similar to: Patch to support timing of Speex stream

Hi all, Please find below the -01 update to draft-herlein-speex-rtp-profile, as submitted to the IETF. Regards Phil <p>-------------------8<-----------------------------------8<--------------------- <p><p>Internet Engineering Task Force Greg Herlein Internet Draft Jean-Marc Valin

Hello, What's the purpose of the 'sr' sdp parameter ? The sample rate is already given in the a=rtpmap line ? Simon Le dim 29/06/2003 à 12:12, philkerr@elec.gla.ac.uk a écrit : > Hi all, > > Please find below an updated Speex Internet Draft document. > > It would be good if we could book some time for discussion on Speex at the IETF > meeting in Vienna (scheduled

Hi all, Please find below an updated Speex Internet Draft document. It would be good if we could book some time for discussion on Speex at the IETF meeting in Vienna (scheduled for 14th July). The cutoff for submission is 9:00am EDT, (GMT -04:00), 30th June. Comments and feedback welcomed! Regards Phil

Hello, On the announcement page for Icecast 2.3 it mentions "Streaming support for ogg speex, ogg flac, ogg midi". I've googled around and not found any free clients that support streaming speex. I looked at svn for libshout2 and it doesn't seem to support speex and skimming through the list archives I only see mention of speex last February, which seemed to indicate it would

As you know, Winamp relies on the file extension to know what input plugin to utilize for the audio decoding. The input plugin for .ogg files (in_vorbis.dll) needs to understand the other ogg container formats but it doesn't as it is a vorbis only decoder. So the only solution is for Ogg Speex to have a different extension so this can be appended to the mountpoint, and an input plugin

Jean-Marc Valin wrote: > Hi, > > I just released unstable version 1.1.2 that contains more fixed-point > work. Though it's still not 100% complete, enough have been done to make > it run in real-time on ARM. In order to do that, compile with > --enable-fixed-point --enable-arm-asm. All narrowband modes work in > real-time with complexity 1 (some work with higher

Hi All, Please find below an updated Vorbis-RTP Internet Draft document for review and discussion at the Xiph IRC meeting on Saturday. The changes in this version have been: Codebook caching mechanism Expanded SDP parameters Expanded MIME section Expanded introduction Packet loss section Minor tweaks and clarity changes to text There are probably some minor tweaks to the formatting needed

Hi all, I haven't found any root cause or workaround for this yet. Can any one help me in underatanding the issue? Regards, Jeevan. On Dec 21, 2017 8:20 PM, "Jeevan Patnaik" <g1patnaik at gmail.com> wrote: > Hi, > > > After running rm -rf on a directory, the files under it got deleted, but > the directory was not deleted and was showing stale file handle

Hi, After running rm -rf on a directory, the files under it got deleted, but the directory was not deleted and was showing stale file handle error. After 18 minutes, I'm able to delete the directory. So could anyone help me in knowing what could have happened or when in general I get such errors. The following is NFS log: [2017-12-21 13:56:01.592256] I [MSGID: 108019]

An ESTALE error usually means the gfid could not be found. Does repeating the "rm -rf" delete the directory? Regards, Nithya On 3 January 2018 at 12:16, Jeevan Patnaik <g1patnaik at gmail.com> wrote: > Hi all, > > I haven't found any root cause or workaround for this yet. Can any one > help me in underatanding the issue? > > Regards, > Jeevan. > >

Hi all, Lots of bugs and support requests in the sourceforge tracker are related to the FLAC frontend, which is bundled with FLAC for Windows with the installer. However, I couldn't find it in git. I really thought the FLAC frontend was part of the project, but it seems it is some piece of freeware last updated 10 years ago: http://members.home.nl/w.speek/index.htm Is this true? Anyone

Hi All. I am seeing the following unexpected behavior with OPUS_SET_MAX_BANDWIDTH. I expect that setting this to OPUS_BANDWIDTH_NARROWBAND would give similar results to passing an 8Khz sample rate stream, but OPUS_SET_MAX_BANDWIDTH has almost no effect with any settings. My test data has 4Khz bandwidth. I am testing the opus encoder (latest versions) with the following opus_encoder_ctl

On Fri, Feb 13, 2015 at 7:12 AM, Lamar Owen <lowen at pari.edu> wrote: > On 02/13/2015 05:41 AM, James Hogarth wrote: > > This is also why the Orange Book and its Rainbow kin exist (Orange Book = > 5200.28-STD, aka DoD Trusted Computer System Evaluation Criteria). > Should anyone care to learn from the Rainbow Books, they are available from the United States of America (USA)

We want to record inbound channels at 16kHz, but send only 8kHz to our peers. I've set our default profile in sip.conf to disallow all but g722, and the peers disallow all but ulaw. We have a proxy in front of Asterisk that is configured to disallow all but G722 also. My test calls show inbound to the proxy is recorded at 16kHz, inbound in Asterisk is only 8kHz, and the peers receive 8kHz. So

Hy everybody, I'm working in an italian company who wan't start using asterisk. My problem: 1. What kind of hardware I need for make a PBX who speek with 3/4 ISDN BRI line and internal use a SIP VoIP telephone? 2. I can install ASTERISK and HYLAFAX on the same machine ? Thanks and sorry for my "english mistakes". Lorenzo Soncini Technoservice S.a.s. 38057 Pergine Valsugana

When using Winamp's diskwriter with the ogg plug-in v1.13 the resulting wav files are only named 01, 02, etc. So the original names are lost. I hope Peter can fix this. Speek --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'vorbis-request@xiph.org' containing only the word

oddcast currently doesn't do speex, or ogg-speex. One of the biggest problems currently with speex (and flac too) is client support. I've just recently added ogg-flac support to oddcast (not yet available) and in my testing found that client support for this format is horrible. On windows, Winamp and Foobar choke on it (most likely assuming that if it's ogg, it must be vorbis)

That's pretty bad. Both DirectSoundCapture and WinMM are capable of recording at 16kHz. I don't know why OpenAL would be incapable of handling it. It's not like it's at all rare or new. I would try 16000 and see if it works. Maybe the docs are wrong? Note that one option to retain high quality is to capture at a higher rate and then downsample using a resampling

v2: Check for PFIFO, don't pause if it's not yet running. This should fix reclocking on boot Signed-off-by: Roy Spliet <rspliet at eclipso.eu> --- drivers/gpu/drm/nouveau/Makefile | 1 + drivers/gpu/drm/nouveau/core/engine/device/nv50.c | 4 +- .../gpu/drm/nouveau/core/include/subdev/clock.h | 4 + drivers/gpu/drm/nouveau/core/subdev/clock/nvaa.c | 439