When I stream Xbox One from my Elgato, it starts out perfect with the audio matched up to the gameplay. But after about an hour or so, the audio starts to lag behind and the gunshots wont match up with the gameplay. Is there a reason for this or a fix?
This is my log for a stream.
This is my log for a stream.
Code:
21:39:20: Open Broadcaster Software v0.592b - 32bit (´・ω・｀)
21:39:20: -------------------------------
21:39:20: CPU Name: Intel(R) Core(TM) i5 CPU 650 @ 3.20GHz
21:39:20: CPU Speed: 3192MHz
21:39:20: Physical Memory: 4095MB Total, 4095MB Free
21:39:20: stepping id: 2, model 37, family 6, type 0, extmodel 1, extfamily 0, HTT 1, logical cores 4, total cores 2
21:39:20: monitor 1: pos={0, 0}, size={1920, 1080}
21:39:20: Windows Version: 6.1 Build 7601 S
21:39:20: Aero is Disabled
21:39:20: -------------------------------
21:39:20: OBS Modules:
21:39:20: Base Address Module
21:39:20: 01380000 OBS.exe
21:39:20: 66090000 OBSApi.dll
21:39:20: 6BBE0000 DShowPlugin.dll
21:39:20: 6DFE0000 GraphicsCapture.dll
21:39:20: 6DC80000 NoiseGate.dll
21:39:20: 67D90000 PSVPlugin.dll
21:39:20: ------------------------------------------
21:39:20: Adapter 1
21:39:20: Video Adapter: NVIDIA GeForce GT 220
21:39:20: Video Adapter Dedicated Video Memory: 1056112640
21:39:20: Video Adapter Shared System Memory: 3238789120
21:39:20: Video Adapter Output 1: pos={0, 0}, size={1920, 1080}, attached=true
21:39:20: =====Stream Start: 2014-01-02, 21:39:20===============================================
21:39:20: Multithreaded optimizations: On
21:39:20: Base resolution: 1280x720
21:39:20: Output resolution: 1280x720
21:39:20: ------------------------------------------
21:39:20: Loading up D3D10 on NVIDIA GeForce GT 220 (Adapter 1)...
21:39:21: ------------------------------------------
21:39:21: Audio Format: 48000hz
21:39:21: Playback device {0.0.0.00000000}.{ce686f9d-f89d-498e-8087-e82be0c9e0a2}
21:39:21: ------------------------------------------
21:39:21: Using desktop audio input: Headset Earphone (ASTRO MixAmp Pro )
21:39:21: ------------------------------------------
21:39:21: Using auxilary audio input: Headset Microphone (ASTRO MixAmp Pro )
21:39:21: ------------------------------------------
21:39:21: Audio Encoding: AAC
21:39:21: bitrate: 128
21:39:21: ------------------------------------------
21:39:21: device: Elgato Game Capture HD,
21:39:21: device id {39F50F4C-99E1-464a-B6F9-D605B4FB5918},
21:39:21: chosen type: UYVY, usingFourCC: false, res: 1280x720 - 1280x720, frameIntervals: 333333-333333
21:39:21: use buffering: false - 0, fourCC: 'UYVY'
21:39:21:
21:39:21: device audio info - bits per sample: 16, channels: 2, samples per sec: 48000, block size: 4
21:39:21: Using directshow input
21:39:21: Scene buffering time set to 2000
21:39:21: ------------------------------------------
21:39:21: Video Encoding: x264
21:39:21: fps: 30
21:39:21: width: 1280, height: 720
21:39:21: preset: veryfast
21:39:21: profile: main
21:39:21: keyint: 60
21:39:21: CBR: yes
21:39:21: CFR: yes
21:39:21: max bitrate: 2500
21:39:21: buffer size: 3000
21:39:21: ------------------------------------------
21:39:21: MMDeviceAudioSource: Frequency for device 'Headset Earphone (ASTRO MixAmp Pro )' is 384000, samples per sec is 48000
21:39:21: MMDeviceAudioSource: Frequency for device 'Headset Microphone (ASTRO MixAmp Pro )' is 192000, samples per sec is 48000
21:39:21: Syncing audio to video time (WARNING: you should not be doing this if you are just having webcam desync, that's a separate issue)
21:40:01: Using RTMP service: Twitch / Justin.tv
21:40:01: Server selection: rtmp://live-ord.justin.tv/app
21:40:01: Interface: Broadcom NetLink (TM) Gigabit Ethernet (ethernet, 1000 mbps)
21:40:01: Completed handshake with rtmp://live-ord.justin.tv/app in 230 ms.
21:40:02: SO_SNDBUF was at 8192
21:40:02: SO_SNDBUF is now 65536
Warning -- MMDeviceAudioSource::GetBuffer: GetNextPacketSize failed, result = 88890004
21:53:48: Total frames encoded: 25947, total frames duplicated: 9365 (36.09%)
21:53:48: Number of frames skipped due to encoder lag: 68 (0.26%)
21:53:48: Total frames rendered: 17952, number of late frames: 4780 (26.63%) (it's okay for some frames to be late)
21:53:50: RTMPPublisher::SocketLoop: Aborting due to WSAEnumNetworkEvents failure, 10038
21:53:51: Average send payload: 12318 bytes, average send interval: 39 ms
21:53:51: Number of times waited to send: 0, Waited for a total of 0 bytes
21:53:51: Number of b-frames dropped: 0 (0%), Number of p-frames dropped: 0 (0%), Total 0 (0%)
21:53:51: Number of bytes sent: 260609562
