similar to: Flac to Amanero

Hi, i have the big pleasure to officially announce that the matroska multimedia container project has finally left alpha status and turned into public beta status last night. By following the links below you will be able to obtain various tools to create, edit and play matroska audio and video files on your computers. Supported Operating Systems are currently Windows and Linux, but it seems

I have one theory about your playback problem: Many flac solutions load the entire flac and decompress the samples before passing them on to the player. In your case, the conversion code may be running out of memory on 24-bit files where it does not run out of memory on 16-bit files of the same duration. Another possibility is that the flac player does not support 24-bit data properly.

Matko Uibo wrote: > There is new Hi-Rez audio format avaliable - MQA - Master Quality > Authenticated > > Have anybody looked deeper into this to confirm is it some kind of drm? Hi Marko, I had never heard of MQA before, and now have only read the blog post you included a link to plus a follow-up post by the same blogger. MQA is not a system of DRM but, instead, uses a watermark

The main problem is in the Ogg layer, in my opinion. Imagine this extreme use-case with __completely made up__ numbers. This is a scenario where the server is encoding to FLAC on-the-fly from a raw PCM input, either from disk or a live stream. Let's say the FLAC block size is 1024 samples, or 23ms at 44100 Hz. Let's say each silent block compresses to 1 byte. Let's also say that

I am forwarding this bug report from the Debian bug tracking system. Please preserve the CC header when responding. To reach the submitter directly, send mail to 179764-submitter@bugs.debian.org. ----- Forwarded message from HJ <inzanekaoz@yahoo.com> ----- Date: Tue, 4 Feb 2003 03:54:36 -0800 (PST) From: HJ <inzanekaoz@yahoo.com> Resent-From: HJ <inzanekaoz@yahoo.com> To:

Has any consideration been given to support for external file attributes, at least on file systems that implement them? At the very least I'd find it VERY useful if the FLAC command simply copied any extended file attributes between source and destination files. My application is a program that records and plays raw I/Q sample streams in a software defined radio system. I/Q signal streams

The actual non made up number for 44100 is 23 seconds. :D 4096 samples, 254 packets in an ogg page. -David On Fri, Jan 7, 2011 at 11:44 PM, Ben Allison <benski at winamp.com> wrote: > The main problem is in the Ogg layer, in my opinion. > > Imagine this extreme use-case with __completely made up__ numbers. ?This > is a scenario where the server is encoding to FLAC on-the-fly

I know I've read about it under the FAQ and other various forums a dozen times, FLAC is able to encode up to 8 channels. But has anyone actually tried to do this? I ran across this nifty 7.1 wav sample courtesy of Microsoft. http://www.microsoft.com/windows/windowsmedia/howto/articles/Multichannel.aspx#link6 For some reason the channel mask was incorrectly set to 0x3f, so I manually changed

When I first install wine using the default repository in Ubuntu 9.04 and run winecfg I get the following error: wine: created the configuration directory '/home/bill/.wine' err:alsa:ALSA_CheckSetVolume Could not find 'PCM Playback Volume' element err:alsa:ALSA_CheckSetVolume Could not find 'PCM Playback Volume' element fixme:system:SetProcessDPIAware stub!

On Tue, 2008-12-23 at 00:55 +0100, Tor-Einar Jarnbjo wrote: > the Android media API (android.media) does not offer the required > capabilities to play software generated PCM samples. The first Android > SDK releases contained non-functional classes from the standard > JavaSound API (javax.sound), but these were removed in later releases. I > opened a bug regarding the JavaSound

On Sat, Apr 27, 2002 at 05:24:10PM -0400, Asheesh Laroia wrote: > XMMS-flac still breaks periodically. It just skips over FLAC files, > playing only the MP3 files in my playlist. Crash is the wrong word. If > anything, the flac plugin might be breaking; all I can tell is XMMS no > longer plays the flac files. I have to kill and re-run XMMS. This > happens every few hours of

On Jan 12, 2013, at 05:30, Martijn van Beurden wrote: > On 12-01-13 08:23, pyth.flac-dev.5.pyt at spamgourmet.com wrote: >> I seem to recall that changes in the second number indicated a minor >> change in the *format* of the file itself (for example, 1.1.x to >> 1.2.x >> introduced a new rice coding option used for 24-bit files). > > Well, the only change in

Hello, I am trying to connect a instance of ices2 to icecast2. It seems that the connection is done well but I do not know how to take the microphone input to send it to the server. Any help is highly appreciated. Manuel The information is this: Part of my ices.xml: <input> <module>alsa</module> <param name="rate">44100</param> <param

hi, I was wondering how perfect silence frames (no audio at all) in a PCM stream are encoded by the flac encoder. Suppose I have a PCM audio file with some perfect silence frames in the beginning and at the end of the audio recording and suppose I have the same audio file but without the silence frames in the beginning and at the end (so the second file is identical except for the silence frames

Hi, I'm seeing a problem when calling into my Asterisk server from a T100P with PRI signalling. I'm hoping someone has seen this as well, however, it's a little hard to explain. When I'm navigating through Asterisk on an inbound call from the PSTN, I'm getting a random 'noise' during the playback of sound files. The best way I can describe this noise, is imagining a

Hi, I've been having a little problem with my asterisk servers, I have 4 identical asterisk servers setup (same hardware, same OS, same config). Once in a while (once or twice a day) one of the server crashes on the cron job reload. But I realized this only happens on 3 of the 4 servers. Tried to spot the difference between that one server that wasn't crashing. The difference I found was

I think the confusion stems partly from the ambiguity of the term "audio disc". Strictly speaking, an "audio disc" would be a Red Book CD-DA disc, which uses a specific format for coding audio. These are the discs that you can buy at the music store and that any music CD player can play. These discs have music in straight linear PCM only. On the other hand, a "data

it's really not complicated I think: only api changes to write on any position in the file if that's not possible already with existing function. I'm not sure if decoding can have multi-core support: you need an api for writing pcm files in different parts then and this is maybe more difficult to check if it's valid pcm data since the decoder can only check for valid flac streams

----- Forwarded message from Adam Ottley <aottley@shaw.ca> ----- Date: Wed, 30 Jul 2003 19:48:16 -0600 From: Adam Ottley <aottley@shaw.ca> Resent-From: Adam Ottley <aottley@shaw.ca> To: Debian Bug Tracking System <submit@bugs.debian.org> Subject: Bug#203581: xmms-flac: Plugin misplays certain valid files Package: xmms-flac Version: 1.1.0-9 Severity: normal Certain

So some research and experimentation I think I found the problem, but I'm not sure how to go about fixing it. If I import one of the wav files into Audacity then export them as "16 bit PCM" then they encode to flac and play fine. The files bitrates are currently at 24. If I export them from audacity as a 24 or 32 (anything higher than 16), flac fails on me. With a bitrate of 24, the