webrtc_m130/pc/jsep_transport_controller_unittest.cc

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/*
* Copyright 2018 The WebRTC Project Authors. All rights reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "pc/jsep_transport_controller.h"
#include <map>
#include <memory>
#include "p2p/base/dtls_transport_factory.h"
#include "p2p/base/fake_dtls_transport.h"
#include "p2p/base/fake_ice_transport.h"
#include "p2p/base/transport_info.h"
#include "rtc_base/gunit.h"
#include "rtc_base/thread.h"
#include "test/gtest.h"
using cricket::Candidate;
using cricket::Candidates;
using cricket::FakeDtlsTransport;
using webrtc::SdpType;
static const int kTimeout = 100;
static const char kIceUfrag1[] = "u0001";
static const char kIcePwd1[] = "TESTICEPWD00000000000001";
static const char kIceUfrag2[] = "u0002";
static const char kIcePwd2[] = "TESTICEPWD00000000000002";
static const char kIceUfrag3[] = "u0003";
static const char kIcePwd3[] = "TESTICEPWD00000000000003";
static const char kAudioMid1[] = "audio1";
static const char kAudioMid2[] = "audio2";
static const char kVideoMid1[] = "video1";
static const char kVideoMid2[] = "video2";
static const char kDataMid1[] = "data1";
namespace webrtc {
class FakeIceTransportFactory : public webrtc::IceTransportFactory {
public:
~FakeIceTransportFactory() override = default;
rtc::scoped_refptr<IceTransportInterface> CreateIceTransport(
const std::string& transport_name,
int component,
IceTransportInit init) override {
return new rtc::RefCountedObject<cricket::FakeIceTransportWrapper>(
std::make_unique<cricket::FakeIceTransport>(transport_name, component));
}
};
class FakeDtlsTransportFactory : public cricket::DtlsTransportFactory {
public:
std::unique_ptr<cricket::DtlsTransportInternal> CreateDtlsTransport(
cricket::IceTransportInternal* ice,
Reland "Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h" Promotes rtc::CryptoOptions to webrtc::CryptoOptions converting it from class that only handles SRTP configuration to a more generic structure that can be used and extended for all per peer connection CryptoOptions that can be on a given PeerConnection. Now all SRTP related options are under webrtc::CryptoOptions::Srtp and can be accessed as crypto_options.srtp.whatever_option_name. This is more inline with other structures we have in WebRTC such as VideoConfig. As additional features are added over time this will allow the structure to remain compartmentalized and concerned components can only request a subset of the overall configuration structure e.g: void MySrtpFunction(const webrtc::CryptoOptions::Srtp& srtp_config); In addition to this it made little sense for sslstreamadapter.h to hold all Srtp related configuration options. The header has become loo large and takes on too many responsibilities and spilting this up will lead to more maintainable code going forward. This will be used in a future CL to enable configuration options for the newly supported Frame Crypto. Reland Fix: - cryptooptions.h - now has enable_aes128_sha1_32_crypto_cipher as an optional root level configuration. - peerconnectionfactory - If this optional is set will now overwrite the underyling value. This along with the other field will be deprecated once dependent projects are updated. TBR=sakal@webrtc.org,kthelgason@webrtc.org,emadomara@webrtc.org,qingsi@webrtc.org Bug: webrtc:9681 Change-Id: Iaa6b741baafb85d352e42f54226119f19d97151d Reviewed-on: https://webrtc-review.googlesource.com/c/105560 Reviewed-by: Benjamin Wright <benwright@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Emad Omara <emadomara@webrtc.org> Commit-Queue: Benjamin Wright <benwright@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25135}
2018-10-11 15:33:17 -07:00
const webrtc::CryptoOptions& crypto_options) override {
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
return std::make_unique<FakeDtlsTransport>(
static_cast<cricket::FakeIceTransport*>(ice));
}
};
class JsepTransportControllerTest : public JsepTransportController::Observer,
public ::testing::Test,
public sigslot::has_slots<> {
public:
JsepTransportControllerTest() : signaling_thread_(rtc::Thread::Current()) {
fake_ice_transport_factory_ = std::make_unique<FakeIceTransportFactory>();
fake_dtls_transport_factory_ = std::make_unique<FakeDtlsTransportFactory>();
}
void CreateJsepTransportController(
JsepTransportController::Config config,
Revert "Remove thread hops from events provided by JsepTransportController." This reverts commit f554b3c577f69fa9ffad5c07155898c2d985ac76. Reason for revert: Parent CL breaks FYI bots. See https://webrtc-review.googlesource.com/c/src/+/206466 Original change's description: > Remove thread hops from events provided by JsepTransportController. > > Events associated with Subscribe* methods in JTC had trampolines that > would use an async invoker to fire the events on the signaling thread. > This was being done for the purposes of PeerConnection but the concept > of a signaling thread is otherwise not applicable to JTC and use of > JTC from PC is inconsistent across threads (as has been flagged in > webrtc:9987). > > This change makes all CallbackList members only accessible from the > network thread and moves the signaling thread related work over to > PeerConnection, which makes hops there more visible as well as making > that class easier to refactor for thread efficiency. > > This CL removes the AsyncInvoker from JTC (webrtc:12339) > > The signaling_thread_ variable is also removed from JTC and more thread > checks added to catch errors. > > Bug: webrtc:12427, webrtc:11988, webrtc:12339 > Change-Id: Id232aedd00dfd5403b2ba0ca147d3eca7c12c7c5 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206062 > Commit-Queue: Tommi <tommi@webrtc.org> > Reviewed-by: Niels Moller <nisse@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33195} TBR=nisse@webrtc.org,tommi@webrtc.org Change-Id: I6134b71b74a9408854b79d44506d513519e9cf4d No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:12427 Bug: webrtc:11988 Bug: webrtc:12339 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206467 Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Commit-Queue: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33203}
2021-02-09 12:26:13 +00:00
rtc::Thread* signaling_thread = rtc::Thread::Current(),
rtc::Thread* network_thread = rtc::Thread::Current(),
cricket::PortAllocator* port_allocator = nullptr) {
config.transport_observer = this;
config.rtcp_handler = [](const rtc::CopyOnWriteBuffer& packet,
int64_t packet_time_us) { RTC_NOTREACHED(); };
config.ice_transport_factory = fake_ice_transport_factory_.get();
config.dtls_transport_factory = fake_dtls_transport_factory_.get();
config.on_dtls_handshake_error_ = [](rtc::SSLHandshakeError s) {};
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
transport_controller_ = std::make_unique<JsepTransportController>(
Revert "Remove thread hops from events provided by JsepTransportController." This reverts commit f554b3c577f69fa9ffad5c07155898c2d985ac76. Reason for revert: Parent CL breaks FYI bots. See https://webrtc-review.googlesource.com/c/src/+/206466 Original change's description: > Remove thread hops from events provided by JsepTransportController. > > Events associated with Subscribe* methods in JTC had trampolines that > would use an async invoker to fire the events on the signaling thread. > This was being done for the purposes of PeerConnection but the concept > of a signaling thread is otherwise not applicable to JTC and use of > JTC from PC is inconsistent across threads (as has been flagged in > webrtc:9987). > > This change makes all CallbackList members only accessible from the > network thread and moves the signaling thread related work over to > PeerConnection, which makes hops there more visible as well as making > that class easier to refactor for thread efficiency. > > This CL removes the AsyncInvoker from JTC (webrtc:12339) > > The signaling_thread_ variable is also removed from JTC and more thread > checks added to catch errors. > > Bug: webrtc:12427, webrtc:11988, webrtc:12339 > Change-Id: Id232aedd00dfd5403b2ba0ca147d3eca7c12c7c5 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206062 > Commit-Queue: Tommi <tommi@webrtc.org> > Reviewed-by: Niels Moller <nisse@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33195} TBR=nisse@webrtc.org,tommi@webrtc.org Change-Id: I6134b71b74a9408854b79d44506d513519e9cf4d No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:12427 Bug: webrtc:11988 Bug: webrtc:12339 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206467 Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Commit-Queue: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33203}
2021-02-09 12:26:13 +00:00
signaling_thread, network_thread, port_allocator,
nullptr /* async_resolver_factory */, config);
ConnectTransportControllerSignals();
}
void ConnectTransportControllerSignals() {
transport_controller_->SubscribeIceConnectionState(
[this](cricket::IceConnectionState s) {
JsepTransportControllerTest::OnConnectionState(s);
});
transport_controller_->SubscribeConnectionState(
[this](PeerConnectionInterface::PeerConnectionState s) {
JsepTransportControllerTest::OnCombinedConnectionState(s);
});
transport_controller_->SubscribeStandardizedIceConnectionState(
[this](PeerConnectionInterface::IceConnectionState s) {
JsepTransportControllerTest::OnStandardizedIceConnectionState(s);
});
transport_controller_->SubscribeIceGatheringState(
[this](cricket::IceGatheringState s) {
JsepTransportControllerTest::OnGatheringState(s);
});
transport_controller_->SubscribeIceCandidateGathered(
[this](const std::string& transport,
const std::vector<cricket::Candidate>& candidates) {
JsepTransportControllerTest::OnCandidatesGathered(transport,
candidates);
});
}
std::unique_ptr<cricket::SessionDescription>
CreateSessionDescriptionWithoutBundle() {
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto description = std::make_unique<cricket::SessionDescription>();
AddAudioSection(description.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
AddVideoSection(description.get(), kVideoMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
return description;
}
std::unique_ptr<cricket::SessionDescription>
CreateSessionDescriptionWithBundleGroup() {
auto description = CreateSessionDescriptionWithoutBundle();
cricket::ContentGroup bundle_group(cricket::GROUP_TYPE_BUNDLE);
bundle_group.AddContentName(kAudioMid1);
bundle_group.AddContentName(kVideoMid1);
description->AddGroup(bundle_group);
return description;
}
Add an alt-protocol to SDP to indicate which m= sections use a plugin transport. The plugin transport parameters (a=x-opaque: lines) relate to how to create and set up a plugin transport. When SDP bundle is used, the x-opaque line needs to be copied into the bundled m= section. This means x-opaque can appear on a section even if the offerer does not intend to use the transport for the media described by that section. Consequently, the answerer cannot currently tell whether the caller is offering an alternate transport for media, data, or both. This change adds an a=x-alt-protocol: line to SDP. The value following this line matches the <protocol> part of the x-opaque:<protocol>:<params> line. However, alt-protocol is not bundled--it only ever applies to the m= section that contains the line. This allows the offerer to express which m= sections should actually use an alternate transport, even in the case of bundle. Note that this is still limited by the available configuration options: datagram transport can be used for media (audio + video) and/or data. It is still not possible to use it for audio but not video, or vice versa. PeerConnection places an alt-protocol line in each media (audio/video) m= section if it is configured to use a datagram transport for media. It places an alt-protocol line in each data m= section if it is configured to use a datagram transport for data channels. PeerConnection leaves alt-protocol in media (audio/video) m= sections of the answer if it is configured to use a datagram transport for media, and in data m= sections of the answer if it is configured to use a datagram transport for data channels. JsepTransport now negotiates use of the datagram transport independently for media and data channels. It only uses it for media if the m= sections for bundled audio/video have an alt-protocol line matching the x-opaque protocol, and only uses it for data channels if a bundled m= section for data has an alt-protocol line matching the x-opaque protocol. Bug: webrtc:9719 Change-Id: I773e4fc10c57d815afcd76a2a74da38dd0c52b3b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154763 Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Seth Hampson <shampson@webrtc.org> Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29351}
2019-09-30 15:12:47 -07:00
std::unique_ptr<cricket::SessionDescription>
CreateSessionDescriptionWithBundledData() {
auto description = CreateSessionDescriptionWithoutBundle();
AddDataSection(description.get(), kDataMid1,
cricket::MediaProtocolType::kSctp, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
cricket::ContentGroup bundle_group(cricket::GROUP_TYPE_BUNDLE);
bundle_group.AddContentName(kAudioMid1);
bundle_group.AddContentName(kVideoMid1);
bundle_group.AddContentName(kDataMid1);
description->AddGroup(bundle_group);
return description;
}
void AddAudioSection(cricket::SessionDescription* description,
const std::string& mid,
const std::string& ufrag,
const std::string& pwd,
cricket::IceMode ice_mode,
cricket::ConnectionRole conn_role,
rtc::scoped_refptr<rtc::RTCCertificate> cert) {
std::unique_ptr<cricket::AudioContentDescription> audio(
new cricket::AudioContentDescription());
// Set RTCP-mux to be true because the default policy is "mux required".
audio->set_rtcp_mux(true);
description->AddContent(mid, cricket::MediaProtocolType::kRtp,
/*rejected=*/false, std::move(audio));
AddTransportInfo(description, mid, ufrag, pwd, ice_mode, conn_role, cert);
}
void AddVideoSection(cricket::SessionDescription* description,
const std::string& mid,
const std::string& ufrag,
const std::string& pwd,
cricket::IceMode ice_mode,
cricket::ConnectionRole conn_role,
rtc::scoped_refptr<rtc::RTCCertificate> cert) {
std::unique_ptr<cricket::VideoContentDescription> video(
new cricket::VideoContentDescription());
// Set RTCP-mux to be true because the default policy is "mux required".
video->set_rtcp_mux(true);
description->AddContent(mid, cricket::MediaProtocolType::kRtp,
/*rejected=*/false, std::move(video));
AddTransportInfo(description, mid, ufrag, pwd, ice_mode, conn_role, cert);
}
void AddDataSection(cricket::SessionDescription* description,
const std::string& mid,
cricket::MediaProtocolType protocol_type,
const std::string& ufrag,
const std::string& pwd,
cricket::IceMode ice_mode,
cricket::ConnectionRole conn_role,
rtc::scoped_refptr<rtc::RTCCertificate> cert) {
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
RTC_CHECK(protocol_type == cricket::MediaProtocolType::kSctp);
std::unique_ptr<cricket::SctpDataContentDescription> data(
new cricket::SctpDataContentDescription());
data->set_rtcp_mux(true);
description->AddContent(mid, protocol_type,
/*rejected=*/false, std::move(data));
AddTransportInfo(description, mid, ufrag, pwd, ice_mode, conn_role, cert);
}
void AddTransportInfo(cricket::SessionDescription* description,
const std::string& mid,
const std::string& ufrag,
const std::string& pwd,
cricket::IceMode ice_mode,
cricket::ConnectionRole conn_role,
rtc::scoped_refptr<rtc::RTCCertificate> cert) {
std::unique_ptr<rtc::SSLFingerprint> fingerprint;
if (cert) {
fingerprint = rtc::SSLFingerprint::CreateFromCertificate(*cert);
}
cricket::TransportDescription transport_desc(std::vector<std::string>(),
ufrag, pwd, ice_mode,
conn_role, fingerprint.get());
description->AddTransportInfo(cricket::TransportInfo(mid, transport_desc));
}
cricket::IceConfig CreateIceConfig(
int receiving_timeout,
cricket::ContinualGatheringPolicy continual_gathering_policy) {
cricket::IceConfig config;
config.receiving_timeout = receiving_timeout;
config.continual_gathering_policy = continual_gathering_policy;
return config;
}
Candidate CreateCandidate(const std::string& transport_name, int component) {
Candidate c;
c.set_transport_name(transport_name);
c.set_address(rtc::SocketAddress("192.168.1.1", 8000));
c.set_component(component);
c.set_protocol(cricket::UDP_PROTOCOL_NAME);
c.set_priority(1);
return c;
}
void CreateLocalDescriptionAndCompleteConnectionOnNetworkThread() {
if (!network_thread_->IsCurrent()) {
network_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
CreateLocalDescriptionAndCompleteConnectionOnNetworkThread();
});
return;
}
auto description = CreateSessionDescriptionWithBundleGroup();
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
transport_controller_->MaybeStartGathering();
auto fake_audio_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kAudioMid1));
auto fake_video_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kVideoMid1));
fake_audio_dtls->fake_ice_transport()->SignalCandidateGathered(
fake_audio_dtls->fake_ice_transport(),
CreateCandidate(kAudioMid1, /*component=*/1));
fake_video_dtls->fake_ice_transport()->SignalCandidateGathered(
fake_video_dtls->fake_ice_transport(),
CreateCandidate(kVideoMid1, /*component=*/1));
fake_audio_dtls->fake_ice_transport()->SetCandidatesGatheringComplete();
fake_video_dtls->fake_ice_transport()->SetCandidatesGatheringComplete();
fake_audio_dtls->fake_ice_transport()->SetConnectionCount(2);
fake_video_dtls->fake_ice_transport()->SetConnectionCount(2);
fake_audio_dtls->SetReceiving(true);
fake_video_dtls->SetReceiving(true);
fake_audio_dtls->SetWritable(true);
fake_video_dtls->SetWritable(true);
fake_audio_dtls->fake_ice_transport()->SetConnectionCount(1);
fake_video_dtls->fake_ice_transport()->SetConnectionCount(1);
}
protected:
void OnConnectionState(cricket::IceConnectionState state) {
Revert "Remove thread hops from events provided by JsepTransportController." This reverts commit f554b3c577f69fa9ffad5c07155898c2d985ac76. Reason for revert: Parent CL breaks FYI bots. See https://webrtc-review.googlesource.com/c/src/+/206466 Original change's description: > Remove thread hops from events provided by JsepTransportController. > > Events associated with Subscribe* methods in JTC had trampolines that > would use an async invoker to fire the events on the signaling thread. > This was being done for the purposes of PeerConnection but the concept > of a signaling thread is otherwise not applicable to JTC and use of > JTC from PC is inconsistent across threads (as has been flagged in > webrtc:9987). > > This change makes all CallbackList members only accessible from the > network thread and moves the signaling thread related work over to > PeerConnection, which makes hops there more visible as well as making > that class easier to refactor for thread efficiency. > > This CL removes the AsyncInvoker from JTC (webrtc:12339) > > The signaling_thread_ variable is also removed from JTC and more thread > checks added to catch errors. > > Bug: webrtc:12427, webrtc:11988, webrtc:12339 > Change-Id: Id232aedd00dfd5403b2ba0ca147d3eca7c12c7c5 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206062 > Commit-Queue: Tommi <tommi@webrtc.org> > Reviewed-by: Niels Moller <nisse@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33195} TBR=nisse@webrtc.org,tommi@webrtc.org Change-Id: I6134b71b74a9408854b79d44506d513519e9cf4d No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:12427 Bug: webrtc:11988 Bug: webrtc:12339 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206467 Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Commit-Queue: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33203}
2021-02-09 12:26:13 +00:00
if (!signaling_thread_->IsCurrent()) {
signaled_on_non_signaling_thread_ = true;
}
connection_state_ = state;
++connection_state_signal_count_;
}
void OnStandardizedIceConnectionState(
PeerConnectionInterface::IceConnectionState state) {
Revert "Remove thread hops from events provided by JsepTransportController." This reverts commit f554b3c577f69fa9ffad5c07155898c2d985ac76. Reason for revert: Parent CL breaks FYI bots. See https://webrtc-review.googlesource.com/c/src/+/206466 Original change's description: > Remove thread hops from events provided by JsepTransportController. > > Events associated with Subscribe* methods in JTC had trampolines that > would use an async invoker to fire the events on the signaling thread. > This was being done for the purposes of PeerConnection but the concept > of a signaling thread is otherwise not applicable to JTC and use of > JTC from PC is inconsistent across threads (as has been flagged in > webrtc:9987). > > This change makes all CallbackList members only accessible from the > network thread and moves the signaling thread related work over to > PeerConnection, which makes hops there more visible as well as making > that class easier to refactor for thread efficiency. > > This CL removes the AsyncInvoker from JTC (webrtc:12339) > > The signaling_thread_ variable is also removed from JTC and more thread > checks added to catch errors. > > Bug: webrtc:12427, webrtc:11988, webrtc:12339 > Change-Id: Id232aedd00dfd5403b2ba0ca147d3eca7c12c7c5 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206062 > Commit-Queue: Tommi <tommi@webrtc.org> > Reviewed-by: Niels Moller <nisse@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33195} TBR=nisse@webrtc.org,tommi@webrtc.org Change-Id: I6134b71b74a9408854b79d44506d513519e9cf4d No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:12427 Bug: webrtc:11988 Bug: webrtc:12339 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206467 Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Commit-Queue: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33203}
2021-02-09 12:26:13 +00:00
if (!signaling_thread_->IsCurrent()) {
signaled_on_non_signaling_thread_ = true;
}
ice_connection_state_ = state;
++ice_connection_state_signal_count_;
}
void OnCombinedConnectionState(
PeerConnectionInterface::PeerConnectionState state) {
RTC_LOG(LS_INFO) << "OnCombinedConnectionState: "
<< static_cast<int>(state);
Revert "Remove thread hops from events provided by JsepTransportController." This reverts commit f554b3c577f69fa9ffad5c07155898c2d985ac76. Reason for revert: Parent CL breaks FYI bots. See https://webrtc-review.googlesource.com/c/src/+/206466 Original change's description: > Remove thread hops from events provided by JsepTransportController. > > Events associated with Subscribe* methods in JTC had trampolines that > would use an async invoker to fire the events on the signaling thread. > This was being done for the purposes of PeerConnection but the concept > of a signaling thread is otherwise not applicable to JTC and use of > JTC from PC is inconsistent across threads (as has been flagged in > webrtc:9987). > > This change makes all CallbackList members only accessible from the > network thread and moves the signaling thread related work over to > PeerConnection, which makes hops there more visible as well as making > that class easier to refactor for thread efficiency. > > This CL removes the AsyncInvoker from JTC (webrtc:12339) > > The signaling_thread_ variable is also removed from JTC and more thread > checks added to catch errors. > > Bug: webrtc:12427, webrtc:11988, webrtc:12339 > Change-Id: Id232aedd00dfd5403b2ba0ca147d3eca7c12c7c5 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206062 > Commit-Queue: Tommi <tommi@webrtc.org> > Reviewed-by: Niels Moller <nisse@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33195} TBR=nisse@webrtc.org,tommi@webrtc.org Change-Id: I6134b71b74a9408854b79d44506d513519e9cf4d No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:12427 Bug: webrtc:11988 Bug: webrtc:12339 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206467 Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Commit-Queue: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33203}
2021-02-09 12:26:13 +00:00
if (!signaling_thread_->IsCurrent()) {
signaled_on_non_signaling_thread_ = true;
}
combined_connection_state_ = state;
++combined_connection_state_signal_count_;
}
void OnGatheringState(cricket::IceGatheringState state) {
Revert "Remove thread hops from events provided by JsepTransportController." This reverts commit f554b3c577f69fa9ffad5c07155898c2d985ac76. Reason for revert: Parent CL breaks FYI bots. See https://webrtc-review.googlesource.com/c/src/+/206466 Original change's description: > Remove thread hops from events provided by JsepTransportController. > > Events associated with Subscribe* methods in JTC had trampolines that > would use an async invoker to fire the events on the signaling thread. > This was being done for the purposes of PeerConnection but the concept > of a signaling thread is otherwise not applicable to JTC and use of > JTC from PC is inconsistent across threads (as has been flagged in > webrtc:9987). > > This change makes all CallbackList members only accessible from the > network thread and moves the signaling thread related work over to > PeerConnection, which makes hops there more visible as well as making > that class easier to refactor for thread efficiency. > > This CL removes the AsyncInvoker from JTC (webrtc:12339) > > The signaling_thread_ variable is also removed from JTC and more thread > checks added to catch errors. > > Bug: webrtc:12427, webrtc:11988, webrtc:12339 > Change-Id: Id232aedd00dfd5403b2ba0ca147d3eca7c12c7c5 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206062 > Commit-Queue: Tommi <tommi@webrtc.org> > Reviewed-by: Niels Moller <nisse@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33195} TBR=nisse@webrtc.org,tommi@webrtc.org Change-Id: I6134b71b74a9408854b79d44506d513519e9cf4d No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:12427 Bug: webrtc:11988 Bug: webrtc:12339 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206467 Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Commit-Queue: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33203}
2021-02-09 12:26:13 +00:00
if (!signaling_thread_->IsCurrent()) {
signaled_on_non_signaling_thread_ = true;
}
gathering_state_ = state;
++gathering_state_signal_count_;
}
void OnCandidatesGathered(const std::string& transport_name,
const Candidates& candidates) {
Revert "Remove thread hops from events provided by JsepTransportController." This reverts commit f554b3c577f69fa9ffad5c07155898c2d985ac76. Reason for revert: Parent CL breaks FYI bots. See https://webrtc-review.googlesource.com/c/src/+/206466 Original change's description: > Remove thread hops from events provided by JsepTransportController. > > Events associated with Subscribe* methods in JTC had trampolines that > would use an async invoker to fire the events on the signaling thread. > This was being done for the purposes of PeerConnection but the concept > of a signaling thread is otherwise not applicable to JTC and use of > JTC from PC is inconsistent across threads (as has been flagged in > webrtc:9987). > > This change makes all CallbackList members only accessible from the > network thread and moves the signaling thread related work over to > PeerConnection, which makes hops there more visible as well as making > that class easier to refactor for thread efficiency. > > This CL removes the AsyncInvoker from JTC (webrtc:12339) > > The signaling_thread_ variable is also removed from JTC and more thread > checks added to catch errors. > > Bug: webrtc:12427, webrtc:11988, webrtc:12339 > Change-Id: Id232aedd00dfd5403b2ba0ca147d3eca7c12c7c5 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206062 > Commit-Queue: Tommi <tommi@webrtc.org> > Reviewed-by: Niels Moller <nisse@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33195} TBR=nisse@webrtc.org,tommi@webrtc.org Change-Id: I6134b71b74a9408854b79d44506d513519e9cf4d No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:12427 Bug: webrtc:11988 Bug: webrtc:12339 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206467 Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Commit-Queue: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33203}
2021-02-09 12:26:13 +00:00
if (!signaling_thread_->IsCurrent()) {
signaled_on_non_signaling_thread_ = true;
}
candidates_[transport_name].insert(candidates_[transport_name].end(),
candidates.begin(), candidates.end());
++candidates_signal_count_;
}
// JsepTransportController::Observer overrides.
Changes to enable use of DatagramTransport as a data channel transport. PeerConnection now has a new setting in RTCConfiguration to enable use of datagram transport for data channels. There is also a corresponding field trial, which has both a kill-switch and a way to change the default value. PeerConnection's interaction with MediaTransport for data channels has been refactored to work with DataChannelTransportInterface instead. Adds a DataChannelState and OnStateChanged() to the DataChannelSink callbacks. This allows PeerConnection to listen to the data channel's state directly, instead of indirectly by monitoring media transport state. This is necessary to enable use of non-media-transport (eg. datagram transport) data channel transports. For now, PeerConnection watches the state through MediaTransport as well. This will persist until MediaTransport implements the new callback. Datagram transport use is negotiated. As such, an offer that requests to use datagram transport for data channels may be rejected by the answerer. If the offer includes DTLS, the data channels will be negotiated as SCTP/DTLS data channels with an extra x-opaque parameter for datagram transport. If the opaque parameter is rejected (by an answerer without datagram support), the offerer may fall back to SCTP. If DTLS is not enabled, there is no viable fallback. In this case, the data channels are negotiated as media transport data channels. If the receiver does not understand the x-opaque line, it will reject these data channels, and the offerer's data channels will be closed. Bug: webrtc:9719 Change-Id: Ic1bf3664c4bcf9d754482df59897f5f72fe68fcc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147702 Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28932}
2019-08-21 10:44:59 -07:00
bool OnTransportChanged(
const std::string& mid,
RtpTransportInternal* rtp_transport,
rtc::scoped_refptr<DtlsTransport> dtls_transport,
Reland "Reland "Refactor SCTP data channels to use DataChannelTransportInterface."" This is a reland of 487f9a17e426fd14bb06b13e861071b3f15d119b Original change's description: > Reland "Refactor SCTP data channels to use DataChannelTransportInterface." > > Also clears SctpTransport before deleting JsepTransport. > > SctpTransport is ref-counted, but the underlying transport is deleted when > JsepTransport clears the rtp_dtls_transport. This results in crashes when > usrsctp attempts to send outgoing packets through a dangling pointer to the > underlying transport. > > Clearing SctpTransport before DtlsTransport removes the pointer to the > underlying transport before it becomes invalid. > > This fixes a crash in chromium's web platform tests (see > https://chromium-review.googlesource.com/c/chromium/src/+/1776711). > > Original change's description: > > Refactor SCTP data channels to use DataChannelTransportInterface. > > > > This change moves SctpTransport to be owned by JsepTransport, which now > > holds a DataChannelTransport implementation for SCTP when it is used for > > data channels. > > > > This simplifies negotiation and fallback to SCTP. Negotiation can now > > use a composite DataChannelTransport, just as negotiation for RTP uses a > > composite RTP transport. > > > > PeerConnection also has one fewer way it needs to manage data channels. > > It now handles SCTP and datagram- or media-transport-based data channels > > the same way. > > > > There are a few leaky abstractions left. For example, PeerConnection > > calls Start() on the SctpTransport at a particular point in negotiation, > > but does not need to call this for other transports. Similarly, PC > > exposes an interface to the SCTP transport directly to the user; there > > is no equivalent for other transports. > > Bug: webrtc:9719 > Change-Id: I64e94b88afb119fdbf5f22750f88c8a084d53937 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151981 > Reviewed-by: Benjamin Wright <benwright@webrtc.org> > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Commit-Queue: Benjamin Wright <benwright@webrtc.org> > Commit-Queue: Bjorn Mellem <mellem@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#29120} Bug: webrtc:9719 Change-Id: I28481a3de64a3506bc57748106383eeba4ef205c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152740 Reviewed-by: Artem Titov <titovartem@webrtc.org> Reviewed-by: Benjamin Wright <benwright@webrtc.org> Reviewed-by: Seth Hampson <shampson@webrtc.org> Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29290}
2019-09-23 14:53:54 -07:00
DataChannelTransportInterface* data_channel_transport) override {
changed_rtp_transport_by_mid_[mid] = rtp_transport;
if (dtls_transport) {
changed_dtls_transport_by_mid_[mid] = dtls_transport->internal();
} else {
changed_dtls_transport_by_mid_[mid] = nullptr;
}
return true;
}
// Information received from signals from transport controller.
cricket::IceConnectionState connection_state_ =
cricket::kIceConnectionConnecting;
PeerConnectionInterface::IceConnectionState ice_connection_state_ =
PeerConnectionInterface::kIceConnectionNew;
PeerConnectionInterface::PeerConnectionState combined_connection_state_ =
PeerConnectionInterface::PeerConnectionState::kNew;
bool receiving_ = false;
cricket::IceGatheringState gathering_state_ = cricket::kIceGatheringNew;
// transport_name => candidates
std::map<std::string, Candidates> candidates_;
// Counts of each signal emitted.
int connection_state_signal_count_ = 0;
int ice_connection_state_signal_count_ = 0;
int combined_connection_state_signal_count_ = 0;
int receiving_signal_count_ = 0;
int gathering_state_signal_count_ = 0;
int candidates_signal_count_ = 0;
// |network_thread_| should be destroyed after |transport_controller_|
std::unique_ptr<rtc::Thread> network_thread_;
std::unique_ptr<FakeIceTransportFactory> fake_ice_transport_factory_;
std::unique_ptr<FakeDtlsTransportFactory> fake_dtls_transport_factory_;
rtc::Thread* const signaling_thread_ = nullptr;
Revert "Remove thread hops from events provided by JsepTransportController." This reverts commit f554b3c577f69fa9ffad5c07155898c2d985ac76. Reason for revert: Parent CL breaks FYI bots. See https://webrtc-review.googlesource.com/c/src/+/206466 Original change's description: > Remove thread hops from events provided by JsepTransportController. > > Events associated with Subscribe* methods in JTC had trampolines that > would use an async invoker to fire the events on the signaling thread. > This was being done for the purposes of PeerConnection but the concept > of a signaling thread is otherwise not applicable to JTC and use of > JTC from PC is inconsistent across threads (as has been flagged in > webrtc:9987). > > This change makes all CallbackList members only accessible from the > network thread and moves the signaling thread related work over to > PeerConnection, which makes hops there more visible as well as making > that class easier to refactor for thread efficiency. > > This CL removes the AsyncInvoker from JTC (webrtc:12339) > > The signaling_thread_ variable is also removed from JTC and more thread > checks added to catch errors. > > Bug: webrtc:12427, webrtc:11988, webrtc:12339 > Change-Id: Id232aedd00dfd5403b2ba0ca147d3eca7c12c7c5 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206062 > Commit-Queue: Tommi <tommi@webrtc.org> > Reviewed-by: Niels Moller <nisse@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33195} TBR=nisse@webrtc.org,tommi@webrtc.org Change-Id: I6134b71b74a9408854b79d44506d513519e9cf4d No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:12427 Bug: webrtc:11988 Bug: webrtc:12339 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206467 Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Commit-Queue: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33203}
2021-02-09 12:26:13 +00:00
bool signaled_on_non_signaling_thread_ = false;
// Used to verify the SignalRtpTransportChanged/SignalDtlsTransportChanged are
// signaled correctly.
std::map<std::string, RtpTransportInternal*> changed_rtp_transport_by_mid_;
std::map<std::string, cricket::DtlsTransportInternal*>
changed_dtls_transport_by_mid_;
// Transport controller needs to be destroyed first, because it may issue
// callbacks that modify the changed_*_by_mid in the destructor.
std::unique_ptr<JsepTransportController> transport_controller_;
};
TEST_F(JsepTransportControllerTest, GetRtpTransport) {
CreateJsepTransportController(JsepTransportController::Config());
auto description = CreateSessionDescriptionWithoutBundle();
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
auto audio_rtp_transport = transport_controller_->GetRtpTransport(kAudioMid1);
auto video_rtp_transport = transport_controller_->GetRtpTransport(kVideoMid1);
EXPECT_NE(nullptr, audio_rtp_transport);
EXPECT_NE(nullptr, video_rtp_transport);
EXPECT_NE(audio_rtp_transport, video_rtp_transport);
// Return nullptr for non-existing ones.
EXPECT_EQ(nullptr, transport_controller_->GetRtpTransport(kAudioMid2));
}
TEST_F(JsepTransportControllerTest, GetDtlsTransport) {
JsepTransportController::Config config;
config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyNegotiate;
CreateJsepTransportController(config);
auto description = CreateSessionDescriptionWithoutBundle();
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
EXPECT_NE(nullptr, transport_controller_->GetDtlsTransport(kAudioMid1));
EXPECT_NE(nullptr, transport_controller_->GetRtcpDtlsTransport(kAudioMid1));
EXPECT_NE(nullptr,
transport_controller_->LookupDtlsTransportByMid(kAudioMid1));
EXPECT_NE(nullptr, transport_controller_->GetDtlsTransport(kVideoMid1));
EXPECT_NE(nullptr, transport_controller_->GetRtcpDtlsTransport(kVideoMid1));
EXPECT_NE(nullptr,
transport_controller_->LookupDtlsTransportByMid(kVideoMid1));
// Lookup for all MIDs should return different transports (no bundle)
EXPECT_NE(transport_controller_->LookupDtlsTransportByMid(kAudioMid1),
transport_controller_->LookupDtlsTransportByMid(kVideoMid1));
// Return nullptr for non-existing ones.
EXPECT_EQ(nullptr, transport_controller_->GetDtlsTransport(kVideoMid2));
EXPECT_EQ(nullptr, transport_controller_->GetRtcpDtlsTransport(kVideoMid2));
EXPECT_EQ(nullptr,
transport_controller_->LookupDtlsTransportByMid(kVideoMid2));
// Take a pointer to a transport, shut down the transport controller,
// and verify that the resulting container is empty.
auto dtls_transport =
transport_controller_->LookupDtlsTransportByMid(kVideoMid1);
webrtc::DtlsTransport* my_transport =
static_cast<DtlsTransport*>(dtls_transport.get());
EXPECT_NE(nullptr, my_transport->internal());
transport_controller_.reset();
EXPECT_EQ(nullptr, my_transport->internal());
}
TEST_F(JsepTransportControllerTest, GetDtlsTransportWithRtcpMux) {
JsepTransportController::Config config;
config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
CreateJsepTransportController(config);
auto description = CreateSessionDescriptionWithoutBundle();
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
EXPECT_NE(nullptr, transport_controller_->GetDtlsTransport(kAudioMid1));
EXPECT_EQ(nullptr, transport_controller_->GetRtcpDtlsTransport(kAudioMid1));
EXPECT_NE(nullptr, transport_controller_->GetDtlsTransport(kVideoMid1));
EXPECT_EQ(nullptr, transport_controller_->GetRtcpDtlsTransport(kVideoMid1));
}
TEST_F(JsepTransportControllerTest, SetIceConfig) {
CreateJsepTransportController(JsepTransportController::Config());
auto description = CreateSessionDescriptionWithoutBundle();
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
transport_controller_->SetIceConfig(
CreateIceConfig(kTimeout, cricket::GATHER_CONTINUALLY));
FakeDtlsTransport* fake_audio_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kAudioMid1));
ASSERT_NE(nullptr, fake_audio_dtls);
EXPECT_EQ(kTimeout,
fake_audio_dtls->fake_ice_transport()->receiving_timeout());
EXPECT_TRUE(fake_audio_dtls->fake_ice_transport()->gather_continually());
// Test that value stored in controller is applied to new transports.
AddAudioSection(description.get(), kAudioMid2, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
fake_audio_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kAudioMid2));
ASSERT_NE(nullptr, fake_audio_dtls);
EXPECT_EQ(kTimeout,
fake_audio_dtls->fake_ice_transport()->receiving_timeout());
EXPECT_TRUE(fake_audio_dtls->fake_ice_transport()->gather_continually());
}
// Tests the getter and setter of the ICE restart flag.
TEST_F(JsepTransportControllerTest, NeedIceRestart) {
CreateJsepTransportController(JsepTransportController::Config());
auto description = CreateSessionDescriptionWithoutBundle();
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
EXPECT_TRUE(transport_controller_
->SetRemoteDescription(SdpType::kAnswer, description.get())
.ok());
// Initially NeedsIceRestart should return false.
EXPECT_FALSE(transport_controller_->NeedsIceRestart(kAudioMid1));
EXPECT_FALSE(transport_controller_->NeedsIceRestart(kVideoMid1));
// Set the needs-ice-restart flag and verify NeedsIceRestart starts returning
// true.
transport_controller_->SetNeedsIceRestartFlag();
EXPECT_TRUE(transport_controller_->NeedsIceRestart(kAudioMid1));
EXPECT_TRUE(transport_controller_->NeedsIceRestart(kVideoMid1));
// For a nonexistent transport, false should be returned.
EXPECT_FALSE(transport_controller_->NeedsIceRestart(kVideoMid2));
// Reset the ice_ufrag/ice_pwd for audio.
auto audio_transport_info = description->GetTransportInfoByName(kAudioMid1);
audio_transport_info->description.ice_ufrag = kIceUfrag2;
audio_transport_info->description.ice_pwd = kIcePwd2;
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
// Because the ICE is only restarted for audio, NeedsIceRestart is expected to
// return false for audio and true for video.
EXPECT_FALSE(transport_controller_->NeedsIceRestart(kAudioMid1));
EXPECT_TRUE(transport_controller_->NeedsIceRestart(kVideoMid1));
}
TEST_F(JsepTransportControllerTest, MaybeStartGathering) {
CreateJsepTransportController(JsepTransportController::Config());
auto description = CreateSessionDescriptionWithBundleGroup();
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
// After setting the local description, we should be able to start gathering
// candidates.
transport_controller_->MaybeStartGathering();
EXPECT_EQ_WAIT(cricket::kIceGatheringGathering, gathering_state_, kTimeout);
EXPECT_EQ(1, gathering_state_signal_count_);
}
TEST_F(JsepTransportControllerTest, AddRemoveRemoteCandidates) {
CreateJsepTransportController(JsepTransportController::Config());
auto description = CreateSessionDescriptionWithoutBundle();
transport_controller_->SetLocalDescription(SdpType::kOffer,
description.get());
transport_controller_->SetRemoteDescription(SdpType::kAnswer,
description.get());
auto fake_audio_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kAudioMid1));
ASSERT_NE(nullptr, fake_audio_dtls);
Candidates candidates;
candidates.push_back(
CreateCandidate(kAudioMid1, cricket::ICE_CANDIDATE_COMPONENT_RTP));
EXPECT_TRUE(
transport_controller_->AddRemoteCandidates(kAudioMid1, candidates).ok());
EXPECT_EQ(1U,
fake_audio_dtls->fake_ice_transport()->remote_candidates().size());
EXPECT_TRUE(transport_controller_->RemoveRemoteCandidates(candidates).ok());
EXPECT_EQ(0U,
fake_audio_dtls->fake_ice_transport()->remote_candidates().size());
}
TEST_F(JsepTransportControllerTest, SetAndGetLocalCertificate) {
CreateJsepTransportController(JsepTransportController::Config());
rtc::scoped_refptr<rtc::RTCCertificate> certificate1 =
rtc::RTCCertificate::Create(
rtc::SSLIdentity::Create("session1", rtc::KT_DEFAULT));
rtc::scoped_refptr<rtc::RTCCertificate> returned_certificate;
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto description = std::make_unique<cricket::SessionDescription>();
AddAudioSection(description.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
certificate1);
// Apply the local certificate.
EXPECT_TRUE(transport_controller_->SetLocalCertificate(certificate1));
// Apply the local description.
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
returned_certificate = transport_controller_->GetLocalCertificate(kAudioMid1);
EXPECT_TRUE(returned_certificate);
EXPECT_EQ(certificate1->identity()->certificate().ToPEMString(),
returned_certificate->identity()->certificate().ToPEMString());
// Should fail if called for a nonexistant transport.
EXPECT_EQ(nullptr, transport_controller_->GetLocalCertificate(kVideoMid1));
// Shouldn't be able to change the identity once set.
rtc::scoped_refptr<rtc::RTCCertificate> certificate2 =
rtc::RTCCertificate::Create(
rtc::SSLIdentity::Create("session2", rtc::KT_DEFAULT));
EXPECT_FALSE(transport_controller_->SetLocalCertificate(certificate2));
}
TEST_F(JsepTransportControllerTest, GetRemoteSSLCertChain) {
CreateJsepTransportController(JsepTransportController::Config());
auto description = CreateSessionDescriptionWithBundleGroup();
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
rtc::FakeSSLCertificate fake_certificate("fake_data");
auto fake_audio_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kAudioMid1));
fake_audio_dtls->SetRemoteSSLCertificate(&fake_certificate);
std::unique_ptr<rtc::SSLCertChain> returned_cert_chain =
transport_controller_->GetRemoteSSLCertChain(kAudioMid1);
ASSERT_TRUE(returned_cert_chain);
ASSERT_EQ(1u, returned_cert_chain->GetSize());
EXPECT_EQ(fake_certificate.ToPEMString(),
returned_cert_chain->Get(0).ToPEMString());
// Should fail if called for a nonexistant transport.
EXPECT_FALSE(transport_controller_->GetRemoteSSLCertChain(kAudioMid2));
}
TEST_F(JsepTransportControllerTest, GetDtlsRole) {
CreateJsepTransportController(JsepTransportController::Config());
auto offer_certificate = rtc::RTCCertificate::Create(
rtc::SSLIdentity::Create("offer", rtc::KT_DEFAULT));
auto answer_certificate = rtc::RTCCertificate::Create(
rtc::SSLIdentity::Create("answer", rtc::KT_DEFAULT));
transport_controller_->SetLocalCertificate(offer_certificate);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto offer_desc = std::make_unique<cricket::SessionDescription>();
AddAudioSection(offer_desc.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
offer_certificate);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto answer_desc = std::make_unique<cricket::SessionDescription>();
AddAudioSection(answer_desc.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
answer_certificate);
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, offer_desc.get())
.ok());
absl::optional<rtc::SSLRole> role =
transport_controller_->GetDtlsRole(kAudioMid1);
// The DTLS role is not decided yet.
EXPECT_FALSE(role);
EXPECT_TRUE(transport_controller_
->SetRemoteDescription(SdpType::kAnswer, answer_desc.get())
.ok());
role = transport_controller_->GetDtlsRole(kAudioMid1);
ASSERT_TRUE(role);
EXPECT_EQ(rtc::SSL_CLIENT, *role);
}
TEST_F(JsepTransportControllerTest, GetStats) {
CreateJsepTransportController(JsepTransportController::Config());
auto description = CreateSessionDescriptionWithBundleGroup();
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
cricket::TransportStats stats;
EXPECT_TRUE(transport_controller_->GetStats(kAudioMid1, &stats));
EXPECT_EQ(kAudioMid1, stats.transport_name);
EXPECT_EQ(1u, stats.channel_stats.size());
// Return false for non-existing transport.
EXPECT_FALSE(transport_controller_->GetStats(kAudioMid2, &stats));
}
TEST_F(JsepTransportControllerTest, SignalConnectionStateFailed) {
CreateJsepTransportController(JsepTransportController::Config());
auto description = CreateSessionDescriptionWithoutBundle();
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
auto fake_ice = static_cast<cricket::FakeIceTransport*>(
transport_controller_->GetDtlsTransport(kAudioMid1)->ice_transport());
fake_ice->SetCandidatesGatheringComplete();
fake_ice->SetConnectionCount(1);
// The connection stats will be failed if there is no active connection.
fake_ice->SetConnectionCount(0);
EXPECT_EQ_WAIT(cricket::kIceConnectionFailed, connection_state_, kTimeout);
EXPECT_EQ(1, connection_state_signal_count_);
EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionFailed,
ice_connection_state_, kTimeout);
EXPECT_EQ(1, ice_connection_state_signal_count_);
EXPECT_EQ_WAIT(PeerConnectionInterface::PeerConnectionState::kFailed,
combined_connection_state_, kTimeout);
EXPECT_EQ(1, combined_connection_state_signal_count_);
}
TEST_F(JsepTransportControllerTest,
SignalConnectionStateConnectedNoMediaTransport) {
CreateJsepTransportController(JsepTransportController::Config());
auto description = CreateSessionDescriptionWithoutBundle();
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
auto fake_audio_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kAudioMid1));
auto fake_video_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kVideoMid1));
// First, have one transport connect, and another fail, to ensure that
// the first transport connecting didn't trigger a "connected" state signal.
// We should only get a signal when all are connected.
fake_audio_dtls->fake_ice_transport()->SetConnectionCount(1);
fake_audio_dtls->SetWritable(true);
fake_audio_dtls->fake_ice_transport()->SetCandidatesGatheringComplete();
// Decrease the number of the connection to trigger the signal.
fake_video_dtls->fake_ice_transport()->SetConnectionCount(1);
fake_video_dtls->fake_ice_transport()->SetConnectionCount(0);
fake_video_dtls->fake_ice_transport()->SetCandidatesGatheringComplete();
EXPECT_EQ_WAIT(cricket::kIceConnectionFailed, connection_state_, kTimeout);
EXPECT_EQ(1, connection_state_signal_count_);
EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionFailed,
ice_connection_state_, kTimeout);
EXPECT_EQ(2, ice_connection_state_signal_count_);
EXPECT_EQ_WAIT(PeerConnectionInterface::PeerConnectionState::kFailed,
combined_connection_state_, kTimeout);
EXPECT_EQ(2, combined_connection_state_signal_count_);
fake_audio_dtls->SetDtlsState(cricket::DTLS_TRANSPORT_CONNECTED);
fake_video_dtls->SetDtlsState(cricket::DTLS_TRANSPORT_CONNECTED);
// Set the connection count to be 2 and the cricket::FakeIceTransport will set
// the transport state to be STATE_CONNECTING.
fake_video_dtls->fake_ice_transport()->SetConnectionCount(2);
fake_video_dtls->SetWritable(true);
EXPECT_EQ_WAIT(cricket::kIceConnectionConnected, connection_state_, kTimeout);
EXPECT_EQ(2, connection_state_signal_count_);
EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionConnected,
ice_connection_state_, kTimeout);
EXPECT_EQ(3, ice_connection_state_signal_count_);
EXPECT_EQ_WAIT(PeerConnectionInterface::PeerConnectionState::kConnected,
combined_connection_state_, kTimeout);
EXPECT_EQ(3, combined_connection_state_signal_count_);
}
TEST_F(JsepTransportControllerTest, SignalConnectionStateComplete) {
CreateJsepTransportController(JsepTransportController::Config());
auto description = CreateSessionDescriptionWithoutBundle();
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
auto fake_audio_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kAudioMid1));
auto fake_video_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kVideoMid1));
// First, have one transport connect, and another fail, to ensure that
// the first transport connecting didn't trigger a "connected" state signal.
// We should only get a signal when all are connected.
fake_audio_dtls->fake_ice_transport()->SetTransportState(
IceTransportState::kCompleted,
cricket::IceTransportState::STATE_COMPLETED);
fake_audio_dtls->SetWritable(true);
fake_audio_dtls->fake_ice_transport()->SetCandidatesGatheringComplete();
EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionChecking,
ice_connection_state_, kTimeout);
EXPECT_EQ(1, ice_connection_state_signal_count_);
EXPECT_EQ_WAIT(PeerConnectionInterface::PeerConnectionState::kConnecting,
combined_connection_state_, kTimeout);
EXPECT_EQ(1, combined_connection_state_signal_count_);
fake_video_dtls->fake_ice_transport()->SetTransportState(
IceTransportState::kFailed, cricket::IceTransportState::STATE_FAILED);
fake_video_dtls->fake_ice_transport()->SetCandidatesGatheringComplete();
EXPECT_EQ_WAIT(cricket::kIceConnectionFailed, connection_state_, kTimeout);
EXPECT_EQ(1, connection_state_signal_count_);
EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionFailed,
ice_connection_state_, kTimeout);
EXPECT_EQ(2, ice_connection_state_signal_count_);
EXPECT_EQ_WAIT(PeerConnectionInterface::PeerConnectionState::kFailed,
combined_connection_state_, kTimeout);
EXPECT_EQ(2, combined_connection_state_signal_count_);
fake_audio_dtls->SetDtlsState(cricket::DTLS_TRANSPORT_CONNECTED);
fake_video_dtls->SetDtlsState(cricket::DTLS_TRANSPORT_CONNECTED);
// Set the connection count to be 1 and the cricket::FakeIceTransport will set
// the transport state to be STATE_COMPLETED.
fake_video_dtls->fake_ice_transport()->SetTransportState(
IceTransportState::kCompleted,
cricket::IceTransportState::STATE_COMPLETED);
fake_video_dtls->SetWritable(true);
EXPECT_EQ_WAIT(cricket::kIceConnectionCompleted, connection_state_, kTimeout);
EXPECT_EQ(3, connection_state_signal_count_);
EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
ice_connection_state_, kTimeout);
EXPECT_EQ(3, ice_connection_state_signal_count_);
EXPECT_EQ_WAIT(PeerConnectionInterface::PeerConnectionState::kConnected,
combined_connection_state_, kTimeout);
EXPECT_EQ(3, combined_connection_state_signal_count_);
}
TEST_F(JsepTransportControllerTest, SignalIceGatheringStateGathering) {
CreateJsepTransportController(JsepTransportController::Config());
auto description = CreateSessionDescriptionWithoutBundle();
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
auto fake_audio_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kAudioMid1));
fake_audio_dtls->fake_ice_transport()->MaybeStartGathering();
// Should be in the gathering state as soon as any transport starts gathering.
EXPECT_EQ_WAIT(cricket::kIceGatheringGathering, gathering_state_, kTimeout);
EXPECT_EQ(1, gathering_state_signal_count_);
}
TEST_F(JsepTransportControllerTest, SignalIceGatheringStateComplete) {
CreateJsepTransportController(JsepTransportController::Config());
auto description = CreateSessionDescriptionWithoutBundle();
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
auto fake_audio_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kAudioMid1));
auto fake_video_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kVideoMid1));
fake_audio_dtls->fake_ice_transport()->MaybeStartGathering();
EXPECT_EQ_WAIT(cricket::kIceGatheringGathering, gathering_state_, kTimeout);
EXPECT_EQ(1, gathering_state_signal_count_);
// Have one transport finish gathering, to make sure gathering
// completion wasn't signalled if only one transport finished gathering.
fake_audio_dtls->fake_ice_transport()->SetCandidatesGatheringComplete();
EXPECT_EQ(1, gathering_state_signal_count_);
fake_video_dtls->fake_ice_transport()->MaybeStartGathering();
EXPECT_EQ_WAIT(cricket::kIceGatheringGathering, gathering_state_, kTimeout);
EXPECT_EQ(1, gathering_state_signal_count_);
fake_video_dtls->fake_ice_transport()->SetCandidatesGatheringComplete();
EXPECT_EQ_WAIT(cricket::kIceGatheringComplete, gathering_state_, kTimeout);
EXPECT_EQ(2, gathering_state_signal_count_);
}
// Test that when the last transport that hasn't finished connecting and/or
// gathering is destroyed, the aggregate state jumps to "completed". This can
// happen if, for example, we have an audio and video transport, the audio
// transport completes, then we start bundling video on the audio transport.
TEST_F(JsepTransportControllerTest,
SignalingWhenLastIncompleteTransportDestroyed) {
CreateJsepTransportController(JsepTransportController::Config());
auto description = CreateSessionDescriptionWithBundleGroup();
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
auto fake_audio_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kAudioMid1));
auto fake_video_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kVideoMid1));
EXPECT_NE(fake_audio_dtls, fake_video_dtls);
fake_audio_dtls->fake_ice_transport()->MaybeStartGathering();
EXPECT_EQ_WAIT(cricket::kIceGatheringGathering, gathering_state_, kTimeout);
EXPECT_EQ(1, gathering_state_signal_count_);
// Let the audio transport complete.
fake_audio_dtls->SetWritable(true);
fake_audio_dtls->fake_ice_transport()->SetCandidatesGatheringComplete();
fake_audio_dtls->fake_ice_transport()->SetConnectionCount(1);
fake_audio_dtls->SetDtlsState(cricket::DTLS_TRANSPORT_CONNECTED);
EXPECT_EQ(1, gathering_state_signal_count_);
// Set the remote description and enable the bundle.
EXPECT_TRUE(transport_controller_
->SetRemoteDescription(SdpType::kAnswer, description.get())
.ok());
// The BUNDLE should be enabled, the incomplete video transport should be
// deleted and the states shoud be updated.
fake_video_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kVideoMid1));
EXPECT_EQ(fake_audio_dtls, fake_video_dtls);
EXPECT_EQ_WAIT(cricket::kIceConnectionCompleted, connection_state_, kTimeout);
EXPECT_EQ(PeerConnectionInterface::kIceConnectionCompleted,
ice_connection_state_);
EXPECT_EQ(PeerConnectionInterface::PeerConnectionState::kConnected,
combined_connection_state_);
EXPECT_EQ_WAIT(cricket::kIceGatheringComplete, gathering_state_, kTimeout);
EXPECT_EQ(2, gathering_state_signal_count_);
}
TEST_F(JsepTransportControllerTest, SignalCandidatesGathered) {
CreateJsepTransportController(JsepTransportController::Config());
auto description = CreateSessionDescriptionWithBundleGroup();
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, description.get())
.ok());
transport_controller_->MaybeStartGathering();
auto fake_audio_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kAudioMid1));
fake_audio_dtls->fake_ice_transport()->SignalCandidateGathered(
fake_audio_dtls->fake_ice_transport(), CreateCandidate(kAudioMid1, 1));
EXPECT_EQ_WAIT(1, candidates_signal_count_, kTimeout);
EXPECT_EQ(1u, candidates_[kAudioMid1].size());
}
Revert "Remove thread hops from events provided by JsepTransportController." This reverts commit f554b3c577f69fa9ffad5c07155898c2d985ac76. Reason for revert: Parent CL breaks FYI bots. See https://webrtc-review.googlesource.com/c/src/+/206466 Original change's description: > Remove thread hops from events provided by JsepTransportController. > > Events associated with Subscribe* methods in JTC had trampolines that > would use an async invoker to fire the events on the signaling thread. > This was being done for the purposes of PeerConnection but the concept > of a signaling thread is otherwise not applicable to JTC and use of > JTC from PC is inconsistent across threads (as has been flagged in > webrtc:9987). > > This change makes all CallbackList members only accessible from the > network thread and moves the signaling thread related work over to > PeerConnection, which makes hops there more visible as well as making > that class easier to refactor for thread efficiency. > > This CL removes the AsyncInvoker from JTC (webrtc:12339) > > The signaling_thread_ variable is also removed from JTC and more thread > checks added to catch errors. > > Bug: webrtc:12427, webrtc:11988, webrtc:12339 > Change-Id: Id232aedd00dfd5403b2ba0ca147d3eca7c12c7c5 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206062 > Commit-Queue: Tommi <tommi@webrtc.org> > Reviewed-by: Niels Moller <nisse@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33195} TBR=nisse@webrtc.org,tommi@webrtc.org Change-Id: I6134b71b74a9408854b79d44506d513519e9cf4d No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:12427 Bug: webrtc:11988 Bug: webrtc:12339 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206467 Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Commit-Queue: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33203}
2021-02-09 12:26:13 +00:00
TEST_F(JsepTransportControllerTest, IceSignalingOccursOnSignalingThread) {
network_thread_ = rtc::Thread::CreateWithSocketServer();
network_thread_->Start();
CreateJsepTransportController(JsepTransportController::Config(),
Revert "Remove thread hops from events provided by JsepTransportController." This reverts commit f554b3c577f69fa9ffad5c07155898c2d985ac76. Reason for revert: Parent CL breaks FYI bots. See https://webrtc-review.googlesource.com/c/src/+/206466 Original change's description: > Remove thread hops from events provided by JsepTransportController. > > Events associated with Subscribe* methods in JTC had trampolines that > would use an async invoker to fire the events on the signaling thread. > This was being done for the purposes of PeerConnection but the concept > of a signaling thread is otherwise not applicable to JTC and use of > JTC from PC is inconsistent across threads (as has been flagged in > webrtc:9987). > > This change makes all CallbackList members only accessible from the > network thread and moves the signaling thread related work over to > PeerConnection, which makes hops there more visible as well as making > that class easier to refactor for thread efficiency. > > This CL removes the AsyncInvoker from JTC (webrtc:12339) > > The signaling_thread_ variable is also removed from JTC and more thread > checks added to catch errors. > > Bug: webrtc:12427, webrtc:11988, webrtc:12339 > Change-Id: Id232aedd00dfd5403b2ba0ca147d3eca7c12c7c5 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206062 > Commit-Queue: Tommi <tommi@webrtc.org> > Reviewed-by: Niels Moller <nisse@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33195} TBR=nisse@webrtc.org,tommi@webrtc.org Change-Id: I6134b71b74a9408854b79d44506d513519e9cf4d No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:12427 Bug: webrtc:11988 Bug: webrtc:12339 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206467 Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Commit-Queue: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33203}
2021-02-09 12:26:13 +00:00
signaling_thread_, network_thread_.get(),
/*port_allocator=*/nullptr);
CreateLocalDescriptionAndCompleteConnectionOnNetworkThread();
// connecting --> connected --> completed
EXPECT_EQ_WAIT(cricket::kIceConnectionCompleted, connection_state_, kTimeout);
EXPECT_EQ(2, connection_state_signal_count_);
// new --> gathering --> complete
EXPECT_EQ_WAIT(cricket::kIceGatheringComplete, gathering_state_, kTimeout);
EXPECT_EQ(2, gathering_state_signal_count_);
EXPECT_EQ_WAIT(1u, candidates_[kAudioMid1].size(), kTimeout);
EXPECT_EQ_WAIT(1u, candidates_[kVideoMid1].size(), kTimeout);
EXPECT_EQ(2, candidates_signal_count_);
Revert "Remove thread hops from events provided by JsepTransportController." This reverts commit f554b3c577f69fa9ffad5c07155898c2d985ac76. Reason for revert: Parent CL breaks FYI bots. See https://webrtc-review.googlesource.com/c/src/+/206466 Original change's description: > Remove thread hops from events provided by JsepTransportController. > > Events associated with Subscribe* methods in JTC had trampolines that > would use an async invoker to fire the events on the signaling thread. > This was being done for the purposes of PeerConnection but the concept > of a signaling thread is otherwise not applicable to JTC and use of > JTC from PC is inconsistent across threads (as has been flagged in > webrtc:9987). > > This change makes all CallbackList members only accessible from the > network thread and moves the signaling thread related work over to > PeerConnection, which makes hops there more visible as well as making > that class easier to refactor for thread efficiency. > > This CL removes the AsyncInvoker from JTC (webrtc:12339) > > The signaling_thread_ variable is also removed from JTC and more thread > checks added to catch errors. > > Bug: webrtc:12427, webrtc:11988, webrtc:12339 > Change-Id: Id232aedd00dfd5403b2ba0ca147d3eca7c12c7c5 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206062 > Commit-Queue: Tommi <tommi@webrtc.org> > Reviewed-by: Niels Moller <nisse@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33195} TBR=nisse@webrtc.org,tommi@webrtc.org Change-Id: I6134b71b74a9408854b79d44506d513519e9cf4d No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:12427 Bug: webrtc:11988 Bug: webrtc:12339 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206467 Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Commit-Queue: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33203}
2021-02-09 12:26:13 +00:00
EXPECT_TRUE(!signaled_on_non_signaling_thread_);
Reland "Fix unsynchronized access to mid_to_transport_ in JsepTransportController" This reverts commit 6b143c1c0686519bc9d73223c1350cee286c8d78. Reason for revert: Relanding with updated expectations for SctpTransport::Information based on TransceiverStateSurfacer in Chromium. Original change's description: > Revert "Fix unsynchronized access to mid_to_transport_ in JsepTransportController" > > This reverts commit 6cd405850467683cf10d05028ea0f644a68a91a4. > > Reason for revert: Breaks WebRTC Chromium FYI Bots > > First failure: > https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Android%20Tests%20%28dbg%29%20%28L%20Nexus5%29/1925 > > Failed tests: > WebRtcDataBrowserTest.CallWithSctpDataAndMedia > WebRtcDataBrowserTest.CallWithSctpDataOnly > > Original change's description: > > Fix unsynchronized access to mid_to_transport_ in JsepTransportController > > > > * Added several thread checks to JTC to help with programmer errors. > > * Avoid a few Invokes() to the network thread here and there such > > as for fetching sctp transport name for getStats(). The transport > > name is now cached when it changes on the network thread. > > * JsepTransportController instances now get deleted on the network > > thread rather than on the signaling thread + issuing an Invoke() > > in the dtor. > > * Moved some thread hops from JTC over to PC which is where the problem > > exists and also (imho) makes it easier to see where hops happen in > > the PC code. > > * The sctp transport is now started asynchronously when we push down the > > media description. > > * PeerConnection proxy calls GetSctpTransport directly on the network > > thread instead of to the signaling thread + blocking on the network > > thread. > > * The above changes simplified things for webrtc::SctpTransport which > > allowed for removing locking from that class and delete some code. > > > > Bug: webrtc:9987, webrtc:12445 > > Change-Id: Ic89a9426e314e1b93c81751d4f732f05fa448fbc > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205620 > > Commit-Queue: Tommi <tommi@webrtc.org> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33191} > > TBR=tommi@webrtc.org,hta@webrtc.org > > Change-Id: I7b2913d5133807589461105cf07eff3e9bb7157e > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9987 > Bug: webrtc:12445 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206466 > Reviewed-by: Guido Urdaneta <guidou@webrtc.org> > Commit-Queue: Guido Urdaneta <guidou@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33204} TBR=tommi@webrtc.org,hta@webrtc.org,guidou@webrtc.org # Not skipping CQ checks because this is a reland. Bug: webrtc:9987 Bug: webrtc:12445 Change-Id: Icb205cbac493ed3b881d71ea3af4fb9018701bf4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206560 Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33219}
2021-02-10 13:05:44 +01:00
network_thread_->Invoke<void>(RTC_FROM_HERE,
[&] { transport_controller_.reset(); });
}
// Test that if the TransportController was created with the
// |redetermine_role_on_ice_restart| parameter set to false, the role is *not*
// redetermined on an ICE restart.
TEST_F(JsepTransportControllerTest, IceRoleNotRedetermined) {
JsepTransportController::Config config;
config.redetermine_role_on_ice_restart = false;
CreateJsepTransportController(config);
// Let the |transport_controller_| be the controlled side initially.
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto remote_offer = std::make_unique<cricket::SessionDescription>();
AddAudioSection(remote_offer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto local_answer = std::make_unique<cricket::SessionDescription>();
AddAudioSection(local_answer.get(), kAudioMid1, kIceUfrag2, kIcePwd2,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
EXPECT_TRUE(transport_controller_
->SetRemoteDescription(SdpType::kOffer, remote_offer.get())
.ok());
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kAnswer, local_answer.get())
.ok());
auto fake_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kAudioMid1));
EXPECT_EQ(cricket::ICEROLE_CONTROLLED,
fake_dtls->fake_ice_transport()->GetIceRole());
// New offer will trigger the ICE restart.
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto restart_local_offer = std::make_unique<cricket::SessionDescription>();
AddAudioSection(restart_local_offer.get(), kAudioMid1, kIceUfrag3, kIcePwd3,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
EXPECT_TRUE(
transport_controller_
->SetLocalDescription(SdpType::kOffer, restart_local_offer.get())
.ok());
EXPECT_EQ(cricket::ICEROLE_CONTROLLED,
fake_dtls->fake_ice_transport()->GetIceRole());
}
// Tests ICE-Lite mode in remote answer.
TEST_F(JsepTransportControllerTest, SetIceRoleWhenIceLiteInRemoteAnswer) {
CreateJsepTransportController(JsepTransportController::Config());
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto local_offer = std::make_unique<cricket::SessionDescription>();
AddAudioSection(local_offer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, local_offer.get())
.ok());
auto fake_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kAudioMid1));
EXPECT_EQ(cricket::ICEROLE_CONTROLLING,
fake_dtls->fake_ice_transport()->GetIceRole());
EXPECT_EQ(cricket::ICEMODE_FULL,
fake_dtls->fake_ice_transport()->remote_ice_mode());
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto remote_answer = std::make_unique<cricket::SessionDescription>();
AddAudioSection(remote_answer.get(), kAudioMid1, kIceUfrag2, kIcePwd2,
cricket::ICEMODE_LITE, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
EXPECT_TRUE(transport_controller_
->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
.ok());
EXPECT_EQ(cricket::ICEROLE_CONTROLLING,
fake_dtls->fake_ice_transport()->GetIceRole());
EXPECT_EQ(cricket::ICEMODE_LITE,
fake_dtls->fake_ice_transport()->remote_ice_mode());
}
// Tests that the ICE role remains "controlling" if a subsequent offer that
// does an ICE restart is received from an ICE lite endpoint. Regression test
// for: https://crbug.com/710760
TEST_F(JsepTransportControllerTest,
IceRoleIsControllingAfterIceRestartFromIceLiteEndpoint) {
CreateJsepTransportController(JsepTransportController::Config());
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto remote_offer = std::make_unique<cricket::SessionDescription>();
AddAudioSection(remote_offer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_LITE, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto local_answer = std::make_unique<cricket::SessionDescription>();
AddAudioSection(local_answer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
// Initial Offer/Answer exchange. If the remote offerer is ICE-Lite, then the
// local side is the controlling.
EXPECT_TRUE(transport_controller_
->SetRemoteDescription(SdpType::kOffer, remote_offer.get())
.ok());
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kAnswer, local_answer.get())
.ok());
auto fake_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kAudioMid1));
EXPECT_EQ(cricket::ICEROLE_CONTROLLING,
fake_dtls->fake_ice_transport()->GetIceRole());
// In the subsequence remote offer triggers an ICE restart.
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto remote_offer2 = std::make_unique<cricket::SessionDescription>();
AddAudioSection(remote_offer2.get(), kAudioMid1, kIceUfrag2, kIcePwd2,
cricket::ICEMODE_LITE, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto local_answer2 = std::make_unique<cricket::SessionDescription>();
AddAudioSection(local_answer2.get(), kAudioMid1, kIceUfrag2, kIcePwd2,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
EXPECT_TRUE(transport_controller_
->SetRemoteDescription(SdpType::kOffer, remote_offer2.get())
.ok());
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kAnswer, local_answer2.get())
.ok());
fake_dtls = static_cast<FakeDtlsTransport*>(
transport_controller_->GetDtlsTransport(kAudioMid1));
// The local side is still the controlling role since the remote side is using
// ICE-Lite.
EXPECT_EQ(cricket::ICEROLE_CONTROLLING,
fake_dtls->fake_ice_transport()->GetIceRole());
}
// Tests that the SDP has more than one audio/video m= sections.
TEST_F(JsepTransportControllerTest, MultipleMediaSectionsOfSameTypeWithBundle) {
CreateJsepTransportController(JsepTransportController::Config());
cricket::ContentGroup bundle_group(cricket::GROUP_TYPE_BUNDLE);
bundle_group.AddContentName(kAudioMid1);
bundle_group.AddContentName(kAudioMid2);
bundle_group.AddContentName(kVideoMid1);
bundle_group.AddContentName(kDataMid1);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto local_offer = std::make_unique<cricket::SessionDescription>();
AddAudioSection(local_offer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
AddAudioSection(local_offer.get(), kAudioMid2, kIceUfrag2, kIcePwd2,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
AddVideoSection(local_offer.get(), kVideoMid1, kIceUfrag3, kIcePwd3,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
AddDataSection(local_offer.get(), kDataMid1,
cricket::MediaProtocolType::kSctp, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto remote_answer = std::make_unique<cricket::SessionDescription>();
AddAudioSection(remote_answer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
AddAudioSection(remote_answer.get(), kAudioMid2, kIceUfrag2, kIcePwd2,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
AddVideoSection(remote_answer.get(), kVideoMid1, kIceUfrag3, kIcePwd3,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
AddDataSection(remote_answer.get(), kDataMid1,
cricket::MediaProtocolType::kSctp, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
local_offer->AddGroup(bundle_group);
remote_answer->AddGroup(bundle_group);
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, local_offer.get())
.ok());
EXPECT_TRUE(transport_controller_
->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
.ok());
// Verify that all the sections are bundled on kAudio1.
auto transport1 = transport_controller_->GetRtpTransport(kAudioMid1);
auto transport2 = transport_controller_->GetRtpTransport(kAudioMid2);
auto transport3 = transport_controller_->GetRtpTransport(kVideoMid1);
auto transport4 = transport_controller_->GetRtpTransport(kDataMid1);
EXPECT_EQ(transport1, transport2);
EXPECT_EQ(transport1, transport3);
EXPECT_EQ(transport1, transport4);
EXPECT_EQ(transport_controller_->LookupDtlsTransportByMid(kAudioMid1),
transport_controller_->LookupDtlsTransportByMid(kVideoMid1));
// Verify the OnRtpTransport/DtlsTransportChanged signals are fired correctly.
auto it = changed_rtp_transport_by_mid_.find(kAudioMid2);
ASSERT_TRUE(it != changed_rtp_transport_by_mid_.end());
EXPECT_EQ(transport1, it->second);
it = changed_rtp_transport_by_mid_.find(kAudioMid2);
ASSERT_TRUE(it != changed_rtp_transport_by_mid_.end());
EXPECT_EQ(transport1, it->second);
it = changed_rtp_transport_by_mid_.find(kVideoMid1);
ASSERT_TRUE(it != changed_rtp_transport_by_mid_.end());
EXPECT_EQ(transport1, it->second);
// Verify the DtlsTransport for the SCTP data channel is reset correctly.
auto it2 = changed_dtls_transport_by_mid_.find(kDataMid1);
ASSERT_TRUE(it2 != changed_dtls_transport_by_mid_.end());
}
// Tests that only a subset of all the m= sections are bundled.
TEST_F(JsepTransportControllerTest, BundleSubsetOfMediaSections) {
CreateJsepTransportController(JsepTransportController::Config());
cricket::ContentGroup bundle_group(cricket::GROUP_TYPE_BUNDLE);
bundle_group.AddContentName(kAudioMid1);
bundle_group.AddContentName(kVideoMid1);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto local_offer = std::make_unique<cricket::SessionDescription>();
AddAudioSection(local_offer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
AddAudioSection(local_offer.get(), kAudioMid2, kIceUfrag2, kIcePwd2,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
AddVideoSection(local_offer.get(), kVideoMid1, kIceUfrag3, kIcePwd3,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto remote_answer = std::make_unique<cricket::SessionDescription>();
AddAudioSection(remote_answer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
AddAudioSection(remote_answer.get(), kAudioMid2, kIceUfrag2, kIcePwd2,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
AddVideoSection(remote_answer.get(), kVideoMid1, kIceUfrag3, kIcePwd3,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
local_offer->AddGroup(bundle_group);
remote_answer->AddGroup(bundle_group);
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, local_offer.get())
.ok());
EXPECT_TRUE(transport_controller_
->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
.ok());
// Verifiy that only |kAudio1| and |kVideo1| are bundled.
auto transport1 = transport_controller_->GetRtpTransport(kAudioMid1);
auto transport2 = transport_controller_->GetRtpTransport(kAudioMid2);
auto transport3 = transport_controller_->GetRtpTransport(kVideoMid1);
EXPECT_NE(transport1, transport2);
EXPECT_EQ(transport1, transport3);
auto it = changed_rtp_transport_by_mid_.find(kVideoMid1);
ASSERT_TRUE(it != changed_rtp_transport_by_mid_.end());
EXPECT_EQ(transport1, it->second);
it = changed_rtp_transport_by_mid_.find(kAudioMid2);
EXPECT_TRUE(transport2 == it->second);
}
// Tests that the initial offer/answer only have data section and audio/video
// sections are added in the subsequent offer.
TEST_F(JsepTransportControllerTest, BundleOnDataSectionInSubsequentOffer) {
CreateJsepTransportController(JsepTransportController::Config());
cricket::ContentGroup bundle_group(cricket::GROUP_TYPE_BUNDLE);
bundle_group.AddContentName(kDataMid1);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto local_offer = std::make_unique<cricket::SessionDescription>();
AddDataSection(local_offer.get(), kDataMid1,
cricket::MediaProtocolType::kSctp, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto remote_answer = std::make_unique<cricket::SessionDescription>();
AddDataSection(remote_answer.get(), kDataMid1,
cricket::MediaProtocolType::kSctp, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
local_offer->AddGroup(bundle_group);
remote_answer->AddGroup(bundle_group);
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, local_offer.get())
.ok());
EXPECT_TRUE(transport_controller_
->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
.ok());
auto data_transport = transport_controller_->GetRtpTransport(kDataMid1);
// Add audio/video sections in subsequent offer.
AddAudioSection(local_offer.get(), kAudioMid1, kIceUfrag2, kIcePwd2,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
AddVideoSection(local_offer.get(), kVideoMid1, kIceUfrag3, kIcePwd3,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
AddAudioSection(remote_answer.get(), kAudioMid1, kIceUfrag2, kIcePwd2,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
AddVideoSection(remote_answer.get(), kVideoMid1, kIceUfrag3, kIcePwd3,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
// Reset the bundle group and do another offer/answer exchange.
bundle_group.AddContentName(kAudioMid1);
bundle_group.AddContentName(kVideoMid1);
local_offer->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
remote_answer->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
local_offer->AddGroup(bundle_group);
remote_answer->AddGroup(bundle_group);
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, local_offer.get())
.ok());
EXPECT_TRUE(transport_controller_
->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
.ok());
auto audio_transport = transport_controller_->GetRtpTransport(kAudioMid1);
auto video_transport = transport_controller_->GetRtpTransport(kVideoMid1);
EXPECT_EQ(data_transport, audio_transport);
EXPECT_EQ(data_transport, video_transport);
}
TEST_F(JsepTransportControllerTest, VideoDataRejectedInAnswer) {
CreateJsepTransportController(JsepTransportController::Config());
cricket::ContentGroup bundle_group(cricket::GROUP_TYPE_BUNDLE);
bundle_group.AddContentName(kAudioMid1);
bundle_group.AddContentName(kVideoMid1);
bundle_group.AddContentName(kDataMid1);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto local_offer = std::make_unique<cricket::SessionDescription>();
AddAudioSection(local_offer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
AddVideoSection(local_offer.get(), kVideoMid1, kIceUfrag2, kIcePwd2,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
AddDataSection(local_offer.get(), kDataMid1,
cricket::MediaProtocolType::kSctp, kIceUfrag3, kIcePwd3,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto remote_answer = std::make_unique<cricket::SessionDescription>();
AddAudioSection(remote_answer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
AddVideoSection(remote_answer.get(), kVideoMid1, kIceUfrag2, kIcePwd2,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
AddDataSection(remote_answer.get(), kDataMid1,
cricket::MediaProtocolType::kSctp, kIceUfrag3, kIcePwd3,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
// Reject video and data section.
remote_answer->contents()[1].rejected = true;
remote_answer->contents()[2].rejected = true;
local_offer->AddGroup(bundle_group);
remote_answer->AddGroup(bundle_group);
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, local_offer.get())
.ok());
EXPECT_TRUE(transport_controller_
->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
.ok());
// Verify the RtpTransport/DtlsTransport is destroyed correctly.
EXPECT_EQ(nullptr, transport_controller_->GetRtpTransport(kVideoMid1));
EXPECT_EQ(nullptr, transport_controller_->GetDtlsTransport(kDataMid1));
// Verify the signals are fired correctly.
auto it = changed_rtp_transport_by_mid_.find(kVideoMid1);
ASSERT_TRUE(it != changed_rtp_transport_by_mid_.end());
EXPECT_EQ(nullptr, it->second);
auto it2 = changed_dtls_transport_by_mid_.find(kDataMid1);
ASSERT_TRUE(it2 != changed_dtls_transport_by_mid_.end());
EXPECT_EQ(nullptr, it2->second);
}
// Tests that changing the bundled MID in subsequent offer/answer exchange is
// not supported.
// TODO(bugs.webrtc.org/6704): Change this test to expect success once issue is
// fixed
TEST_F(JsepTransportControllerTest, ChangeBundledMidNotSupported) {
CreateJsepTransportController(JsepTransportController::Config());
cricket::ContentGroup bundle_group(cricket::GROUP_TYPE_BUNDLE);
bundle_group.AddContentName(kAudioMid1);
bundle_group.AddContentName(kVideoMid1);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto local_offer = std::make_unique<cricket::SessionDescription>();
AddAudioSection(local_offer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
AddVideoSection(local_offer.get(), kVideoMid1, kIceUfrag2, kIcePwd2,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto remote_answer = std::make_unique<cricket::SessionDescription>();
AddAudioSection(remote_answer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
AddVideoSection(remote_answer.get(), kVideoMid1, kIceUfrag2, kIcePwd2,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
local_offer->AddGroup(bundle_group);
remote_answer->AddGroup(bundle_group);
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, local_offer.get())
.ok());
EXPECT_TRUE(transport_controller_
->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
.ok());
EXPECT_EQ(transport_controller_->GetRtpTransport(kAudioMid1),
transport_controller_->GetRtpTransport(kVideoMid1));
// Reorder the bundle group.
EXPECT_TRUE(bundle_group.RemoveContentName(kAudioMid1));
bundle_group.AddContentName(kAudioMid1);
// The answerer uses the new bundle group and now the bundle mid is changed to
// |kVideo1|.
remote_answer->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
remote_answer->AddGroup(bundle_group);
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, local_offer.get())
.ok());
EXPECT_FALSE(transport_controller_
->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
.ok());
}
// Test that rejecting only the first m= section of a BUNDLE group is treated as
// an error, but rejecting all of them works as expected.
TEST_F(JsepTransportControllerTest, RejectFirstContentInBundleGroup) {
CreateJsepTransportController(JsepTransportController::Config());
cricket::ContentGroup bundle_group(cricket::GROUP_TYPE_BUNDLE);
bundle_group.AddContentName(kAudioMid1);
bundle_group.AddContentName(kVideoMid1);
bundle_group.AddContentName(kDataMid1);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto local_offer = std::make_unique<cricket::SessionDescription>();
AddAudioSection(local_offer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
AddVideoSection(local_offer.get(), kVideoMid1, kIceUfrag2, kIcePwd2,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
AddDataSection(local_offer.get(), kDataMid1,
cricket::MediaProtocolType::kSctp, kIceUfrag3, kIcePwd3,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto remote_answer = std::make_unique<cricket::SessionDescription>();
AddAudioSection(remote_answer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
AddVideoSection(remote_answer.get(), kVideoMid1, kIceUfrag2, kIcePwd2,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
AddDataSection(remote_answer.get(), kDataMid1,
cricket::MediaProtocolType::kSctp, kIceUfrag3, kIcePwd3,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
// Reject audio content in answer.
remote_answer->contents()[0].rejected = true;
local_offer->AddGroup(bundle_group);
remote_answer->AddGroup(bundle_group);
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, local_offer.get())
.ok());
EXPECT_FALSE(transport_controller_
->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
.ok());
// Reject all the contents.
remote_answer->contents()[1].rejected = true;
remote_answer->contents()[2].rejected = true;
EXPECT_TRUE(transport_controller_
->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
.ok());
EXPECT_EQ(nullptr, transport_controller_->GetRtpTransport(kAudioMid1));
EXPECT_EQ(nullptr, transport_controller_->GetRtpTransport(kVideoMid1));
EXPECT_EQ(nullptr, transport_controller_->GetDtlsTransport(kDataMid1));
}
// Tests that applying non-RTCP-mux offer would fail when kRtcpMuxPolicyRequire
// is used.
TEST_F(JsepTransportControllerTest, ApplyNonRtcpMuxOfferWhenMuxingRequired) {
JsepTransportController::Config config;
config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
CreateJsepTransportController(config);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto local_offer = std::make_unique<cricket::SessionDescription>();
AddAudioSection(local_offer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
local_offer->contents()[0].media_description()->set_rtcp_mux(false);
// Applying a non-RTCP-mux offer is expected to fail.
EXPECT_FALSE(transport_controller_
->SetLocalDescription(SdpType::kOffer, local_offer.get())
.ok());
}
// Tests that applying non-RTCP-mux answer would fail when kRtcpMuxPolicyRequire
// is used.
TEST_F(JsepTransportControllerTest, ApplyNonRtcpMuxAnswerWhenMuxingRequired) {
JsepTransportController::Config config;
config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
CreateJsepTransportController(config);
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto local_offer = std::make_unique<cricket::SessionDescription>();
AddAudioSection(local_offer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, local_offer.get())
.ok());
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto remote_answer = std::make_unique<cricket::SessionDescription>();
AddAudioSection(remote_answer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
nullptr);
// Applying a non-RTCP-mux answer is expected to fail.
remote_answer->contents()[0].media_description()->set_rtcp_mux(false);
EXPECT_FALSE(transport_controller_
->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
.ok());
}
// This tests that the BUNDLE group in answer should be a subset of the offered
// group.
TEST_F(JsepTransportControllerTest,
AddContentToBundleGroupInAnswerNotSupported) {
CreateJsepTransportController(JsepTransportController::Config());
auto local_offer = CreateSessionDescriptionWithoutBundle();
auto remote_answer = CreateSessionDescriptionWithoutBundle();
cricket::ContentGroup offer_bundle_group(cricket::GROUP_TYPE_BUNDLE);
offer_bundle_group.AddContentName(kAudioMid1);
local_offer->AddGroup(offer_bundle_group);
cricket::ContentGroup answer_bundle_group(cricket::GROUP_TYPE_BUNDLE);
answer_bundle_group.AddContentName(kAudioMid1);
answer_bundle_group.AddContentName(kVideoMid1);
remote_answer->AddGroup(answer_bundle_group);
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, local_offer.get())
.ok());
EXPECT_FALSE(transport_controller_
->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
.ok());
}
// This tests that the BUNDLE group with non-existing MID should be rejectd.
TEST_F(JsepTransportControllerTest, RejectBundleGroupWithNonExistingMid) {
CreateJsepTransportController(JsepTransportController::Config());
auto local_offer = CreateSessionDescriptionWithoutBundle();
auto remote_answer = CreateSessionDescriptionWithoutBundle();
cricket::ContentGroup invalid_bundle_group(cricket::GROUP_TYPE_BUNDLE);
// The BUNDLE group is invalid because there is no data section in the
// description.
invalid_bundle_group.AddContentName(kDataMid1);
local_offer->AddGroup(invalid_bundle_group);
remote_answer->AddGroup(invalid_bundle_group);
EXPECT_FALSE(transport_controller_
->SetLocalDescription(SdpType::kOffer, local_offer.get())
.ok());
EXPECT_FALSE(transport_controller_
->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
.ok());
}
// This tests that an answer shouldn't be able to remove an m= section from an
// established group without rejecting it.
TEST_F(JsepTransportControllerTest, RemoveContentFromBundleGroup) {
CreateJsepTransportController(JsepTransportController::Config());
auto local_offer = CreateSessionDescriptionWithBundleGroup();
auto remote_answer = CreateSessionDescriptionWithBundleGroup();
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, local_offer.get())
.ok());
EXPECT_TRUE(transport_controller_
->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
.ok());
// Do an re-offer/answer.
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, local_offer.get())
.ok());
auto new_answer = CreateSessionDescriptionWithoutBundle();
cricket::ContentGroup new_bundle_group(cricket::GROUP_TYPE_BUNDLE);
// The answer removes video from the BUNDLE group without rejecting it is
// invalid.
new_bundle_group.AddContentName(kAudioMid1);
new_answer->AddGroup(new_bundle_group);
// Applying invalid answer is expected to fail.
EXPECT_FALSE(transport_controller_
->SetRemoteDescription(SdpType::kAnswer, new_answer.get())
.ok());
// Rejected the video content.
auto video_content = new_answer->GetContentByName(kVideoMid1);
ASSERT_TRUE(video_content);
video_content->rejected = true;
EXPECT_TRUE(transport_controller_
->SetRemoteDescription(SdpType::kAnswer, new_answer.get())
.ok());
}
// Test that the JsepTransportController can process a new local and remote
// description that changes the tagged BUNDLE group with the max-bundle policy
// specified.
// This is a regression test for bugs.webrtc.org/9954
TEST_F(JsepTransportControllerTest, ChangeTaggedMediaSectionMaxBundle) {
CreateJsepTransportController(JsepTransportController::Config());
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
auto local_offer = std::make_unique<cricket::SessionDescription>();
AddAudioSection(local_offer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
cricket::ContentGroup bundle_group(cricket::GROUP_TYPE_BUNDLE);
bundle_group.AddContentName(kAudioMid1);
local_offer->AddGroup(bundle_group);
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, local_offer.get())
.ok());
std::unique_ptr<cricket::SessionDescription> remote_answer(
local_offer->Clone());
EXPECT_TRUE(transport_controller_
->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
.ok());
std::unique_ptr<cricket::SessionDescription> local_reoffer(
local_offer->Clone());
local_reoffer->contents()[0].rejected = true;
AddVideoSection(local_reoffer.get(), kVideoMid1, kIceUfrag1, kIcePwd1,
cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
nullptr);
local_reoffer->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
cricket::ContentGroup new_bundle_group(cricket::GROUP_TYPE_BUNDLE);
new_bundle_group.AddContentName(kVideoMid1);
local_reoffer->AddGroup(new_bundle_group);
EXPECT_TRUE(transport_controller_
->SetLocalDescription(SdpType::kOffer, local_reoffer.get())
.ok());
std::unique_ptr<cricket::SessionDescription> remote_reanswer(
local_reoffer->Clone());
EXPECT_TRUE(
transport_controller_
->SetRemoteDescription(SdpType::kAnswer, remote_reanswer.get())
.ok());
}
} // namespace webrtc