mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-14 14:20:45 +01:00
7bca8ca4e2
This CL separates the files under sdk/objc into logical directories, replacing the previous file layout under Framework/. A long term goal is to have some system set up to generate the files under sdk/objc/api (the PeerConnection API wrappers) from the C++ code. In the shorter term the goal is to abstract out shared concepts from these classes in order to make them as uniform as possible. The separation into base/, components/, and helpers/ are to differentiate between the base layer's common protocols, various utilities and the actual platform specific components. The old directory layout that resembled a framework's internal layout is not necessary, since it is generated by the framework target when building it. Bug: webrtc:9627 Change-Id: Ib084fd83f050ae980649ca99e841f4fb0580bd8f Reviewed-on: https://webrtc-review.googlesource.com/94142 Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Anders Carlsson <andersc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24493}
68 lines
2.2 KiB
Text
68 lines
2.2 KiB
Text
/*
|
|
* 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 RTCAudioTrack
|
|
|
|
@synthesize source = _source;
|
|
|
|
- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
|
|
source:(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);
|
|
if (self = [self initWithFactory:factory nativeTrack:track type:RTCMediaStreamTrackTypeAudio]) {
|
|
_source = source;
|
|
}
|
|
return self;
|
|
}
|
|
|
|
- (instancetype)initWithFactory:(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];
|
|
}
|
|
|
|
|
|
- (RTCAudioSource *)source {
|
|
if (!_source) {
|
|
rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
|
|
self.nativeAudioTrack->GetSource();
|
|
if (source) {
|
|
_source =
|
|
[[RTCAudioSource alloc] initWithFactory:self.factory nativeAudioSource:source.get()];
|
|
}
|
|
}
|
|
return _source;
|
|
}
|
|
|
|
#pragma mark - Private
|
|
|
|
- (rtc::scoped_refptr<webrtc::AudioTrackInterface>)nativeAudioTrack {
|
|
return static_cast<webrtc::AudioTrackInterface *>(self.nativeTrack.get());
|
|
}
|
|
|
|
@end
|