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

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

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

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

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

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

On Sun, Jun 08, 2003 at 02:06:18AM +0200, Paul Seelig wrote: > This is what i get when trying to encode a WAV file: > > ------------- snip ----------------- > [pseelig]/tmp > flac -o YouGotMail.flac YouGotMail.wav > > flac 1.1.0, Copyright (C) 2000,2001,2002,2003 Josh Coalson > flac comes with ABSOLUTELY NO WARRANTY. This is free software, and you > are welcome to

Hello, Here's a script i use to join multiple PCM source files and feed them to FLAC. In the future i plan to add tag support from static CDDB (files). It would be nice if this feature could be incorporated to the native encoder. Another idea is to have flac/metaflac support joining FLAC files themselves (but this is much heavier to implement) -- and may pose issues with meta info like

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

[Woops, forgot to reply to the list ; just sent the following to Ralph] Thankyou guys. > All Speex files (.spx) are encoded in Ogg, so speexenc is all you need for that. > The standard "flac" utility will encode into Ogg with the --ogg switch. D'oh! Have been RTFM'ing an awful lot lately, but I completely missed that flac has an --ogg switch > You may also wish to

can you file a bug on sourceforge: http://sourceforge.net/tracker/?group_id=13478&atid=113478 and attach the first 100k of each file? it will help to debug since I don't have any wave files that cause this behavior. Josh --- Free Lunch <freelunch@gmail.com> wrote: > Okay.. I love flac but just had a rather serious failure that really > shakes my confidence. It resulted

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:

Hi Matt! On Sat, Jun 07, 2003 at 10:09:20PM -0400, Matt Zimmerman wrote: > Run file(1) on the .wav file and send the output. Particularly, what are the > sampling rates of the input files? > "RIFF (little-endian) data, WAVE audio, Microsoft PCM, 8 bit, mono 11025 Hz" All the WAV files i tried before have been the ones shipped with the openwebmail Debian package, showing

Okay.. I love flac but just had a rather serious failure that really shakes my confidence. It resulted in the near loss of a master audio recording. Fortunately, I have a backup. Though there may have been other cases where I have lost original material because I have been compressing a lot of originals and deleting them after doing a 'flac -t' on them. Basically, flac failed with a

Hi Kerry, On 9/16/05, Kerry Hoath <kerry@gotss.net> wrote: > There is no standard for wav64 and a .wav in its standard form can only hold > 2gb of data. In this case the WAV files were less than 2 GB. > libsndfile can handle larger .wav files and so can other audio applications > but usually nonstandard subchunk data means that the wav reader in flac > can't understand

First, Thank you for your answers. I using the following code to try simply decode a flac file and write the decoded data raw PCM file. The resulting file is just noise and pops, so is the decoded data in a different format than PCM? struct flacData { FILE *inputFile; FILE *outputFile; char *filename; }; FLAC__StreamDecoderReadStatus flac_DecoderReadCallback(const FLAC__StreamDecoder

I'm looking for sample code for using flac in my program. I want to be able to decode a flac file to PCM. Something like: bool decodeFile( const char* pathNameOfFile, voud** data ); How would I complete the above function? TIA