WebRTC音视频通话-iOS端调用ossrs直播拉流

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：

@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

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.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");        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

学习记录，每天不停进步。

来源地址：https://blog.csdn.net/gloryFlow/article/details/132417602