similar to: Flac in Sox

Hello, trying to compile a project with MSVC++ and libFLAC_static.lib I have a CMakeLists.txt with this: set(CMAKE_REQUIRED_LIBRARIES ${CMAKE_REQUIRED_LIBRARIES} libFLAC_static.lib) optional(HAVE_FLAC FLAC/all.h libFLAC_static FLAC__stream_encoder_new flac) optional(STATIC_FLAC FLAC/all.h libFLAC_static FLAC__stream_encoder_new flac) CMake answer (the *.lib is always found): Looking for

On Tue, 9 Feb 2016, Larry Turnbull wrote: > One of the streams is an old time radio stream and I use ezstream to run the > prerecorded shows. > > Is there a package or some sort of way I can apply dynamic compression to > the stream? Ezstream really isn't designed for this. It's primarily meant for streaming files as they are, which is a very light-weight operation, with

Thanks Geoff, I will give it a try. I am reencoding all files to a 128K mp3, 44100 sample rate stereo stream for consistency. So yes I have the decode and encode lines in the xml file as you outlined. I will check out sox and see how it does. I am also exploring mp3gain to run across the entire library and see if getting all files to the same level will help. Larry -----Original

Hey, Sesse asked me to do a patch to the auto* set-up to allow for FLAC to be added. The patch touches configure.in and oggenc/Makefile.am It is disabled by default, use --enable-flac as an arg to configure to enable it. You might want to maybe add more checks, since FLAC's API tends to change more than is good for anyone ;) Hope that helps, Thomas -- The Dave/Dina Project : future

> Hi Daniel, > > I'm intrigued by this and wanted to try it out - but I'm wondering how > you get Asterisk to call sox at all during Voicemail()? Our server > doesn't even have sox installed, so I'm not sure how to go about > tricking Asterisk into running a different one. To do anything useful you would have to get sox installed on your server. But to get

I need to convert a bunch of .flac files to .mp3 files, but the sox and flac command tools thats in the base repo won't do that. EPEL doesn't seem to have much either. I went looking at ATRPMS but am getting database errors from their home page :( I know rpmforge used to have this stuff, but its virtually unmaintained today so I've written it off. any suggestions? I

For those of you who want to be able to decode Ogg Vorbis files into other formats, SoX (the swiss army knife of audio programs) now has Ogg Vorbis support in CVS. It can read Ogg Vorbis files and export the audio in any format SoX supports. You can also write Ogg Vorbis with it, but, because of limitations in the design of SoX, only output at 128kbps is supported. Note: Instructions on how to

Hi, Can someone please comment about the below issue [root at host0040 kaushal]# file obd-demo.mp3 obd-demo.mp3: MPEG ADTS, layer III, v1, 256 kBits, 44.1 kHz, Monaural [root at host0040 kaushal]# sox obd-demo.mp3 -e stat sox: Failed reading obd-demo.mp3: Do not understand format type: mp3 [root at host0040 kaushal]# sox -V obd-demo.mp3 -r 8000 -c 1 -t ul -w vm-intro.ulaw sox: Failed reading

Hi, I'm trying to convert a high(er) quality wav/ulaw/alaw file (captured with Record()) to a gsm file and can't get the bugger to work. The example on the page http://www.voip-info.org/wiki-Asterisk+sound+files says that: $ sox inputfile.wav -r 8000 -c 1 outputfile.gsm resample -ql Should work, but I get the error message: sox: Input and Output rates must be different to use resample

CentOS Errata and Bugfix Advisory 2010:0916 Upstream details at : https://rhn.redhat.com/errata/RHBA-2010-0916.html The following updated files have been uploaded and are currently syncing to the mirrors: ( md5sum Filename ) x86_64: 545dcd3e62e21f8bd7fd45d14ce8bd1e sox-12.18.1-1.el5_5.1.x86_64.rpm 3ccf7b508e00bdc6075dab22b1fe4732 sox-devel-12.18.1-1.el5_5.1.i386.rpm

Hi, I'm currently running an * version 1.2.13 and sox version 12.17.5. I want to upgrade sox to the newest release ( 12.18.2 ); need mp3 support. But how do I make the upgrade. Do I need to recompile asterisk afterwards? If I make a " sox -h" after a reboot I can see the new version is running but is that enough? _________________________________________________________________

CentOS Errata and Bugfix Advisory 2010:0916 Upstream details at : https://rhn.redhat.com/errata/RHBA-2010-0916.html The following updated files have been uploaded and are currently syncing to the mirrors: ( md5sum Filename ) i386: 19963a55a13ed174dad36aea4a1fa152 sox-12.18.1-1.el5_5.1.i386.rpm 54ed365f21dceaf8ff0ad601c9d6780c sox-devel-12.18.1-1.el5_5.1.i386.rpm Source:

Would anyone out there be able to confirm (or deny) if ogg audio file previewing works properly in nautilus? I can preview wave (*.wav) files and mp3 (*.mp3) files (I had to recompile sox for this to work), but not ogg files. On another note, I take it that recompilation of the sox source rpm (with libmad-devel and lame-devel rpms installed) is the "right" way of making sox

Hi all, Lance got married and is a little distracted, at the moment. :) so I built sox, samba and ipsec-tools for centos. They're available here: http://linux.duke.edu/centos/samba/ - RHSA-2004:259-23 http://linux.duke.edu/centos/sox/ - RHSA-2004:409-05 http://linux.duke.edu/centos/ipsec-tools/ - RHSA-2004:308-06 These are all for Centos-3 They're signed with my gpg key. As soon as

I believe I used to pull sox for el5 from dag's site, but it doesn't appear to be there anymore. Nor on rpmforge, atrpms, epel. Does any (trusted/reliable) repo out there have sox built for el5? I started a rebuild of fedora-12 version, but it's a pretty major dependency hell. johnny

hello, if I pipe raw audio in and a wav from sox to oggenc, sox sets the time to the maximum possible value. I don't know the wav header, could it be that it's 15 bit for the time and oggenc exits after this time (546min)? --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to

On 11.3.2013 13:05, Erik de Castro Lopo wrote: >> It includes Ben Allison's MSVC changes and JonY's MinGW changes with >> some tweaks to make both environments happy. > Please don't do that. Adding bits of other patches makes it more > difficult to evaluate and review this patch which is already difficult > to review because of its size. Sorry, I'm new at

I'm trying create a feature that allows a callers to add more speech to his recording. I think this can be done inside a dialplan, but I can't find an example of how to do this. Basically,after he records the primary message, a menu would play asking if he wants to append to this message. If yes, then he would record a temp file with the additional message and when done, I want SOX to

Does anyone know what options you need to use with "sox" to output the audio in the WAV49 format that Asterisk uses.

I recorded some live stuff with a new sound card (which defaults to 48000kHz) and because I didn't set sox right (-t wav wasn't also forced to 44100), I have an ogg that think's it's longer than it really is, and is therefore too low and slow. Is there a quick and dirty hex edit or something I can run it through to fix that, other than decoding and encoding again? Thanks,