similar to: Header

Some interesting domain names have been registered recently. Looks like someone has grabbed vorbister.com, sounds like a possible Napster clone for vorbis files! Regards, Adam Whitehead Netherworld Media E-mail: adam@nwmedia.com.au Phone: +61 (889) 279 898 FAX: +61 (889) 273 889 AudioPhilez (http://www.audiophilez.com) --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project

Hi all, I have been putting a small (simple?)dedicated email server together to (ultimately) host a couple of domains. I am currently getting the configuration to work with one domain w/local mbox users and then plan to extend it to virtual users and an additional domain. Dovecot has been a problem to integrate in the configuration. The basic pieces are FC5, Postfix, Dovecot, with

>>I realize that I may have unintentionally been misleading you by leaving some information out. That error report from ogginfo keeps going, repeating the same 3 errors (as above) over and over again. It ends after a stream_truncated=true, not giving a final header_integrity=fail. (Perhaps because there's not another packet?) Then it reports a total playtime of 0. > >Yes,

Hi, Searched in Google, browsed xiph vorbis sites but found no good information about how to get the current playtime out of the granuleposition of the ogg_packet. Anyone knows if there is any 'Programming with libvorbis' documentation out there, cause on xiph's site (http://www.xiph.org/vorbis/doc/) there's nothing :( Any help? THX, Tom

The last two days have been amazing for Vorbis. Here's a list I've collected of our press: In Print: Wall Street Journal, Market section, back of the first page On the net: http://www.inside.com/story/Story_Cached/0,2770,6015,00.html http://www.news.com/news/0-1005-200-2091466.html http://news.webnoize.com/item.rs?ID=9484 http://slashdot.org/article.pl?sid=00/06/16/2228244

The last two days have been amazing for Vorbis. Here's a list I've collected of our press: In Print: Wall Street Journal, Market section, back of the first page On the net: http://www.inside.com/story/Story_Cached/0,2770,6015,00.html http://www.news.com/news/0-1005-200-2091466.html http://news.webnoize.com/item.rs?ID=9484 http://slashdot.org/article.pl?sid=00/06/16/2228244

Created corresponding unit test, and the optimization is bit exact with C function. --- silk/SigProc_FIX.h | 7 ++- silk/arm/arm_silk_map.c | 12 ++++ silk/arm/inner_prod_aligned_arm.h | 58 +++++++++++++++++++ silk/arm/inner_prod_aligned_neon_intr.c | 66 ++++++++++++++++++++++ silk/enc_API.c

Hi All- Sorry for this Off-Topic message but is there a method for unsubscribing other than sending email to vorbis-request@xiph.org. My mail server thinks vorbis-request is an invalid username and will not send the mail. Regards, Adam Whitehead E-mail: adam@nwmedia.com.au Phone: +61 (411) 241 120 FAX: +61 (889) 272 965 --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project

Does anyone have any idea how to fix the vixie crontab root exploit under Redhat 4.2? It seems to still exist. I have tried several methods to fix this but I haven''t been able to succeed. -ds ********************************************* Dark Shadow Director of Internet Services Netherworld Online Systems telnet://netherworld.net / Data: 703-383-3280 33.6 / FAX: 703-503-5945

Greetings list, I am working on a project on which we wish to use Speex with Google Automatic Speech Recognition (ASR) to transcribe Speex audio being sent on to Google ASR service and return us the text of the spoken audio in the Speex audio stream. However, Google ASR's Speex support requires the off-standard Speex-with-header-byte format, and my group cannot find any worthwhile

Greetings list, I am working on a project on which we wish to use Speex with Google Automatic Speech Recognition (ASR) to transcribe Speex audio being sent on to Google ASR service and return us the text of the spoken audio in the Speex audio stream. However, Google ASR's Speex support requires the off-standard Speex-with-header-byte format, and my group cannot find any worthwhile

Hi, I have 8 E1 PRI Lines and i have 200 phone numbers and 200 channels (25 channels per PRI). is there a utility available in Asterisk to dial out 200 numbers and run a campaign for 200 numbers concurrently and play a mp3 file ? Please suggest/guide Regards Kaushal

Hi, We are looking at the VAD implementation used in opus. We are looking at the code where speech probability is calculated based on which SNR is estimated. Below is the part of the code I am talking about. /*********************************/ /* Speech Probability Estimation */ /*********************************/ SA_Q15 = silk_sigm_Q15( silk_SMULWB( VAD_SNR_FACTOR_Q16, pSNR_dB_Q7 ) -

OEM software means no CD/DVD, no packing case, no booklets and no overhead cost! So OEM is synonym for lowest price. Buy directly from the manufacturer, pay for software ONLY and save 75-90%! Check discounts and special offers! Find software for home and office! TOP ITEMS Adobe Photoshop CS2 V9.0 $69 Macromedia Flash Prof 8 $49 Adobe Premiere 2.0 $59 MS Office

OEM software means no CD/DVD, no packing case, no booklets and no overhead cost! So OEM is synonym for lowest price. Buy directly from the manufacturer, pay for software ONLY and save 75-90%! Check discounts and special offers! Find software for home and office! TOP ITEMS Adobe Photoshop CS2 V9.0 $69 Macromedia Flash Prof 8 $49 Adobe Premiere 2.0 $59 MS Office

Yes, yes, I can reproduce it now, but only on platforms that define a 16-bit int by default (SA_Q15 is an opus_int rather than opus_int32). What system are you compiling this for? On Tue, Nov 21, 2017 at 8:34 PM, Chandrakala Madhira < chandrakala.madhira at soctronics.com> wrote: > Hi Logan, > > Please find attached the input stream we are using testing. > > Thank you, >

Sorry, long holiday weekend in America. I can say with pretty high certainty that there is an overflow occurring and it is flipping smooth_coef_Q16 to be negative when it probably shouldn't be. I had originally thought it was only an issue where it was overflowing the 15th bit but not the 16th, which might still preserve the intended value for operations that ignore the sign bit (in cases

Just for fun, I tried to reproduce such an overflow. I turned on all debug macros, assertions, and checked arithmetic and then encoded 2 hours of mixed speech/audio with these parameters: Sample rate = 48000 Channels = 1 Application = OPUS_APPLICATION_AUDIO Bitrate = 24 KB/s Force Mode = MODE_SILK_ONLY Signal Type = OPUS_SIGNAL_AUTO Complexity = 10 Frame size = 480 samples (10ms) No errors came

Hi Chandrakala, Logan, Can you confirm that the attached patch fixes the overflow problem? Koen, can you confirm the fix makes sense? Cheers, Jean-Marc On 11/27/2017 12:10 PM, Logan Stromberg wrote: > Sorry, long holiday weekend in America. > I can say with pretty high certainty that there is an overflow occurring > and it is flipping smooth_coef_Q16 to be negative when it probably

Hi, Can anyone let me know if this is a bug? Thank you, Chandrakala ----- Original Message ----- From: "Logan Stromberg" <loganstromberg at gmail.com> To: "Chandrakala Madhira" <chandrakala.madhira at soctronics.com> Cc: opus at xiph.org Sent: Wednesday, November 22, 2017 12:12:39 PM Subject: Re: [opus] Reg an issue with smoothing factor in VAD