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?
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:128
和a=mid:audio
之间追加a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
。