similar to: [jamie@audible.transient.net: Bug#160155: gapless playback]

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>

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

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

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

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 have a simplex microwave radio ethernet bridge that can transport IP network traffic in one direction only. I need to get secondary audio streams to a transmitter site and Icecast source client seems like a good idea but the docs show me that it wants to establish a connection before starting the UDP stream. Can this be tweaked to just send the UDP steam to an ip address and port number. If not

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

-----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. > >

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 Andreas, So if I understand your question correctly, what you want is really short "files" that are independent, but yet create a glitchless stream when concatenated, right. For Ogg, this can be implemented with libopusenc and chaining. It works pretty well (even for really tiny files). For WebM, I'm not sure how to handle the details at the container level, but for how to handle

It will not let me authorize my books. I saw a post on SSL issue. But, it did not mention the current wine ver or Audible ver. I am using Wine 1.3.1 Audible 5.5.0.4

Hi Andreas, 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 where the file boundaries are. Cheers, Jean-Marc On

Hi Andreas, So I encoded your file in chunks with a slightly modified version of opusenc_example and I can't hear anything wrong. Maybe there's a problem in the tools you used? I uploaded the files at: https://jmvalin.ca/misc_stuff/continuous.opus (one file) https://jmvalin.ca/misc_stuff/continuous.wav (one file, decoded) https://jmvalin.ca/misc_stuff/chained.opus (many small files)

Actually, cross-fading will work even better than what libopusenc does. The reason I did not do it is because the Ogg Opus spec provides a preskip, but no crossfade option. This means you will not be able to get standard players to play your files (which may be OK). BTW, there may be a way to implement what libopusenc does in parallel. All you'd need to do is start each parallel chunk with

Hi guys, I had the pleasure of interviewing Thomas Fuchs for "Audible Ajax", and thought that some of you may be interested to hear it: http://ajaxian.com/archives/audible-ajax-episode-12-thomas-fuchs-of-scriptac ulous Cheers, Dion Almaer Founder, Ajaxian.com http://ajaxian.com Cleaning up the web with Ajax

Hello, I'm a student now using libvorbis in my project. I aim to record voice through APIs in winmm.lib in wav buffer, and than encode the raw data in wav buffer and save as ogg file. The wav buffer is offered in a pipeline scheme, and in order to save memory, when a wav buffer is full, I start a new thread to process encoding through libvorbis. Problem is, when wav buffer is full and

I came across an interesting test sample the other day while trying to compress some of my music and trying out different encoders. You can grab the sample in question from here: http://www.animus-facticius.org drone_clip.zip is a 9 second clip of the track where the artifact is most apparent, drone_short.pac is a 1 minute clip of track. The sample is from the song "Drone" from

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