similar to: Idea for gapless tracks

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

In light of the recent study - http://www.airwindows.com/encoders/index.html - posted at slashdot, I've come to realize how little I know about this whole field. So, I'm making an appeal to the audio experts here. I want 'jukebox' quality, meaning ~0 artifacts while maintaining a good compression ratio, but not quite 'archival' (read high bitrate) as I don't mind

does ogg vorbis or any other format currently allow say, a whole cd recorded and compressed into one file, but with points stating where one track begins and ends, so you can just play certain songs out of this file, or possibly cut songs out of the file into seperate tracks. or is there any format that will do this in the future? (like ogg vorbis maybe). just wondering, thanks! Andy

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

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

-----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 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, 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

Assuming I have a WAV file that represents a continous performance, e.g. a recording of a live concert, which I want to play continously on a CD but nevertheless have track marks so that individual items can be selected some WAV splitting programs offer the ability to split the file into pieces which are an exact multiple of the CD sector size so that when recording all these files to CD with no

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

On Tue, Jan 17, 2012 at 09:44:25AM +0100, rs at noveltech.de wrote: > 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. Are you aiming for maximum compatibility with how the source CD would be played on an average CD audio

I wonder what psycho-acoustic model is used in ogg vorbis. Did they create their own model (this requires big listening tests) ? Or is it something that was already there, like gpsycho ? <p>--- >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-dev-request@xiph.org'

Hi everyone, For those who like to play with experimental stuff, I've got something new to test. If you compile with --enable-vorbis-psy , Speex uses a variant of the Vorbis psycho-acoustic model to shape the coding noise. So far, I've obtained a slight increase in quality, but I'm interested in feedback from others. This still needs a lot of tuning and has received only a minimal

Hi, I encoded some songs into 350 kbit/s with beta2 and beta3 too, and I don't know why, but it seems the b2 sounds better (makes a better quality ogg file)... The b2 contains more the b3 contains less high sounds. And with the b3, the size of file is smaller with 4-5%... Do you know this problem, or do you think so, that the sound quality is good (or better in the b3) ?? Please be carefull

Hello there folks, i was having a look through bits of the code to see if there was anything to do with pitch control, or perhaps i should call it speed control. For instance, say you took a track off a DVD which is 48khz, forgot to run it through sox to downsample it to 44khz, yet encoded it anyway. It will sound slower (therefore, the pitch has dropped). Whilst dj-ing (though i am no

Jamie, I hear what you're saying. I don't believe this *should* be a plugin's responsibility, though it sounds like with XMMS it is. But I don't know how to fix it. Probably with enough archaeology into the XMMS source and other plugins I could find out. I'll file it in the feature requests and hope someone can get to it. Josh --- Matt Zimmerman <mdz@debian.org>

hello, i noticed that when i play flac files with xmms there is a gap between tracks, even on a continuous cd. is this supposed to happen? i thought that the file format was suppose to allow continuous playback. my first thoughts were that the program i used to rip (cdparanoia) was to blame. i uncompressed the flac to wav and spliced it together only find it playing correctly. my next

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.

Alright, I'm trying this message again. I'm receiving tons of traffic on the Vorbis list, but so far my submissions have been silently rejected. I'm very concerned about a serious usability feature lacking from the current MP3 spec, and possibly from Ogg Vorbis. It has nothing to do with bitrate, but everything to do with quality. I listen to progressive rock and live albums quite