similar to: Transcode lossy to further reduced lossy to stream over Icecast

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

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

On 1/7/20 4:13 AM, Adrian Ambrożewicz wrote: > Actual variable holding content length is int64_t, but it was assigned > by explicit cast to size_t. On 32-bit systems it's a lossy conversion, > so it was replaced by casting to int64_t instead. > > Signed-off-by: Adrian Ambrożewicz <adrian.ambrozewicz@linux.intel.com> > --- >  plugins/curl/curl.c | 2 +- >  1 file

W dniu 1/9/2020 o 14:07, Richard W.M. Jones pisze: > On Wed, Jan 08, 2020 at 07:19:56AM -0600, Eric Blake wrote: >> On 1/7/20 4:13 AM, Adrian Ambrożewicz wrote: >>> Actual variable holding content length is int64_t, but it was assigned >>> by explicit cast to size_t. On 32-bit systems it's a lossy conversion, >>> so it was replaced by casting to int64_t

Thanks - I have pushed this now. Rich. -- Richard Jones, Virtualization Group, Red Hat http://people.redhat.com/~rjones Read my programming and virtualization blog: http://rwmj.wordpress.com virt-df lists disk usage of guests without needing to install any software inside the virtual machine. Supports Linux and Windows. http://people.redhat.com/~rjones/virt-df/

From a120342e0d3d20962396e6bf5bd5ac30c66b5983 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Adrian=20Ambro=C5=BCewicz?= <adrian.ambrozewicz@linux.intel.com> Date: Tue, 7 Jan 2020 11:07:08 +0100 Subject: [PATCH 1/1] Fix lossy conversion of Content-Length MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Actual variable holding content length is int64_t,

On Sat, Jan 08, 2011 at 12:54:01AM +0100, jorgen at anion.no wrote: > > 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. It would be more versatile to check if the

Any comments? > http://slashdot.org/article.pl?sid=01/07/13/1558239&mode=nocomment > > Lossy Music Formats Compared > Posted by michael on Friday July 13, @12:45PM > from the is-it-real-or-mp3 dept. > > Nicholas writes: "Today's Washington Post has an article detailing > the results of having a "a diverse panel of listeners: two members > of the

Does anybody have a 'net or hardcopy reference for basic information about the sorts of compression Vorbis uses? It sounds like there's a lot of interesting math hiding in there... Thanks, d. --- >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'

Brian Willoughby <brianw at sounds.wa.com> wrote: > On Jan 7, 2011, at 16:27, Declan Kelly wrote: ... >> As the human hearing can't really tell direction with lower >> frequencies, >> it's not as essential. This same shortcut is why most movie "surround >> sound" systems have only one sub bass channel. > > In this case, you have been misled

On Thursday, 9 January 2020 17:41:58 CET Adrian Ambrożewicz wrote: > W dniu 1/9/2020 o 14:07, Richard W.M. Jones pisze: > > On Wed, Jan 08, 2020 at 07:19:56AM -0600, Eric Blake wrote: > >> On 1/7/20 4:13 AM, Adrian Ambrożewicz wrote: > >>> Actual variable holding content length is int64_t, but it was assigned > >>> by explicit cast to size_t. On 32-bit

Hello everyone, I'm a student at the Universtiy of Delaware, and will be soon starting some research on the effects of lossy audio compression on speech sounds. I will be preforming test with both mp3 and vorbis. First of all, if I use the '--ogg' switch to lame, does lame use GPSYCHO to encode the wave, or some other psychoacoustic model (perhaps one designed for

Actual variable holding content length is int64_t, but it was assigned by explicit cast to size_t. On 32-bit systems it's a lossy conversion, so it was replaced by casting to int64_t instead. Signed-off-by: Adrian Ambro?ewicz <adrian.ambrozewicz at linux.intel.com> --- plugins/curl/curl.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/plugins/curl/curl.c

On Wed, Jan 08, 2020 at 07:19:56AM -0600, Eric Blake wrote: > On 1/7/20 4:13 AM, Adrian Ambrożewicz wrote: > >Actual variable holding content length is int64_t, but it was assigned > >by explicit cast to size_t. On 32-bit systems it's a lossy conversion, > >so it was replaced by casting to int64_t instead. > > > >Signed-off-by: Adrian Ambrożewicz

Monkey's Audio lossless compressor (currently win32 only, free but not open-source except decoder) author is thinking to implement a kind of audiophile-quality lossy compression which would filter "noise bits" that are hard to encode lossless but which are (or should be) inaudible and thus improve lossless compression (avg. 300-450kbps). I think that implementing something like this

I mention metaflac because there are a few shell scripts that use it to write RG tags in a flac music library on Linux. With support for Ebu R128 gain in metaflac (the calculation according to specification, not an external program) it would be easy to use, just change the cmd line for metaflac in the script. Now that metaflac supports sample rates higher than 48kHz this would be a good thing in

I have asterisk 1.8.2.3 + A102D Sangoma card 2 port T1. when i am starting asterisk i am getting this error on console. func_callerid.so => (Party ID related dialplan functions (Caller-ID, Connected-line, Redirecting)) == Registered application 'PrivacyManager' app_privacy.so => (Require phone number to be entered, if no CallerID sent) == Registered custom function

Is there a way to not have a transcode happen when saving voicemail? ie. the voicemail gets stored in the same format that the channel calling it is using - g711, gsm, g729, etc. ? Just playing with some really weedy processors and wanting to avoid transcodes at all costs... Gordon

Does anyone know how to do this? Just curious, ie SIP callflow A -- Asterisk -- B, RTP goes directly from A to B .. Matt

Hi guys, I don't know if this is the question ... But I have my softphones set up to use ilbc (because I found that it is better for me) and I'm trying to connect them to my provider that provides me termination through g.729. I don't have any T1 or whatever card in my server. But everytime I try to connect them they use gsm ... I don't know why ... So, that's the reason of my