webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCMediaSource.mm
Mirko Bonadei 92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
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}
2017-09-15 05:02:56 +00:00

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2.2 KiB
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/*
* 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 "RTCMediaSource+Private.h"
#include "rtc_base/checks.h"
@implementation RTCMediaSource {
RTCMediaSourceType _type;
}
@synthesize nativeMediaSource = _nativeMediaSource;
- (instancetype)initWithNativeMediaSource:
(rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
type:(RTCMediaSourceType)type {
RTC_DCHECK(nativeMediaSource);
if (self = [super init]) {
_nativeMediaSource = nativeMediaSource;
_type = type;
}
return self;
}
- (RTCSourceState)state {
return [[self class] sourceStateForNativeState:_nativeMediaSource->state()];
}
#pragma mark - Private
+ (webrtc::MediaSourceInterface::SourceState)nativeSourceStateForState:
(RTCSourceState)state {
switch (state) {
case RTCSourceStateInitializing:
return webrtc::MediaSourceInterface::kInitializing;
case RTCSourceStateLive:
return webrtc::MediaSourceInterface::kLive;
case RTCSourceStateEnded:
return webrtc::MediaSourceInterface::kEnded;
case RTCSourceStateMuted:
return webrtc::MediaSourceInterface::kMuted;
}
}
+ (RTCSourceState)sourceStateForNativeState:
(webrtc::MediaSourceInterface::SourceState)nativeState {
switch (nativeState) {
case webrtc::MediaSourceInterface::kInitializing:
return RTCSourceStateInitializing;
case webrtc::MediaSourceInterface::kLive:
return RTCSourceStateLive;
case webrtc::MediaSourceInterface::kEnded:
return RTCSourceStateEnded;
case webrtc::MediaSourceInterface::kMuted:
return RTCSourceStateMuted;
}
}
+ (NSString *)stringForState:(RTCSourceState)state {
switch (state) {
case RTCSourceStateInitializing:
return @"Initializing";
case RTCSourceStateLive:
return @"Live";
case RTCSourceStateEnded:
return @"Ended";
case RTCSourceStateMuted:
return @"Muted";
}
}
@end