similar to: FLAC and external file attributes

Hello, The previous message seems to confirm that I'm posting to the right place - apologies if not. I have a project where I'm streaming live software-generated PCM audio to icecast via ices 2.0 stdinpcm - this works great for ogg vorbis (thanks very much guys!) I haven't been able to find a stdin-based solution for PCM->mp3->icecast, is there one? From the docs, it looks

I have been tyring to use the preprocessor standalone by feeding it pcm data from a headerless pcm file (mono, 22050hz) and then write the result to another file. Unfortunately, all I get back is noise (first frame is set to 0, rest is noise). Frame size is set to 440. I just modified testdenoise.c slightly to read/write from a file (see below) but I can't get useful results. What am I

Greetings LLVMers, I would like to add the licensed part of AnghaBench (http://cuda.dcc.ufmg.br/angha/) as an External Test in the LLVM test-suite. We have 128,411 files with their original licenses. (http://www.dcc.ufmg.br/~fernando/coisas/c_files_with_licenses.tar.gz) Each file is a single function, that compiles as is---no dependencies are needed. For instance, the commands below work for each

Hi: Well, your main problem is that only 1 process will be able to read from the soundcard. I can think of a bit of hackery that might work though. If you're running ALSA, I believe there's a switch for arecord that will cause the left channel to go to 1 file and the right to go to another. If you got it to write these to FIFOs then maybe ices could use its wav input? hmmm, the header

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 openssh-1.2.1pre20 has been released at: http://violet.ibs.com.au/openssh/files/ This release integrates more of Andre Lucas' portability patch, Ben Taylor's utmpx patch and some cleanups and bugfixes of my own. The auth-passwd failures should be fixed, as should lastlog support on NetBSD. Since Andre Lucas' patch included platform

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 openssh-1.2.1pre20 has been released at: http://violet.ibs.com.au/openssh/files/ This release integrates more of Andre Lucas' portability patch, Ben Taylor's utmpx patch and some cleanups and bugfixes of my own. The auth-passwd failures should be fixed, as should lastlog support on NetBSD. Since Andre Lucas' patch included platform

<speex-dev at xiph.org> Hi all, I'm a begginer with DSP and i need your help and suggestions. I'm trying to use the Speex and a DSP DM642 to implement a solution with voice! I'm using the 1.2 beta 3 distribution and the TI's Code Composer Studio v3.1 simulator. I coosed speex_C64_test.pjt and modified the speex_C64_test.cmd to only use the DM642 external RAM memory. I'm

Hello :-) Recently I have switched to Creative Audigy 2 ZS [SB0350] PCI audio card from SoundBlaster Live! The new audio device although using the same kernel driver have some problems with audio/video streams - sound does not resemble original at all and the video player hangs (until audio buffer is flushed I guess). Did anyone enountered similar problem? I guess there is something wrong with

Martin Leese wrote: > Just so it is clear in my own mind. > > You are offering to add support for extended > file attributes to flac, the command-line > wrapper around libFLAC to encode and > decode .flac files. Presumably such support > would reside in libFLAC, and so libFLAC++ > and metaflac should also be updated. (Not > sure whether the various music player input

Scenario: Use the ssh-keygen utility in openssh-1.2pre17 to generate a host key Kill and restart sshd Remove the old host key from ~/.ssh/known_hosts Connect to the host using ssh. I get this: homer.ka9q.ampr.org$ ssh 199.106.106.3 who The authenticity of host '199.106.106.3' can't be established. Key fingerprint is 1024 a0:8d:17:f0:fa:a9:9f:6f:b5:d0:1c:d6:02:92:bd:5e. Are you sure

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

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

There's one vulnerability that's bugged me for some time. It applies to nearly all crypto software, including ssh. That's the swapping of sensitive info (such as keys and key equivalents) onto hard drives where they could possibly be recovered later. The Linux kernel provides a system call, mlock(), that inhibits swapping of a specified region of virtual memory. It locks it into real

Hi. When I discovered rsync, it immediately became one of my most indispensable utilities. It's a real godsend on bandwidth-limited links, especially digital cellular. It works remarkably well in the general case, but I think the algorithm could be improved for one very important special case. Many (or even most) of the updated files I transfer with rsync change only by stuff being appending

Everyone we have a few new things to give back to the asterisk community. http://bugs.digium.com/bug_view_page.php?bug_id=0002379 http://bugs.digium.com/bug_view_page.php?bug_id=0002380 http://bugs.digium.com/bug_view_page.php?bug_id=0002381 These include app_chanspy, the ability to spy on ANY bridged call taking place inside asterisk. NOT just ZAP as with ZapScan/Barge. Native format_* files

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

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

> Hello FLAC list. > > As far as I know 24 bit FLAC support is broken. It often doesn't > compress the audio at all, but instead stores the chunks as verbatim > type (although the FLAC format supports 24 bit). Perhaps this is fixed? > If so, do let me know. Hi Josh Green, 24-bit FLAC works perfectly, and has done so for years. I regularly make live recordings in

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.

The attached file contains build files for TI's Code Composer Studio (CCS) for the C54x, C55x, and C6x DSPs. I had intended to post this a couple of months ago, but it took a long time to get around to doing the little bit of cleanup required. This is a file I/O loopback application suitable for running with the CCS simulators, for evaluating memory and MIPs requirements for these