similar to: Call Quality Issues with IAX?

please tell me that what sip based softphone will beused with Asterisk so that i can trasmit and receive video on my LAN . --------------------------------- Yahoo! Mail Use Photomail to share photos without annoying attachments. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Situation:? * remote virtual server with very little storage (estimate: I can spare about 40G for music) * local music collection of ~80G in all sorts of formats - lossy in varying quality, some lossless too Vision: * stream my whole music collection randomized so I can listen to it anywhere Plan/Idea: * Locally transcode everything to one format that results in files that are?

> Good point. File extensions normally represent groups of related > formats. I don't propose differentiating everything (e.g. standalone > FLAC from Ogg FLAC). I do want as a minimum to tell apart these > categories: > > - Lossy audio: Vorbis, Speex. But speech is useful to distinguish > from music, so making Speex separate is not a bad idea. > - Lossless audio:

Opus or AAC will give you comparable results at reasonable bitrates (~128k). Though, I would suggest finding a way to get more storage. You could upload to Backblaze B2 or AWS S3 for pennies, if your current host won't let you upgrade. On Sat, Apr 15, 2023 at 3:36?PM D.T. <ohnonot-github at posteo.de> wrote: > Situation: > > - remote virtual server with very little

Hello, I have a large collection of audio files contains music in mp3 format, due to need to free space of hard disk, I need to reduce their size. It seems opus is the best format for this purpose, in order to have the quality of original mp3 files, currently I use ffmpeg command to convert them to FLAC and then use opusenc, the official opus encoder, to convert FLAC files to opus. By using one

Curt Sampson wrote: > > > > I've thought of doing lossy compression before on instruments, > but I'd > > > > much rather stick to lossless, at least for now. > > Honestly, stick to lossless. I mean, to the point where you can get > your exact samples back. Sure, an S900 sample is not so great quality, > but having come from the era where I did the

On Aug 8, 2009, at 23:11, Didier Dambrin wrote: > Electronic music quite often doesn't leave a computer these days. > And it > mainly consists of drums, synths & vocals/effects. Drums are often > samples > sequenced at sample (not sub-sample) accuracy, thus repeated (of > course if > the song was post-resampled, there will be sub-sample times). Good point. I

I created some test samples and transcoded to FDK AAC and libopus at fairly low bitrates - I cannot recreate what bothered me about Opus & noisy music previously. It also seems I cannot tease ffmpeg into encoding FDK's AAC with VBR. As it stands, Opus clearly wins in this scenario.* Q: Is it possible to stream in variable bitrate? * ffmpeg -i "$track" -vn -ac 2 -c:a libfdk_aac

> Steve Lhomme wrote: > > You might try to contact the makers of FruityLoops. Right now their > samples packs are in Ogg format (stored in a WAV IIRC). They like open > and free formats (they also use the LAME encoder). I've thought of doing lossy compression before on instruments, but I'd much rather stick to lossless, at least for now. There are issues to be sorted out

Has anyone found an ITSP that will relay T.38 fax to an asterisk 1.6.x instance AND do it reliably? If so, I can think of a number of locations with copper loops that could be scrapped. I'm actually quite surprised at what an underwhelming number of ITSP's that say they support T.38 (zero so far among my normal go-to companies). For locations that just want to be able to send

are any tool, script, software, that let me simulate network traffic, like trasmit for 2 hours at 100Mbs? -- Patricio Bruna http://www.linuxcenterla.com Ingeniero de Proyectos Canada # 239 Piso 5 Red Hat Certified Engineer Providencia, Santiago - CHILE Linux Center Latinoamerica Fono: +56 2 2745000, Fax : +56 22747075

Hi All, What is the status of the oggpcm project? I'm investigation solutions to the following problem: losslessly encode double-precision mutli-channel timeseries data in a format that is compatible with free (libre) internet streaming technologies and that permits diverse metadata to be encoded with the stream. flac isn't suitable because it only supports integer data, lossy

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On 15.03.2013 12:09, Erik de Castro Lopo wrote: > Marcus Johnson wrote: >> but no matter what you choose to do FLAC MUST REMAIN LOSSLESS. > > That has always been and will remain goal #1 for FLAC. FLAC stands for Free Lossless Audio Codec, so "lossy FLAC" is like "honest politician", or "fast turtle".

Hi Chris, > I just meant -O3 as an example. I'd expect all -O levels to have the > same behavior. -O3 may run passes which are more "lossy" than -O1 > does though, and I'd expect us to put the most effort into making > passes run at -O1 update debug info. I'm not really sure that you could divide passes into "lossy" and "not so lossy"

using iTunes i've noticed that AAC is very good at re-encoding own lossy sound. let's test Opus! neroaacenc.exe -q 0.75 -if 000.wav -of 001.m4a neroaacdec.exe -if 001.m4a -of aac001.wav wavdiff.exe 000.wav aac001.wav Comparing 000.wav - aac001.wav... Max diff: -17.3867dB RMS diff: -33.0851dB Mean diff: -32.4582dB opusenc.exe --bitrate 512 "000.wav" 001.opus opusdec.exe 001.opus

Hello, I am hoping that perhaps you know the answer to my car audio question, or can at least point me in the right direction. I own a 2013 Audi S4 that has the 3G-Plus Multi Media Interface (MMI) with the Bang & Olufsen Sound System. The MMI is able to playback lossy, compressed audio files (MP3, WMA and AAC), but does not have native support for FLAC, ALAC, or WMA Lossless.

Asterisk 1.2.7 RedHat 9.0 Hi, I've run into some voice degradation problems with IAX2: I frequently have calls come in on a DiD provided by an ITSP. I often have to redirect these calls back out to the PSTN (i.e. to a cell phone). When this happens, I don't want my server in the media path, I want to hand it off to my ITSP instead and let them handle both ends of the call. I've

Hi folks! Due to the fact that more and more users increasingly use MP3 < 320kbps as their source for encoding music, and publish it as flac files, I suggest that something is done in the flac encoder to possible avoid this. My idea is kinda easy/stupid, but might work; Implement a function that use a FFT to check if the input has frequencies > 16kHz, and informs the user that the file

Cool, thanks for all the great comments. I think we agree now on that the "find mp3 before encoding" feature would not be a good idea to implement in the flac core. As Brian pointed out, it might be a better idea to create a program that automatically checks if a flac might have been an mp3 source. My first suggestion was to use FFT, because I know that 128kbps mp3 have a low-pass

On Fri, Aug 24, 2012 at 4:07 PM, Villmow, Micah <Micah.Villmow at amd.com> wrote: > > >> -----Original Message----- >> From: Villmow, Micah >> Sent: Friday, August 24, 2012 2:56 PM >> To: 'Eli Friedman' >> Cc: LLVM Developers Mailing List >> Subject: RE: [LLVMdev] RFC: Supporting different sized address space >> arithmetic >>