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/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();
