Question / Help How to know what to tune in in config

[C4nt3r]

Hi all,

Yesterday, I did my first streaming with OBS.
People told me that streaming was smooth except a strange problem with volume.
My surprise, was when I finished, and started to see the video at twitch.
Example: http://www.twitch.tv/c4nt3r/c/2019591

As you can see, at game (around minute 20), is not smooth with a video very lagged. Game and all was ok while streaming, and as I wrote, I confirmed with one viewer, streaming was smooth too (without taking account of volume issues, I don't know why this difference of volume yet), but if you see the recorded streaming, is far from reality.

It's my problem? It's twitch.tv problem?

How to know, what to tune to get a smooth video too ? I'm newbie in streaming world.

My config is:
OBS version, 473b t11 64bits
Windows 7 64b
Laptop:

Spoiler

CPU Name: Intel(R) Core(TM) i5-2410M CPU @ 2.30GHz
CPU Speed: 2294MHz
Physical Memory: 3947MB Total, 972MB Free
stepping id: 7, model 10, family 6, type 0, extmodel 1, extfamily 0, HTT 1, logical cores 4, total cores 2
monitor 1: pos={0, 0}, size={1920, 1080}
monitor 2: pos={-1366, 0}, size={1366, 768}
Windows Version: 6.1 Build 7601 S
Aero is Enabled
------------------------------------------
Adapter 1
Video Adapter: NVIDIA GeForce GT 540M
Video Adapter Dedicated Video Memory: 2112421888
Video Adapter Shared System Memory: 1801379840
------------------------------------------
Adapter 2
Video Adapter: NVIDIA GeForce GT 540M
Video Adapter Dedicated Video Memory: 2112421888
Video Adapter Shared System Memory: 1801379840

Internet connection: 50Mb down/~6Mb up
Connected with wifi to router
Encoding: Use CBR, Max Bitrate 2500
Broadcast: twitchtv with minifying network impact
Video: 1920x1080 without down scaling (to get 1080 videos for YouTube), 30fps, Aero enabled
Audio: desktop boost 5 (this could be my audio problem, boost should be in mic not desktop, true?)
Advanced: use multithreaded optimizations,use higher quality resampling

Had 2 scenes with various sources, but only owed one scene, with game capture only source as stretch image to screen. Resolution of the game, 1600x900

 

[C4nt3r]

log
=====Stream Start=====================================================================
Multithreaded optimizations: On
Base resolution: 1920x1080
Output resolution: 1920x1080
------------------------------------------
Loading up D3D10...
Playback device {0.0.0.00000000}.{2fbf0ca2-7546-4531-860f-dfc1cef6c27b}
------------------------------------------
Using desktop audio input: Altavoces (Realtek High Definition Audio)
------------------------------------------
Using auxilary audio input: Micrófono (Realtek High Definition Audio)
------------------------------------------
Audio Encoding: AAC
bitrate: 128
------------------------------------------
device: 1.3M HD WebCam,
device id \\?\usb#vid_04f2&pid_b23d&mi_00#7&34be53fb&0&0000#{65e8773d-8f56-11d0-a3b9-00a0c9223196}\global,
chosen type: YUY2, usingFourCC: false, res: 176x144 - 176x144, frameIntervals: 333333-666666, fourCC: 'YUY2'

Using directshow input
Using graphics capture
x264: using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX
x264: profile High, level 4.0
------------------------------------------
Video Encoding: x264
fps: 30
width: 1920, height: 1080
preset: veryfast
CBR: yes
CFR: no
max bitrate: 2500
------------------------------------------
SharedTexCapture hooked
Using RTMP service: Twitch / Justin.tv
Server selection: rtmp://live-fra.justin.tv/app
SO_SNDBUF was at 8192
SO_SNDBUF is now 65536
Audio timestamp for device 'Micrófono (Realtek High Definition Audio)' was behind target timestamp by 10! Had to delete audio segment.
RTMPPublisher::BufferedSend: Increasing socket send buffer to ISB 131072
Audio timestamp for device 'Micrófono (Realtek High Definition Audio)' was behind target timestamp by 20! Had to delete audio segment.
Audio timestamp for device 'Micrófono (Realtek High Definition Audio)' was behind target timestamp by 10! Had to delete audio segment.
Audio timestamp for device 'Micrófono (Realtek High Definition Audio)' was behind target timestamp by 10! Had to delete audio segment.
Total frames rendered: 34549, number of frames that lagged: 5285 (15.30%) (it's okay for some frames to lag)
Total duplicated frames: 31671 (91.67%)
RTMPPublisher::SocketLoop: Aborting due to bStopping
Number of times waited to send: 61, Waited for a total of 222131 bytes
Number of b-frames dropped: 69 (0.1%), Number of p-frames dropped: 65 (0.099%), Total 134 (0.2%)
x264: frame I:361 Avg QP:22.63 size:168460
x264: frame P:19727 Avg QP:19.57 size: 21401
x264: frame B:45799 Avg QP:22.24 size: 4439
x264: consecutive B-frames: 5.8% 3.1% 4.7% 86.5%
x264: mb I I16..4: 24.0% 28.4% 47.6%
x264: mb P I16..4: 2.2% 1.5% 0.6% P16..4: 19.3% 3.6% 2.0% 0.0% 0.0% skip:70.9%
x264: mb B I16..4: 0.1% 0.0% 0.0% B16..8: 1.8% 0.5% 0.1% direct: 1.0% skip:96.5% L0:37.8% L1:52.7% BI: 9.5%
x264: 8x8 transform intra:31.8% inter:27.6%
x264: coded y,uvDC,uvAC intra: 50.5% 47.6% 29.7% inter: 3.4% 3.3% 1.1%
x264: i16 v,h,dc,p: 31% 63% 4% 2%
x264: i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 16% 39% 16% 5% 5% 4% 5% 4% 6%
x264: i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 23% 30% 10% 6% 7% 6% 7% 6% 6%
x264: i8c dc,h,v,p: 45% 36% 15% 4%
x264: Weighted P-Frames: Y:0.9% UV:0.8%
x264: kb/s:2499.85
=====Stream End=======================================================================

 

[hilalpro]

with that cpu. you would have to downscale to 720p. maybe also lower the fps to 25.

pick a different server than frankfurt. try (uk or amesterdam).
about the other issue, unless the viewers are having a hard time hearing your game/music you wanna leave the desktop boost a 1.

 

[C4nt3r]

with that cpu. you would have to downscale to 720p. maybe also lower the fps to 25.

pick a different server than frankfurt. try (uk or amesterdam).
about the other issue, unless the viewers are having a hard time hearing your game/music you wanna leave the desktop boost a 1.

I will try your suggestions thanks.