WebRTC音视频通话-iOS端调用ossrs直播拉流
之前实现iOS端调用ossrs服务，文中提到了推流。没有写拉流流程，所以会用到文中的WebRTCClient。请详细查看：https://blog.csdn.net/gloryFlow/article/details/132262724
最近有朋友问过，我发现之前少了一块拉流流程，这里补充一下。
拉流实现时候设置WebRTCClient时候初始化，这里isPublish为false哦
#pragma mark - Lazy
- (WebRTCClient *)webRTCClient {
if (!_webRTCClient) {
_webRTCClient = [[WebRTCClient alloc] initWithPublish:NO];
}
return _webRTCClient;
}
之前的文中摄像头画面显示使用的是startCaptureLocalVideo，但是拉流需要设置remoteRenderView
WebRTCClient中有定义：
/**
RTCVideoRenderer
*/
@property (nonatomic, weak) id<RTCVideoRenderer> remoteRenderView;
设置拉流显示的画面View
#import "RTCPlayView.h"
@interface RTCPlayView ()
@property (nonatomic, strong) WebRTCClient *webRTCClient;
@property (nonatomic, strong) RTCEAGLVideoView *remoteRenderer;
@end
@implementation RTCPlayView
- (instancetype)initWithFrame:(CGRect)frame webRTCClient:(WebRTCClient *)webRTCClient {
self = [super initWithFrame:frame];
if (self) {
self.webRTCClient = webRTCClient;
self.remoteRenderer = [[RTCEAGLVideoView alloc] initWithFrame:CGRectZero];
self.remoteRenderer.contentMode = UIViewContentModeScaleAspectFit;
[self addSubview:self.remoteRenderer];
self.webRTCClient.remoteRenderView = self.remoteRenderer;
}
return self;
}
- (void)layoutSubviews {
[super layoutSubviews];
self.remoteRenderer.frame = self.bounds;
NSLog(@"self.remoteRenderer frame:%@", NSStringFromCGRect(self.remoteRenderer.frame));
}
@end
这里使用的创建RTCEAGLVideoView，设置self.webRTCClient.remoteRenderView为self.remoteRenderer
实现拉流调用流程和推流类似，这里不再说明，请查看 https://blog.csdn.net/gloryFlow/article/details/132262724
具体方法如下
- (void)playBtnClick {
__weak typeof(self) weakSelf = self;
[self.webRTCClient offer:^(RTCSessionDescription *sdp) {
[weakSelf.webRTCClient changeSDP2Server:sdp urlStr:@"https://192.168.10.102:1990/rtc/v1/play/" streamUrl:@"webrtc://192.168.10.102:1990/live/livestream" closure:^(BOOL isServerRetSuc) {
NSLog(@"isServerRetSuc:%@",(isServerRetSuc?@"YES":@"NO"));
}];
}];
}
完整的Controller代码如下
#import "RTCPlayViewController.h"
@interface RTCPlayViewController ()<WebRTCClientDelegate>
@property (nonatomic, strong) WebRTCClient *webRTCClient;
@property (nonatomic, strong) RTCPlayView *rtcPlayView;
@property (nonatomic, strong) UIButton *playBtn;
@end
@implementation RTCPlayViewController
- (void)viewDidLoad {
[super viewDidLoad];
// Do any additional setup after loading the view.
self.view.backgroundColor = [UIColor whiteColor];
self.rtcPlayView = [[RTCPlayView alloc] initWithFrame:CGRectZero webRTCClient:self.webRTCClient];
[self.view addSubview: self.rtcPlayView];
self.rtcPlayView.backgroundColor = [UIColor lightGrayColor];
self.rtcPlayView.frame = self.view.bounds;
CGFloat screenWidth = CGRectGetWidth(self.view.bounds);
CGFloat screenHeight = CGRectGetHeight(self.view.bounds);
self.playBtn = [UIButton buttonWithType:UIButtonTypeCustom];
self.playBtn.frame = CGRectMake(50, screenHeight - 160, screenWidth - 2*50, 46);
self.playBtn.layer.cornerRadius = 4;
self.playBtn.backgroundColor = [UIColor grayColor];
[self.playBtn setTitle:@"publish" forState:UIControlStateNormal];
[self.playBtn addTarget:self action:@selector(playBtnClick) forControlEvents:UIControlEventTouchUpInside];
[self.view addSubview:self.playBtn];
self.webRTCClient.delegate = self;
}
- (void)playBtnClick {
__weak typeof(self) weakSelf = self;
[self.webRTCClient offer:^(RTCSessionDescription *sdp) {
[weakSelf.webRTCClient changeSDP2Server:sdp urlStr:@"https://192.168.10.102:1990/rtc/v1/play/" streamUrl:@"webrtc://192.168.10.102:1990/live/livestream" closure:^(BOOL isServerRetSuc) {
NSLog(@"isServerRetSuc:%@",(isServerRetSuc?@"YES":@"NO"));
}];
}];
}
#pragma mark - WebRTCClientDelegate
- (void)webRTCClient:(WebRTCClient *)client didDiscoverLocalCandidate:(RTCIceCandidate *)candidate {
NSLog(@"webRTCClient didDiscoverLocalCandidate");
}
- (void)webRTCClient:(WebRTCClient *)client didChangeConnectionState:(RTCIceConnectionState)state {
NSLog(@"webRTCClient didChangeConnectionState");
/**
RTCIceConnectionStateNew,
RTCIceConnectionStateChecking,
RTCIceConnectionStateConnected,
RTCIceConnectionStateCompleted,
RTCIceConnectionStateFailed,
RTCIceConnectionStateDisconnected,
RTCIceConnectionStateClosed,
RTCIceConnectionStateCount,
*/
UIColor *textColor = [UIColor blackColor];
BOOL openSpeak = NO;
switch (state) {
case RTCIceConnectionStateCompleted:
case RTCIceConnectionStateConnected:
textColor = [UIColor greenColor];
openSpeak = YES;
break;
case RTCIceConnectionStateDisconnected:
textColor = [UIColor orangeColor];
break;
case RTCIceConnectionStateFailed:
case RTCIceConnectionStateClosed:
textColor = [UIColor redColor];
break;
case RTCIceConnectionStateNew:
case RTCIceConnectionStateChecking:
case RTCIceConnectionStateCount:
textColor = [UIColor blackColor];
break;
default:
break;
}
dispatch_async(dispatch_get_main_queue(), ^{
NSString *text = [NSString stringWithFormat:@"%ld", state];
[self.playBtn setTitle:text forState:UIControlStateNormal];
[self.playBtn setTitleColor:textColor forState:UIControlStateNormal];
if (openSpeak) {
[self.webRTCClient speakOn];
}
// if textColor == .green {
// self?.webRTCClient.speakerOn()
// }
});
}
- (void)webRTCClient:(WebRTCClient *)client didReceiveData:(NSData *)data {
NSLog(@"webRTCClient didReceiveData");
}
#pragma mark - Lazy
- (WebRTCClient *)webRTCClient {
if (!_webRTCClient) {
_webRTCClient = [[WebRTCClient alloc] initWithPublish:NO];
}
return _webRTCClient;
}
@end
至此，可以实现iOS端调用的ossrs视频通话拉流
其他
之前搭建ossrs服务，可以查看：https://blog.csdn.net/gloryFlow/article/details/132257196
之前实现iOS端调用ossrs音视频通话，可以查看：https://blog.csdn.net/gloryFlow/article/details/132262724
之前WebRTC音视频通话高分辨率不显示画面问题，可以查看：https://blog.csdn.net/gloryFlow/article/details/132240952
修改SDP中的码率Bitrate，可以查看：https://blog.csdn.net/gloryFlow/article/details/132263021
GPUImage视频通话视频美颜滤镜，可以查看：https://blog.csdn.net/gloryFlow/article/details/132265842
RTC直播本地视频或相册视频，可以查看：https://blog.csdn.net/gloryFlow/article/details/132267068
WebRTC音视频通话-iOS端调用ossrs直播拉流。用到了WebRTC调用ossrs实现推拉流效果。内容较多，描述可能不准确，请见谅。
https://blog.csdn.net/gloryFlow/article/details/132417602
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