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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/sdk/objc/api/peerconnection/RTCAudioTrack.mm | |
parent | Initial commit. (diff) | |
download | thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.tar.xz thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to '')
-rw-r--r-- | third_party/libwebrtc/sdk/objc/api/peerconnection/RTCAudioTrack.mm | 67 |
1 files changed, 67 insertions, 0 deletions
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCAudioTrack.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCAudioTrack.mm new file mode 100644 index 0000000000..5c1736f436 --- /dev/null +++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCAudioTrack.mm @@ -0,0 +1,67 @@ +/* + * 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 "RTCAudioTrack+Private.h" + +#import "RTCAudioSource+Private.h" +#import "RTCMediaStreamTrack+Private.h" +#import "RTCPeerConnectionFactory+Private.h" +#import "helpers/NSString+StdString.h" + +#include "rtc_base/checks.h" + +@implementation RTC_OBJC_TYPE (RTCAudioTrack) + +@synthesize source = _source; + +- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory + source:(RTC_OBJC_TYPE(RTCAudioSource) *)source + trackId:(NSString *)trackId { + RTC_DCHECK(factory); + RTC_DCHECK(source); + RTC_DCHECK(trackId.length); + + std::string nativeId = [NSString stdStringForString:trackId]; + rtc::scoped_refptr<webrtc::AudioTrackInterface> track = + factory.nativeFactory->CreateAudioTrack(nativeId, source.nativeAudioSource.get()); + if (self = [self initWithFactory:factory nativeTrack:track type:RTCMediaStreamTrackTypeAudio]) { + _source = source; + } + return self; +} + +- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory + nativeTrack:(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack + type:(RTCMediaStreamTrackType)type { + NSParameterAssert(factory); + NSParameterAssert(nativeTrack); + NSParameterAssert(type == RTCMediaStreamTrackTypeAudio); + return [super initWithFactory:factory nativeTrack:nativeTrack type:type]; +} + +- (RTC_OBJC_TYPE(RTCAudioSource) *)source { + if (!_source) { + rtc::scoped_refptr<webrtc::AudioSourceInterface> source(self.nativeAudioTrack->GetSource()); + if (source) { + _source = [[RTC_OBJC_TYPE(RTCAudioSource) alloc] initWithFactory:self.factory + nativeAudioSource:source]; + } + } + return _source; +} + +#pragma mark - Private + +- (rtc::scoped_refptr<webrtc::AudioTrackInterface>)nativeAudioTrack { + return rtc::scoped_refptr<webrtc::AudioTrackInterface>( + static_cast<webrtc::AudioTrackInterface *>(self.nativeTrack.get())); +} + +@end |