similar to: Switching Left Right Channel

> On Dec 17, 2009, at 8:50 PM, dathead2 at gmail.com wrote: > > On Dec 17, 2009, at 8:42 PM, Ron Decline <rutlecorps at gmail.com> wrote: > >> Is it possible to switch the Left / Right channel when encoding in FLAC? >> (I have some flac files with incorrect left/right channel assignment that I would like to re-encode with flac) >> _____________________________

On Dec 18, 2009, at 12:12 AM, Brian Willoughby wrote: > > On Dec 17, 2009, at 20:42, Ron Decline wrote: >> Anyway, as you and dathead2 suggested, I can just flac -d, then swap the channels on the wav with SoX, and then flac -a, I can then use metaflac to transfer the tags. Easy enough to write a script to do the work. > > > flac -a will analyze, not encode. Apart from

On Dec 17, 2009, at 11:02 PM, Brian Willoughby wrote: > > On Dec 17, 2009, at 19:00, Ron Decline wrote: >> On Dec 17, 2009, at 8:50 PM, dathead2 at gmail.com wrote: >>> On Dec 17, 2009, at 8:42 PM, Ron Decline <rutlecorps at gmail.com> wrote: >>> >>>> Is it possible to switch the Left / Right channel when encoding in FLAC? >>>> (I have

On Dec 17, 2009, at 19:00, Ron Decline wrote: > On Dec 17, 2009, at 8:50 PM, dathead2 at gmail.com wrote: >> On Dec 17, 2009, at 8:42 PM, Ron Decline <rutlecorps at gmail.com> >> wrote: >> >>> Is it possible to switch the Left / Right channel when encoding >>> in FLAC? >>> (I have some flac files with incorrect left/right channel

I have FLAC files that I would like to re-encode with --replay-gain. I have my Flac's organized in Artist/Album folders, so I should have no problem getting an album gain/peak per album. Do I need to first de-code the flac's to wav files and then "flac --replay-gain *.wav", or can this be accomplished by going to each Album directory and "flac -f --replay-gain

What is "Channels change program for processing sound"? On Dec 17, 2009, at 9:11 PM, Maciej M. wrote: > Decode to WAV. Channels change program for processing sound. > > 2009/12/18 Ron Decline <rutlecorps at gmail.com> > Is it possible to switch the Left / Right channel when encoding in FLAC? > (I have some flac files with incorrect left/right channel assignment

Hi! Playing with Opus 1.3 I converted a tone sweep with a sample rate of 96kHz (just for fun). Before I had converted that from WAV to FLAC, and to Vorbis without problems. With Opus I noticed that the file size for 48kHz and 48 kbps compared to 96kHz Vorbis at 31kbps is about double the size and it sounds even worse (than Vorbis) (there is a lot of noise in the lower frequencies when a low

I'd like to increase the speed of the Asterisk sound files. Miss Alison talks a bit slow. I can use sox to increase the speed, but then the pitch changes and she starts to sound like a chipmunk. Any audio experts out there know how I can increase the speed a little bit, and change the pitch accordingly so that she sounds ... normal? Thanks Doug.

By the way, I was delighted to find out that I could join .ogg files with cat, i.e. cat {file1.ogg} file2.ogg > outputfile.ogg <p>I successfully used this capability, together with oggenc and sox, to join two .WAV files in the following manner: <p>oggenc -q9 file1.wav ... oggenc -q9 file2.wav --- cat file1.ogg file2.ogg > outfile.ogg ox outfile.ogg outfile.wav I get

Can anyone provide the settings in Audacity to create a proper wav file without having to do additional conversion in the cli? Has to be a way to do this with less steps. ---------------------------------------------------------------------------- David Shauger Vice President Sollos Technology Solutions 678-317-9444 x5 - voice 404-886-7603 - cell 772-679-5830 - fax dave at sollos.com

I recorded some sound files using Asterisk record() app as ulaw file. I need to edit these sound files. What kind of audio editor can I use to edit these files? Gary Chen -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070710/8320ad04/attachment.htm

I am using Audacity to record some voice prompts. The .wav files I'm producing are of stellar quality. However, once I turn them into .gsm, they sound buzzy and muffled. I know that some of this comes with the territory, but I wonder if there is anyone out there who does this routinely, and who can advise me as to the MO I could use that results in the highest quality in the resulting

Does anybody have (or know of) a command line application that would: ) Eliminate pops and other random loud noises. ) Trim leading and trailing silence. ) Trim pauses exceeding x milliseconds to y milliseconds. ) Normalize what's left. I know about normalize and have figured out how to trim leading and trailing silence in sox, but I'm looking for more :) Thanks in advance,

I have a bunch of sounds that I've converted into gsm from (Indexed NMS) vox files. There's only a single utility that I've found that can read and convert vox files. My conversion process is to use this utility to convert the index vox file in to a series of wave files and then use sox to convert the wave files to gsm files. Over all this works really well, the problem is that about

On 5 June 2016 at 21:34, Frank Esposito <fpesposito at gmail.com> wrote: > Hello -- > > I have a flac file encode 92/24bit -- what tools are available to > convert these files to wave 44.1/16 bit so then I can convert to > mp3? Thanks > > -- > Frank Esposito Don't you mean 96/24? Also, which platform (*nix, Mac, Windows)? Off the top of my head: Audacity,

Hi all I have a small problem here. I asked this question on another asterisk mailing list, but nobody seemed to be able to help me there. We are running * Asterisk 1.4.17 * Libpri 1.4.3 * Zaptel 1.4.8 on a 1.6 dual core, 2GB ram and a digium TDM800P wildcard, hardware echo cancelation and a quad FXO card. We have 4 analog lines, one of which is a Cellphone line for least cost

I am, as this question will make very obvious, not exactly a guru when it comes to audio under Linux. I'd like to record some prompts using a nice microphone and my Ensoniq sound card, and then convert them to gsm/wav for use with asterisk. What Linux command would I use to a) do the recordings, and then b) convert them to the appropriate formats? I know there are

I used ChanSpy( ) recorded some test conversations. It has .raw extension. What kind of audio file is this? How can I play it? Gary -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071113/dea0f584/attachment.htm

Hello All, I have several soundfiles that are recorded ub 44100Hz, 16-bit Mono. What is the best way and right tools to use to downsample these to 8000Hz so that they can be used with Asterisk. I've tried using sox with the -r switch and Audacity on the mac and Goldwave on Windows and they all generate files that sound like a bad acid trip. I tried increasing the speed 551.25 percent after

Hi all, Here is the script i want to make - Caller call to a number to record a message - Asterisk answer and start recording message as following + User press * to start recording + Record is finished if: + User press # + OR message duration reach 60 second + Hangup How do you counting down 60s, and how to detect # (i make a test using Read() but it cant read #) Thanks in advance