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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
66 lines
2 KiB
Text
66 lines
2 KiB
Text
/*
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* Copyright 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#import "RTCAudioTrack+Private.h"
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#import "NSString+StdString.h"
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#import "RTCAudioSource+Private.h"
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#import "RTCMediaStreamTrack+Private.h"
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#import "RTCPeerConnectionFactory+Private.h"
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#include "rtc_base/checks.h"
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@implementation RTCAudioTrack
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@synthesize source = _source;
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- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
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source:(RTCAudioSource *)source
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trackId:(NSString *)trackId {
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RTC_DCHECK(factory);
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RTC_DCHECK(source);
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RTC_DCHECK(trackId.length);
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std::string nativeId = [NSString stdStringForString:trackId];
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rtc::scoped_refptr<webrtc::AudioTrackInterface> track =
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factory.nativeFactory->CreateAudioTrack(nativeId, source.nativeAudioSource);
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if ([self initWithNativeTrack:track type:RTCMediaStreamTrackTypeAudio]) {
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_source = source;
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}
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return self;
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}
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- (instancetype)initWithNativeTrack:
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(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack
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type:(RTCMediaStreamTrackType)type {
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NSParameterAssert(nativeTrack);
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NSParameterAssert(type == RTCMediaStreamTrackTypeAudio);
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return [super initWithNativeTrack:nativeTrack type:type];
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}
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- (RTCAudioSource *)source {
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if (!_source) {
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rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
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self.nativeAudioTrack->GetSource();
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if (source) {
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_source = [[RTCAudioSource alloc] initWithNativeAudioSource:source.get()];
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}
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}
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return _source;
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}
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#pragma mark - Private
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- (rtc::scoped_refptr<webrtc::AudioTrackInterface>)nativeAudioTrack {
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return static_cast<webrtc::AudioTrackInterface *>(self.nativeTrack.get());
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}
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@end
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