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  • How to keep source frame rate with mencoder/ffmpeg?

    - by Sandra
    I would like to crop and rotate a video, and then encode it to mp4 or mkv. mencoder video.mp4 -vf rotate=1,crop=720:1280:0:0 -oac pcm -ovc x264 -x264encopts preset=veryslow:tune=film:crf=15:frameref=15:fast_pskip=0:threads=auto -lavfopts format=matroska -o test.mkv But when I do the above encoding, the frame rate is way too fast. The encoding options were something I found, so I don't know if that is the problem. Question All I want is to crop and rotate the video, and keep the audio/video quality as good as possible. Have anyone tried this?

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  • guvcview recording video and audio out of synchronisation in Ubuntu 10.10

    - by SIJAR
    I finally got Guvcview, a great software for Logitech webcam and it does all the stuff that one wants out of it. But I'm not satisfy with the video recording, video and audio out of synchronisation also video seems to be in slow motion. Please help so that I can tweak in and get a good video recording with the webcam. Below is the log of Guvcview ------------------------------------------------------------------------------- guvcview 1.4.1 video_device: /dev/video0 vid_sleep: 0 cap_meth: 1 resolution: 640 x 480 windowsize: 1024 x 715 vert pane: 578 spin behavior: 0 mode: mjpg fps: 1/25 Display Fps: 0 bpp: 0 hwaccel: 1 avi_format: 4 sound: 1 sound Device: 4 sound samp rate: 0 sound Channels: 0 Sound delay: 0 nanosec Sound Format: 85 Pan Step: 2 degrees Tilt Step: 2 degrees Video Filter Flags: 0 image inc: 0 profile(default):/home/sijar/default.gpfl starting portaudio... bt_audio_service_open: connect() failed: Connection refused (111) bt_audio_service_open: connect() failed: Connection refused (111) bt_audio_service_open: connect() failed: Connection refused (111) bt_audio_service_open: connect() failed: Connection refused (111) Cannot connect to server socket err = No such file or directory Cannot connect to server socket jack server is not running or cannot be started language catalog= dir:/usr/share/locale type:UTF-8 lang:en_US.utf8 cat:guvcview.mo mjpg: setting format to 1196444237 capture method = 1 video device: /dev/video0 libv4lconvert: warning more framesizes then I can handle! libv4lconvert: warning more framesizes then I can handle! /dev/video0 - device 1 libv4lconvert: warning more framesizes then I can handle! libv4lconvert: warning more framesizes then I can handle! Init. UVC Camera (046d:0825) (location: usb-0000:00:1d.7-5) { pixelformat = 'YUYV', description = 'YUV 4:2:2 (YUYV)' } { discrete: width = 640, height = 480 } Time interval between frame: 1/30, 1/25, 1/20, 1/15, 1/10, 1/5, { discrete: width = 160, height = 120 } Time interval between frame: 1/30, 1/25, 1/20, 1/15, 1/10, 1/5, { discrete: width = 176, height = 144 } Time interval between frame: 1/30, 1/25, 1/20, 1/15, 1/10, 1/5, { discrete: width = 320, height = 176 } Time interval between frame: 1/30, 1/25, 1/20, 1/15, 1/10, 1/5, { discrete: width = 320, height = 240 } Time interval between frame: 1/30, 1/25, 1/20, 1/15, 1/10, 1/5, { discrete: width = 352, height = 288 } Time interval between frame: 1/30, 1/25, 1/20, 1/15, 1/10, 1/5, { discrete: width = 432, height = 240 } Time interval between frame: 1/30, 1/25, 1/20, 1/15, 1/10, 1/5, { discrete: width = 544, height = 288 } Time interval between frame: 1/30, 1/25, 1/20, 1/15, 1/10, 1/5, { discrete: width = 640, height = 360 } Time interval between frame: 1/30, 1/25, 1/20, 1/15, 1/10, 1/5, ... repeats a couple of times ... vid:046d pid:0825 driver:uvcvideo Adding control for Pan (relative) UVCIOC_CTRL_ADD - Error: Operation not permitted checking format: 1196444237 VIDIOC_G_COMP:: Invalid argument compression control not supported fps is set to 1/25 drawing controls control[0]: 0x980900 Brightness, 0:255:1, default 128 control[0]: 0x980901 Contrast, 0:255:1, default 32 control[0]: 0x980902 Saturation, 0:255:1, default 32 control[0]: 0x98090c White Balance Temperature, Auto, 0:1:1, default 1 control[0]: 0x980913 Gain, 0:255:1, default 0 control[0]: 0x980918 Power Line Frequency, 0:2:1, default 2 control[0]: 0x98091a White Balance Temperature, 0:10000:10, default 4000 control[0]: 0x98091b Sharpness, 0:255:1, default 24 control[0]: 0x98091c Backlight Compensation, 0:1:1, default 1 control[0]: 0x9a0901 Exposure, Auto, 0:3:1, default 3 control[0]: 0x9a0902 Exposure (Absolute), 1:10000:1, default 166 control[0]: 0x9a0903 Exposure, Auto Priority, 0:1:1, default 0 resolutions of format(2) = 19 frame rates of 1º resolution=6 Def. Res: 0 numb. fps:6 --------------------------------------- device #0 Name = Intel 82801DB-ICH4: Intel 82801DB-ICH4 (hw:0,0) Host API = ALSA Max inputs = 2, Max outputs = 2 Def. low input latency = 0.012 Def. low output latency = 0.012 Def. high input latency = 0.046 Def. high output latency = 0.046 Def. sample rate = 44100.00 --------------------------------------- device #1 Name = Intel 82801DB-ICH4: Intel 82801DB-ICH4 - MIC ADC (hw:0,1) Host API = ALSA Max inputs = 2, Max outputs = 0 Def. low input latency = 0.011 Def. low output latency = -1.000 Def. high input latency = 0.043 Def. high output latency = -1.000 Def. sample rate = 48000.00 --------------------------------------- device #2 Name = Intel 82801DB-ICH4: Intel 82801DB-ICH4 - MIC2 ADC (hw:0,2) Host API = ALSA Max inputs = 2, Max outputs = 0 Def. low input latency = 0.011 Def. low output latency = -1.000 Def. high input latency = 0.043 Def. high output latency = -1.000 Def. sample rate = 48000.00 --------------------------------------- device #3 Name = Intel 82801DB-ICH4: Intel 82801DB-ICH4 - ADC2 (hw:0,3) Host API = ALSA Max inputs = 2, Max outputs = 0 Def. low input latency = 0.011 Def. low output latency = -1.000 Def. high input latency = 0.043 Def. high output latency = -1.000 Def. sample rate = 48000.00 --------------------------------------- device #4 Name = Intel 82801DB-ICH4: Intel 82801DB-ICH4 - IEC958 (hw:0,4) Host API = ALSA Max inputs = 0, Max outputs = 2 Def. low input latency = -1.000 Def. low output latency = 0.011 Def. high input latency = -1.000 Def. high output latency = 0.043 Def. sample rate = 48000.00 --------------------------------------- device #5 Name = USB Device 0x46d:0x825: USB Audio (hw:1,0) Host API = ALSA Max inputs = 1, Max outputs = 0 Def. low input latency = 0.011 Def. low output latency = -1.000 Def. high input latency = 0.043 Def. high output latency = -1.000 Def. sample rate = 48000.00 --------------------------------------- device #6 Name = front Host API = ALSA Max inputs = 0, Max outputs = 2 Def. low input latency = -1.000 Def. low output latency = 0.012 Def. high input latency = -1.000 Def. high output latency = 0.046 Def. sample rate = 44100.00 --------------------------------------- device #7 Name = iec958 Host API = ALSA Max inputs = 0, Max outputs = 2 Def. low input latency = -1.000 Def. low output latency = 0.011 Def. high input latency = -1.000 Def. high output latency = 0.043 Def. sample rate = 48000.00 --------------------------------------- device #8 Name = spdif Host API = ALSA Max inputs = 0, Max outputs = 2 Def. low input latency = -1.000 Def. low output latency = 0.011 Def. high input latency = -1.000 Def. high output latency = 0.043 Def. sample rate = 48000.00 --------------------------------------- device #9 Name = pulse Host API = ALSA Max inputs = 32, Max outputs = 32 Def. low input latency = 0.012 Def. low output latency = 0.012 Def. high input latency = 0.046 Def. high output latency = 0.046 Def. sample rate = 44100.00 --------------------------------------- device #10 Name = dmix Host API = ALSA Max inputs = 0, Max outputs = 2 Def. low input latency = -1.000 Def. low output latency = 0.043 Def. high input latency = -1.000 Def. high output latency = 0.043 Def. sample rate = 48000.00 --------------------------------------- device #11 [ Default Input, Default Output ] Name = default Host API = ALSA Max inputs = 32, Max outputs = 32 Def. low input latency = 0.012 Def. low output latency = 0.012 Def. high input latency = 0.046 Def. high output latency = 0.046 Def. sample rate = 44100.00 ---------------------------------------------- SampleRate:0 Channels:0 Video driver: x11 A window manager is available VIDIOC_S_EXT_CTRLS for multiple controls failed (error -1) using VIDIOC_S_CTRL for user class controls control(0x0098091a) "White Balance Temperature" failed to set (error -1) VIDIOC_S_EXT_CTRLS for multiple controls failed (error -1) using VIDIOC_S_EXT_CTRLS on single controls for class: 0x009a0000 control(0x009a0902) "Exposure (Absolute)" failed to set (error -1) VIDIOC_S_EXT_CTRLS for multiple controls failed (error -1) using VIDIOC_S_CTRL for user class controls control(0x0098091a) "White Balance Temperature" failed to set (error -1) VIDIOC_S_EXT_CTRLS for multiple controls failed (error -1) using VIDIOC_S_EXT_CTRLS on single controls for class: 0x009a0000 control(0x009a0902) "Exposure (Absolute)" failed to set (error -1) Cap Video toggled: 1 (/home/sijar/Videos/Webcam) 25371756K bytes free on a total of 39908968K (used: 36 %) treshold=51200K using audio codec: 0x0055 Audio frame size is 1152 samples for selected codec IO thread started...OK [libx264 @ 0x8cbd8b0]using cpu capabilities: MMX2 SSE2 Cache64 [libx264 @ 0x8cbd8b0]profile Baseline, level 3.0 [libx264 @ 0x8cbd8b0]non-strictly-monotonic PTS shift sound by -9 ms shift sound by -9 ms shift sound by -9 ms AUDIO: droping audio data AUDIO: droping audio data AUDIO: droping audio data AUDIO: droping audio data AUDIO: droping audio data ... repeats a couple of times ... AUDIO: droping audio data (/home/sijar/Videos/Webcam) 25371748K bytes free on a total of 39908968K (used: 36 %) treshold=51200K AUDIO: droping audio data AUDIO: droping audio data ... repeats a couple of times ... Cap Video toggled: 0 Shuting Down IO Thread AUDIO: droping audio data stop= 4426644744000 start=4416533023000 VIDEO: 146 frames in 10111.000000 ms = 14.439719 fps Stoping audio stream Closing audio stream... close avi Last message repeated 145 times [libx264 @ 0x8cbd8b0]frame I:2 Avg QP:14.10 size: 24492 [libx264 @ 0x8cbd8b0]frame P:103 Avg QP:16.06 size: 20715 [libx264 @ 0x8cbd8b0]mb I I16..4: 48.4% 0.0% 51.6% [libx264 @ 0x8cbd8b0]mb P I16..4: 57.5% 0.0% 0.0% P16..4: 40.2% 0.0% 0.0% 0.0% 0.0% skip: 2.3% [libx264 @ 0x8cbd8b0]final ratefactor: 62.05 [libx264 @ 0x8cbd8b0]coded y,uvDC,uvAC intra: 79.7% 92.2% 68.4% inter: 62.4% 87.5% 48.0% [libx264 @ 0x8cbd8b0]i16 v,h,dc,p: 23% 17% 41% 19% [libx264 @ 0x8cbd8b0]i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 30% 24% 26% 2% 5% 3% 3% 3% 4% [libx264 @ 0x8cbd8b0]i8c dc,h,v,p: 53% 20% 23% 4% [libx264 @ 0x8cbd8b0]ref P L0: 63.0% 37.0% [libx264 @ 0x8cbd8b0]kb/s:-0.00 total frames encoded: 0 total audio frames encoded: 0 IO thread finished...OK IO Thread finished enabling controls Cap Video toggled: 1 (/home/sijar/Videos/Webcam) 25379744K bytes free on a total of 39908968K (used: 36 %) treshold=51200K using audio codec: 0x0055 Audio frame size is 1152 samples for selected codec IO thread started...OK [libx264 @ 0x8cfba20]using cpu capabilities: MMX2 SSE2 Cache64 [libx264 @ 0x8cfba20]profile Baseline, level 3.0 [libx264 @ 0x8cfba20]non-strictly-monotonic PTS shift sound by -236 ms shift sound by -236 ms shift sound by -236 ms (/home/sijar/Videos/Webcam) 25377044K bytes free on a total of 39908968K (used: 36 %) treshold=51200K (/home/sijar/Videos/Webcam) 25373408K bytes free on a total of 39908968K (used: 36 %) treshold=51200K AUDIO: droping audio data AUDIO: droping audio data AUDIO: droping audio data AUDIO: droping audio data AUDIO: droping audio data AUDIO: droping audio data ... repeats a couple of times ... (/home/sijar/Videos/Webcam) 25370696K bytes free on a total of 39908968K (used: 36 %) treshold=51200K AUDIO: droping audio data AUDIO: droping audio data AUDIO: droping audio data ... repeats a couple of times ... (/home/sijar/Videos/Webcam) 25367680K bytes free on a total of 39908968K (used: 36 %) treshold=51200K (/home/sijar/Videos/Webcam) 25364052K bytes free on a total of 39908968K (used: 36 %) treshold=51200K (/home/sijar/Videos/Webcam) 25360312K bytes free on a total of 39908968K (used: 36 %) treshold=51200K (/home/sijar/Videos/Webcam) 25356628K bytes free on a total of 39908968K (used: 36 %) treshold=51200K (/home/sijar/Videos/Webcam) 25352908K bytes free on a total of 39908968K (used: 36 %) treshold=51200K (/home/sijar/Videos/Webcam) 25349316K bytes free on a total of 39908968K (used: 36 %) treshold=51200K (/home/sijar/Videos/Webcam) 25345552K bytes free on a total of 39908968K (used: 36 %) treshold=51200K (/home/sijar/Videos/Webcam) 25341828K bytes free on a total of 39908968K (used: 36 %) treshold=51200K (/home/sijar/Videos/Webcam) 25338092K bytes free on a total of 39908968K (used: 36 %) treshold=51200K (/home/sijar/Videos/Webcam) 25334412K bytes free on a total of 39908968K (used: 36 %) treshold=51200K Cap Video toggled: 0 Shuting Down IO Thread stop= 4708817235000 start=4578624714000 VIDEO: 1604 frames in 130192.000000 ms = 12.320265 fps Stoping audio stream Closing audio stream... close avi Last message repeated 1603 times [libx264 @ 0x8cfba20]frame I:16 Avg QP:14.78 size: 42627 [libx264 @ 0x8cfba20]frame P:1547 Avg QP:16.44 size: 28599 [libx264 @ 0x8cfba20]mb I I16..4: 21.6% 0.0% 78.4% [libx264 @ 0x8cfba20]mb P I16..4: 28.1% 0.0% 0.0% P16..4: 70.5% 0.0% 0.0% 0.0% 0.0% skip: 1.4% [libx264 @ 0x8cfba20]final ratefactor: 88.17 [libx264 @ 0x8cfba20]coded y,uvDC,uvAC intra: 74.4% 95.8% 83.2% inter: 75.2% 94.6% 69.2% [libx264 @ 0x8cfba20]i16 v,h,dc,p: 27% 17% 40% 16% [libx264 @ 0x8cfba20]i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 25% 25% 21% 3% 6% 4% 5% 4% 7% [libx264 @ 0x8cfba20]i8c dc,h,v,p: 61% 18% 18% 4% [libx264 @ 0x8cfba20]ref P L0: 64.0% 36.0% [libx264 @ 0x8cfba20]kb/s:-0.00 total frames encoded: 0 total audio frames encoded: 0 IO thread finished...OK IO Thread finished enabling controls Shuting Down Thread Thread terminated... cleaning Thread allocations: 100% SDL Quit Video Thread finished write /home/sijar/.guvcviewrc OK free audio mutex closed v4l2 strutures free controls free controls - vidState cleaned allocations - 100% Closing portaudio ...OK Closing GTK... OK

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  • VLC 2.0.3 on Lubuntu 12.04: No audio?

    - by drezabek
    I am on Lubuntu 12.04, and I have installed VLC media player version 2.0.3. When I try and play an audio file, it appears to load fine, and the media position bar displays the progress, and it says it is playing, but I can't here any thing through my speakers. I can hear game audio, web audio, and audio from SMPlayer just fine, but with VLC, I can't here anything. Below is the "Messages" output with the verbosity option set to "2 (debug)" main debug: processing request item: The Bottom, node: Playlist, skip: 0 main debug: resyncing on The Bottom main debug: The Bottom is at 0 main debug: starting playback of the new playlist item main debug: resyncing on The Bottom main debug: The Bottom is at 0 main debug: creating new input thread main debug: Creating an input for 'The Bottom' main debug: TIMER input launching for 'Floex - Machinarium Soundtrack - 01 The Bottom.flac' : 23.706 ms - Total 23.706 ms / 1 intvls (Avg 23.706 ms) main debug: using timeshift granularity of 50 MiB, in path '/tmp' main debug: `file:///home/doug/Music/unsorted/Floex%20-%20Machinarium%20Soundtrack/Floex%20-%20Machinarium%20Soundtrack%20-%2001%20The%20Bottom.flac' gives access `file' demux `' path `/home/doug/Music/unsorted/Floex%20-%20Machinarium%20Soundtrack/Floex%20-%20Machinarium%20Soundtrack%20-%2001%20The%20Bottom.flac' main debug: creating demux: access='file' demux='' location='/home/doug/Music/unsorted/Floex%20-%20Machinarium%20Soundtrack/Floex%20-%20Machinarium%20Soundtrack%20-%2001%20The%20Bottom.flac' file='/home/doug/Music/unsorted/Floex - Machinarium Soundtrack/Floex - Machinarium Soundtrack - 01 The Bottom.flac' main debug: looking for access_demux module: 3 candidates main debug: no access_demux module matching "file" could be loaded main debug: TIMER module_need() : 2.332 ms - Total 2.332 ms / 1 intvls (Avg 2.332 ms) main debug: creating access 'file' location='/home/doug/Music/unsorted/Floex%20-%20Machinarium%20Soundtrack/Floex%20-%20Machinarium%20Soundtrack%20-%2001%20The%20Bottom.flac', path='/home/doug/Music/unsorted/Floex - Machinarium Soundtrack/Floex - Machinarium Soundtrack - 01 The Bottom.flac' main debug: looking for access module: 2 candidates filesystem debug: opening file `/home/doug/Music/unsorted/Floex - Machinarium Soundtrack/Floex - Machinarium Soundtrack - 01 The Bottom.flac' main debug: using access module "filesystem" main debug: TIMER module_need() : 0.762 ms - Total 0.762 ms / 1 intvls (Avg 0.762 ms) main debug: Using stream method for AStream* main debug: starting pre-buffering main debug: received first data after 0 ms main debug: pre-buffering done 1024 bytes in 0s - 43478 KiB/s main debug: looking for stream_filter module: 7 candidates main debug: no stream_filter module matching "any" could be loaded main debug: TIMER module_need() : 0.236 ms - Total 0.236 ms / 1 intvls (Avg 0.236 ms) main debug: looking for stream_filter module: 1 candidate main debug: using stream_filter module "stream_filter_record" main debug: TIMER module_need() : 0.156 ms - Total 0.156 ms / 1 intvls (Avg 0.156 ms) main debug: creating demux: access='file' demux='' location='/home/doug/Music/unsorted/Floex%20-%20Machinarium%20Soundtrack/Floex%20-%20Machinarium%20Soundtrack%20-%2001%20The%20Bottom.flac' file='/home/doug/Music/unsorted/Floex - Machinarium Soundtrack/Floex - Machinarium Soundtrack - 01 The Bottom.flac' main debug: looking for demux module: 54 candidates flacsys debug: Picture type=3 mime=image/png description='' file length=679371 qt4 debug: IM: Setting an input main debug: looking for packetizer module: 21 candidates main debug: using packetizer module "packetizer_flac" main debug: TIMER module_need() : 0.211 ms - Total 0.211 ms / 1 intvls (Avg 0.211 ms) main debug: using demux module "flacsys" main debug: TIMER module_need() : 4.023 ms - Total 4.023 ms / 1 intvls (Avg 4.023 ms) main debug: looking for a subtitle file in /home/doug/Music/unsorted/Floex - Machinarium Soundtrack/ main debug: looking for meta reader module: 2 candidates main debug: using meta reader module "taglib" main debug: TIMER module_need() : 5.245 ms - Total 5.245 ms / 1 intvls (Avg 5.245 ms) main debug: removing module "taglib" main debug: `file:///home/doug/Music/unsorted/Floex%20-%20Machinarium%20Soundtrack/Floex%20-%20Machinarium%20Soundtrack%20-%2001%20The%20Bottom.flac' successfully opened main debug: selecting program id=0 main debug: looking for decoder module: 30 candidates main debug: using decoder module "flac" main debug: TIMER module_need() : 0.442 ms - Total 0.442 ms / 1 intvls (Avg 0.442 ms) main debug: Buffering 0% flac debug: decode STREAMINFO flac debug: channels:2 samplerate:44100 bitspersamples:16 flac debug: STREAMINFO decoded main debug: Buffering 30% main debug: recycling audio output main debug: looking for audio output module: 3 candidates main debug: Buffering 61% pulse debug: using stereo channel map pulse debug: using library version 1.1.0 pulse debug: (compiled with version 1.1.0, protocol 26) main debug: Buffering 92% main debug: Stream buffering done (371 ms in 2 ms) pulse debug: connected locally to unix:/home/doug/.pulse/dce22254e867f905188a2ce200000003-runtime/native as client #14 pulse debug: using protocol 26, server protocol 26 pulse debug: using buffer metrics: maxlength=4194304, tlength=9880, prebuf=0, minreq=3528 pulse debug: connected to sink 0: alsa_output.pci-0000_00_14.2.analog-stereo main debug: using audio output module "pulse" main debug: TIMER module_need() : 4.571 ms - Total 4.571 ms / 1 intvls (Avg 4.571 ms) main debug: output 's16l' 44100 Hz Stereo frame=1 samples/4 bytes main debug: mixer 'f32l' 44100 Hz Stereo frame=1 samples/8 bytes main debug: filter(s) 'f32l'->'s16l' 44100 Hz->44100 Hz Stereo->Stereo main debug: looking for audio filter module: 14 candidates audio_format debug: f32l->s16l, bits per sample: 32->16 main debug: using audio filter module "audio_format" main debug: TIMER module_need() : 0.187 ms - Total 0.187 ms / 1 intvls (Avg 0.187 ms) main debug: conversion pipeline completed main debug: looking for audio mixer module: 2 candidates main debug: using audio mixer module "float32_mixer" main debug: TIMER module_need() : 0.125 ms - Total 0.125 ms / 1 intvls (Avg 0.125 ms) main debug: input 's16l' 44100 Hz Stereo frame=1 samples/4 bytes main debug: looking for audio filter module: 1 candidate scaletempo debug: format: 44100 rate, 2 nch, 4 bps, fl32 scaletempo debug: params: 30 stride, 0.200 overlap, 14 search scaletempo debug: 1.000 scale, 1323.000 stride_in, 1323 stride_out, 1059 standing, 264 overlap, 617 search, 2204 queue, fl32 mode main debug: using audio filter module "scaletempo" main debug: TIMER module_need() : 0.233 ms - Total 0.233 ms / 1 intvls (Avg 0.233 ms) main debug: filter(s) 's16l'->'f32l' 44100 Hz->44100 Hz Stereo->Stereo pulse debug: listing sink alsa_output.pci-0000_00_14.2.analog-stereo (0): Built-in Audio Analog Stereo main debug: looking for audio filter module: 14 candidates audio_format debug: s16l->f32l, bits per sample: 16->32 main debug: using audio filter module "audio_format" main debug: TIMER module_need() : 0.147 ms - Total 0.147 ms / 1 intvls (Avg 0.147 ms) main debug: conversion pipeline completed pulse debug: base volume: 65536 main debug: looking for audio filter module: 1 candidate equalizer debug: equalizer loaded for 44100 Hz with 10 bands 2 pass equalizer debug: 60 Hz -> factor:0.000000 alpha:0.003013 beta:0.993973 gamma:1.993901 equalizer debug: 170 Hz -> factor:0.000000 alpha:0.008490 beta:0.983019 gamma:1.982437 equalizer debug: 310 Hz -> factor:0.000000 alpha:0.015374 beta:0.969252 gamma:1.967331 equalizer debug: 600 Hz -> factor:0.000000 alpha:0.029328 beta:0.941343 gamma:1.934254 equalizer debug: 1000 Hz -> factor:0.000000 alpha:0.047918 beta:0.904163 gamma:1.884869 equalizer debug: 3000 Hz -> factor:0.000000 alpha:0.130408 beta:0.739184 gamma:1.582718 equalizer debug: 6000 Hz -> factor:0.000000 alpha:0.226555 beta:0.546889 gamma:1.015267 equalizer debug: 12000 Hz -> factor:0.000000 alpha:0.344937 beta:0.310127 gamma:-0.181410 equalizer debug: 14000 Hz -> factor:0.000000 alpha:0.366438 beta:0.267123 gamma:-0.521151 equalizer debug: 16000 Hz -> factor:0.000000 alpha:0.379009 beta:0.241981 gamma:-0.808451 main debug: using audio filter module "equalizer" main debug: TIMER module_need() : 0.353 ms - Total 0.353 ms / 1 intvls (Avg 0.353 ms) main debug: filter(s) 'f32l'->'f32l' 44100 Hz->44100 Hz Stereo->Stereo main debug: conversion pipeline completed main debug: looking for visualization2 module: 1 candidate main debug: looking for text renderer module: 2 candidates freetype debug: Building font databases. freetype debug: Took 0 microseconds freetype debug: Using Serif Bold as font from file /usr/share/fonts/truetype/ttf-dejavu/DejaVuSans.ttf freetype debug: using fontsize: 2 main debug: using text renderer module "freetype" main debug: TIMER module_need() : 3.278 ms - Total 3.278 ms / 1 intvls (Avg 3.278 ms) main debug: looking for video filter2 module: 18 candidates swscale debug: 32x32 chroma: YUVA -> 16x16 chroma: RGBA with scaling using Bicubic (good quality) main debug: using video filter2 module "swscale" main debug: TIMER module_need() : 1.037 ms - Total 1.037 ms / 1 intvls (Avg 1.037 ms) main debug: looking for video filter2 module: 18 candidates yuvp debug: YUVP to YUVA converter main debug: using video filter2 module "yuvp" main debug: TIMER module_need() : 0.156 ms - Total 0.156 ms / 1 intvls (Avg 0.156 ms) main debug: Deinterlacing available main debug: deinterlace 0, mode blend, is_needed 0 main debug: Opening vout display wrapper main debug: looking for vout display module: 6 candidates main debug: looking for vout window xid module: 4 candidates qt4 debug: requesting video... qt4 debug: Video was requested 0, 0 main debug: using vout window xid module "qt4" main debug: TIMER module_need() : 61.671 ms - Total 61.671 ms / 1 intvls (Avg 61.671 ms) main debug: looking for inhibit module: 2 candidates main debug: using inhibit module "xdg_screensaver" main debug: TIMER module_need() : 0.336 ms - Total 0.336 ms / 1 intvls (Avg 0.336 ms) xdg_screensaver debug: started xdg-screensaver (PID = 6682) xcb_xv debug: connected to X11.0 server xcb_xv debug: vendor : The X.Org Foundation xcb_xv debug: version: 11103000 xcb_xv debug: using screen 0x15a xcb_xv debug: using XVideo extension v2.2 xcb_xv debug: using adaptor NV17 Video Texture xcb_xv debug: using port 310 xcb_xv debug: using image format 0x30323449 xcb_xv debug: using X11 visual ID 0x21 (depth: 24) xcb_xv debug: using X11 window 0x03400000 xcb_xv debug: using X11 graphic context 0x03400002 main debug: VoutDisplayEvent 'fullscreen' 0 main debug: VoutDisplayEvent 'resize' 800x500 window main debug: using vout display module "xcb_xv" main debug: TIMER module_need() : 69.890 ms - Total 69.890 ms / 1 intvls (Avg 69.890 ms) main debug: original format sz 800x500, of (0,0), vsz 800x500, 4cc I420, sar 1:1, msk r0x0 g0x0 b0x0 main debug: removing module "freetype" main debug: looking for text renderer module: 2 candidates freetype debug: Building font databases. freetype debug: Took 0 microseconds freetype debug: Using Serif Bold as font from file /usr/share/fonts/truetype/ttf-dejavu/DejaVuSans.ttf freetype debug: using fontsize: 2 main debug: using text renderer module "freetype" main debug: TIMER module_need() : 4.552 ms - Total 4.552 ms / 1 intvls (Avg 4.552 ms) main debug: using visualization2 module "visual" main debug: TIMER module_need() : 84.104 ms - Total 84.104 ms / 1 intvls (Avg 84.104 ms) main debug: filter(s) 'f32l'->'f32l' 44100 Hz->44100 Hz Stereo->Stereo main debug: conversion pipeline completed main debug: filter(s) 'f32l'->'f32l' 44100 Hz->44100 Hz Stereo->Stereo main debug: conversion pipeline completed main debug: filter(s) 'f32l'->'f32l' 48510 Hz->44100 Hz Stereo->Stereo main debug: looking for audio filter module: 14 candidates main debug: using audio filter module "samplerate" main debug: TIMER module_need() : 0.375 ms - Total 0.375 ms / 1 intvls (Avg 0.375 ms) main debug: conversion pipeline completed main debug: End of audio preroll main debug: Decoder buffering done in 91 ms main warning: PTS is out of range (-9269), dropping buffer pulse debug: deferring start (190703 us) main debug: looking for video blending module: 1 candidate main debug: using video blending module "blend" main debug: TIMER module_need() : 0.275 ms - Total 0.275 ms / 1 intvls (Avg 0.275 ms) main debug: Detected interlaced video main debug: deinterlace 0, mode blend, is_needed 1 xcb_xv debug: display is visible pulse debug: starting deferred pulse warning: too late by 93760 us pulse debug: changed sample rate to 44186 Hz pulse debug: started pulse warning: too late by 94474 us pulse debug: changed sample rate to 44229 Hz pulse warning: too late by 93532 us pulse debug: changed sample rate to 44272 Hz pulse warning: too late by 92829 us pulse debug: changed sample rate to 44315 Hz pulse warning: too late by 92132 us pulse debug: changed sample rate to 44358 Hz xcb_xv debug: display is visible pulse warning: too late by 91534 us pulse debug: changed sample rate to 44401 Hz xcb_xv debug: display is visible pulse warning: too late by 89482 us pulse debug: changed sample rate to 44440 Hz xcb_xv debug: display is visible xcb_xv debug: display is visible pulse warning: too late by 87529 us pulse debug: changed sample rate to 44479 Hz pulse warning: too late by 84577 us pulse debug: changed sample rate to 44504 Hz main debug: auto hiding mouse cursor pulse warning: too late by 78562 us pulse debug: changed sample rate to 44492 Hz pulse warning: too late by 68015 us pulse debug: changed sample rate to 44422 Hz xcb_xv debug: display is visible xcb_xv debug: display is visible xcb_xv debug: display is visible xcb_xv debug: display is visible main debug: auto hiding mouse cursor pulse debug: changed sample rate to 44336 Hz xcb_xv debug: display is visible xcb_xv debug: display is visible xcb_xv debug: display is visible main debug: auto hiding mouse cursor I have had issues with VLC in the past- the audio quality was extremely crackly, as if the headphone jack was plugged in only half way, and the sounds were extremely sharp and caused my speakers to make a ringing/vibrating noise... It would eventually start working after I messed around with the audio settings, but it happened every restart. I eventually switched to SMPlayer, but now I need some of the features that VLC offers, but I still can't use VLC. At this point, the audio can not be heard at all, and the method I used before, messing around with the audio settings, isn't getting me anywhere. (note, I reposted this on VideoLan's forums, link is here: http://forum.videolan.org/viewtopic.php?f=13&t=104726) Please let me know if you need more information, or are confused by something I posted! Thanks!

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  • Visual Studio 2008 doesn't create *.refresh files for external DLL references... what am I missing?

    - by Cory Larson
    Hi all-- I've got a question about something that's just been irritating me. A colleague and I are building a support framework for our current client that we want to reference in other projects. The DLL we want as a reference in our project would be an external reference. We're adding it by doing "Add Reference...", then browsing to the location of the .dll. What I want Visual Studio to do is only add the .xml, .pdb, and a .dll.refresh file, but instead it copies the actual .dll (and .xml and .pdb) into the bin. When we rebuild the framework project, the other project that uses its .dll gets all out of whack until we drop and re-add the reference. Everything I've read online says that VS2008 is supposed to create the .dll.refresh files for you, but it never does. Any ideas? Am I missing something or doing something wrong? At this point I'm ready to add a pre-build event to simply copy the framework .dll into my bin, but the .refresh file seems like less of a hassle if it would just work. Thanks, Cory

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  • How to do partial page refresh using struts2-jquery plugin in struts2?

    - by user1703710
    I want to do partial page refresh with the help of this. Take a scenario, we have a dropdown list according to select option of it, I want to refresh a div section of a page with data populated according to dropdown selection . How to do this? i have tried this: JSP Code: On this Dropdown selection i want to populate (refresh) div. <s:form id="RoleListForm"> <s:label value="Roles"/> <s:url id="fetchJsonRoleListUrl" action="fetchJsonRoleList" namespace="/RolesPrivilegesJson"/> <sj:select name="idRoleInfo" id="idRoleInfoList" href="%{fetchJsonRoleListUrl}" list="roleNameList" onChangeTopics="reloadRolePrivilegesDiv" listKey="idRoleInfo" listValue="roleName" emptyOption="true"/> </s:form> Here is the div code that i want to populate according to DD selection: <s:url id="roleDetailsUrl" action="roleDetailsAction" /> <sj:div href="%{roleDetailsUrl}" formIds="RoleListForm" reloadTopics="reloadRolePrivilegesDiv"> <s:textfield id="idRoleName" name="roleName" /> <s:textfield id="idRolePrivileges" name="privileges"/> </sj:div>

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  • How to Refresh / Reload a KML layer in OpenLayers. Dynamic KML Layer.

    - by Ozaki
    TLDR See my answer below on how to refresh the layer. So far I have tried action function as follows: function RefreshKMLData(layer) { layer.loaded = false; layer.setVisibility(true); layer.redraw({ force: true }); } set interval of the function: window.setInterval(RefreshKMLData, 5000, KMLLAYER); the layer itself: var KMLLAYER = new OpenLayers.Layer.Vector("MYKMLLAYER", { projection: new OpenLayers.Projection("EPSG:4326"), strategies: [new OpenLayers.Strategy.Fixed()], protocol: new OpenLayers.Protocol.HTTP({ url: MYKMLURL, format: new OpenLayers.Format.KML({ extractStyles: true, extractAttributes: true }) }) }); the url for KMLLAYER with Math random so it doesnt cache: var MYKMLURL = var currentanchorpositionurl = 'http://' + host + '/data?_salt=' + Math.random(); I would have thought that this would Refresh the layer. As by setting its loaded to false unloads it. Visibility to true reloads it and with the Math random shouldn't allow it to cache? So has anyone done this before or know how I can get this to work? TLDR See my answer below on how to refresh the layer.

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  • Jquery tabs with cookie support restore wrong tab position after page refresh.

    - by zenonych
    Hello, all. I have tricky problem which I can't completely understand... It's jquery tabs with cookie support. I've following code: $(document).ready(function() { var $tabs = $("#tabs").tabs(); $tabs.tabs('select', $.cookie("tabNumber")); $('#tabs ul li a').click(function() { $.cookie("tabNumber", $tabs.tabs('option', 'selected')); }); $('#btnSelect').click(function() { //alert($.cookie("tabNumber")); //$tabs.tabs('select', 2); $tabs.tabs('select', $.cookie("tabNumber")); }); }); So, I've 3 tabs (with positions 0,1,2) inside div named "tabs". When user selects one tab, then tab position stores in cookie. After that if user refresh page, active tab position must be restored. But each time I refresh page I get active tab in previous position (if I select 2nd tab, then after refresh I got active tab in position 1, etc.). I add some test in code (button btnSelect with onclick handler which duplicates load position functionality). So, if I uncomment and use $tabs.tabs('select', 2); Then after I click btnSelect I've got right position. Ok, that's right. Then I comment that line and uncomment next one: alert($.cookie("tabNumber")); So, I select tab, click button, get dialog message "2", and after that tab in position 1 became active. Why?? In both cases I call 'select' method with parameter 2... I know I can use aliases for tabs, but I want to understate why my code doesn't work properly.

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  • How do I force all Tree itemrenderers to refresh?

    - by Richard Haven
    I have item renderers in an mx.controls.Tree that I need to refresh on demand. I have code in the updateDisplayList that fires for only some of the visible nodes no matter what I do. I've tried triggering a change that they should all be listening for; I have tried clearing and resetting the dataProvider and the itemRenderer properties. private function forceCategoryTreeRefresh(event : Event = null) : void { trace("forceCategoryTreeRefresh"); var prevDataProvider : Object = CategoryTree.dataProvider; CategoryTree.dataProvider = null; CategoryTree.validateNow(); CategoryTree.dataProvider = prevDataProvider; var prevItemRenderer : IFactory = CategoryTree.itemRenderer; CategoryTree.itemRenderer = null; CategoryTree.itemRenderer = prevItemRenderer as IFactory; _categoriesChangeDispatcher.dispatchEvent(new Event(Event.CHANGE)); } The nodes refresh properly when I scroll them into view (e.g. the .data gets set), but I cannot force the ones that already exist to refresh or reset themselves. Any ideas?

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  • Meteor: how to update DOM via Javascript without a page refresh?

    - by strack
    UPDATE: It looks like this script may be the catalyst I'm looking for. Will test it and answer/close this question if so. I'm sure I'll find the answer to this eventually, but I'm putting it out there now, in case someone else knows it right off... I am using RaphaelJS to manipulate the DOM (adds a bunch of SVG tags to an identified DIV), and I want to re-render those specific DOM parts, if there's an update to the MongoDB collection. As an example: -Let's say that I have a collection called PiePieces. -When the page is first rendered/ called, let's pretend that the number of pieces in the collection is 4. -I programmatically add a pie piece using console: PiePieces.insert({...}) -I want the page to update like it would for a standard handlebars binding situation, but the problem is, the new entry needs to go through the Raphael script, which performs direct DOM manipulation. So, the logic would go something like this: MongoDB collection update event - Client function call to manipulate DOM - DOM modified/ page updated without a refresh. I've tried implementing this by reading values from the DOM itself, and I can get the updated DOM, but the entire page refreshes and/or I have to manually refresh the page, OR the DOM tree isn't completed yet, and so it's blank until I refresh. Can you point me in the right direction, maybe with a small code snippet/example? (if something similar already exists, just tell me where and I'll go digging) Thanks in advance! (I am LOVING Meteor so far...)

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  • When can Java produce a NaN (with specific code question)

    - by Brent
    I'm a bit perplexed by some code I'm currently writing. I am trying to preform a specific gradient descent (main loop included below) and depending on the initial conditions I will alternatively get good looking results (perhaps 20% of the time) or everything becomes NaN (the other 80% of the time). However it seems to me that none of the operations in my code could produce NaN's when given honest numbers! My main loop is: // calculate errors delta = m1 + m2 - M; eta = f1 + f2 - F; for (int i = 0; i < numChildren; i++) { epsilon[i] = p[i]*m1+(1-p[i])*m2+q[i]*f1+(1-q[i])*f2-C[i]; } // use errors in gradient descent // set aside differences for the p's and q's float mDiff = m1 - m2; float fDiff = f1 - f2; // first update m's and f's m1 -= rate*delta; m2 -= rate*delta; f1 -= rate*eta; f2 -= rate*eta; for (int i = 0; i < numChildren; i++) { m1 -= rate*epsilon[i]*p[i]; m2 -= rate*epsilon[i]*(1-p[i]); f1 -= rate*epsilon[i]*q[i]; f2 -= rate*epsilon[i]*(1-q[i]); } // now update the p's and q's for (int i = 0; i < numChildren; i++) { p[i] -= rate*epsilon[i]*mDiff; q[i] -= rate*epsilon[i]*fDiff; } This behavior can be seen when we have rate = 0.01; M = 30; F = 30; C = {15, 25, 35, 45}; with the p[i] and q[i] chosen randomly uniformly between 0 and 1, m1 and m2 chosen randomly uniformly to add to M, and f1 and f2 chosen randomly uniformly to add up to F. Does anyone see anything that could create these NaN's?

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  • APC not caching many files

    - by tetranz
    Hello I have a Drupal site running on a VPS at Linode with PHP 5.2.10 and APC 3.1.6. It never caches more than about 25 files and barely uses any of its available memory. Drupal has hundreds of php files. I have another server where APC seems to work well and does indeed cache hundreds of files. The only difference with that site is that it runs Ubuntu 10.04 and php 5.3.2. The config settings are the same. What could be wrong? I'll paste the config from apc.php below. This is after hitting multiple parts of Drupal. Thanks APC Version 3.1.6 PHP Version 5.2.10-2ubuntu6.5 APC Host xxx.example.com Server Software Apache/2.2.12 (Ubuntu) Shared Memory 1 Segment(s) with 32.0 MBytes (mmap memory, pthread mutex locking) Start Time 2010/12/02 11:32:17 Uptime 3 minutes File Upload Support 1 File Cache Information Cached Files 21 ( 1.4 MBytes) Hits 169 Misses 21 Request Rate (hits, misses) 1.00 cache requests/second Hit Rate 0.89 cache requests/second Miss Rate 0.11 cache requests/second Insert Rate 0.17 cache requests/second Cache full count 0 User Cache Information Cached Variables 0 ( 0.0 Bytes) Hits 0 Misses 0 Request Rate (hits, misses) 0.00 cache requests/second Hit Rate 0.00 cache requests/second Miss Rate 0.00 cache requests/second Insert Rate 0.00 cache requests/second Cache full count 0 Runtime Settings apc.cache_by_default 1 apc.canonicalize 1 apc.coredump_unmap 0 apc.enable_cli 0 apc.enabled 1 apc.file_md5 0 apc.file_update_protection 2 apc.filters apc.gc_ttl 3600 apc.include_once_override 0 apc.lazy_classes 0 apc.lazy_functions 0 apc.max_file_size 1M apc.mmap_file_mask apc.num_files_hint 1000 apc.preload_path apc.report_autofilter 0 apc.rfc1867 0 apc.rfc1867_freq 0 apc.rfc1867_name APC_UPLOAD_PROGRESS apc.rfc1867_prefix upload_ apc.rfc1867_ttl 3600 apc.shm_segments 1 apc.shm_size 32M apc.slam_defense 1 apc.stat 1 apc.stat_ctime 0 apc.ttl 0 apc.use_request_time 1 apc.user_entries_hint 4096 apc.user_ttl 0 apc.write_lock 1

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  • This .mpg video clip doesn't play well

    - by Roey
    I've installed K-lite mega codec pack v6.9.0 with playback essentials without player. My default and only media player is windows media player. here are the clip's media info: General Complete name : D:\Users\Roey\Downloads\B384MV.mpg Format : MPEG-PS File size : 273 MiB Duration : 4mn 59s Overall bit rate : 7 643 Kbps Video ID : 224 (0xE0) Format : MPEG Video Format version : Version 2 Format profile : Main@High Format settings, BVOP : No Format settings, Matrix : Default Format settings, GOP : M=1, N=15 Duration : 4mn 57s Bit rate mode : Variable Bit rate : 7 363 Kbps Nominal bit rate : 9 000 Kbps Width : 1 920 pixels Height : 1 080 pixels Display aspect ratio : 16:9 Frame rate : 25.000 fps Color space : YUV Chroma subsampling : 4:2:0 Bit depth : 8 bits Scan type : Progressive Compression mode : Lossy Bits/(Pixel*Frame) : 0.142 Stream size : 261 MiB (96%) Audio ID : 192 (0xC0) Format : MPEG Audio Format version : Version 1 Format profile : Layer 3 Mode : Joint stereo Duration : 4mn 59s Bit rate mode : Constant Bit rate : 128 Kbps Channel(s) : 2 channels Sampling rate : 44.1 KHz Compression mode : Lossy Stream size : 4.56 MiB (2%) Menu When I play it there is no sound (just a little "kahhhh" noise every 10-20 seconds) and the frames are moving very slow - it "jumps" frames. A blue tray icon [FFa] "ffdshow audio decoder" pops with the following details: Input:MP3, stereo, 44100 Hz (libavocodec) Output:PCM, stereo, 44100 Hz, 16-bit integer Any help will be much appreciated. Thanks

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  • Apache APC (Windows) Can I optimize these APC settings more?

    - by ar099968
    I would like to optimize APC some more but I am not sure where I could do something. First here is the stats after 1 week of running with the current configuration: General Cache Information APC Version 3.1.9 PHP Version 5.4.4 APC Host XXXXXXXXXXXXXXXXXXXXXXXXXXXXXX Server Software Apache Shared Memory 1 Segment(s) with 128.0 MBytes (IPC shared memory, Windows Slim RWLOCK (native) locking) Start Time 2014/06/08 05:00:00 Uptime 6 days, 11 hours and 55 minutes File Upload Support 1 Host Status Diagrams Memory Usage Free: 99.7 MBytes (77.9%) Used: 28.3 MBytes (22.1%) Hits & Misses Hits: 510818 (99.9%) Misses: 608 (0.1%) Detailed Memory Usage and Fragmentation Fragmentation: 0.60% (609.8 KBytes out of 99.7 MBytes in 83 fragments) File Cache Information Cached Files 693 ( 35.4 MBytes) Hits 5143359 Misses 1087 Request Rate (hits, misses) 13.24 cache requests/second Hit Rate 13.24 cache requests/second Miss Rate 0.00 cache requests/second Insert Rate 0.01 cache requests/second Cache full count 0 User Cache Information Cached Variables 0 ( 0.0 Bytes) Hits 0 Misses 0 Request Rate (hits, misses) 0.00 cache requests/second Hit Rate 0.00 cache requests/second Miss Rate 0.00 cache requests/second Insert Rate 0.00 cache requests/second Cache full count 0 Runtime Settings apc.cache_by_default 1 apc.canonicalize 1 apc.coredump_unmap 0 apc.enable_cli 0 apc.enabled 1 apc.file_md5 0 apc.file_update_protection 2 apc.filters -/apc.php$, -/apc_clean.php$, -.tpl.cache.php$, -.tpl.php$, -.string.cache.php$, -.string.php$ apc.gc_ttl 3600 apc.include_once_override 0 apc.lazy_classes 0 apc.lazy_functions 0 apc.max_file_size 2M apc.num_files_hint 7000 apc.preload_path apc.report_autofilter 0 apc.rfc1867 0 apc.rfc1867_freq 0 apc.rfc1867_name APC_UPLOAD_PROGRESS apc.rfc1867_prefix upload_ apc.rfc1867_ttl 3600 apc.serializer default apc.shm_segments 1 apc.shm_size 128M apc.shm_strings_buffer 4M apc.slam_defense 0 apc.stat 1 apc.stat_ctime 0 apc.ttl 7200 apc.use_request_time 1 apc.user_entries_hint 4096 apc.user_ttl 7200 apc.write_lock 1

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  • How do you update an Excel file (Data Refresh and update formulas) WITHOUT opening the file?

    - by Alex
    I have an Excel file that want to update and save automatically with out having to open it or manually interact with. Manually, I open the file up and hit data refresh which goes and does a SQL query and then hit F9 for the formulas to update and then I just close/save. (I then would mail the file out to people using a perl script or use SAS JMP to run some numbers/charts and also mail them out. Basically I need to script some things which require the XLS file to be updated.)

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  • Issue with javascript array object

    - by ezhil
    I have the below JSON response. I am using $.getJSON method to loads JSON data and using callback function to do some manipulation by checking whether it is array as below. { "r": [{ "IsDefault": false, "re": { "Name": "Depo" }, "Valid": "Oct8, 2013", "Clg": [{ "Name": "james", "Rate": 0.05 }, { "Name": "Jack", "Rate": 0.55 }, { "Name": "Mcd", "Rate": 0.01, }], }, { "IsDefault": false, "re": { "Name": "Depo" }, "Valid": "Oct8, 2013", "Clg": [{ "Name": "james", "Rate": 0.05 }, { "Name": "Jack", "Rate": 0.55 }, { "Name": "Mcd", "Rate": 0.01, }], }, { "IsDefault": false, "re": { "Name": "Depo" }, "Valid": "Oct8, 2013", "Clg": [{ "Name": "james", "Rate": 0.05 }, { "Name": "Jack", "Rate": 0.55 }, { "Name": "Mcd", "Rate": 0.01, }], }] } I am passing the json responses on both loadFromJson1 and loadFromJson2 function as "input" as parameter as below. var tablesResult = loadFromJson1(resultstest.r[0].Clg); loadFromJson1 = function (input) { if (_.isArray(input)) { alert("loadFromJson1: Inside array function"); var collection = new CompeCollection(); _.each(input, function (modelData) { collection.add(loadFromJson1(modelData)); }); return collection; } return new CompeModel({ compeRates: loadFromJson2(input), compName: input.Name }); }; loadFromJson2 = function (input) // here is the problem, the 'input' is not an array object so it is not going to IF condition of the isArray method. { if (_.isArray(input)) { alert("loadFromJson2: Inside array function"); //alert is not coming here though it is an array var rcollect = new rateCollection(); _.each(input, function (modelData) { rcollect.add(modelData); }); return rcollect; } }; The above code i am passing json responses for both loadFromJson1 and loadFromJson2 function as "input". isArray is getting true on only loadFromJson1 function and giving alert inside the if condition but not coming in loadFromJson2 function though i am passing the same parameter. can anyone tell me why loadFromJson2 function is not getting the alert inside if condition though i pass array object?

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  • Upload using python script takes very long on one laptop as compared to another

    - by Engr Am
    I have a python 2.7 code which uses STORBINARY function for uploading files to an ftp server and RETRBINARY for downloading from this server. However, the issue is the upload is taking a very long time on three laptops from different brands as compared to a Dell laptop. The strange part is when I manually upload any file, it takes the same time on all the systems. The manual upload rate and upload rate with the python script is the same on the Dell Laptop. However, on every other brand of laptop (I have tried with IBM, Toshiba, Fujitsu-Siemens) the python script has a very low upload rate than the manual attempt. Also, on all these other laptops, the upload rate using the python script is the same (1Mbit/s) while the manual upload rate is approx. 8 Mbit/s. I have tried to vary the filesize for the upload to no avail. TCP Optimizer improved the download rate on all the systems but had no effect on the upload rate. Download rate using this script on all the systems is fine and same as the manual download rate. I have checked the server and it has more than 90% free space. The network connection is the same for all the laptops, and I try uploading only with one laptop at a time. All the laptops have almost the same system configurations, same operating system and approximately the same free drive space. If anything the Dell laptop is a little less in terms of processing power and RAM than 2 of the others, but I suppose this has no effect as I have checked many times to see how much was the CPU usage and network usage during these uploads and downloads, and I am sure that no other virus or program has been eating up my bandwidth. Here is the code ('ftp' and 'file_path' are inputs to the function): path,filename=os.path.split(file_path) filesize=os.path.getsize(file_path) deffilesize=(filesize/1024)/1024 f = open(file_path, "rb") upstart = time.clock() print ftp.storbinary("STOR "+filename, f) upende = time.clock()-upstart outname="Upload " f.close() return upende, deffilesize, outname

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  • Javascript and Twitter API rate limitation? (Changing variable values in a loop)

    - by Pablo
    Hello, I have adapted an script from an example of http://github.com/remy/twitterlib. It´s a script that makes one query each 10 seconds to my Twitter timeline, to get only the messages that begin with a musical notation. It´s already working, but I don´t know it is the better way to do this... The Twitter API has a rate limit of 150 IP access per hour (queries from the same user). At this time, my Twitter API is blocked at 25 minutes because the 10 seconds frecuency between posts. If I set up a frecuency of 25 seconds between post, I am below the rate limit per hour, but the first 10 posts are shown so slowly. I think this way I can guarantee to be below the Twitter API rate limit and show the first 10 posts at normal speed: For the first 10 posts, I would like to set a frecuency of 5 seconds between queries. For the rest of the posts, I would like to set a frecuency of 25 seconds between queries. I think if making somewhere in the code a loop with the previous sentences, setting the "frecuency" value from 5000 to 25000 after the 10th query (or after 50 seconds, it´s the same), that´s it... Can you help me on modify this code below to make it work? Thank you in advance. var Queue = function (delay, callback) { var q = [], timer = null, processed = {}, empty = null, ignoreRT = twitterlib.filter.format('-"RT @"'); function process() { var item = null; if (q.length) { callback(q.shift()); } else { this.stop(); setTimeout(empty, 5000); } return this; } return { push: function (item) { var green = [], i; if (!(item instanceof Array)) { item = [item]; } if (timer == null && q.length == 0) { this.start(); } for (i = 0; i < item.length; i++) { if (!processed[item[i].id] && twitterlib.filter.match(item[i], ignoreRT)) { processed[item[i].id] = true; q.push(item[i]); } } q = q.sort(function (a, b) { return a.id > b.id; }); return this; }, start: function () { if (timer == null) { timer = setInterval(process, delay); } return this; }, stop: function () { clearInterval(timer); timer = null; return this; }, empty: function (fn) { empty = fn; return this; }, q: q, next: process }; }; $.extend($.expr[':'], { below: function (a, i, m) { var y = m[3]; return $(a).offset().top y; } }); function renderTweet(data) { var html = ''; html += ''; html += twitterlib.ify.clean(data.text); html += ''; since_id = data.id; return html; } function passToQueue(data) { if (data.length) { twitterQueue.push(data.reverse()); } } var frecuency = 10000; // The lapse between each new Queue var since_id = 1; var run = function () { twitterlib .timeline('twitteruser', { filter : "'?'", limit: 10 }, passToQueue) }; var twitterQueue = new Queue(frecuency, function (item) { var tweet = $(renderTweet(item)); var tweetClone = tweet.clone().hide().css({ visibility: 'hidden' }).prependTo('#tweets').slideDown(1000); tweet.css({ top: -200, position: 'absolute' }).prependTo('#tweets').animate({ top: 0 }, 1000, function () { tweetClone.css({ visibility: 'visible' }); $(this).remove(); }); $('#tweets p:below(' + window.innerHeight + ')').remove(); }).empty(run); run();

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  • Will this increase my Virtual private Server failing rate ?

    - by Spencer Lim
    Will this increase my Virtual private Server failing rate if i :- install Microsoft Window Server 2008 Enterprise install SQL server enterprise 2008 install IIS 7.5 install ASP.Net Mvc 2 install Microsoft Exchange << should live inside MWS2008 ? or standalone without OS? install Team foundation server << should live inside MWS2008 ? or standalone without OS? on one mini VPS with specification of DELL Poweredge R710 shared plan DDR3 ECC RAMs 16GB and -- 1GB for this VPS using DELL PERC 6i raid controller (this thing alone about 1.5k-2k) and the SAS HDD (15K RPM) (146GB) -- 33GB to this VPS each hdd is freaking fast over 300MB read / write possible with proper tuning the motherboard is a DELL and it has twin redundant PSU (870watt 85%eff) its running on Intel Xeon 5502 (Quad Core) x2 so about 8 physical proc (fairly share) is there any ruler to measure for this about one VPS can only install what what what service ? because of my resource is limited =.@ may i know if it is install in this way,maybe it seem like defeat the way of "VPS"... what will happen ? or any guideline on this issue (fully configuring the window server 2008 R2) ? Thx for reply

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  • Converting LINQ to Twitter to Twitter API v1.1

    - by Joe Mayo
    Twitter recently updated their API to v1.1 (Current status: API v1.1). Naturally, LINQ to Twitter  needed to be updated too. This blog post outlines the changes made to LINQ to Twitter during this conversion and highlights important features that LINQ to Twitter developers will want to know. Overall Impact Generally speaking, Twitter API v1.1 is semantically very much the same as it’s predecessor. The base URL changed and so did a few resource segments, but the resources themselves are still intact. The good news is that LINQ to Twitter has always shielded the developer from this plumbing, so the entities, types, and filters didn’t change much at all.  The following sections describe what did  change. Authentication In Twitter API v1.0 authentication was not required for some resources, such as user timelines and search. However, that’s all changed because *all* queries must be authenticated in Twitter API v1.1. LINQ to Twitter has various types of authorizers you can use, supporting whatever OAuth options are available via Twitter.  You can see the LINQ to Twitter documentation, Securing Your Applications, for more info on OAuth support. The New Search One of the larger changes to the API was Search. To be more specific, the Search entity now contains a List<Status>, named Statuses, to hold results.  Additionally, any meta-data associated with the search is now in a property named SearchMetaData. The change to the Search entity and responses is the big change, but the good news is that your Search query syntax doesn’t change. Different Rate Limits The issue of rate limits itself is contentious, but this discussion is focused on the coding experience and I’ll leave the politics to those who prefer to engage in that activity. What’s important here is that both headers and resources have changed. You should review Twitter’s Rate Limit documentation to understand what the changes mean.  A quick explanation is that rate limits are applied individually to each resource in 15 minute time intervals. In LINQ to Twitter these changes surface on the Help entity, via HelpType.RateLimits. The RateLimits query has a Resources filter where you can specify a comma-separated list of categories to return rate limit info for.  The results materialize in the RateLimits dictionary, keyed on category. The Help entity also has a RateLimitsAuthorizationContext, holding the Access Token for the user performing queries – and to whom the rate limits apply. In addition to the new RateLimits query, there are new RateLimit headers that appear in the query response, whose HTTP header name is of the form X-Rate-Limit… which is different from the previous header name. LINQ to Twitter surfaces these headers via the existing properties of the TwitterContext instance. For anyone who retrieved rate limit information via the Headers property of TwitterContext, you should be aware of the new header names.  I haven’t done anything with Feature rate limit properties yet, but they appear to no longer be available – this will require more follow-up. Error Handling Twitter API v1.1 has a new format for Error Codes & Responses. LINQ to Twitter wraps these messages in the TwitterQueryException, which has been updated appropriately. The Message property of TwitterQueryException now reflects the Twitter error message, when available. There’s also a new ErrorCode that’s populated with the message error code. Parameters Most parameters stayed the same, but one of interest is Include Entities (different from LINQ to Twitter data object entities). Entities are metadata hanging off tweets, that provide start/end position in the tweet and other information for mentions, urls, hash tags, and media. Entities used to not be included unless you specified you wanted them. Now, in v1.1, entities are included by default for all APIs that return a Status.  If you were always setting IncludeEntities to true, then you won’t see a change. However, be aware that you’ll now be receiving additional data in your response from Twitter, which will explain a sudden increase in bandwidth utilization. This might or might not  matter to you  depending on the requirements of your application, but you should be aware of it. Everything Else There might be small changes here and there that I haven’t mentioned, but these were the ones you should be most aware of.  Streams didn’t change, but Twitter will be deprecating username/password authentication on public streams, in favor of OAuth, so you’ll be seeing me make that change some time in the future.  Also, Twitter will continue to evolve the API and you can expect that LINQ to Twitter will change accordingly. Summary The big changes to Twitter API were Authentication, Search, Rate Limits, and Error Handling. All API calls must be authenticated. You’ll need to change your code to read Search results differently, but the query is much the same as you use now. There’s a new RateLimits API, one of the Help queries.  Also, the new error messages are integrated into TwitterQueryException. Besides these changes, I expect  most others to be small or affect a smaller percentage of developers.  You can get the latest version of LINQ to Twitter from NuGet or visit the LINQ to Twitter download page at CodePlex.com.   @JoeMayo

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  • Hardware refresh of Solaris 10 systems? Try this!

    - by mgerdts
    I've been seeing quite an uptick in the people that are wanting to install Solaris 11 when they are doing hardware refreshes.  I applaud that effort - Solaris 11 (and 11.1) have great improvements that advance the state of the art and make the best use of the latest hardware. Sometimes, however, you really don't want to disturb the OS or upgrade to the a later version of an application that is certified with Solaris 11.  That's a great use of Solaris 10 Zones.  If you are already using Solaris Cluster, or would like to have more protection as you put more eggs in an ever growing basket, check out solaris10 Brand Zone Clusters.

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  • Moving two objects proportionally

    - by SSL
    I'm trying to move two objects away from each other at a proportional distance, but on different scales. I'm not quite sure how to do this. Object A can go from position 0.1 to 1. Object B has no limits. If object B is decreasing, then Object A should be decreasing at rate R. Likewise, if Object B is increasing, then Object A increases at rate R. How can I tie these two Object positions together so that in an update loop, they automatically update their positions? I tried using: ObjA.Pos += 0.001f * ObjB.VelocityY; //0.001f is the rate This works but there's an error each time it runs. ObjA starts off at its max position 1 but then the next time it will stop at 0.97, 0.94, 0.91 etc.. This is due to the 0.001f rate I put in. Is there a way to control the rate, yet not end up with the rounding error?

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  • are projects with high developer turn over rate really a bad thing?

    - by John
    I've inherited a lot of web projects that experienced high developer turn over rates. Sometimes these web projects are a horrible patchwork of band aid solutions. Other times they can be somewhat maintainable mozaics of half-done features each built with a different architectural style. Everytime I inherit these projects, I wish the previous developers could explain to me why things got so bad. What puzzles me is the reaction of the owners (either a manager, a middle man company, or a client). They seem to think, "Well, if you leave, I'll just find another developer." Or they think, "Oh, it costs that much money to refactor the system? I know another developer who can do it at half the price. I'll hire him if I can't afford you." I'm guessing that the high developer turn over rate is related to the owner's mentality of "If you think it's a bad idea to build this, I'll just find another (possibly cheaper) developer to do what I want". For the owners, the approach seems to work because their business is thriving. Unfortunately, it's no fun for the developers that go AWOL 3-4 months after working with poor code, strict timelines, and little feedback. So my question is the following: Are the following symptoms of a project really such a bad thing for business? high developer turn over rate poorly built technology - often a patchwork of different and inappropriately used architectural styles owners without a clear roadmap for their web project, and they request features on a whim I've seen numerous businesses prosper while experiencing the symptoms above. So as a programmer, even though my instincts tell me the above points are terrible, I'm forced to take a step back and ask, "are things really that bad in the grand scheme of things?" If not, I will re-evaluate my approach to these projects.

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  • How to rate-limit concurrent sessions with nginx or haproxy?

    - by bantic
    I'm currently using nginx to reverse-proxy requests from web clients that are doing long-polling to an upstream. Since we're doing long polling (as opposed to websockets), when a client connects it will make multiple http connections to the server in serial, re-establishing a connection every time the server sends it some data (or timing out and re-establishing if the server has nothing to say for 10 seconds). What I'd like to do is limit the number of concurrent web clients. Since the clients are constantly making new HTTP requests instead of keeping a single request open, it's a little tricky to count the total number of web clients (because it's not the same as total number of concurrently connected http clients). The method I've come up with is to track http requests by the originating IP address, and store the IP address somewhere with a TTL of 20 seconds. If a request comes in whose IP isn't recognized, then we check the total number of unexpired stored IP addresses; if that's less than the maximum then we allow this request through. And if a request comes in with an IP address that we can find in the look-up table that hasn't yet expired, then it is allowed through as well. All requests that are allowed through have their IPs added to the table (if not there before) and the TTL refreshed to 20 seconds again. I had actually whipped something together that worked correctly this way using nginx along with the Redis 2.0 Nginx Module (and the nginx lua module to simplify the conditional branching), using redis to store my IP addresses with a TTL (the SETEX command), and checking the table size with the DBSIZE command. This worked but the performance was horrible. nginx and redis ended up using lots of cpu and the machine could only handle a very small number of concurrent requests. The new stick-table and tracking counters that were added to Haproxy in version 1.5 (via a commission from serverfault) seem like they might be ideal to implement exactly this sort of rate limiting, because the stick-table can track IP addresses and automatically expire entries. However, I don't see an easy way to get a total count of the unexpired entries in the stick table, which would be necessary to know the number of connected web clients. I'm curious if anyone has any suggestions, for nginx or haproxy or even for something else not mentioned here that I haven't thought of yet.

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  • How to make a server.transfer() with a Response.AddHeader("refresh", "seconds") and not get a 404 er

    - by Unlimited071
    Hi all, so this is the scenario: I have a base class for all login-controlled pages: public class SessionControlledPage : Page { protected virtual void Page_Load(Object sender, EventArgs e) { Response.AddHeader("Refresh", Convert.ToString(Session.Timeout * 60 + 5)); if (Session.IsNewSession) { Response.Redirect("~/login.aspx"); } } } And a regularpage.aspx page that inherints from the base class: public partial class RegularPage : SessionControlledPage { override protected void Page_Load(Object sender, EventArgs e) { base.Page_Load(sender, e); Server.Transfer("~/otherpage.aspx"); } } Now let's say the original URL was http://localhost/regularpage.aspx and that Session.Timeout = 5. The problem appears when the refresh happens after the transfer takes place. I'm getting a 404 Not Found error and the URL changes for http://localhost/305. Notice that 5 * 60 + 5 = 305. Do you know why is this happening? got a solution? PD: I use transfer because I need to access some properties of regularpage.aspx on otherpage.aspx, maybe you could provide another solution for this too.

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