similar to: Help!

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.

(I've already posted this message month ago, but nobody answered, may be it was not delivered to newsgroup?) Hello, All. I wrote an encoder-decoder based on example from OGG-Vorbis SDK. This encoder can encode a large amount of small WAV-files with equal parameters into one sound archive. Then I decode this sound archive back into large amount of small WAV-files. Theese files are

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

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-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 all, I routinesly rips my CDs to WAV and then convert to ogg vorbis format for use in my car and portable player. I don't usually notice anything amiss, but on the last track of Mike Oldfield's "Music of the Spheres" album ("Musica Universalis", at the very end crescendo), the converted .ogg file exhibits terrible distortion (sounds like digital clipping). This

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! 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 Aug 7, 2009, at 21:48, Didier Dambrin wrote: > FLAC doesn't preserve every chunk? I thought it did. I only gave a > quick try > but it seemed to have preserved even the most obscure chunks. > Let me check: it even seems to preserve "MIDI note associated to > marker", > which is a very unknown metadata used by SoundForge (& even defined > in a >

Hi All I have a problem with IVR scripts which consist mainly of Playback of audio files, driven from an AGI application. There are clicks every few seconds or more frequently that is audible on the remote end (PSTN), but not on the Asterisk recording of the call. If I record the remote end and compare it to the local recording, it appears to be about 5%-7% too fast - i.e. if I synchronise the

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

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, 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

Hello, I'm still working on the native .NET ogg tagging library. It already works well for non-multiplexed ogg vorbis, ogg flac and native flac files. Concerning multiplexed files, I've got two short questions: 1) Is it valid to insert an "empty" page into the stream which simply contains the binary value 0 for the page_segments counter, and which therefore contains no segment

I'm collaborating with and electronic musician to experiment on the production of music from number sequences. As I'm an R user I started playing around with the sound library and I found it very useful. However there are several things I do not understand (I'm not an expert in acustic nor audio signal treatment). The first thing I'd like to understand is: let s be a normalized

Hi Jim, First of all - thanks, turning the highpass filter off was what I needed, and the waveforms match now. But, when i did the PESQ tests again I found an interesting result : version 1.0.5 still got a slightly better average score, but the standard deviation on version 1.2 beta1 was much smaller. The cause for that is this - on some samples versions 1.0.5 and 1.2beta2 produced a single

Hi, ??Forgive my poor English, I'm trying to implement a net phone using vorbis codec, which need realtime encode and decode the stream. The problem is, I use vorbis_analysis_buffer() and vorbis_analysis_wrote() push the PCM data in, then use vorbis_analysis_blockout(&vd, &vb) to output a block, it can't output all the data. It seems that some data which length is constant, must

Hi. I'm trying to implement Ogg/Vorbis support for the MPEG4IP project. The goal is to support Ogg/Vorbis audio for MPEG-4 streaming. So far I have managed to make the encoder save Ogg packets as an Audio object in an .mp4 file. As a side effect, it can also save Ogg pages in an .ogg file playable by xmms, but that's no big deal. So what I'm doing is this. First the init part. a)

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