diff options
Diffstat (limited to 'third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpSender.mm')
-rw-r--r-- | third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpSender.mm | 132 |
1 files changed, 132 insertions, 0 deletions
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpSender.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpSender.mm new file mode 100644 index 0000000000..4fadb30f49 --- /dev/null +++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpSender.mm @@ -0,0 +1,132 @@ +/* + * Copyright 2016 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 "RTCRtpSender+Private.h" + +#import "RTCDtmfSender+Private.h" +#import "RTCMediaStreamTrack+Private.h" +#import "RTCRtpParameters+Private.h" +#import "RTCRtpSender+Native.h" +#import "base/RTCLogging.h" +#import "helpers/NSString+StdString.h" + +#include "api/media_stream_interface.h" + +@implementation RTC_OBJC_TYPE (RTCRtpSender) { + RTC_OBJC_TYPE(RTCPeerConnectionFactory) * _factory; + rtc::scoped_refptr<webrtc::RtpSenderInterface> _nativeRtpSender; +} + +@synthesize dtmfSender = _dtmfSender; + +- (NSString *)senderId { + return [NSString stringForStdString:_nativeRtpSender->id()]; +} + +- (RTC_OBJC_TYPE(RTCRtpParameters) *)parameters { + return [[RTC_OBJC_TYPE(RTCRtpParameters) alloc] + initWithNativeParameters:_nativeRtpSender->GetParameters()]; +} + +- (void)setParameters:(RTC_OBJC_TYPE(RTCRtpParameters) *)parameters { + if (!_nativeRtpSender->SetParameters(parameters.nativeParameters).ok()) { + RTCLogError(@"RTC_OBJC_TYPE(RTCRtpSender)(%p): Failed to set parameters: %@", self, parameters); + } +} + +- (RTC_OBJC_TYPE(RTCMediaStreamTrack) *)track { + rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> nativeTrack( + _nativeRtpSender->track()); + if (nativeTrack) { + return [RTC_OBJC_TYPE(RTCMediaStreamTrack) mediaTrackForNativeTrack:nativeTrack + factory:_factory]; + } + return nil; +} + +- (void)setTrack:(RTC_OBJC_TYPE(RTCMediaStreamTrack) *)track { + if (!_nativeRtpSender->SetTrack(track.nativeTrack.get())) { + RTCLogError(@"RTC_OBJC_TYPE(RTCRtpSender)(%p): Failed to set track %@", self, track); + } +} + +- (NSArray<NSString *> *)streamIds { + std::vector<std::string> nativeStreamIds = _nativeRtpSender->stream_ids(); + NSMutableArray *streamIds = [NSMutableArray arrayWithCapacity:nativeStreamIds.size()]; + for (const auto &s : nativeStreamIds) { + [streamIds addObject:[NSString stringForStdString:s]]; + } + return streamIds; +} + +- (void)setStreamIds:(NSArray<NSString *> *)streamIds { + std::vector<std::string> nativeStreamIds; + for (NSString *streamId in streamIds) { + nativeStreamIds.push_back([streamId UTF8String]); + } + _nativeRtpSender->SetStreams(nativeStreamIds); +} + +- (NSString *)description { + return [NSString + stringWithFormat:@"RTC_OBJC_TYPE(RTCRtpSender) {\n senderId: %@\n}", self.senderId]; +} + +- (BOOL)isEqual:(id)object { + if (self == object) { + return YES; + } + if (object == nil) { + return NO; + } + if (![object isMemberOfClass:[self class]]) { + return NO; + } + RTC_OBJC_TYPE(RTCRtpSender) *sender = (RTC_OBJC_TYPE(RTCRtpSender) *)object; + return _nativeRtpSender == sender.nativeRtpSender; +} + +- (NSUInteger)hash { + return (NSUInteger)_nativeRtpSender.get(); +} + +#pragma mark - Native + +- (void)setFrameEncryptor:(rtc::scoped_refptr<webrtc::FrameEncryptorInterface>)frameEncryptor { + _nativeRtpSender->SetFrameEncryptor(frameEncryptor); +} + +#pragma mark - Private + +- (rtc::scoped_refptr<webrtc::RtpSenderInterface>)nativeRtpSender { + return _nativeRtpSender; +} + +- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory + nativeRtpSender:(rtc::scoped_refptr<webrtc::RtpSenderInterface>)nativeRtpSender { + NSParameterAssert(factory); + NSParameterAssert(nativeRtpSender); + if (self = [super init]) { + _factory = factory; + _nativeRtpSender = nativeRtpSender; + if (_nativeRtpSender->media_type() == cricket::MEDIA_TYPE_AUDIO) { + rtc::scoped_refptr<webrtc::DtmfSenderInterface> nativeDtmfSender( + _nativeRtpSender->GetDtmfSender()); + if (nativeDtmfSender) { + _dtmfSender = + [[RTC_OBJC_TYPE(RTCDtmfSender) alloc] initWithNativeDtmfSender:nativeDtmfSender]; + } + } + RTCLogInfo(@"RTC_OBJC_TYPE(RTCRtpSender)(%p): created sender: %@", self, self.description); + } + return self; +} + +@end |