我怎么能压缩webrtc ios中的视频文件?

时间:2016-11-25 13:17:40

标签: ios iphone video-streaming webrtc

我正在使用Webrtc视频流。

- (RTCMediaConstraints *)defaultMediaStreamConstraints {
NSArray *mandatoryConstraints = @[
                                  [[RTCPair alloc] initWithKey:@"maxWidth" value:@"640"],
                                  [[RTCPair alloc] initWithKey:@"maxHeight" value:@"480"],
                                  [[RTCPair alloc] initWithKey:@"maxFrameRate" value:@"15"],
                                   ];

NSArray *optionalConstraints = @[];
RTCMediaConstraints* constraints1 =
[[RTCMediaConstraints alloc]
 initWithMandatoryConstraints:mandatoryConstraints
 optionalConstraints:nil];
return constraints1;
}
- (RTCMediaConstraints *)defaultPeerConnectionConstraints {
     NSArray *mandatoryConstraints = @[ [[RTCPair alloc] initWithKey:@"maxHeight" value:[NSString stringWithFormat:@"%@",@"640"]],
                                   [[RTCPair alloc] initWithKey:@"maxWidth" value:[NSString stringWithFormat:@"%@",@"480"]],
                                   [[RTCPair alloc] initWithKey:@"maxFrameRate" value:[NSString stringWithFormat:@"%@",@"15"]]
                                   ];
     NSArray *optionalConstraints = @[];

     RTCMediaConstraints* constraints1 =
                  [[RTCMediaConstraints alloc]
                  initWithMandatoryConstraints:mandatoryConstraints
                  optionalConstraints:optionalConstraints];
return constraints1;
}


但是,它没有工作。它设置了高质量的视频流 如何压缩质量?

1 个答案:

答案 0 :(得分:0)

您应该在将SDP设置为本地会话描述并将其发送给对等方之前,直接在SDP内设置视频带宽。

查找a=mid:video\r\n并将b=AS:128\r\n附加到其中。

您可以通过查找a=mid:audio\r\n来为音频流执行相同操作。

例如,给出以下SDP:

v=0 
o=- 487255629242026503 2 IN IP4 127.0.0.1 
s=- 
t=0 0 

a=rtcp:9 IN IP4 0.0.0.0 
a=ice-ufrag:8a1/LJqQMzBmYtes 
a=ice-pwd:sbfskHYHACygyHW1wVi8GZM+ 
a=ice-options:google-ice 
a=fingerprint:sha-256 28:4C:19:10:97:56:FB:22:57:9E:5A:88:28:F3:04:
   DF:37:D0:7D:55:C3:D1:59:B0:B2:81 :FB:9D:DF:CB:15:A8 
a=setup:actpass 
a=mid:audio 
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level 
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time 

您必须在b=AS:128a=mid:audio之间追加a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level