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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 09:22:09 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 09:22:09 +0000
commit43a97878ce14b72f0981164f87f2e35e14151312 (patch)
tree620249daf56c0258faa40cbdcf9cfba06de2a846 /third_party/libwebrtc/sdk/objc/api/peerconnection/RTCAudioTrack.mm
parentInitial commit. (diff)
downloadfirefox-43a97878ce14b72f0981164f87f2e35e14151312.tar.xz
firefox-43a97878ce14b72f0981164f87f2e35e14151312.zip
Adding upstream version 110.0.1.upstream/110.0.1upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/sdk/objc/api/peerconnection/RTCAudioTrack.mm')
-rw-r--r--third_party/libwebrtc/sdk/objc/api/peerconnection/RTCAudioTrack.mm67
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