search for: 32000

...0.6% 16000, 2, 7, 49300, 49600, 0.6% 16000, 2, 8, 57300, 57600, 0.5% 16000, 2, 9, 70100, 70400, 0.4% 16000, 2, 10, 86100, 86400, 0.3% Ultrawideband, Mono 32000, 1, 0, 8300, 8800, 5.6% 32000, 1, 1, 15100, 15200, 0.6% 32000, 1, 2, 19100, 19200, 0.5% 32000, 1, 3, 23200, 23200, 0.0% 32000, 1, 4, 29200,...

...,pivotColumns[level]], drop=T), function(x) lapplyFunctionRecurse(x, level+1, ...))) } } result = lapplyFunctionRecurse(data, ...) matrix2 <- do.call('rbind', lapply(result, function(x) do.call('rbind',x))) return(matrix2) } dat <- data.frame(D=sample(32000:33000, 666000, T),Fid=sample(1:10,666000,T), A=sample(1:5,666000,T)) > temp = proc.time(); ret = fedb.ddplyWrapper2(dat, c("D", "Fid"), function(x) c(sum(x[,"A"], na.rm=T), sum(x[,"A"], na.rm=T))); proc.time()-temp user system elapsed 4.616 0.006...

Hi all, I wrote a function that actually does what I want it to do, but it tends to be very slow for large amount of data. On my computer it takes 5.37 seconds for 16000 data points and 21.95 seconds for 32000 data points. As my real data consists of 18000000 data points it would take ages to use the function as it is now. Could someone help me to speed up the calculation? Thank you, Tonja system.time({ x <- runif(32000) y <- runif(32000) xy <- cbind(x,y) outer <- function(z){ !any(x &g...

I have a 32000 x 14 matrix (M) where entry (x, y) corresponds to person x at a discrete time y. I then have a matrix (M2) of 60000 x 2 where each entry is a an event by a person that is in the above 32000 and at a time that is in the range of the discrete time points above. I want to populate the another matrix (...

...ress to 1Mbit tc qdisc add dev eth0 parent 1:0 handle 2:0 htb tc class add dev eth0 parent 2:0 classid 2:1 htb rate 1Mbit ceil 1Mbit #create 13 gred''s tc qdisc add dev eth0 parent 2:1 gred setup DPs 13 default 13 grio #ef tc qdisc change dev eth0 parent 2:1 gred limit 512000 min 24000 max 32000 \ avpkt 1000 burst 40 probability 0.01 bandwidth 1024 DP 1 prio 1 #af41 tc qdisc change dev eth0 parent 2:1 gred limit 512000 min 24000 max 32000 \ avpkt 1000 burst 40 probability 0.04 bandwidth 1024 DP 2 prio 2 #af42 tc qdisc change dev eth0 parent 2:1 gred limit 512000 min 24000 max 32000 \ avpkt...

...ation: Android Based Client SDP Parameters v=0 o=- 1325786904 1325786904 IN IP4 172.16.130.47 s=Polycom RealPresence c=IN IP4 172.16.130.47 b=AS:1920 t=0 0 a=sendrecv m=audio 3230 RTP/AVP 118 115 114 113 0 8 119 a=rtpmap:118 SIRENLPR/48000 a=fmtp:118 bitrate=64000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:114 G7221/32000 a=fmtp:114 bitrate=32000 a=rtpmap:113 G7221/32000 a=fmtp:113 bitrate=24000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:119 telephone-event/8000 a=fmtp:119 0-15 m=video 3232 RTP/AVP 109 110 a=rtcp-fb:* ccm fir tmmbr a=rtpmap:109 H2...

we have following setup on sysctl.conf: kernel.sem = 256 32000 128 142 I tried to startup database and have error happen. I knew problem come from SEMMNS (32000) is not enough. Does anyone know how to calculate how many semaphores already use on system? Thanks. %ipcs -s ------ Semaphore Arrays -------- key semid owner perms nsems...

...adpcm:8000 : No Translation Path speex:8000 To slin:8000 : No Translation Path speex:8000 To slin:12000 : No Translation Path speex:8000 To slin:16000 : No Translation Path speex:8000 To slin:24000 : No Translation Path speex:8000 To slin:32000 : No Translation Path speex:8000 To slin:44100 : No Translation Path speex:8000 To slin:48000 : No Translation Path speex:8000 To slin:96000 : No Translation Path speex:8000 To slin:192000 : No Translation Path speex:8000 To lpc10:8000...

...in it: 8 drives are 2 Tb each and 4 are 1 Tb each. All of them are in array RAID 1+0. This storage is connected to two servers which use data, stored on the storage. So I'm using OCFS2 on these two nodes. Today, after long time of successfull work with it, I've found that it has a limit of 32000 subdirs. The trouble is I have more than 32000 subdirs and amount of subdirs will only increase. Is there a way to overcome this limit? I've found this topic: http://www.mail-archive.com/ocfs2-users at oss.oracle.com/msg03151.html which says that there is an option named "indexed director...

Using FFmpeg, I can stream to a file on disk okay: c:\apps\ffmpeg\bin\ffmpeg.exe -f dshow -i audio="Line In (Realtek High Definition Audio)" -c:a libopus -ac 1 -b:a 32000 live.caf But, if I add Icey metadata, FFmpeg throws errors: c:\apps\ffmpeg\bin\ffmpeg.exe -f dshow -i audio="Line In (Realtek High Definition Audio)" -c:a libopus -ac 1 -b:a 32000^ -ice_name "live broadcast" -ice_description "desc"^ -ice_genre "genre"...

...: (opus at 48000)->(slin at 48000 )->(slin at 12000) opus:48000 To slin:16000 : (opus at 48000)->(slin at 48000 )->(slin at 16000) opus:48000 To slin:24000 : (opus at 48000)->(slin at 48000 )->(slin at 24000) opus:48000 To slin:32000 : (opus at 48000)->(slin at 48000 )->(slin at 32000) opus:48000 To slin:44100 : (opus at 48000)->(slin at 48000 )->(slin at 44100) opus:48000 To slin:48000 : (opus at 48000)->(slin at 48000) opus:48000 To slin:96000 : (opu...

I think there is a problem. If we are coding PCM 16 mono, then, when we decode the samples we should obtain PCM 16 mono again, but wave data over 16bits is extracted from the decoder. If the input level is reduced we could loss wave information. Thanks for your help <p>Jean-Marc Valin wrote: >> we are using speex 1.0.3 under windows. We code samples with

...; Is this the right name? And the INVITE comes from the Polycom softphone with an SDP of: ... User-Agent: Polycom VV 8.0.4.4035. ... m=audio 12386 RTP/AVP 99 98 97 102 101 103 9 15 18 0 8. a=rtpmap:99 SIREN14/16000. a=fmtp:99 bitrate=48000. a=rtpmap:98 SIREN14/16000. a=fmtp:98 bitrate=32000. a=rtpmap:97 SIREN14/16000. a=fmtp:97 bitrate=24000. a=rtpmap:102 G7221/16000. a=fmtp:102 bitrate=32000. a=rtpmap:101 G7221/16000. a=fmtp:101 bitrate=24000. a=rtpmap:103 G7221/16000. a=fmtp:103 bitrate=16000. a=rtpmap:9 G722/8000. a=rtpmap:15 G728/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a...

...HDMI eld_version [0x2] CEA-861D or below edid_version [0x3] CEA-861-B, C or D manufacture_id 0x2d4c product_id 0x69b port_id 0x0 support_hdcp 0 support_ai 1 audio_sync_delay 0 speakers [0xf] FL/FR LFE FC RL/RR sad_count 4 sad0_coding_type [0x1] LPCM sad0_channels 2 sad0_rates [0x6e0] 32000 44100 48000 88200 96000 sad0_bits [0xe0000] 16 20 24 sad1_coding_type [0x1] LPCM sad1_channels 6 sad1_rates [0x6e0] 32000 44100 48000 88200 96000 sad1_bits [0xe0000] 16 20 24 sad2_coding_type [0x2] AC-3 sad2_channels 6 sad2_rates [0xe0] 32000 44100 48000 sad2_max_bitrate 640000 sad3_co...

Hello all, I'm using gentoo, with kernel 3.17.0-p1-pf and at some point, a patch was included in this branch of the gentoo kernel that broke hdmi audio. I've checked with the latest 3.19 vanilla kernel, and I still have the same problem. I cannot output multichannel sound over hdmi. After some investigations, I've narrowed down the issue to the following lines in the file

Hello, we are using speex 1.0.3 under windows. We code samples with values from -32.000 to 32.000 (signed short), but when we decode these samples the values range from -45.000 to 45.000. What is the problem? <p> Thanks --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message

I have having problems trying to take a file recorded with Monitor and convert it to MP3. When I use 'play' to play the .wav file, it sounds fine. After bladenc'ing it, it plays at lightening speed, and the voices are all high pitch. I tried using sox to resample to 32000 before encoding, but that didnt work either. Do any of you convert your .wav files to mp3? Monitor call: Monitor(wav|test) 'file' output: test.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz Sox resample: sox test.wav -r 32000 newtest.wav Bladeenc cal...

Can anyone show me how to add a log Pearson type III plot using the evdistq() command to an extreme value plot using the lmom package? Attached sample code below... Thanks in advance, Dave library(lmom) # annual maximum daily streamflows Mackenzie River mackenzieRiver = c(26600, 30300, 34000, 32000, 29200, 28300, 28600, 26400, 28300, 28800, 29000, 22100, 32900, 31800, 21600, 32100, 27000, 24800, 28000, 35000, 32000, 25000, 15800, 28800, 29900, 28000, 25600, 19700, 25700, 29500, 26800, 30000, 29500) # estimate moments moments = samlmu(mackenzieRiver, sort.data = TRUE) log.moments <- samlmu...

...onnection_type HDMI eld_version [0x2] CEA-861D or below edid_version [0x3] CEA-861-B, C or D manufacture_id 0x2d4c product_id 0x69b port_id 0x0 support_hdcp 0 support_ai 1 audio_sync_delay 0 speakers [0xf] FL/FR LFE FC RL/RR sad_count 4 sad0_coding_type [0x1] LPCM sad0_channels 2 sad0_rates [0x6e0] 32000 44100 48000 88200 96000 sad0_bits [0xe0000] 16 20 24 sad1_coding_type [0x1] LPCM sad1_channels 6 sad1_rates [0x6e0] 32000 44100 48000 88200 96000 sad1_bits [0xe0000] 16 20 24 sad2_coding_type [0x2] AC-3 sad2_channels 6 sad2_rates [0xe0] 32000 44100 48000 sad2_max_bitrate 640000 sad3_coding_type [0x...

Hi all, Asterisk is great but I'm having issues with setting up realtime for our call center, which is needed for login integration with the rest of our applications (telephonists' web interface, etc.). I have reviewed a large number of previous posts to the mailing list and the voip-info wiki to no avail. Setup is as follows: Linux 2.6.23 (gentoo) / AMD Athlon(tm) 64 Processor 3000+ /