similar to: 1.1.6 is out

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On 03.02.2012 1:11, Declan Kelly wrote: > Many people who use FLAC to archive entire CDs (as opposed to > "albums" of tracks that may or may not be on the same CD) will rip > the entire disc and store it in a single FLAC file, with the CUE > sheet either as a separate file, embedded in the FLAC metadata, or > both. > >

Hi, i?m not part of the FLAC project, but i have a question regarding FLAC and Gapless support I hope, I get an answer from some of you ;-) We are currently try to add Gapless support on our device If we rip an CD with our device, we can find out, that one track follow after another so we can recognize, that the tracks are gapless or not. But how can we find that out on already existing

I am forwarding your request to the FLAC development mailing list. ----- Forwarded message from Jamie Heilman <jamie@audible.transient.net> ----- Date: Sun, 8 Sep 2002 16:13:32 -0700 From: Jamie Heilman <jamie@audible.transient.net> Resent-From: Jamie Heilman <jamie@audible.transient.net> To: submit@bugs.debian.org Subject: Bug#160155: gapless playback Package: xmms-flac

On 2015-02-20 7:25 AM, Daniel James wrote: > I don't understand why this has to be so limited, because the basic > idea, as I understand it, is to extend the .m3u playlist format so that > stream listeners can automatically choose alternative sources for the > same content. That could be implemented in a codec-agnostic way. Stitching together compressed media streams for gapless

We all know that currently Vorbis is the only codec out there that doesn't add any silence to the decoded file, so when you decode your .ogg file the resulting .wav will be exactly the same length as the original one. This is very useful when encoding gapless songs (like live concerts...). But since Vorbis is a lossy codec, even though there is no silence added, sometimes you can still hear a

On Mon, Sep 26, 2005 at 12:41:00PM +0200, Michael Smith wrote: > On 9/23/05, Paul Martin <pm@nowster.zetnet.co.uk> wrote: > > Mike... did you look into this? I posted a waveform from a file > > generated by ogg123 dumping to a wav (so that audio driver problems were > > eliminated). > > > > It looks to me like ices2 is restarting the vorbis encoder in a >

Hi, Right now, I'm thinking of adding a speex_lib_ctl() call that would support SPEEX_GET_VERSION (and return a string) or SPEEX_GET_MAJOR_VERSION and SPEEX_GET_MINOR_VERSION (and return ints). I'm open to other suggestions though. If there's anything you'd like to see in the API for 1.2, say it now. ...and no, I won't add a speex_do_all_the_work_for_me() call :)

Mike... did you look into this? I posted a waveform from a file generated by ogg123 dumping to a wav (so that audio driver problems were eliminated). It looks to me like ices2 is restarting the vorbis encoder in a non-gapless way when it stops and starts the encoding (to change the stream id and update the metadata). Because I'm doing crossfades from one track to another, this shows up.

On Sat, 2015-10-24 at 22:16 -0700, Thomas Daede wrote: > Everything above 96kbps on that table is speculative, as the highest > multi-participant listening testing done was at 96kbps. Here's the > results from that test, if you're curious: > > http://listening-test.coresv.net/results.htm > > As you can see, at that rate Opus ranged from slightly perceptible to >

Richard Polton (Richard.Polton@morganstanley.com) wrote: > Is there a way, in either cdparanoia or ogg vorbis, to encode a live > album which has tracks but no gaps between them? Is the only present > solution to concatenate all the extracted wav files and then to encode > it as a single ogg? You can't cat wav files together. But you can tell cdparanoia to rip the whole CD into

Hi, I am using speex 1.2beta2 on a narrowband 16-bit, 8khz system that has a severe program space problem and will not fit speex in its normal operation. In an attempt to shrink speex I placed a breakpoint in every function and ran a decode and encode and removed the breakpoints that I hit. in the functions that had a breakpoint that I didn't hit I commented out those functions (as well as

>> Oh, your table looked fine. I think it's mine that was wrong for the >> "null mode" (which nobody ever uses I guess!). > > I'm not sure, but I guess it may be used in VBR or DTX mode? > At least that's why I kept it. Of course it *can* be used (and needs fixing). It just seems few people did :-) > So, what's the next step towards

Hi Jean-Mark, thank you for your answer! Yes, you understood my question correctly. I was just about to compose a reply to my original question, where I described how I solved my problem. As you've already suggested, I've switched to Ogg/Opus, which is better supported, but does not work with the Media Source Extensions. I'll have a look whether disabling prediction will help with

Hello Bernard, Friday, December 13, 2002, 7:22:54 AM, you wrote: Bernard> I've one small request - an option on speexenc that allows you to Bernard> specify a speex file to append to, allowing you to concatenate Bernard> streams without losing quality by decoding & encoding. Ideally, it Bernard> would: But you can `cat speex1.ogg speex2.ogg> unionspeex.og` and still have

Hi all, I finally understand how lipopusenc is capable of producing chainable Opus files (in contrast to my program), and I managed to successfully implement the method [1]. Essentially, the last frame of a file is marked as a "keyframe" by disabling prediction for this frame in libopus. This encoded keyframe frame is then copied in verbatim to the next file, with the pre-skip set to

Hi Jean-Marc (and everyone else who replied), > Considering you're switching to Ogg, I think you should give libopusenc> a try. It does a really good job at getting rid of *all* discontinuities> -- to the point where you can chop a song into files less than one> millisecond each and it still sounds good. It's also pretty simple to> use. You just feed it audio and tell it

Hi Jean-Marc, thank you for looking into this. I'm using the current release version of libopus (1.2.1) and version 0.1.10 of opus-tools, both built from source. Libopusenc is the current git master. Indeed I cannot hear any glitches in continuous.opus, continuous.wav, and chained.wav. chained.opus decodes just fine with opusdec as well, but doesn't play properly in any standard audio

Hi list, I am just wondering if anyone knows a way to stream with automated crossfading under Linux. I know that you can stream to icecast from windows using something like otsjuke, or WinAmp with the SQR crossfading plugin, but am not aware of anything for Linux that will allow me to stream with gapless output. Any suggestions greatly appreciated as always.

Hi! Short version of my question: How to produce Opus frames which can be safely concatenated and how to embed them into a WebM file? Long version: I'm currently implementing a web-based audio player which streams audio as opus/WebM using the HTML5 media source extensions. Currently, the server decodes a set of input files to a fixed RAW audio format (stereo, 48000 kHz) and encodes the

for the bits init I am using speex_bits_set_bit_buffer and I don't use the write to or read from because the data is already in the buffer I am reading from and I am writing to the final buffer so I don't need to move arrays around. what part is the vocoder part of the decode? Thanks for your help! -Mike >>> Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> 08/06/07