CreatePeerConnectionWrapper(
const std::string& debug_name,
- const MediaConstraintsInterface* constraints,
const PeerConnectionFactory::Options* options,
const RTCConfiguration* config,
webrtc::PeerConnectionDependencies dependencies,
@@ -1171,9 +1159,9 @@
std::unique_ptr<PeerConnectionWrapper> client(
new PeerConnectionWrapper(debug_name));
- if (!client->Init(constraints, options, &modified_config,
- std::move(dependencies), network_thread_.get(),
- worker_thread_.get(), std::move(event_log_factory))) {
+ if (!client->Init(options, &modified_config, std::move(dependencies),
+ network_thread_.get(), worker_thread_.get(),
+ std::move(event_log_factory))) {
return nullptr;
}
return client;
@@ -1182,13 +1170,12 @@
std::unique_ptr<PeerConnectionWrapper>
CreatePeerConnectionWrapperWithFakeRtcEventLog(
const std::string& debug_name,
- const MediaConstraintsInterface* constraints,
const PeerConnectionFactory::Options* options,
const RTCConfiguration* config,
webrtc::PeerConnectionDependencies dependencies) {
std::unique_ptr<webrtc::FakeRtcEventLogFactory> event_log_factory(
new webrtc::FakeRtcEventLogFactory(rtc::Thread::Current()));
- return CreatePeerConnectionWrapper(debug_name, constraints, options, config,
+ return CreatePeerConnectionWrapper(debug_name, options, config,
std::move(dependencies),
std::move(event_log_factory));
}
@@ -1210,37 +1197,24 @@
SdpSemantics original_semantics = sdp_semantics_;
sdp_semantics_ = caller_semantics;
caller_ = CreatePeerConnectionWrapper(
- "Caller", nullptr, nullptr, nullptr,
- webrtc::PeerConnectionDependencies(nullptr), nullptr);
+ "Caller", nullptr, nullptr, webrtc::PeerConnectionDependencies(nullptr),
+ nullptr);
sdp_semantics_ = callee_semantics;
callee_ = CreatePeerConnectionWrapper(
- "Callee", nullptr, nullptr, nullptr,
- webrtc::PeerConnectionDependencies(nullptr), nullptr);
+ "Callee", nullptr, nullptr, webrtc::PeerConnectionDependencies(nullptr),
+ nullptr);
sdp_semantics_ = original_semantics;
return caller_ && callee_;
}
- bool CreatePeerConnectionWrappersWithConstraints(
- MediaConstraintsInterface* caller_constraints,
- MediaConstraintsInterface* callee_constraints) {
- caller_ = CreatePeerConnectionWrapper(
- "Caller", caller_constraints, nullptr, nullptr,
- webrtc::PeerConnectionDependencies(nullptr), nullptr);
- callee_ = CreatePeerConnectionWrapper(
- "Callee", callee_constraints, nullptr, nullptr,
- webrtc::PeerConnectionDependencies(nullptr), nullptr);
-
- return caller_ && callee_;
- }
-
bool CreatePeerConnectionWrappersWithConfig(
const PeerConnectionInterface::RTCConfiguration& caller_config,
const PeerConnectionInterface::RTCConfiguration& callee_config) {
caller_ = CreatePeerConnectionWrapper(
- "Caller", nullptr, nullptr, &caller_config,
+ "Caller", nullptr, &caller_config,
webrtc::PeerConnectionDependencies(nullptr), nullptr);
callee_ = CreatePeerConnectionWrapper(
- "Callee", nullptr, nullptr, &callee_config,
+ "Callee", nullptr, &callee_config,
webrtc::PeerConnectionDependencies(nullptr), nullptr);
return caller_ && callee_;
}
@@ -1251,10 +1225,10 @@
const PeerConnectionInterface::RTCConfiguration& callee_config,
webrtc::PeerConnectionDependencies callee_dependencies) {
caller_ =
- CreatePeerConnectionWrapper("Caller", nullptr, nullptr, &caller_config,
+ CreatePeerConnectionWrapper("Caller", nullptr, &caller_config,
std::move(caller_dependencies), nullptr);
callee_ =
- CreatePeerConnectionWrapper("Callee", nullptr, nullptr, &callee_config,
+ CreatePeerConnectionWrapper("Callee", nullptr, &callee_config,
std::move(callee_dependencies), nullptr);
return caller_ && callee_;
}
@@ -1263,10 +1237,10 @@
const PeerConnectionFactory::Options& caller_options,
const PeerConnectionFactory::Options& callee_options) {
caller_ = CreatePeerConnectionWrapper(
- "Caller", nullptr, &caller_options, nullptr,
+ "Caller", &caller_options, nullptr,
webrtc::PeerConnectionDependencies(nullptr), nullptr);
callee_ = CreatePeerConnectionWrapper(
- "Callee", nullptr, &callee_options, nullptr,
+ "Callee", &callee_options, nullptr,
webrtc::PeerConnectionDependencies(nullptr), nullptr);
return caller_ && callee_;
}
@@ -1274,10 +1248,10 @@
bool CreatePeerConnectionWrappersWithFakeRtcEventLog() {
PeerConnectionInterface::RTCConfiguration default_config;
caller_ = CreatePeerConnectionWrapperWithFakeRtcEventLog(
- "Caller", nullptr, nullptr, &default_config,
+ "Caller", nullptr, &default_config,
webrtc::PeerConnectionDependencies(nullptr));
callee_ = CreatePeerConnectionWrapperWithFakeRtcEventLog(
- "Callee", nullptr, nullptr, &default_config,
+ "Callee", nullptr, &default_config,
webrtc::PeerConnectionDependencies(nullptr));
return caller_ && callee_;
}
@@ -1290,7 +1264,7 @@
webrtc::PeerConnectionDependencies dependencies(nullptr);
dependencies.cert_generator = std::move(cert_generator);
- return CreatePeerConnectionWrapper("New Peer", nullptr, nullptr, nullptr,
+ return CreatePeerConnectionWrapper("New Peer", nullptr, nullptr,
std::move(dependencies), nullptr);
}
@@ -2906,10 +2880,10 @@
// This test sets up a call between two parties with audio, video and an RTP
// data channel.
TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithRtpDataChannel) {
- FakeConstraints setup_constraints;
- setup_constraints.SetAllowRtpDataChannels();
- ASSERT_TRUE(CreatePeerConnectionWrappersWithConstraints(&setup_constraints,
- &setup_constraints));
+ PeerConnectionInterface::RTCConfiguration rtc_config;
+ rtc_config.enable_rtp_data_channel = true;
+ rtc_config.enable_dtls_srtp = false;
+ ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(rtc_config, rtc_config));
ConnectFakeSignaling();
// Expect that data channel created on caller side will show up for callee as
// well.
@@ -2942,10 +2916,10 @@
TEST_P(PeerConnectionIntegrationTest,
RtpDataChannelSignaledClosedInCalleeOffer) {
// Same procedure as above test.
- FakeConstraints setup_constraints;
- setup_constraints.SetAllowRtpDataChannels();
- ASSERT_TRUE(CreatePeerConnectionWrappersWithConstraints(&setup_constraints,
- &setup_constraints));
+ PeerConnectionInterface::RTCConfiguration rtc_config;
+ rtc_config.enable_rtp_data_channel = true;
+ rtc_config.enable_dtls_srtp = false;
+ ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(rtc_config, rtc_config));
ConnectFakeSignaling();
caller()->CreateDataChannel();
caller()->AddAudioVideoTracks();
@@ -2984,10 +2958,10 @@
virtual_socket_server()->set_delay_mean(5); // 5 ms per hop.
virtual_socket_server()->UpdateDelayDistribution();
- FakeConstraints constraints;
- constraints.SetAllowRtpDataChannels();
- ASSERT_TRUE(
- CreatePeerConnectionWrappersWithConstraints(&constraints, &constraints));
+ PeerConnectionInterface::RTCConfiguration rtc_config;
+ rtc_config.enable_rtp_data_channel = true;
+ rtc_config.enable_dtls_srtp = false;
+ ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(rtc_config, rtc_config));
ConnectFakeSignaling();
caller()->CreateDataChannel();
caller()->CreateAndSetAndSignalOffer();
@@ -3023,16 +2997,15 @@
// This test sets up a call between two parties with audio, video and but only
// the caller client supports RTP data channels.
TEST_P(PeerConnectionIntegrationTest, RtpDataChannelsRejectedByCallee) {
- FakeConstraints setup_constraints_1;
- setup_constraints_1.SetAllowRtpDataChannels();
+ PeerConnectionInterface::RTCConfiguration rtc_config_1;
+ rtc_config_1.enable_rtp_data_channel = true;
// Must disable DTLS to make negotiation succeed.
- setup_constraints_1.SetMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
- false);
- FakeConstraints setup_constraints_2;
- setup_constraints_2.SetMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
- false);
- ASSERT_TRUE(CreatePeerConnectionWrappersWithConstraints(
- &setup_constraints_1, &setup_constraints_2));
+ rtc_config_1.enable_dtls_srtp = false;
+ PeerConnectionInterface::RTCConfiguration rtc_config_2;
+ rtc_config_2.enable_dtls_srtp = false;
+ rtc_config_2.enable_dtls_srtp = false;
+ ASSERT_TRUE(
+ CreatePeerConnectionWrappersWithConfig(rtc_config_1, rtc_config_2));
ConnectFakeSignaling();
caller()->CreateDataChannel();
caller()->AddAudioVideoTracks();
@@ -3049,10 +3022,10 @@
// This test sets up a call between two parties with audio, and video. When
// audio and video is setup and flowing, an RTP data channel is negotiated.
TEST_P(PeerConnectionIntegrationTest, AddRtpDataChannelInSubsequentOffer) {
- FakeConstraints setup_constraints;
- setup_constraints.SetAllowRtpDataChannels();
- ASSERT_TRUE(CreatePeerConnectionWrappersWithConstraints(&setup_constraints,
- &setup_constraints));
+ PeerConnectionInterface::RTCConfiguration rtc_config;
+ rtc_config.enable_rtp_data_channel = true;
+ rtc_config.enable_dtls_srtp = false;
+ ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(rtc_config, rtc_config));
ConnectFakeSignaling();
// Do initial offer/answer with audio/video.
caller()->AddAudioVideoTracks();
diff --git a/pc/peerconnection_rampup_tests.cc b/pc/peerconnection_rampup_tests.cc
index 6d6a9fc..850212b 100644
--- a/pc/peerconnection_rampup_tests.cc
+++ b/pc/peerconnection_rampup_tests.cc
@@ -11,7 +11,6 @@
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/stats/rtcstats_objects.h"
-#include "api/test/fakeconstraints.h"
#include "api/video_codecs/builtin_video_decoder_factory.h"
#include "api/video_codecs/builtin_video_encoder_factory.h"
#include "p2p/base/testturnserver.h"
diff --git a/pc/peerconnectionendtoend_unittest.cc b/pc/peerconnectionendtoend_unittest.cc
index 7a8c2db..427529c 100644
--- a/pc/peerconnectionendtoend_unittest.cc
+++ b/pc/peerconnectionendtoend_unittest.cc
@@ -76,16 +76,13 @@
}
void CreatePcs(
- const MediaConstraintsInterface* pc_constraints,
rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory1,
rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory1,
rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory2,
rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory2) {
- EXPECT_TRUE(caller_->CreatePc(pc_constraints, config_,
- audio_encoder_factory1,
+ EXPECT_TRUE(caller_->CreatePc(config_, audio_encoder_factory1,
audio_decoder_factory1));
- EXPECT_TRUE(callee_->CreatePc(pc_constraints, config_,
- audio_encoder_factory2,
+ EXPECT_TRUE(callee_->CreatePc(config_, audio_encoder_factory2,
audio_decoder_factory2));
PeerConnectionTestWrapper::Connect(caller_.get(), callee_.get());
@@ -96,10 +93,9 @@
}
void CreatePcs(
- const MediaConstraintsInterface* pc_constraints,
rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory,
rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory) {
- CreatePcs(pc_constraints, audio_encoder_factory, audio_decoder_factory,
+ CreatePcs(audio_encoder_factory, audio_decoder_factory,
audio_encoder_factory, audio_decoder_factory);
}
@@ -117,7 +113,10 @@
callee_->GetAndAddUserMedia(audio, audio_options, video, video_constraints);
}
- void Negotiate() { caller_->CreateOffer(NULL); }
+ void Negotiate() {
+ caller_->CreateOffer(
+ webrtc::PeerConnectionInterface::RTCOfferAnswerOptions());
+ }
void WaitForCallEstablished() {
caller_->WaitForCallEstablished();
@@ -373,18 +372,16 @@
TEST_P(PeerConnectionEndToEndTest, Call) {
rtc::scoped_refptr<webrtc::AudioDecoderFactory> real_decoder_factory =
webrtc::CreateBuiltinAudioDecoderFactory();
- CreatePcs(nullptr, webrtc::CreateBuiltinAudioEncoderFactory(),
+ CreatePcs(webrtc::CreateBuiltinAudioEncoderFactory(),
CreateForwardingMockDecoderFactory(real_decoder_factory.get()));
GetAndAddUserMedia();
Negotiate();
WaitForCallEstablished();
}
-TEST_P(PeerConnectionEndToEndTest, CallWithLegacySdp) {
- FakeConstraints pc_constraints;
- pc_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
- false);
- CreatePcs(&pc_constraints, webrtc::CreateBuiltinAudioEncoderFactory(),
+TEST_P(PeerConnectionEndToEndTest, CallWithSdesKeyNegotiation) {
+ config_.enable_dtls_srtp = false;
+ CreatePcs(webrtc::CreateBuiltinAudioEncoderFactory(),
webrtc::CreateBuiltinAudioDecoderFactory());
GetAndAddUserMedia();
Negotiate();
@@ -446,8 +443,7 @@
std::vector<webrtc::AudioCodecPairId> encoder_id1, encoder_id2, decoder_id1,
decoder_id2;
- CreatePcs(nullptr,
- rtc::scoped_refptr<webrtc::AudioEncoderFactory>(
+ CreatePcs(rtc::scoped_refptr<webrtc::AudioEncoderFactory>(
new rtc::RefCountedObject<IdLoggingAudioEncoderFactory>(
webrtc::CreateAudioEncoderFactory<
AudioEncoderUnicornSparklesRainbow>(),
@@ -486,7 +482,7 @@
// Verifies that a DataChannel created before the negotiation can transition to
// "OPEN" and transfer data.
TEST_P(PeerConnectionEndToEndTest, CreateDataChannelBeforeNegotiate) {
- CreatePcs(nullptr, webrtc::CreateBuiltinAudioEncoderFactory(),
+ CreatePcs(webrtc::CreateBuiltinAudioEncoderFactory(),
webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
webrtc::DataChannelInit init;
@@ -511,7 +507,7 @@
// Verifies that a DataChannel created after the negotiation can transition to
// "OPEN" and transfer data.
TEST_P(PeerConnectionEndToEndTest, CreateDataChannelAfterNegotiate) {
- CreatePcs(nullptr, webrtc::CreateBuiltinAudioEncoderFactory(),
+ CreatePcs(webrtc::CreateBuiltinAudioEncoderFactory(),
webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
webrtc::DataChannelInit init;
@@ -543,7 +539,7 @@
// Verifies that a DataChannel created can transfer large messages.
TEST_P(PeerConnectionEndToEndTest, CreateDataChannelLargeTransfer) {
- CreatePcs(nullptr, webrtc::CreateBuiltinAudioEncoderFactory(),
+ CreatePcs(webrtc::CreateBuiltinAudioEncoderFactory(),
webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
webrtc::DataChannelInit init;
@@ -577,7 +573,7 @@
// Verifies that DataChannel IDs are even/odd based on the DTLS roles.
TEST_P(PeerConnectionEndToEndTest, DataChannelIdAssignment) {
- CreatePcs(nullptr, webrtc::CreateBuiltinAudioEncoderFactory(),
+ CreatePcs(webrtc::CreateBuiltinAudioEncoderFactory(),
webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
webrtc::DataChannelInit init;
@@ -605,7 +601,7 @@
// there are multiple DataChannels.
TEST_P(PeerConnectionEndToEndTest,
MessageTransferBetweenTwoPairsOfDataChannels) {
- CreatePcs(nullptr, webrtc::CreateBuiltinAudioEncoderFactory(),
+ CreatePcs(webrtc::CreateBuiltinAudioEncoderFactory(),
webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
webrtc::DataChannelInit init;
@@ -645,7 +641,7 @@
// channel, and the closed channel was incorrectly still assigned to the ID.
TEST_P(PeerConnectionEndToEndTest,
DataChannelFromOpenWorksAfterPreviousChannelClosed) {
- CreatePcs(nullptr, webrtc::CreateBuiltinAudioEncoderFactory(),
+ CreatePcs(webrtc::CreateBuiltinAudioEncoderFactory(),
webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
webrtc::DataChannelInit init;
@@ -678,7 +674,7 @@
// closing before creating the second one.
TEST_P(PeerConnectionEndToEndTest,
DataChannelFromOpenWorksWhilePreviousChannelClosing) {
- CreatePcs(nullptr, webrtc::CreateBuiltinAudioEncoderFactory(),
+ CreatePcs(webrtc::CreateBuiltinAudioEncoderFactory(),
webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
webrtc::DataChannelInit init;
@@ -709,7 +705,7 @@
// reference count), no memory access violation will occur.
// See: https://code.google.com/p/chromium/issues/detail?id=565048
TEST_P(PeerConnectionEndToEndTest, CloseDataChannelRemotelyWhileNotReferenced) {
- CreatePcs(nullptr, webrtc::CreateBuiltinAudioEncoderFactory(),
+ CreatePcs(webrtc::CreateBuiltinAudioEncoderFactory(),
webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
webrtc::DataChannelInit init;
diff --git a/pc/peerconnectionfactory.cc b/pc/peerconnectionfactory.cc
index d407228..b78ed8d 100644
--- a/pc/peerconnectionfactory.cc
+++ b/pc/peerconnectionfactory.cc
@@ -331,23 +331,6 @@
rtc::scoped_refptr<PeerConnectionInterface>
PeerConnectionFactory::CreatePeerConnection(
- const PeerConnectionInterface::RTCConfiguration& configuration_in,
- const MediaConstraintsInterface* constraints,
- std::unique_ptr<cricket::PortAllocator> allocator,
- std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
- PeerConnectionObserver* observer) {
- RTC_DCHECK(signaling_thread_->IsCurrent());
-
- // We merge constraints and configuration into a single configuration.
- PeerConnectionInterface::RTCConfiguration configuration = configuration_in;
- CopyConstraintsIntoRtcConfiguration(constraints, &configuration);
-
- return CreatePeerConnection(configuration, std::move(allocator),
- std::move(cert_generator), observer);
-}
-
-rtc::scoped_refptr<PeerConnectionInterface>
-PeerConnectionFactory::CreatePeerConnection(
const PeerConnectionInterface::RTCConfiguration& configuration,
std::unique_ptr<cricket::PortAllocator> allocator,
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
diff --git a/pc/peerconnectionfactory.h b/pc/peerconnectionfactory.h
index 82fa508..f36a4e0 100644
--- a/pc/peerconnectionfactory.h
+++ b/pc/peerconnectionfactory.h
@@ -42,14 +42,6 @@
void SetOptions(const Options& options) override;
- // Deprecated, use version without constraints.
- rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
- const PeerConnectionInterface::RTCConfiguration& configuration,
- const MediaConstraintsInterface* constraints,
- std::unique_ptr<cricket::PortAllocator> allocator,
- std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
- PeerConnectionObserver* observer) override;
-
rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
const PeerConnectionInterface::RTCConfiguration& configuration,
std::unique_ptr<cricket::PortAllocator> allocator,
diff --git a/pc/peerconnectionfactory_unittest.cc b/pc/peerconnectionfactory_unittest.cc
index 773ba01..84c828f 100644
--- a/pc/peerconnectionfactory_unittest.cc
+++ b/pc/peerconnectionfactory_unittest.cc
@@ -178,7 +178,7 @@
std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
new FakeRTCCertificateGenerator());
rtc::scoped_refptr<PeerConnectionInterface> pc(factory->CreatePeerConnection(
- config, nullptr, nullptr, std::move(cert_generator), &observer));
+ config, nullptr, std::move(cert_generator), &observer));
EXPECT_TRUE(pc.get() != nullptr);
}
@@ -264,9 +264,9 @@
config.servers.push_back(ice_server);
std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
new FakeRTCCertificateGenerator());
- rtc::scoped_refptr<PeerConnectionInterface> pc(factory_->CreatePeerConnection(
- config, nullptr, std::move(port_allocator_), std::move(cert_generator),
- &observer_));
+ rtc::scoped_refptr<PeerConnectionInterface> pc(
+ factory_->CreatePeerConnection(config, std::move(port_allocator_),
+ std::move(cert_generator), &observer_));
ASSERT_TRUE(pc.get() != NULL);
cricket::ServerAddresses stun_servers;
rtc::SocketAddress stun1("stun.l.google.com", 19302);
@@ -294,9 +294,9 @@
config.servers.push_back(ice_server);
std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
new FakeRTCCertificateGenerator());
- rtc::scoped_refptr<PeerConnectionInterface> pc(factory_->CreatePeerConnection(
- config, nullptr, std::move(port_allocator_), std::move(cert_generator),
- &observer_));
+ rtc::scoped_refptr<PeerConnectionInterface> pc(
+ factory_->CreatePeerConnection(config, std::move(port_allocator_),
+ std::move(cert_generator), &observer_));
ASSERT_TRUE(pc.get() != NULL);
cricket::ServerAddresses stun_servers;
rtc::SocketAddress stun1("stun.l.google.com", 19302);
@@ -323,9 +323,9 @@
config.servers.push_back(ice_server);
std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
new FakeRTCCertificateGenerator());
- rtc::scoped_refptr<PeerConnectionInterface> pc(factory_->CreatePeerConnection(
- config, nullptr, std::move(port_allocator_), std::move(cert_generator),
- &observer_));
+ rtc::scoped_refptr<PeerConnectionInterface> pc(
+ factory_->CreatePeerConnection(config, std::move(port_allocator_),
+ std::move(cert_generator), &observer_));
ASSERT_TRUE(pc.get() != NULL);
std::vector<cricket::RelayServerConfig> turn_servers;
cricket::RelayServerConfig turn("test.com", 1234, kTurnUsername,
@@ -344,9 +344,9 @@
config.servers.push_back(ice_server);
std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
new FakeRTCCertificateGenerator());
- rtc::scoped_refptr<PeerConnectionInterface> pc(factory_->CreatePeerConnection(
- config, nullptr, std::move(port_allocator_), std::move(cert_generator),
- &observer_));
+ rtc::scoped_refptr<PeerConnectionInterface> pc(
+ factory_->CreatePeerConnection(config, std::move(port_allocator_),
+ std::move(cert_generator), &observer_));
ASSERT_TRUE(pc.get() != NULL);
std::vector<cricket::RelayServerConfig> turn_servers;
cricket::RelayServerConfig turn("hello.com", kDefaultStunPort, "test",
@@ -369,9 +369,9 @@
config.servers.push_back(ice_server);
std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
new FakeRTCCertificateGenerator());
- rtc::scoped_refptr<PeerConnectionInterface> pc(factory_->CreatePeerConnection(
- config, nullptr, std::move(port_allocator_), std::move(cert_generator),
- &observer_));
+ rtc::scoped_refptr<PeerConnectionInterface> pc(
+ factory_->CreatePeerConnection(config, std::move(port_allocator_),
+ std::move(cert_generator), &observer_));
ASSERT_TRUE(pc.get() != NULL);
std::vector<cricket::RelayServerConfig> turn_servers;
cricket::RelayServerConfig turn1("hello.com", kDefaultStunTlsPort, "test",
@@ -404,9 +404,9 @@
config.servers.push_back(ice_server);
std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
new FakeRTCCertificateGenerator());
- rtc::scoped_refptr<PeerConnectionInterface> pc(factory_->CreatePeerConnection(
- config, nullptr, std::move(port_allocator_), std::move(cert_generator),
- &observer_));
+ rtc::scoped_refptr<PeerConnectionInterface> pc(
+ factory_->CreatePeerConnection(config, std::move(port_allocator_),
+ std::move(cert_generator), &observer_));
ASSERT_TRUE(pc.get() != NULL);
cricket::ServerAddresses stun_servers;
rtc::SocketAddress stun1("1.2.3.4", 1234);
diff --git a/pc/peerconnectioninterface_unittest.cc b/pc/peerconnectioninterface_unittest.cc
index 1beb347..2f6c0ef 100644
--- a/pc/peerconnectioninterface_unittest.cc
+++ b/pc/peerconnectioninterface_unittest.cc
@@ -22,7 +22,6 @@
#include "api/peerconnectioninterface.h"
#include "api/rtpreceiverinterface.h"
#include "api/rtpsenderinterface.h"
-#include "api/test/fakeconstraints.h"
#include "api/video_codecs/builtin_video_decoder_factory.h"
#include "api/video_codecs/builtin_video_encoder_factory.h"
#include "logging/rtc_event_log/output/rtc_event_log_output_file.h"
@@ -417,7 +416,6 @@
using webrtc::AudioTrackInterface;
using webrtc::DataBuffer;
using webrtc::DataChannelInterface;
-using webrtc::FakeConstraints;
using webrtc::IceCandidateInterface;
using webrtc::MediaConstraintsInterface;
using webrtc::MediaStream;
@@ -707,30 +705,23 @@
}
void CreatePeerConnection() {
- CreatePeerConnection(PeerConnectionInterface::RTCConfiguration(), nullptr);
+ CreatePeerConnection(PeerConnectionInterface::RTCConfiguration());
}
// DTLS does not work in a loopback call, so is disabled for most of the
// tests in this file.
void CreatePeerConnectionWithoutDtls() {
- FakeConstraints no_dtls_constraints;
- no_dtls_constraints.AddMandatory(
- webrtc::MediaConstraintsInterface::kEnableDtlsSrtp, false);
+ RTCConfiguration config;
+ config.enable_dtls_srtp = false;
- CreatePeerConnection(PeerConnectionInterface::RTCConfiguration(),
- &no_dtls_constraints);
- }
-
- void CreatePeerConnection(webrtc::MediaConstraintsInterface* constraints) {
- CreatePeerConnection(PeerConnectionInterface::RTCConfiguration(),
- constraints);
+ CreatePeerConnection(config);
}
void CreatePeerConnectionWithIceTransportsType(
PeerConnectionInterface::IceTransportsType type) {
PeerConnectionInterface::RTCConfiguration config;
config.type = type;
- return CreatePeerConnection(config, nullptr);
+ return CreatePeerConnection(config);
}
void CreatePeerConnectionWithIceServer(const std::string& uri,
@@ -740,12 +731,10 @@
server.uri = uri;
server.password = password;
config.servers.push_back(server);
- CreatePeerConnection(config, nullptr);
+ CreatePeerConnection(config);
}
- void CreatePeerConnection(
- const PeerConnectionInterface::RTCConfiguration& config,
- webrtc::MediaConstraintsInterface* constraints) {
+ void CreatePeerConnection(const RTCConfiguration& config) {
std::unique_ptr<cricket::FakePortAllocator> port_allocator(
new cricket::FakePortAllocator(rtc::Thread::Current(), nullptr));
port_allocator_ = port_allocator.get();
@@ -753,19 +742,16 @@
// Create certificate generator unless DTLS constraint is explicitly set to
// false.
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator;
- bool dtls;
- if (FindConstraint(constraints,
- webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
- &dtls, nullptr) &&
- dtls) {
+
+ if (config.enable_dtls_srtp.value_or(true)) {
fake_certificate_generator_ = new FakeRTCCertificateGenerator();
cert_generator.reset(fake_certificate_generator_);
}
RTCConfiguration modified_config = config;
modified_config.sdp_semantics = sdp_semantics_;
pc_ = pc_factory_->CreatePeerConnection(
- modified_config, constraints, std::move(port_allocator),
- std::move(cert_generator), &observer_);
+ modified_config, std::move(port_allocator), std::move(cert_generator),
+ &observer_);
ASSERT_TRUE(pc_.get() != NULL);
observer_.SetPeerConnectionInterface(pc_.get());
EXPECT_EQ(PeerConnectionInterface::kStable, observer_.state_);
@@ -778,8 +764,7 @@
config.servers.push_back(server);
config.sdp_semantics = sdp_semantics_;
rtc::scoped_refptr<PeerConnectionInterface> pc =
- pc_factory_->CreatePeerConnection(config, nullptr, nullptr, nullptr,
- &observer_);
+ pc_factory_->CreatePeerConnection(config, nullptr, nullptr, &observer_);
EXPECT_EQ(nullptr, pc);
}
@@ -895,14 +880,14 @@
}
bool DoCreateOfferAnswer(std::unique_ptr<SessionDescriptionInterface>* desc,
- bool offer,
- MediaConstraintsInterface* constraints) {
+ const RTCOfferAnswerOptions* options,
+ bool offer) {
rtc::scoped_refptr<MockCreateSessionDescriptionObserver> observer(
new rtc::RefCountedObject<MockCreateSessionDescriptionObserver>());
if (offer) {
- pc_->CreateOffer(observer, constraints);
+ pc_->CreateOffer(observer, options ? *options : RTCOfferAnswerOptions());
} else {
- pc_->CreateAnswer(observer, constraints);
+ pc_->CreateAnswer(observer, options ? *options : RTCOfferAnswerOptions());
}
EXPECT_EQ_WAIT(true, observer->called(), kTimeout);
*desc = observer->MoveDescription();
@@ -910,13 +895,13 @@
}
bool DoCreateOffer(std::unique_ptr<SessionDescriptionInterface>* desc,
- MediaConstraintsInterface* constraints) {
- return DoCreateOfferAnswer(desc, true, constraints);
+ const RTCOfferAnswerOptions* options) {
+ return DoCreateOfferAnswer(desc, options, true);
}
bool DoCreateAnswer(std::unique_ptr<SessionDescriptionInterface>* desc,
- MediaConstraintsInterface* constraints) {
- return DoCreateOfferAnswer(desc, false, constraints);
+ const RTCOfferAnswerOptions* options) {
+ return DoCreateOfferAnswer(desc, options, false);
}
bool DoSetSessionDescription(
@@ -1359,7 +1344,7 @@
config.candidate_network_policy =
PeerConnectionInterface::kCandidateNetworkPolicyLowCost;
config.ice_candidate_pool_size = 1;
- CreatePeerConnection(config, nullptr);
+ CreatePeerConnection(config);
const cricket::FakePortAllocatorSession* session =
static_cast<const cricket::FakePortAllocatorSession*>(
@@ -1411,8 +1396,8 @@
webrtc::CreateBuiltinVideoDecoderFactory(), nullptr /* audio_mixer */,
nullptr /* audio_processing */));
rtc::scoped_refptr<PeerConnectionInterface> pc(
- pc_factory->CreatePeerConnection(
- config, nullptr, std::move(port_allocator), nullptr, &observer_));
+ pc_factory->CreatePeerConnection(config, std::move(port_allocator),
+ nullptr, &observer_));
// Now validate that the config fields set above were applied to the
// PortAllocator, as flags or otherwise.
@@ -1430,7 +1415,7 @@
TEST_P(PeerConnectionInterfaceTest, GetConfigurationAfterCreatePeerConnection) {
PeerConnectionInterface::RTCConfiguration config;
config.type = PeerConnectionInterface::kRelay;
- CreatePeerConnection(config, nullptr);
+ CreatePeerConnection(config);
PeerConnectionInterface::RTCConfiguration returned_config =
pc_->GetConfiguration();
@@ -1930,9 +1915,10 @@
// This test setup two RTP data channels in loop back.
TEST_P(PeerConnectionInterfaceTest, TestDataChannel) {
- FakeConstraints constraints;
- constraints.SetAllowRtpDataChannels();
- CreatePeerConnection(&constraints);
+ RTCConfiguration config;
+ config.enable_rtp_data_channel = true;
+ config.enable_dtls_srtp = false;
+ CreatePeerConnection(config);
rtc::scoped_refptr<DataChannelInterface> data1 =
pc_->CreateDataChannel("test1", NULL);
rtc::scoped_refptr<DataChannelInterface> data2 =
@@ -1977,9 +1963,10 @@
// This test verifies that sendnig binary data over RTP data channels should
// fail.
TEST_P(PeerConnectionInterfaceTest, TestSendBinaryOnRtpDataChannel) {
- FakeConstraints constraints;
- constraints.SetAllowRtpDataChannels();
- CreatePeerConnection(&constraints);
+ RTCConfiguration config;
+ config.enable_rtp_data_channel = true;
+ config.enable_dtls_srtp = false;
+ CreatePeerConnection(config);
rtc::scoped_refptr<DataChannelInterface> data1 =
pc_->CreateDataChannel("test1", NULL);
rtc::scoped_refptr<DataChannelInterface> data2 =
@@ -2007,9 +1994,10 @@
// This test setup a RTP data channels in loop back and test that a channel is
// opened even if the remote end answer with a zero SSRC.
TEST_P(PeerConnectionInterfaceTest, TestSendOnlyDataChannel) {
- FakeConstraints constraints;
- constraints.SetAllowRtpDataChannels();
- CreatePeerConnection(&constraints);
+ RTCConfiguration config;
+ config.enable_rtp_data_channel = true;
+ config.enable_dtls_srtp = false;
+ CreatePeerConnection(config);
rtc::scoped_refptr<DataChannelInterface> data1 =
pc_->CreateDataChannel("test1", NULL);
std::unique_ptr<MockDataChannelObserver> observer1(
@@ -2029,9 +2017,11 @@
// This test that if a data channel is added in an answer a receive only channel
// channel is created.
TEST_P(PeerConnectionInterfaceTest, TestReceiveOnlyDataChannel) {
- FakeConstraints constraints;
- constraints.SetAllowRtpDataChannels();
- CreatePeerConnection(&constraints);
+ RTCConfiguration config;
+ config.enable_rtp_data_channel = true;
+ config.enable_dtls_srtp = false;
+
+ CreatePeerConnection(config);
std::string offer_label = "offer_channel";
rtc::scoped_refptr<DataChannelInterface> offer_channel =
@@ -2069,9 +2059,9 @@
// requested.
// TODO(perkj): Remove this test once reliable channels are implemented.
TEST_P(PeerConnectionInterfaceTest, CreateReliableRtpDataChannelShouldFail) {
- FakeConstraints constraints;
- constraints.SetAllowRtpDataChannels();
- CreatePeerConnection(&constraints);
+ RTCConfiguration rtc_config;
+ rtc_config.enable_rtp_data_channel = true;
+ CreatePeerConnection(rtc_config);
std::string label = "test";
webrtc::DataChannelInit config;
@@ -2083,9 +2073,9 @@
// Verifies that duplicated label is not allowed for RTP data channel.
TEST_P(PeerConnectionInterfaceTest, RtpDuplicatedLabelNotAllowed) {
- FakeConstraints constraints;
- constraints.SetAllowRtpDataChannels();
- CreatePeerConnection(&constraints);
+ RTCConfiguration config;
+ config.enable_rtp_data_channel = true;
+ CreatePeerConnection(config);
std::string label = "test";
rtc::scoped_refptr<DataChannelInterface> channel =
@@ -2100,9 +2090,9 @@
// This tests that a SCTP data channel is returned using different
// DataChannelInit configurations.
TEST_P(PeerConnectionInterfaceTest, CreateSctpDataChannel) {
- FakeConstraints constraints;
- constraints.SetAllowDtlsSctpDataChannels();
- CreatePeerConnection(&constraints);
+ RTCConfiguration rtc_config;
+ rtc_config.enable_dtls_srtp = true;
+ CreatePeerConnection(rtc_config);
webrtc::DataChannelInit config;
@@ -2138,9 +2128,9 @@
// maxRetransmitTime are set for SCTP data channels.
TEST_P(PeerConnectionInterfaceTest,
CreateSctpDataChannelShouldFailForInvalidConfig) {
- FakeConstraints constraints;
- constraints.SetAllowDtlsSctpDataChannels();
- CreatePeerConnection(&constraints);
+ RTCConfiguration rtc_config;
+ rtc_config.enable_dtls_srtp = true;
+ CreatePeerConnection(rtc_config);
std::string label = "test";
webrtc::DataChannelInit config;
@@ -2156,9 +2146,9 @@
// or out of range should fail.
TEST_P(PeerConnectionInterfaceTest,
CreateSctpDataChannelWithInvalidIdShouldFail) {
- FakeConstraints constraints;
- constraints.SetAllowDtlsSctpDataChannels();
- CreatePeerConnection(&constraints);
+ RTCConfiguration rtc_config;
+ rtc_config.enable_dtls_srtp = true;
+ CreatePeerConnection(rtc_config);
webrtc::DataChannelInit config;
rtc::scoped_refptr<DataChannelInterface> channel;
@@ -2183,10 +2173,9 @@
// Verifies that duplicated label is allowed for SCTP data channel.
TEST_P(PeerConnectionInterfaceTest, SctpDuplicatedLabelAllowed) {
- FakeConstraints constraints;
- constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
- true);
- CreatePeerConnection(&constraints);
+ RTCConfiguration rtc_config;
+ rtc_config.enable_dtls_srtp = true;
+ CreatePeerConnection(rtc_config);
std::string label = "test";
rtc::scoped_refptr<DataChannelInterface> channel =
@@ -2201,9 +2190,10 @@
// This test verifies that OnRenegotiationNeeded is fired for every new RTP
// DataChannel.
TEST_P(PeerConnectionInterfaceTest, RenegotiationNeededForNewRtpDataChannel) {
- FakeConstraints constraints;
- constraints.SetAllowRtpDataChannels();
- CreatePeerConnection(&constraints);
+ RTCConfiguration rtc_config;
+ rtc_config.enable_rtp_data_channel = true;
+ rtc_config.enable_dtls_srtp = false;
+ CreatePeerConnection(rtc_config);
rtc::scoped_refptr<DataChannelInterface> dc1 =
pc_->CreateDataChannel("test1", NULL);
@@ -2217,9 +2207,10 @@
// This test that a data channel closes when a PeerConnection is deleted/closed.
TEST_P(PeerConnectionInterfaceTest, DataChannelCloseWhenPeerConnectionClose) {
- FakeConstraints constraints;
- constraints.SetAllowRtpDataChannels();
- CreatePeerConnection(&constraints);
+ RTCConfiguration rtc_config;
+ rtc_config.enable_rtp_data_channel = true;
+ rtc_config.enable_dtls_srtp = false;
+ CreatePeerConnection(rtc_config);
rtc::scoped_refptr<DataChannelInterface> data1 =
pc_->CreateDataChannel("test1", NULL);
@@ -2242,9 +2233,10 @@
// This tests that RTP data channels can be rejected in an answer.
TEST_P(PeerConnectionInterfaceTest, TestRejectRtpDataChannelInAnswer) {
- FakeConstraints constraints;
- constraints.SetAllowRtpDataChannels();
- CreatePeerConnection(&constraints);
+ RTCConfiguration rtc_config;
+ rtc_config.enable_rtp_data_channel = true;
+ rtc_config.enable_dtls_srtp = false;
+ CreatePeerConnection(rtc_config);
rtc::scoped_refptr<DataChannelInterface> offer_channel(
pc_->CreateDataChannel("offer_channel", NULL));
@@ -2272,8 +2264,8 @@
TEST_P(PeerConnectionInterfaceTest, DISABLED_TestRejectSctpDataChannelInAnswer)
#endif
{
- FakeConstraints constraints;
- CreatePeerConnection(&constraints);
+ RTCConfiguration rtc_config;
+ CreatePeerConnection(rtc_config);
rtc::scoped_refptr<DataChannelInterface> offer_channel(
pc_->CreateDataChannel("offer_channel", NULL));
@@ -2298,10 +2290,9 @@
// FireFox, use it as a remote session description, generate an answer and use
// the answer as a local description.
TEST_P(PeerConnectionInterfaceTest, ReceiveFireFoxOffer) {
- FakeConstraints constraints;
- constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
- true);
- CreatePeerConnection(&constraints);
+ RTCConfiguration rtc_config;
+ rtc_config.enable_dtls_srtp = true;
+ CreatePeerConnection(rtc_config);
AddAudioTrack("audio_label");
AddVideoTrack("video_label");
std::unique_ptr<SessionDescriptionInterface> desc(
@@ -2333,10 +2324,9 @@
// The fallback was previously supported but was removed to simplify the code
// and because it's non-standard.
TEST_P(PeerConnectionInterfaceTest, DtlsSdesFallbackNotSupported) {
- FakeConstraints constraints;
- constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
- true);
- CreatePeerConnection(&constraints);
+ RTCConfiguration rtc_config;
+ rtc_config.enable_dtls_srtp = true;
+ CreatePeerConnection(rtc_config);
// Wait for fake certificate to be generated. Previously, this is what caused
// the "a=crypto" lines to be rejected.
AddAudioTrack("audio_label");
@@ -2378,10 +2368,9 @@
// Test that if we're receiving (but not sending) a track, subsequent offers
// will have m-lines with a=recvonly.
TEST_P(PeerConnectionInterfaceTest, CreateSubsequentRecvOnlyOffer) {
- FakeConstraints constraints;
- constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
- true);
- CreatePeerConnection(&constraints);
+ RTCConfiguration rtc_config;
+ rtc_config.enable_dtls_srtp = true;
+ CreatePeerConnection(rtc_config);
CreateAndSetRemoteOffer(GetSdpStringWithStream1());
CreateAnswerAsLocalDescription();
@@ -2405,10 +2394,9 @@
// offerToReceiveVideo/offerToReceiveAudio constraints are explicitly set to
// false, the generated m-lines will be a=inactive.
TEST_P(PeerConnectionInterfaceTest, CreateSubsequentInactiveOffer) {
- FakeConstraints constraints;
- constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
- true);
- CreatePeerConnection(&constraints);
+ RTCConfiguration rtc_config;
+ rtc_config.enable_dtls_srtp = true;
+ CreatePeerConnection(rtc_config);
CreateAndSetRemoteOffer(GetSdpStringWithStream1());
CreateAnswerAsLocalDescription();
@@ -2416,12 +2404,10 @@
// A new offer would be recvonly, but we'll set the "no receive" constraints
// to make it inactive.
std::unique_ptr<SessionDescriptionInterface> offer;
- FakeConstraints offer_constraints;
- offer_constraints.AddMandatory(
- webrtc::MediaConstraintsInterface::kOfferToReceiveVideo, false);
- offer_constraints.AddMandatory(
- webrtc::MediaConstraintsInterface::kOfferToReceiveAudio, false);
- DoCreateOffer(&offer, &offer_constraints);
+ RTCOfferAnswerOptions options;
+ options.offer_to_receive_audio = 0;
+ options.offer_to_receive_video = 0;
+ DoCreateOffer(&offer, &options);
const cricket::ContentInfo* video_content =
cricket::GetFirstVideoContent(offer->description());
@@ -2461,7 +2447,7 @@
TEST_P(PeerConnectionInterfaceTest, SetConfigurationChangesPruneTurnPortsFlag) {
PeerConnectionInterface::RTCConfiguration config;
config.prune_turn_ports = false;
- CreatePeerConnection(config, nullptr);
+ CreatePeerConnection(config);
config = pc_->GetConfiguration();
EXPECT_FALSE(port_allocator_->prune_turn_ports());
@@ -2476,7 +2462,7 @@
TEST_P(PeerConnectionInterfaceTest, SetConfigurationChangesIceCheckInterval) {
PeerConnectionInterface::RTCConfiguration config;
config.ice_check_min_interval = absl::nullopt;
- CreatePeerConnection(config, nullptr);
+ CreatePeerConnection(config);
config = pc_->GetConfiguration();
config.ice_check_min_interval = 100;
EXPECT_TRUE(pc_->SetConfiguration(config));
@@ -2573,7 +2559,7 @@
config.bundle_policy = PeerConnectionInterface::kBundlePolicyBalanced;
config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyNegotiate;
config.continual_gathering_policy = PeerConnectionInterface::GATHER_ONCE;
- CreatePeerConnection(config, nullptr);
+ CreatePeerConnection(config);
PeerConnectionInterface::RTCConfiguration modified_config =
pc_->GetConfiguration();
@@ -2603,7 +2589,7 @@
TEST_P(PeerConnectionInterfaceTest,
SetConfigurationReturnsRangeErrorForBadCandidatePoolSize) {
PeerConnectionInterface::RTCConfiguration config;
- CreatePeerConnection(config, nullptr);
+ CreatePeerConnection(config);
config = pc_->GetConfiguration();
config.ice_candidate_pool_size = -1;
@@ -2622,7 +2608,7 @@
TEST_P(PeerConnectionInterfaceTest,
SetConfigurationReturnsSyntaxErrorFromBadIceUrls) {
PeerConnectionInterface::RTCConfiguration config;
- CreatePeerConnection(config, nullptr);
+ CreatePeerConnection(config);
config = pc_->GetConfiguration();
PeerConnectionInterface::IceServer bad_server;
@@ -2638,7 +2624,7 @@
TEST_P(PeerConnectionInterfaceTest,
SetConfigurationReturnsInvalidParameterIfCredentialsMissing) {
PeerConnectionInterface::RTCConfiguration config;
- CreatePeerConnection(config, nullptr);
+ CreatePeerConnection(config);
config = pc_->GetConfiguration();
PeerConnectionInterface::IceServer bad_server;
@@ -2753,10 +2739,9 @@
// SDP string is created. In this test the two separate MediaStreams are
// signaled.
TEST_P(PeerConnectionInterfaceTest, UpdateRemoteStreams) {
- FakeConstraints constraints;
- constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
- true);
- CreatePeerConnection(&constraints);
+ RTCConfiguration config;
+ config.enable_dtls_srtp = true;
+ CreatePeerConnection(config);
CreateAndSetRemoteOffer(GetSdpStringWithStream1());
rtc::scoped_refptr<StreamCollection> reference(CreateStreamCollection(1, 1));
@@ -2780,10 +2765,9 @@
// specific behavior.
TEST_F(PeerConnectionInterfaceTestPlanB,
AddRemoveTrackFromExistingRemoteMediaStream) {
- FakeConstraints constraints;
- constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
- true);
- CreatePeerConnection(&constraints);
+ RTCConfiguration config;
+ config.enable_dtls_srtp = true;
+ CreatePeerConnection(config);
std::unique_ptr<SessionDescriptionInterface> desc_ms1 =
CreateSessionDescriptionAndReference(1, 1);
EXPECT_TRUE(DoSetRemoteDescription(std::move(desc_ms1)));
@@ -2824,10 +2808,9 @@
// This tests that remote tracks are ended if a local session description is set
// that rejects the media content type.
TEST_P(PeerConnectionInterfaceTest, RejectMediaContent) {
- FakeConstraints constraints;
- constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
- true);
- CreatePeerConnection(&constraints);
+ RTCConfiguration config;
+ config.enable_dtls_srtp = true;
+ CreatePeerConnection(config);
// First create and set a remote offer, then reject its video content in our
// answer.
CreateAndSetRemoteOffer(kSdpStringWithStream1PlanB);
@@ -2874,10 +2857,9 @@
// of PeerConnection and then PeerConnection tries to reject the track.
// Don't run under Unified Plan since the stream API is not available.
TEST_F(PeerConnectionInterfaceTestPlanB, RemoveTrackThenRejectMediaContent) {
- FakeConstraints constraints;
- constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
- true);
- CreatePeerConnection(&constraints);
+ RTCConfiguration config;
+ config.enable_dtls_srtp = true;
+ CreatePeerConnection(config);
CreateAndSetRemoteOffer(GetSdpStringWithStream1());
MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
remote_stream->RemoveTrack(remote_stream->GetVideoTracks()[0]);
@@ -2901,10 +2883,9 @@
// will be created, even if the description contains SSRCs/MSIDs.
// See: https://code.google.com/p/webrtc/issues/detail?id=5054
TEST_P(PeerConnectionInterfaceTest, RecvonlyDescriptionDoesntCreateStream) {
- FakeConstraints constraints;
- constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
- true);
- CreatePeerConnection(&constraints);
+ RTCConfiguration config;
+ config.enable_dtls_srtp = true;
+ CreatePeerConnection(config);
std::string recvonly_offer = GetSdpStringWithStream1();
rtc::replace_substrs(kSendrecv, strlen(kSendrecv), kRecvonly,
@@ -2920,10 +2901,9 @@
// in a subsequent session description.
// Don't run under Unified Plan since this behavior is Plan B specific.
TEST_F(PeerConnectionInterfaceTestPlanB, SdpWithoutMsidCreatesDefaultStream) {
- FakeConstraints constraints;
- constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
- true);
- CreatePeerConnection(&constraints);
+ RTCConfiguration config;
+ config.enable_dtls_srtp = true;
+ CreatePeerConnection(config);
CreateAndSetRemoteOffer(kSdpStringWithoutStreamsAudioOnly);
ASSERT_EQ(1u, observer_.remote_streams()->count());
@@ -2950,10 +2930,9 @@
// Don't run under Unified Plan since this behavior is Plan B specific.
TEST_F(PeerConnectionInterfaceTestPlanB,
SendOnlySdpWithoutMsidCreatesDefaultStream) {
- FakeConstraints constraints;
- constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
- true);
- CreatePeerConnection(&constraints);
+ RTCConfiguration config;
+ config.enable_dtls_srtp = true;
+ CreatePeerConnection(config);
CreateAndSetRemoteOffer(kSdpStringSendOnlyWithoutStreams);
ASSERT_EQ(1u, observer_.remote_streams()->count());
@@ -2968,10 +2947,9 @@
// a remote track that as already been removed from the MediaStream.
// Don't run under Unified Plan since this behavior is Plan B specific.
TEST_F(PeerConnectionInterfaceTestPlanB, RemoveAlreadyGoneRemoteStream) {
- FakeConstraints constraints;
- constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
- true);
- CreatePeerConnection(&constraints);
+ RTCConfiguration config;
+ config.enable_dtls_srtp = true;
+ CreatePeerConnection(config);
CreateAndSetRemoteOffer(GetSdpStringWithStream1());
MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
remote_stream->RemoveTrack(remote_stream->GetAudioTracks()[0]);
@@ -2988,10 +2966,9 @@
// Don't run under Unified Plan since this behavior is Plan B specific.
TEST_F(PeerConnectionInterfaceTestPlanB,
SdpWithoutMsidAndStreamsCreatesDefaultStream) {
- FakeConstraints constraints;
- constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
- true);
- CreatePeerConnection(&constraints);
+ RTCConfiguration config;
+ config.enable_dtls_srtp = true;
+ CreatePeerConnection(config);
CreateAndSetRemoteOffer(kSdpStringWithoutStreams);
ASSERT_EQ(1u, observer_.remote_streams()->count());
@@ -3004,10 +2981,9 @@
// description doesn't contain any streams but does support MSID.
// Don't run under Unified Plan since this behavior is Plan B specific.
TEST_F(PeerConnectionInterfaceTestPlanB, SdpWithMsidDontCreatesDefaultStream) {
- FakeConstraints constraints;
- constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
- true);
- CreatePeerConnection(&constraints);
+ RTCConfiguration config;
+ config.enable_dtls_srtp = true;
+ CreatePeerConnection(config);
CreateAndSetRemoteOffer(kSdpStringWithMsidWithoutStreams);
EXPECT_EQ(0u, observer_.remote_streams()->count());
}
@@ -3018,10 +2994,9 @@
// Don't run under Unified Plan since this behavior is Plan B specific.
TEST_F(PeerConnectionInterfaceTestPlanB,
DefaultTracksNotDestroyedAndRecreated) {
- FakeConstraints constraints;
- constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
- true);
- CreatePeerConnection(&constraints);
+ RTCConfiguration config;
+ config.enable_dtls_srtp = true;
+ CreatePeerConnection(config);
CreateAndSetRemoteOffer(kSdpStringWithoutStreamsAudioOnly);
ASSERT_EQ(1u, observer_.remote_streams()->count());
@@ -3040,10 +3015,9 @@
// description is updated to not have any MediaStreams.
// Don't run under Unified Plan since this behavior is Plan B specific.
TEST_F(PeerConnectionInterfaceTestPlanB, VerifyDefaultStreamIsNotCreated) {
- FakeConstraints constraints;
- constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
- true);
- CreatePeerConnection(&constraints);
+ RTCConfiguration config;
+ config.enable_dtls_srtp = true;
+ CreatePeerConnection(config);
CreateAndSetRemoteOffer(GetSdpStringWithStream1());
rtc::scoped_refptr<StreamCollection> reference(CreateStreamCollection(1, 1));
EXPECT_TRUE(
@@ -3057,10 +3031,9 @@
// an endpoint that doesn't signal SSRCs, but signals media stream IDs.
TEST_F(PeerConnectionInterfaceTestPlanB,
SdpWithMsidWithoutSsrcCreatesDefaultStream) {
- FakeConstraints constraints;
- constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
- true);
- CreatePeerConnection(&constraints);
+ RTCConfiguration config;
+ config.enable_dtls_srtp = true;
+ CreatePeerConnection(config);
std::string sdp_string = kSdpStringWithoutStreamsAudioOnly;
// Add a=msid lines to simulate a Unified Plan endpoint that only
// signals stream IDs with a=msid lines.
@@ -3081,10 +3054,9 @@
// a=ssrc msid and a=msid lines for interop signaling with Plan B.
TEST_F(PeerConnectionInterfaceTestPlanB,
SdpWithEmptyMsidAndSsrcCreatesDefaultStreamId) {
- FakeConstraints constraints;
- constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
- true);
- CreatePeerConnection(&constraints);
+ RTCConfiguration config;
+ config.enable_dtls_srtp = true;
+ CreatePeerConnection(config);
// Add a a=msid line to the SDP. This is prioritized when parsing the SDP, so
// the sender's stream ID will be interpreted as no stream IDs.
std::string sdp_string = kSdpStringWithStream1AudioTrackOnly;
@@ -3120,10 +3092,9 @@
// an RtpSender's lifetime isn't determined by when a local description is set.
// Don't run under Unified Plan since this behavior is Plan B specific.
TEST_F(PeerConnectionInterfaceTestPlanB, LocalDescriptionChanged) {
- FakeConstraints constraints;
- constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
- true);
- CreatePeerConnection(&constraints);
+ RTCConfiguration config;
+ config.enable_dtls_srtp = true;
+ CreatePeerConnection(config);
// Create an offer with 1 stream with 2 tracks of each type.
rtc::scoped_refptr<StreamCollection> stream_collection =
@@ -3160,10 +3131,9 @@
// Don't run under Unified Plan since this behavior is Plan B specific.
TEST_F(PeerConnectionInterfaceTestPlanB,
AddLocalStreamAfterLocalDescriptionChanged) {
- FakeConstraints constraints;
- constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
- true);
- CreatePeerConnection(&constraints);
+ RTCConfiguration config;
+ config.enable_dtls_srtp = true;
+ CreatePeerConnection(config);
rtc::scoped_refptr<StreamCollection> stream_collection =
CreateStreamCollection(1, 2);
@@ -3190,10 +3160,9 @@
// changed when SetLocalDescription is called.
TEST_P(PeerConnectionInterfaceTest,
ChangeSsrcOnTrackInLocalSessionDescription) {
- FakeConstraints constraints;
- constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
- true);
- CreatePeerConnection(&constraints);
+ RTCConfiguration config;
+ config.enable_dtls_srtp = true;
+ CreatePeerConnection(config);
AddAudioTrack(kAudioTracks[0]);
AddVideoTrack(kVideoTracks[0]);
@@ -3245,10 +3214,9 @@
// Don't run under Unified Plan since the stream API is not available.
TEST_F(PeerConnectionInterfaceTestPlanB,
SignalSameTracksInSeparateMediaStream) {
- FakeConstraints constraints;
- constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
- true);
- CreatePeerConnection(&constraints);
+ RTCConfiguration config;
+ config.enable_dtls_srtp = true;
+ CreatePeerConnection(config);
rtc::scoped_refptr<StreamCollection> stream_collection =
CreateStreamCollection(2, 1);
@@ -3283,10 +3251,9 @@
// This tests that PeerConnectionObserver::OnAddTrack is correctly called.
TEST_P(PeerConnectionInterfaceTest, OnAddTrackCallback) {
- FakeConstraints constraints;
- constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
- true);
- CreatePeerConnection(&constraints);
+ RTCConfiguration config;
+ config.enable_dtls_srtp = true;
+ CreatePeerConnection(config);
CreateAndSetRemoteOffer(kSdpStringWithStream1AudioTrackOnly);
EXPECT_EQ(observer_.num_added_tracks_, 1);
EXPECT_EQ(observer_.last_added_track_label_, kAudioTracks[0]);
@@ -3302,9 +3269,7 @@
TEST_P(PeerConnectionInterfaceTest, SetConfigurationCausingIceRestart) {
PeerConnectionInterface::RTCConfiguration config;
config.type = PeerConnectionInterface::kRelay;
- // Need to pass default constraints to prevent disabling of DTLS...
- FakeConstraints default_constraints;
- CreatePeerConnection(config, &default_constraints);
+ CreatePeerConnection(config);
config = pc_->GetConfiguration();
AddAudioTrack(kAudioTracks[0], {kStreamId1});
AddVideoTrack(kVideoTracks[0], {kStreamId1});
@@ -3338,9 +3303,7 @@
TEST_P(PeerConnectionInterfaceTest, SetConfigurationNotCausingIceRestart) {
PeerConnectionInterface::RTCConfiguration config;
config.type = PeerConnectionInterface::kRelay;
- // Need to pass default constraints to prevent disabling of DTLS...
- FakeConstraints default_constraints;
- CreatePeerConnection(config, &default_constraints);
+ CreatePeerConnection(config);
config = pc_->GetConfiguration();
AddAudioTrack(kAudioTracks[0]);
AddVideoTrack(kVideoTracks[0]);
@@ -3374,9 +3337,7 @@
TEST_P(PeerConnectionInterfaceTest, SetConfigurationCausingPartialIceRestart) {
PeerConnectionInterface::RTCConfiguration config;
config.type = PeerConnectionInterface::kRelay;
- // Need to pass default constraints to prevent disabling of DTLS...
- FakeConstraints default_constraints;
- CreatePeerConnection(config, &default_constraints);
+ CreatePeerConnection(config);
config = pc_->GetConfiguration();
AddAudioTrack(kAudioTracks[0], {kStreamId1});
AddVideoTrack(kVideoTracks[0], {kStreamId1});
@@ -3534,11 +3495,11 @@
CreatePeerConnection();
// First, create an offer with audio/video.
- FakeConstraints constraints;
- constraints.SetMandatoryReceiveAudio(true);
- constraints.SetMandatoryReceiveVideo(true);
+ RTCOfferAnswerOptions options;
+ options.offer_to_receive_audio = 1;
+ options.offer_to_receive_video = 1;
std::unique_ptr<SessionDescriptionInterface> offer;
- ASSERT_TRUE(DoCreateOffer(&offer, &constraints));
+ ASSERT_TRUE(DoCreateOffer(&offer, &options));
cricket::SessionDescription* desc = offer->description();
ASSERT_EQ(2u, desc->transport_infos().size());
EXPECT_TRUE(desc->transport_infos()[0].description.HasOption("trickle"));
@@ -3547,7 +3508,7 @@
// Apply the offer as a remote description, then create an answer.
EXPECT_TRUE(DoSetRemoteDescription(std::move(offer)));
std::unique_ptr<SessionDescriptionInterface> answer;
- ASSERT_TRUE(DoCreateAnswer(&answer, &constraints));
+ ASSERT_TRUE(DoCreateAnswer(&answer, &options));
desc = answer->description();
ASSERT_EQ(2u, desc->transport_infos().size());
EXPECT_TRUE(desc->transport_infos()[0].description.HasOption("trickle"));
@@ -3559,7 +3520,7 @@
TEST_P(PeerConnectionInterfaceTest, IceRenominationNotOffered) {
PeerConnectionInterface::RTCConfiguration config;
config.enable_ice_renomination = false;
- CreatePeerConnection(config, nullptr);
+ CreatePeerConnection(config);
AddAudioTrack("foo");
std::unique_ptr<SessionDescriptionInterface> offer;
@@ -3575,7 +3536,7 @@
TEST_P(PeerConnectionInterfaceTest, IceRenominationOptionInOfferAndAnswer) {
PeerConnectionInterface::RTCConfiguration config;
config.enable_ice_renomination = true;
- CreatePeerConnection(config, nullptr);
+ CreatePeerConnection(config);
AddAudioTrack("foo");
std::unique_ptr<SessionDescriptionInterface> offer;
@@ -3602,11 +3563,11 @@
TEST_P(PeerConnectionInterfaceTest, CreateOfferWithOfferToReceiveConstraints) {
CreatePeerConnection();
- FakeConstraints constraints;
- constraints.SetMandatoryReceiveAudio(true);
- constraints.SetMandatoryReceiveVideo(true);
+ RTCOfferAnswerOptions options;
+ options.offer_to_receive_audio = 1;
+ options.offer_to_receive_video = 1;
std::unique_ptr<SessionDescriptionInterface> offer;
- ASSERT_TRUE(DoCreateOffer(&offer, &constraints));
+ ASSERT_TRUE(DoCreateOffer(&offer, &options));
cricket::SessionDescription* desc = offer->description();
const cricket::ContentInfo* audio = cricket::GetFirstAudioContent(desc);
@@ -3627,18 +3588,18 @@
// First, create an offer with audio/video and apply it as a remote
// description.
- FakeConstraints constraints;
- constraints.SetMandatoryReceiveAudio(true);
- constraints.SetMandatoryReceiveVideo(true);
+ RTCOfferAnswerOptions options;
+ options.offer_to_receive_audio = 1;
+ options.offer_to_receive_video = 1;
std::unique_ptr<SessionDescriptionInterface> offer;
- ASSERT_TRUE(DoCreateOffer(&offer, &constraints));
+ ASSERT_TRUE(DoCreateOffer(&offer, &options));
EXPECT_TRUE(DoSetRemoteDescription(std::move(offer)));
// Now create answer that rejects audio/video.
- constraints.SetMandatoryReceiveAudio(false);
- constraints.SetMandatoryReceiveVideo(false);
+ options.offer_to_receive_audio = 0;
+ options.offer_to_receive_video = 0;
std::unique_ptr<SessionDescriptionInterface> answer;
- ASSERT_TRUE(DoCreateAnswer(&answer, &constraints));
+ ASSERT_TRUE(DoCreateAnswer(&answer, &options));
cricket::SessionDescription* desc = answer->description();
const cricket::ContentInfo* audio = cricket::GetFirstAudioContent(desc);
@@ -3659,7 +3620,7 @@
#endif // HAVE_SCTP
PeerConnectionInterface::RTCConfiguration config;
config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
- CreatePeerConnection(config, nullptr);
+ CreatePeerConnection(config);
// First, create an offer with only a data channel and apply it as a remote
// description.
@@ -3745,7 +3706,7 @@
config.ice_regather_interval_range.emplace(1000, 2000);
config.continual_gathering_policy =
PeerConnectionInterface::ContinualGatheringPolicy::GATHER_CONTINUALLY;
- CreatePeerConnection(config, nullptr);
+ CreatePeerConnection(config);
}
// The current bitrate from BitrateSettings is currently clamped
@@ -3990,10 +3951,9 @@
pcf_->Initialize();
}
const cricket::MediaConfig TestCreatePeerConnection(
- const PeerConnectionInterface::RTCConfiguration& config,
- const MediaConstraintsInterface* constraints) {
- rtc::scoped_refptr<PeerConnectionInterface> pc(pcf_->CreatePeerConnection(
- config, constraints, nullptr, nullptr, &observer_));
+ const RTCConfiguration& config) {
+ rtc::scoped_refptr<PeerConnectionInterface> pc(
+ pcf_->CreatePeerConnection(config, nullptr, nullptr, &observer_));
EXPECT_TRUE(pc.get());
return pc->GetConfiguration().media_config;
}
@@ -4006,10 +3966,8 @@
// default RTCConfiguration.
TEST_F(PeerConnectionMediaConfigTest, TestDefaults) {
PeerConnectionInterface::RTCConfiguration config;
- FakeConstraints constraints;
- const cricket::MediaConfig& media_config =
- TestCreatePeerConnection(config, &constraints);
+ const cricket::MediaConfig& media_config = TestCreatePeerConnection(config);
EXPECT_FALSE(media_config.enable_dscp);
EXPECT_TRUE(media_config.video.enable_cpu_adaptation);
@@ -4018,42 +3976,13 @@
EXPECT_FALSE(media_config.video.experiment_cpu_load_estimator);
}
-// This test verifies the DSCP constraint is recognized and passed to
-// the PeerConnection.
-TEST_F(PeerConnectionMediaConfigTest, TestDscpConstraintTrue) {
- PeerConnectionInterface::RTCConfiguration config;
- FakeConstraints constraints;
-
- constraints.AddOptional(webrtc::MediaConstraintsInterface::kEnableDscp, true);
- const cricket::MediaConfig& media_config =
- TestCreatePeerConnection(config, &constraints);
-
- EXPECT_TRUE(media_config.enable_dscp);
-}
-
-// This test verifies the cpu overuse detection constraint is
-// recognized and passed to the PeerConnection.
-TEST_F(PeerConnectionMediaConfigTest, TestCpuOveruseConstraintFalse) {
- PeerConnectionInterface::RTCConfiguration config;
- FakeConstraints constraints;
-
- constraints.AddOptional(
- webrtc::MediaConstraintsInterface::kCpuOveruseDetection, false);
- const cricket::MediaConfig media_config =
- TestCreatePeerConnection(config, &constraints);
-
- EXPECT_FALSE(media_config.video.enable_cpu_adaptation);
-}
-
// This test verifies that the enable_prerenderer_smoothing flag is
// propagated from RTCConfiguration to the PeerConnection.
TEST_F(PeerConnectionMediaConfigTest, TestDisablePrerendererSmoothingTrue) {
PeerConnectionInterface::RTCConfiguration config;
- FakeConstraints constraints;
config.set_prerenderer_smoothing(false);
- const cricket::MediaConfig& media_config =
- TestCreatePeerConnection(config, &constraints);
+ const cricket::MediaConfig& media_config = TestCreatePeerConnection(config);
EXPECT_FALSE(media_config.video.enable_prerenderer_smoothing);
}
@@ -4062,31 +3991,13 @@
// propagated from RTCConfiguration to the PeerConnection.
TEST_F(PeerConnectionMediaConfigTest, TestEnableExperimentCpuLoadEstimator) {
PeerConnectionInterface::RTCConfiguration config;
- FakeConstraints constraints;
config.set_experiment_cpu_load_estimator(true);
- const cricket::MediaConfig& media_config =
- TestCreatePeerConnection(config, &constraints);
+ const cricket::MediaConfig& media_config = TestCreatePeerConnection(config);
EXPECT_TRUE(media_config.video.experiment_cpu_load_estimator);
}
-// This test verifies the suspend below min bitrate constraint is
-// recognized and passed to the PeerConnection.
-TEST_F(PeerConnectionMediaConfigTest,
- TestSuspendBelowMinBitrateConstraintTrue) {
- PeerConnectionInterface::RTCConfiguration config;
- FakeConstraints constraints;
-
- constraints.AddOptional(
- webrtc::MediaConstraintsInterface::kEnableVideoSuspendBelowMinBitrate,
- true);
- const cricket::MediaConfig media_config =
- TestCreatePeerConnection(config, &constraints);
-
- EXPECT_TRUE(media_config.video.suspend_below_min_bitrate);
-}
-
// Tests a few random fields being different.
TEST(RTCConfigurationTest, ComparisonOperators) {
PeerConnectionInterface::RTCConfiguration a;
diff --git a/pc/rtcstats_integrationtest.cc b/pc/rtcstats_integrationtest.cc
index 355d1f7a8..987f49c 100644
--- a/pc/rtcstats_integrationtest.cc
+++ b/pc/rtcstats_integrationtest.cc
@@ -109,11 +109,9 @@
PeerConnectionInterface::IceServer ice_server;
ice_server.uri = "stun:1.1.1.1:3478";
config.servers.push_back(ice_server);
- EXPECT_TRUE(caller_->CreatePc(nullptr, config,
- CreateBuiltinAudioEncoderFactory(),
+ EXPECT_TRUE(caller_->CreatePc(config, CreateBuiltinAudioEncoderFactory(),
CreateBuiltinAudioDecoderFactory()));
- EXPECT_TRUE(callee_->CreatePc(nullptr, config,
- CreateBuiltinAudioEncoderFactory(),
+ EXPECT_TRUE(callee_->CreatePc(config, CreateBuiltinAudioEncoderFactory(),
CreateBuiltinAudioDecoderFactory()));
PeerConnectionTestWrapper::Connect(caller_.get(), callee_.get());
@@ -129,7 +127,7 @@
callee_->CreateDataChannel("data", init);
// Negotiate and wait for call to establish
- caller_->CreateOffer(nullptr);
+ caller_->CreateOffer(PeerConnectionInterface::RTCOfferAnswerOptions());
caller_->WaitForCallEstablished();
callee_->WaitForCallEstablished();
}
diff --git a/pc/test/fakepeerconnectionbase.h b/pc/test/fakepeerconnectionbase.h
index db8596c..93380c0 100644
--- a/pc/test/fakepeerconnectionbase.h
+++ b/pc/test/fakepeerconnectionbase.h
@@ -140,17 +140,11 @@
}
void CreateOffer(CreateSessionDescriptionObserver* observer,
- const MediaConstraintsInterface* constraints) override {}
-
- void CreateOffer(CreateSessionDescriptionObserver* observer,
const RTCOfferAnswerOptions& options) override {}
void CreateAnswer(CreateSessionDescriptionObserver* observer,
const RTCOfferAnswerOptions& options) override {}
- void CreateAnswer(CreateSessionDescriptionObserver* observer,
- const MediaConstraintsInterface* constraints) override {}
-
void SetLocalDescription(SetSessionDescriptionObserver* observer,
SessionDescriptionInterface* desc) override {}
diff --git a/pc/test/peerconnectiontestwrapper.cc b/pc/test/peerconnectiontestwrapper.cc
index 145a095..fc699d3 100644
--- a/pc/test/peerconnectiontestwrapper.cc
+++ b/pc/test/peerconnectiontestwrapper.cc
@@ -69,7 +69,6 @@
PeerConnectionTestWrapper::~PeerConnectionTestWrapper() {}
bool PeerConnectionTestWrapper::CreatePc(
- const MediaConstraintsInterface* constraints,
const webrtc::PeerConnectionInterface::RTCConfiguration& config,
rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory,
rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory) {
@@ -95,8 +94,7 @@
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator(
new FakeRTCCertificateGenerator());
peer_connection_ = peer_connection_factory_->CreatePeerConnection(
- config, constraints, std::move(port_allocator), std::move(cert_generator),
- this);
+ config, std::move(port_allocator), std::move(cert_generator), this);
return peer_connection_.get() != NULL;
}
@@ -153,21 +151,21 @@
}
void PeerConnectionTestWrapper::CreateOffer(
- const MediaConstraintsInterface* constraints) {
+ const webrtc::PeerConnectionInterface::RTCOfferAnswerOptions& options) {
RTC_LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_ << ": CreateOffer.";
- peer_connection_->CreateOffer(this, constraints);
+ peer_connection_->CreateOffer(this, options);
}
void PeerConnectionTestWrapper::CreateAnswer(
- const MediaConstraintsInterface* constraints) {
+ const webrtc::PeerConnectionInterface::RTCOfferAnswerOptions& options) {
RTC_LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
<< ": CreateAnswer.";
- peer_connection_->CreateAnswer(this, constraints);
+ peer_connection_->CreateAnswer(this, options);
}
void PeerConnectionTestWrapper::ReceiveOfferSdp(const std::string& sdp) {
SetRemoteDescription(SdpType::kOffer, sdp);
- CreateAnswer(NULL);
+ CreateAnswer(webrtc::PeerConnectionInterface::RTCOfferAnswerOptions());
}
void PeerConnectionTestWrapper::ReceiveAnswerSdp(const std::string& sdp) {
diff --git a/pc/test/peerconnectiontestwrapper.h b/pc/test/peerconnectiontestwrapper.h
index f82bcf2..21ba89a 100644
--- a/pc/test/peerconnectiontestwrapper.h
+++ b/pc/test/peerconnectiontestwrapper.h
@@ -35,7 +35,6 @@
virtual ~PeerConnectionTestWrapper();
bool CreatePc(
- const webrtc::MediaConstraintsInterface* constraints,
const webrtc::PeerConnectionInterface::RTCConfiguration& config,
rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory,
rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory);
@@ -66,8 +65,10 @@
void OnSuccess(webrtc::SessionDescriptionInterface* desc) override;
void OnFailure(webrtc::RTCError) override {}
- void CreateOffer(const webrtc::MediaConstraintsInterface* constraints);
- void CreateAnswer(const webrtc::MediaConstraintsInterface* constraints);
+ void CreateOffer(
+ const webrtc::PeerConnectionInterface::RTCOfferAnswerOptions& options);
+ void CreateAnswer(
+ const webrtc::PeerConnectionInterface::RTCOfferAnswerOptions& options);
void ReceiveOfferSdp(const std::string& sdp);
void ReceiveAnswerSdp(const std::string& sdp);
void AddIceCandidate(const std::string& sdp_mid,
diff --git a/sdk/android/src/jni/pc/peerconnection.cc b/sdk/android/src/jni/pc/peerconnection.cc
index 31a0689..79da797 100644
--- a/sdk/android/src/jni/pc/peerconnection.cc
+++ b/sdk/android/src/jni/pc/peerconnection.cc
@@ -451,7 +451,9 @@
rtc::scoped_refptr<CreateSdpObserverJni> observer(
new rtc::RefCountedObject<CreateSdpObserverJni>(jni, j_observer,
std::move(constraints)));
- ExtractNativePC(jni, j_pc)->CreateOffer(observer, observer->constraints());
+ PeerConnectionInterface::RTCOfferAnswerOptions options;
+ CopyConstraintsIntoOfferAnswerOptions(observer->constraints(), &options);
+ ExtractNativePC(jni, j_pc)->CreateOffer(observer, options);
}
static void JNI_PeerConnection_CreateAnswer(
@@ -464,7 +466,9 @@
rtc::scoped_refptr<CreateSdpObserverJni> observer(
new rtc::RefCountedObject<CreateSdpObserverJni>(jni, j_observer,
std::move(constraints)));
- ExtractNativePC(jni, j_pc)->CreateAnswer(observer, observer->constraints());
+ PeerConnectionInterface::RTCOfferAnswerOptions options;
+ CopyConstraintsIntoOfferAnswerOptions(observer->constraints(), &options);
+ ExtractNativePC(jni, j_pc)->CreateAnswer(observer, options);
}
static void JNI_PeerConnection_SetLocalDescription(
diff --git a/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection.mm b/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection.mm
index d0bb39b..11c3a6d 100644
--- a/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection.mm
+++ b/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection.mm
@@ -448,7 +448,10 @@
rtc::scoped_refptr<webrtc::CreateSessionDescriptionObserverAdapter>
observer(new rtc::RefCountedObject
<webrtc::CreateSessionDescriptionObserverAdapter>(completionHandler));
- _peerConnection->CreateOffer(observer, constraints.nativeConstraints.get());
+ webrtc::PeerConnectionInterface::RTCOfferAnswerOptions options;
+ CopyConstraintsIntoOfferAnswerOptions(constraints.nativeConstraints.get(), &options);
+
+ _peerConnection->CreateOffer(observer, options);
}
- (void)answerForConstraints:(RTCMediaConstraints *)constraints
@@ -458,7 +461,10 @@
rtc::scoped_refptr<webrtc::CreateSessionDescriptionObserverAdapter>
observer(new rtc::RefCountedObject
<webrtc::CreateSessionDescriptionObserverAdapter>(completionHandler));
- _peerConnection->CreateAnswer(observer, constraints.nativeConstraints.get());
+ webrtc::PeerConnectionInterface::RTCOfferAnswerOptions options;
+ CopyConstraintsIntoOfferAnswerOptions(constraints.nativeConstraints.get(), &options);
+
+ _peerConnection->CreateAnswer(observer, options);
}
- (void)setLocalDescription:(RTCSessionDescription *)sdp
