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/*
* Copyright 2015 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.
*/
#import "RTCPeerConnection+Private.h"
#import "NSString+StdString.h"
#import "RTCConfiguration+Private.h"
#import "RTCDataChannel+Private.h"
#import "RTCIceCandidate+Private.h"
#import "RTCLegacyStatsReport+Private.h"
#import "RTCMediaConstraints+Private.h"
#import "RTCMediaStream+Private.h"
#import "RTCPeerConnectionFactory+Private.h"
#import "RTCRtpReceiver+Private.h"
#import "RTCRtpSender+Private.h"
#import "RTCSessionDescription+Private.h"
#import "WebRTC/RTCLogging.h"
#include <memory>
#include "webrtc/api/jsepicecandidate.h"
#include "webrtc/rtc_base/checks.h"
NSString * const kRTCPeerConnectionErrorDomain =
@"org.webrtc.RTCPeerConnection";
int const kRTCPeerConnnectionSessionDescriptionError = -1;
namespace webrtc {
class CreateSessionDescriptionObserverAdapter
: public CreateSessionDescriptionObserver {
public:
CreateSessionDescriptionObserverAdapter(
void (^completionHandler)(RTCSessionDescription *sessionDescription,
NSError *error)) {
completion_handler_ = completionHandler;
}
~CreateSessionDescriptionObserverAdapter() {
completion_handler_ = nil;
}
void OnSuccess(SessionDescriptionInterface *desc) override {
RTC_DCHECK(completion_handler_);
std::unique_ptr<webrtc::SessionDescriptionInterface> description =
std::unique_ptr<webrtc::SessionDescriptionInterface>(desc);
RTCSessionDescription* session =
[[RTCSessionDescription alloc] initWithNativeDescription:
description.get()];
completion_handler_(session, nil);
completion_handler_ = nil;
}
void OnFailure(const std::string& error) override {
RTC_DCHECK(completion_handler_);
NSString* str = [NSString stringForStdString:error];
NSError* err =
[NSError errorWithDomain:kRTCPeerConnectionErrorDomain
code:kRTCPeerConnnectionSessionDescriptionError
userInfo:@{ NSLocalizedDescriptionKey : str }];
completion_handler_(nil, err);
completion_handler_ = nil;
}
private:
void (^completion_handler_)
(RTCSessionDescription *sessionDescription, NSError *error);
};
class SetSessionDescriptionObserverAdapter :
public SetSessionDescriptionObserver {
public:
SetSessionDescriptionObserverAdapter(void (^completionHandler)
(NSError *error)) {
completion_handler_ = completionHandler;
}
~SetSessionDescriptionObserverAdapter() {
completion_handler_ = nil;
}
void OnSuccess() override {
RTC_DCHECK(completion_handler_);
completion_handler_(nil);
completion_handler_ = nil;
}
void OnFailure(const std::string& error) override {
RTC_DCHECK(completion_handler_);
NSString* str = [NSString stringForStdString:error];
NSError* err =
[NSError errorWithDomain:kRTCPeerConnectionErrorDomain
code:kRTCPeerConnnectionSessionDescriptionError
userInfo:@{ NSLocalizedDescriptionKey : str }];
completion_handler_(err);
completion_handler_ = nil;
}
private:
void (^completion_handler_)(NSError *error);
};
PeerConnectionDelegateAdapter::PeerConnectionDelegateAdapter(
RTCPeerConnection *peerConnection) {
peer_connection_ = peerConnection;
}
PeerConnectionDelegateAdapter::~PeerConnectionDelegateAdapter() {
peer_connection_ = nil;
}
void PeerConnectionDelegateAdapter::OnSignalingChange(
PeerConnectionInterface::SignalingState new_state) {
RTCSignalingState state =
[[RTCPeerConnection class] signalingStateForNativeState:new_state];
RTCPeerConnection *peer_connection = peer_connection_;
[peer_connection.delegate peerConnection:peer_connection
didChangeSignalingState:state];
}
void PeerConnectionDelegateAdapter::OnAddStream(
rtc::scoped_refptr<MediaStreamInterface> stream) {
Revert of Add the OnAddTrack callback for Objective-C wrapper. (patchset #7 id:240001 of https://codereview.webrtc.org/2513063003/ ) Reason for revert: This CL breaks iOS AppRTCMobile. We don't have any automatic tests running on the bots yet, so please try AppRTCMobile locally before relanding. Stack trace: * thread #15: tid = 0x20e933, 0x0000000100488440 AppRTCMobile`webrtc::AudioRtpReceiver::OnFirstPacketReceived(this=0x0000000170156c60, channel=0x000000010511a600) + 48 at rtpreceiver.cc:133, name = 'Thread 0x0x10421b2a0', stop reason = EXC_BAD_ACCESS (code=1, address=0x1a1aac71979) * frame #0: 0x0000000100488440 AppRTCMobile`webrtc::AudioRtpReceiver::OnFirstPacketReceived(this=0x0000000170156c60, channel=0x000000010511a600) + 48 at rtpreceiver.cc:133 frame #1: 0x000000010048a3f8 AppRTCMobile`void sigslot::_opaque_connection::emitter<webrtc::AudioRtpReceiver, cricket::BaseChannel*>(self=0x000000017424b380, args=0x000000010511a600) + 184 at sigslot.h:391 frame #2: 0x00000001005a30ec AppRTCMobile`void sigslot::_opaque_connection::emit<cricket::BaseChannel*>(this=0x000000017424b380, args=0x000000010511a600) const + 56 at sigslot.h:381 frame #3: 0x00000001005a3094 AppRTCMobile`sigslot::signal_with_thread_policy<sigslot::single_threaded, cricket::BaseChannel*>::emit(this=0x000000010511a678, args=0x000000010511a600) + 504 at sigslot.h:615 frame #4: 0x000000010057ef5c AppRTCMobile`sigslot::signal_with_thread_policy<sigslot::single_threaded, cricket::BaseChannel*>::operator(this=0x000000010511a678, args=0x000000010511a600)(cricket::BaseChannel*) + 32 at sigslot.h:621 frame #5: 0x000000010057ef00 AppRTCMobile`cricket::BaseChannel::OnMessage(this=0x000000010511a600, pmsg=0x000000016e676db0) + 600 at channel.cc:1494 frame #6: 0x0000000100584a58 AppRTCMobile`cricket::VoiceChannel::OnMessage(this=0x000000010511a600, pmsg=0x000000016e676db0) + 152 at channel.cc:1909 frame #7: 0x000000010017c0dc AppRTCMobile`rtc::MessageQueue::Dispatch(this=0x000000010421b2a0, pmsg=0x000000016e676db0) + 336 at messagequeue.cc:538 frame #8: 0x00000001001d8efc AppRTCMobile`rtc::Thread::ProcessMessages(this=0x000000010421b2a0, cmsLoop=-1) + 228 at thread.cc:496 frame #9: 0x00000001001d8e08 AppRTCMobile`rtc::Thread::Run(this=0x000000010421b2a0) + 28 at thread.cc:327 frame #10: 0x00000001001d8b0c AppRTCMobile`rtc::Thread::PreRun(pv=0x000000017000f030) + 300 at thread.cc:316 frame #11: 0x00000001843f1850 libsystem_pthread.dylib`_pthread_body + 240 frame #12: 0x00000001843f1760 libsystem_pthread.dylib`_pthread_start + 284 frame #13: 0x00000001843eed94 libsystem_pthread.dylib`thread_start + 4 Original issue's description: > Add the OnAddTrack callback for Objective-C wrapper. > > Created an Obj-C wrapper for the callback OnAddTrack in this CL since it has been added to native C++ API > The callback function is called when a track is signaled by remote side and a new RtpReceiver is created. > The application can tell when tracks are added to the streams by listening to this callback. > > BUG=webrtc:6112 > > Review-Url: https://codereview.webrtc.org/2513063003 > Cr-Commit-Position: refs/heads/master@{#16835} > Committed: https://chromium.googlesource.com/external/webrtc/+/633f6fe0046131ed815098298b9a3120bac1d7a0 TBR=tkchin@webrtc.org,deadbeef@webrtc.org,zhihuang@webrtc.org # Not skipping CQ checks because original CL landed more than 1 days ago. BUG=webrtc:6112 Review-Url: https://codereview.webrtc.org/2720753002 Cr-Commit-Position: refs/heads/master@{#16871}
2017-02-27 07:04:25 -08:00
RTCMediaStream *mediaStream =
[[RTCMediaStream alloc] initWithNativeMediaStream:stream];
RTCPeerConnection *peer_connection = peer_connection_;
[peer_connection.delegate peerConnection:peer_connection
didAddStream:mediaStream];
}
void PeerConnectionDelegateAdapter::OnRemoveStream(
rtc::scoped_refptr<MediaStreamInterface> stream) {
RTCMediaStream *mediaStream =
[[RTCMediaStream alloc] initWithNativeMediaStream:stream];
RTCPeerConnection *peer_connection = peer_connection_;
[peer_connection.delegate peerConnection:peer_connection
didRemoveStream:mediaStream];
}
void PeerConnectionDelegateAdapter::OnDataChannel(
rtc::scoped_refptr<DataChannelInterface> data_channel) {
RTCDataChannel *dataChannel =
[[RTCDataChannel alloc] initWithNativeDataChannel:data_channel];
RTCPeerConnection *peer_connection = peer_connection_;
[peer_connection.delegate peerConnection:peer_connection
didOpenDataChannel:dataChannel];
}
void PeerConnectionDelegateAdapter::OnRenegotiationNeeded() {
RTCPeerConnection *peer_connection = peer_connection_;
[peer_connection.delegate peerConnectionShouldNegotiate:peer_connection];
}
void PeerConnectionDelegateAdapter::OnIceConnectionChange(
PeerConnectionInterface::IceConnectionState new_state) {
RTCIceConnectionState state =
[[RTCPeerConnection class] iceConnectionStateForNativeState:new_state];
RTCPeerConnection *peer_connection = peer_connection_;
[peer_connection.delegate peerConnection:peer_connection
didChangeIceConnectionState:state];
}
void PeerConnectionDelegateAdapter::OnIceGatheringChange(
PeerConnectionInterface::IceGatheringState new_state) {
RTCIceGatheringState state =
[[RTCPeerConnection class] iceGatheringStateForNativeState:new_state];
RTCPeerConnection *peer_connection = peer_connection_;
[peer_connection.delegate peerConnection:peer_connection
didChangeIceGatheringState:state];
}
void PeerConnectionDelegateAdapter::OnIceCandidate(
const IceCandidateInterface *candidate) {
RTCIceCandidate *iceCandidate =
[[RTCIceCandidate alloc] initWithNativeCandidate:candidate];
RTCPeerConnection *peer_connection = peer_connection_;
[peer_connection.delegate peerConnection:peer_connection
didGenerateIceCandidate:iceCandidate];
}
void PeerConnectionDelegateAdapter::OnIceCandidatesRemoved(
const std::vector<cricket::Candidate>& candidates) {
NSMutableArray* ice_candidates =
[NSMutableArray arrayWithCapacity:candidates.size()];
for (const auto& candidate : candidates) {
std::unique_ptr<JsepIceCandidate> candidate_wrapper(
new JsepIceCandidate(candidate.transport_name(), -1, candidate));
RTCIceCandidate* ice_candidate = [[RTCIceCandidate alloc]
initWithNativeCandidate:candidate_wrapper.get()];
[ice_candidates addObject:ice_candidate];
}
RTCPeerConnection* peer_connection = peer_connection_;
[peer_connection.delegate peerConnection:peer_connection
didRemoveIceCandidates:ice_candidates];
}
} // namespace webrtc
@implementation RTCPeerConnection {
NSMutableArray<RTCMediaStream *> *_localStreams;
std::unique_ptr<webrtc::PeerConnectionDelegateAdapter> _observer;
rtc::scoped_refptr<webrtc::PeerConnectionInterface> _peerConnection;
std::unique_ptr<webrtc::MediaConstraints> _nativeConstraints;
BOOL _hasStartedRtcEventLog;
}
@synthesize delegate = _delegate;
- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
configuration:(RTCConfiguration *)configuration
constraints:(RTCMediaConstraints *)constraints
delegate:(id<RTCPeerConnectionDelegate>)delegate {
NSParameterAssert(factory);
std::unique_ptr<webrtc::PeerConnectionInterface::RTCConfiguration> config(
[configuration createNativeConfiguration]);
if (!config) {
return nil;
}
if (self = [super init]) {
_observer.reset(new webrtc::PeerConnectionDelegateAdapter(self));
_nativeConstraints = constraints.nativeConstraints;
CopyConstraintsIntoRtcConfiguration(_nativeConstraints.get(),
config.get());
_peerConnection =
Revert of Replacing DtlsIdentityStoreInterface with RTCCertificateGeneratorInterface. (patchset #2 id:20001 of https://codereview.webrtc.org/2013523002/ ) Reason for revert: There are more CreatePeerConnection calls than I anticipated/had found in Chromium, like remoting/protocol/webrtc_transport.cc. Reverting due to broken Chromium FYI bots. Original issue's description: > Replacing DtlsIdentityStoreInterface with RTCCertificateGeneratorInterface. > > The store was used in WebRtcSessionDescriptionFactory to generate certificates, > now a generator is used instead (new API). PeerConnection[Factory][Interface], > and WebRtcSession are updated to pass generators all the way down to the > WebRtcSessionDescriptionFactory instead of stores. > > The webrtc implementation of a generator, RTCCertificateGenerator, is used as > the default generator (peerconnectionfactory.cc:189) instead of the webrtc > implementation of a store, DtlsIdentityStoreImpl. > The generator is fully parameterized and does not generate RSA-1024 unless you > ask for it (which makes sense not to do beforehand since ECDSA is now default). > The store was not fully parameterized (known filed bug). > > The "top" layer, PeerConnectionFactoryInterface::CreatePeerConnection, is > updated to take a generator instead of a store. But as to not break Chromium, > the old function signature taking a store is kept. It is implemented to invoke > the generator version by wrapping the store in an > RTCCertificateGeneratorStoreWrapper. As soon as Chromium is updated to use the > new function signature we can remove the old CreatePeerConnection. > Due to having multiple CreatePeerConnection signatures, some calling places > are updated to resolve the ambiguity introduced. > > BUG=webrtc:5707, webrtc:5708 > R=phoglund@webrtc.org, tommi@webrtc.org > TBR=tkchin@webrc.org > > Committed: https://chromium.googlesource.com/external/webrtc/+/400781a2091d09a725b32c6953247036b22478e8 TBR=tkchin@webrtc.org,tommi@webrtc.org,phoglund@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:5707, webrtc:5708 Review-Url: https://codereview.webrtc.org/2020633002 Cr-Commit-Position: refs/heads/master@{#12948}
2016-05-27 06:08:53 -07:00
factory.nativeFactory->CreatePeerConnection(*config,
nullptr,
nullptr,
_observer.get());
if (!_peerConnection) {
return nil;
}
_localStreams = [[NSMutableArray alloc] init];
_delegate = delegate;
}
return self;
}
- (NSArray<RTCMediaStream *> *)localStreams {
return [_localStreams copy];
}
- (RTCSessionDescription *)localDescription {
const webrtc::SessionDescriptionInterface *description =
_peerConnection->local_description();
return description ?
[[RTCSessionDescription alloc] initWithNativeDescription:description]
: nil;
}
- (RTCSessionDescription *)remoteDescription {
const webrtc::SessionDescriptionInterface *description =
_peerConnection->remote_description();
return description ?
[[RTCSessionDescription alloc] initWithNativeDescription:description]
: nil;
}
- (RTCSignalingState)signalingState {
return [[self class]
signalingStateForNativeState:_peerConnection->signaling_state()];
}
- (RTCIceConnectionState)iceConnectionState {
return [[self class] iceConnectionStateForNativeState:
_peerConnection->ice_connection_state()];
}
- (RTCIceGatheringState)iceGatheringState {
return [[self class] iceGatheringStateForNativeState:
_peerConnection->ice_gathering_state()];
}
- (BOOL)setConfiguration:(RTCConfiguration *)configuration {
std::unique_ptr<webrtc::PeerConnectionInterface::RTCConfiguration> config(
[configuration createNativeConfiguration]);
if (!config) {
return NO;
}
CopyConstraintsIntoRtcConfiguration(_nativeConstraints.get(),
config.get());
return _peerConnection->SetConfiguration(*config);
}
- (RTCConfiguration *)configuration {
webrtc::PeerConnectionInterface::RTCConfiguration config =
_peerConnection->GetConfiguration();
return [[RTCConfiguration alloc] initWithNativeConfiguration:config];
}
- (void)close {
_peerConnection->Close();
}
- (void)addIceCandidate:(RTCIceCandidate *)candidate {
std::unique_ptr<const webrtc::IceCandidateInterface> iceCandidate(
candidate.nativeCandidate);
_peerConnection->AddIceCandidate(iceCandidate.get());
}
- (void)removeIceCandidates:(NSArray<RTCIceCandidate *> *)iceCandidates {
std::vector<cricket::Candidate> candidates;
for (RTCIceCandidate *iceCandidate in iceCandidates) {
std::unique_ptr<const webrtc::IceCandidateInterface> candidate(
iceCandidate.nativeCandidate);
if (candidate) {
candidates.push_back(candidate->candidate());
// Need to fill the transport name from the sdp_mid.
candidates.back().set_transport_name(candidate->sdp_mid());
}
}
if (!candidates.empty()) {
_peerConnection->RemoveIceCandidates(candidates);
}
}
- (void)addStream:(RTCMediaStream *)stream {
if (!_peerConnection->AddStream(stream.nativeMediaStream)) {
RTCLogError(@"Failed to add stream: %@", stream);
return;
}
[_localStreams addObject:stream];
}
- (void)removeStream:(RTCMediaStream *)stream {
_peerConnection->RemoveStream(stream.nativeMediaStream);
[_localStreams removeObject:stream];
}
- (void)offerForConstraints:(RTCMediaConstraints *)constraints
completionHandler:
(void (^)(RTCSessionDescription *sessionDescription,
NSError *error))completionHandler {
rtc::scoped_refptr<webrtc::CreateSessionDescriptionObserverAdapter>
observer(new rtc::RefCountedObject
<webrtc::CreateSessionDescriptionObserverAdapter>(completionHandler));
_peerConnection->CreateOffer(observer, constraints.nativeConstraints.get());
}
- (void)answerForConstraints:(RTCMediaConstraints *)constraints
completionHandler:
(void (^)(RTCSessionDescription *sessionDescription,
NSError *error))completionHandler {
rtc::scoped_refptr<webrtc::CreateSessionDescriptionObserverAdapter>
observer(new rtc::RefCountedObject
<webrtc::CreateSessionDescriptionObserverAdapter>(completionHandler));
_peerConnection->CreateAnswer(observer, constraints.nativeConstraints.get());
}
- (void)setLocalDescription:(RTCSessionDescription *)sdp
completionHandler:(void (^)(NSError *error))completionHandler {
rtc::scoped_refptr<webrtc::SetSessionDescriptionObserverAdapter> observer(
new rtc::RefCountedObject<webrtc::SetSessionDescriptionObserverAdapter>(
completionHandler));
_peerConnection->SetLocalDescription(observer, sdp.nativeDescription);
}
- (void)setRemoteDescription:(RTCSessionDescription *)sdp
completionHandler:(void (^)(NSError *error))completionHandler {
rtc::scoped_refptr<webrtc::SetSessionDescriptionObserverAdapter> observer(
new rtc::RefCountedObject<webrtc::SetSessionDescriptionObserverAdapter>(
completionHandler));
_peerConnection->SetRemoteDescription(observer, sdp.nativeDescription);
}
- (BOOL)startRtcEventLogWithFilePath:(NSString *)filePath
maxSizeInBytes:(int64_t)maxSizeInBytes {
RTC_DCHECK(filePath.length);
RTC_DCHECK_GT(maxSizeInBytes, 0);
RTC_DCHECK(!_hasStartedRtcEventLog);
if (_hasStartedRtcEventLog) {
RTCLogError(@"Event logging already started.");
return NO;
}
int fd = open(filePath.UTF8String, O_WRONLY | O_CREAT | O_TRUNC,
S_IRUSR | S_IWUSR);
if (fd < 0) {
RTCLogError(@"Error opening file: %@. Error: %d", filePath, errno);
return NO;
}
_hasStartedRtcEventLog =
_peerConnection->StartRtcEventLog(fd, maxSizeInBytes);
return _hasStartedRtcEventLog;
}
- (void)stopRtcEventLog {
_peerConnection->StopRtcEventLog();
_hasStartedRtcEventLog = NO;
}
- (RTCRtpSender *)senderWithKind:(NSString *)kind
streamId:(NSString *)streamId {
std::string nativeKind = [NSString stdStringForString:kind];
std::string nativeStreamId = [NSString stdStringForString:streamId];
rtc::scoped_refptr<webrtc::RtpSenderInterface> nativeSender(
_peerConnection->CreateSender(nativeKind, nativeStreamId));
return nativeSender ?
[[RTCRtpSender alloc] initWithNativeRtpSender:nativeSender]
: nil;
}
- (NSArray<RTCRtpSender *> *)senders {
std::vector<rtc::scoped_refptr<webrtc::RtpSenderInterface>> nativeSenders(
_peerConnection->GetSenders());
NSMutableArray *senders = [[NSMutableArray alloc] init];
for (const auto &nativeSender : nativeSenders) {
RTCRtpSender *sender =
[[RTCRtpSender alloc] initWithNativeRtpSender:nativeSender];
[senders addObject:sender];
}
return senders;
}
- (NSArray<RTCRtpReceiver *> *)receivers {
std::vector<rtc::scoped_refptr<webrtc::RtpReceiverInterface>> nativeReceivers(
_peerConnection->GetReceivers());
NSMutableArray *receivers = [[NSMutableArray alloc] init];
for (const auto &nativeReceiver : nativeReceivers) {
RTCRtpReceiver *receiver =
[[RTCRtpReceiver alloc] initWithNativeRtpReceiver:nativeReceiver];
[receivers addObject:receiver];
}
return receivers;
}
#pragma mark - Private
+ (webrtc::PeerConnectionInterface::SignalingState)nativeSignalingStateForState:
(RTCSignalingState)state {
switch (state) {
case RTCSignalingStateStable:
return webrtc::PeerConnectionInterface::kStable;
case RTCSignalingStateHaveLocalOffer:
return webrtc::PeerConnectionInterface::kHaveLocalOffer;
case RTCSignalingStateHaveLocalPrAnswer:
return webrtc::PeerConnectionInterface::kHaveLocalPrAnswer;
case RTCSignalingStateHaveRemoteOffer:
return webrtc::PeerConnectionInterface::kHaveRemoteOffer;
case RTCSignalingStateHaveRemotePrAnswer:
return webrtc::PeerConnectionInterface::kHaveRemotePrAnswer;
case RTCSignalingStateClosed:
return webrtc::PeerConnectionInterface::kClosed;
}
}
+ (RTCSignalingState)signalingStateForNativeState:
(webrtc::PeerConnectionInterface::SignalingState)nativeState {
switch (nativeState) {
case webrtc::PeerConnectionInterface::kStable:
return RTCSignalingStateStable;
case webrtc::PeerConnectionInterface::kHaveLocalOffer:
return RTCSignalingStateHaveLocalOffer;
case webrtc::PeerConnectionInterface::kHaveLocalPrAnswer:
return RTCSignalingStateHaveLocalPrAnswer;
case webrtc::PeerConnectionInterface::kHaveRemoteOffer:
return RTCSignalingStateHaveRemoteOffer;
case webrtc::PeerConnectionInterface::kHaveRemotePrAnswer:
return RTCSignalingStateHaveRemotePrAnswer;
case webrtc::PeerConnectionInterface::kClosed:
return RTCSignalingStateClosed;
}
}
+ (NSString *)stringForSignalingState:(RTCSignalingState)state {
switch (state) {
case RTCSignalingStateStable:
return @"STABLE";
case RTCSignalingStateHaveLocalOffer:
return @"HAVE_LOCAL_OFFER";
case RTCSignalingStateHaveLocalPrAnswer:
return @"HAVE_LOCAL_PRANSWER";
case RTCSignalingStateHaveRemoteOffer:
return @"HAVE_REMOTE_OFFER";
case RTCSignalingStateHaveRemotePrAnswer:
return @"HAVE_REMOTE_PRANSWER";
case RTCSignalingStateClosed:
return @"CLOSED";
}
}
+ (webrtc::PeerConnectionInterface::IceConnectionState)
nativeIceConnectionStateForState:(RTCIceConnectionState)state {
switch (state) {
case RTCIceConnectionStateNew:
return webrtc::PeerConnectionInterface::kIceConnectionNew;
case RTCIceConnectionStateChecking:
return webrtc::PeerConnectionInterface::kIceConnectionChecking;
case RTCIceConnectionStateConnected:
return webrtc::PeerConnectionInterface::kIceConnectionConnected;
case RTCIceConnectionStateCompleted:
return webrtc::PeerConnectionInterface::kIceConnectionCompleted;
case RTCIceConnectionStateFailed:
return webrtc::PeerConnectionInterface::kIceConnectionFailed;
case RTCIceConnectionStateDisconnected:
return webrtc::PeerConnectionInterface::kIceConnectionDisconnected;
case RTCIceConnectionStateClosed:
return webrtc::PeerConnectionInterface::kIceConnectionClosed;
case RTCIceConnectionStateCount:
return webrtc::PeerConnectionInterface::kIceConnectionMax;
}
}
+ (RTCIceConnectionState)iceConnectionStateForNativeState:
(webrtc::PeerConnectionInterface::IceConnectionState)nativeState {
switch (nativeState) {
case webrtc::PeerConnectionInterface::kIceConnectionNew:
return RTCIceConnectionStateNew;
case webrtc::PeerConnectionInterface::kIceConnectionChecking:
return RTCIceConnectionStateChecking;
case webrtc::PeerConnectionInterface::kIceConnectionConnected:
return RTCIceConnectionStateConnected;
case webrtc::PeerConnectionInterface::kIceConnectionCompleted:
return RTCIceConnectionStateCompleted;
case webrtc::PeerConnectionInterface::kIceConnectionFailed:
return RTCIceConnectionStateFailed;
case webrtc::PeerConnectionInterface::kIceConnectionDisconnected:
return RTCIceConnectionStateDisconnected;
case webrtc::PeerConnectionInterface::kIceConnectionClosed:
return RTCIceConnectionStateClosed;
case webrtc::PeerConnectionInterface::kIceConnectionMax:
return RTCIceConnectionStateCount;
}
}
+ (NSString *)stringForIceConnectionState:(RTCIceConnectionState)state {
switch (state) {
case RTCIceConnectionStateNew:
return @"NEW";
case RTCIceConnectionStateChecking:
return @"CHECKING";
case RTCIceConnectionStateConnected:
return @"CONNECTED";
case RTCIceConnectionStateCompleted:
return @"COMPLETED";
case RTCIceConnectionStateFailed:
return @"FAILED";
case RTCIceConnectionStateDisconnected:
return @"DISCONNECTED";
case RTCIceConnectionStateClosed:
return @"CLOSED";
case RTCIceConnectionStateCount:
return @"COUNT";
}
}
+ (webrtc::PeerConnectionInterface::IceGatheringState)
nativeIceGatheringStateForState:(RTCIceGatheringState)state {
switch (state) {
case RTCIceGatheringStateNew:
return webrtc::PeerConnectionInterface::kIceGatheringNew;
case RTCIceGatheringStateGathering:
return webrtc::PeerConnectionInterface::kIceGatheringGathering;
case RTCIceGatheringStateComplete:
return webrtc::PeerConnectionInterface::kIceGatheringComplete;
}
}
+ (RTCIceGatheringState)iceGatheringStateForNativeState:
(webrtc::PeerConnectionInterface::IceGatheringState)nativeState {
switch (nativeState) {
case webrtc::PeerConnectionInterface::kIceGatheringNew:
return RTCIceGatheringStateNew;
case webrtc::PeerConnectionInterface::kIceGatheringGathering:
return RTCIceGatheringStateGathering;
case webrtc::PeerConnectionInterface::kIceGatheringComplete:
return RTCIceGatheringStateComplete;
}
}
+ (NSString *)stringForIceGatheringState:(RTCIceGatheringState)state {
switch (state) {
case RTCIceGatheringStateNew:
return @"NEW";
case RTCIceGatheringStateGathering:
return @"GATHERING";
case RTCIceGatheringStateComplete:
return @"COMPLETE";
}
}
+ (webrtc::PeerConnectionInterface::StatsOutputLevel)
nativeStatsOutputLevelForLevel:(RTCStatsOutputLevel)level {
switch (level) {
case RTCStatsOutputLevelStandard:
return webrtc::PeerConnectionInterface::kStatsOutputLevelStandard;
case RTCStatsOutputLevelDebug:
return webrtc::PeerConnectionInterface::kStatsOutputLevelDebug;
}
}
- (rtc::scoped_refptr<webrtc::PeerConnectionInterface>)nativePeerConnection {
return _peerConnection;
}
@end