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之前实现iOS端调用ossrs服务#xff0c;文中提到了推流。没有写拉流流程#xff0c;所以会用到文中的WebRTCClient。请详细查看#xff1a;https://blog.csdn.net/gloryFlow/article/details/132262724
一、iOS播放端拉流效果 二…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
一、iOS播放端拉流效果 二、实现iOS端调用ossrs拉流
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。
2.1、拉流实现时候设置WebRTCClient
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;
}2.2、设置拉流显示的画面View。
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) idRTCVideoRenderer remoteRenderView;设置拉流显示的画面View
#import RTCPlayView.hinterface RTCPlayView ()property (nonatomic, strong) WebRTCClient *webRTCClient;
property (nonatomic, strong) RTCEAGLVideoView *remoteRenderer;endimplementation 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
2.3、调用ossrs服务play接口为rtc/v1/play/
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.hinterface RTCPlayViewController ()WebRTCClientDelegateproperty (nonatomic, strong) WebRTCClient *webRTCClient;property (nonatomic, strong) RTCPlayView *rtcPlayView;property (nonatomic, strong) UIButton *playBtn;endimplementation 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
三、小结
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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