similar to: [PATCH 12]

Sorry for the delay; I have been waiting for SourceForge to come back on-line. lvqcl wrote: > 1) It seems that some programs (eac3to) write the value of > WAVEFORMATEXTENSIBLE_CHANNEL_MASK tag with uppercase 'x' (e.g. 0X3) > (see <http://forum.doom9.org/showthread.php?p=1728852#post1728852>) > > Also, from MediaInfo changelog

Maybe I don't understand you correctly, but 0000 is mono, not undefined. It says: 0000-0111 : (number of independent channels)-1 I agree that the use of WAVEFORMATEXTENSIBLE_CHANNEL_MASK should be documented. I can probably take a look at it this week or so, if no one hasn't already? 2015-07-05 13:54 GMT+02:00 lvqcl <lvqcl.mail at gmail.com>: > An issue was raised at < >

Martin Leese wrote: > Your proposed wording was: > 0000-0111 : (number of independent channels)-1. The channel order > follows SMPTE/ITU-R recommendations. The assignments are as follows: > > The channel order might not follow SMPTE/ITU-R > recommendations, so this proposed wording > seems misleading to me. But this text describes only those 4 bits in frame header. IMHO

An issue was raised at <http://forum.doom9.org/showthread.php?p=1728923#post1728923> - FLAC uses WAVEFORMATEXTENSIBLE_CHANNEL_MASK tag to describe non-standard layout, but it isn't mentioned anywhere in FLAC format. Channel assignment is described at <https://xiph.org/flac/format.html#frame_header>: "Where defined, the channel order follows SMPTE/ITU-R recommendations."

I am trying to record a call into a stereo mp3 in Asterisk 1.4, but I can't seem to get it to work correct. Could someone point me to what I need to do? I have attached what I believe are the relevant parts. [globals] ; script to be executed when monitoring has been finished MONITOR_EXEC=/usr/local/bin/2wav2mp3 ; uncomment this line if you are using Ogg Vorbis

/var/lib/asterisk/sounds/soundfile.alaw /var/lib/asterisk/sounds/soundfile.wav to go from alaw to mp3, first convert to wav, then use lame <options> /var/lib/asterisk/sounds/soundfile.wav /var/lib/asterisk/sounds/soundfile.mp3 sox looks like it can ogg/vorbis, but mine doesn't list mp3. You might fetch the source for sox and see if it can do mp3; lame is probably just as easy to obtain

Martijn van Beurden wrote: > Op 16-07-15 om 17:58 schreef lvqcl: >> But this text describes only those 4 bits in frame header. >> IMHO this sectoin should not describe WAVEFORMATEXTENSIBLE_CHANNEL_MASK, >> it should be described somewhere in vorbis comments section. > > I would propose: 0000-0111 : (number of independent channels)-1. > The channel order is defined

I have having problems trying to take a file recorded with Monitor and convert it to MP3. When I use 'play' to play the .wav file, it sounds fine. After bladenc'ing it, it plays at lightening speed, and the voices are all high pitch. I tried using sox to resample to 32000 before encoding, but that didnt work either. Do any of you convert your .wav files to mp3? Monitor call:

um, where is sox? is there any lind of user guide, list of switches/methods for oggenc for CLI-challenged? I gave it a try last night and it refused my source - was that cos it's mono? (44khz 16bit wav) anyway I fedexed the wavs up to gnu.org on a couple of cd's with their vhs.. they are on the case now. I have re-recorded the audio as stereo for CD and am going to give oggenc another

Hi, how can I convert .wav files to .WAV: # file greet.* greet.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz greet.WAV: RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz using 'sox'? Thanks -- Domenico Viggiani

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

lvqcl wrote: ... > From FLAC 1.2.1 documentation: > > 0000-0111 : (number of independent channels)-1. Where defined, the channel > order follows SMPTE/ITU-R recommendations. The assignments are as follows: > 1 channel: mono > 2 channels: left, right > 3 channels: left, right, center > 4 channels: left, right, back left, back right > 5 channels: left, right, center,

Hello list, it seems that Asterisk is unable to convert a wav-file into an alaw-file : [root at asterisk testing]# asterisk -rx "file convert testExtended2.wav testExtended2.alaw" Unable to open input file: testExtended2.wav [root at asterisk testing]# asterisk -rx "file convert testLong2.wav testLong2.alaw" Unable to open input file: testLong2.wav The wav-file is MONO,

lvqcl wrote: > Martin Leese wrote: >> Why place restrictions on which speaker a >> user can use? ... >> Finally to answer your question, for a >> single-channel file, FLAC should accept any >> one of the three masks 0x00000001 (FL), >> 0x00000002 (FR), 0x00000004 (FC), plus any >> one of the 15 other single-bit masks, plus zero. Please note that the

This is equivalent and just makes the code shorter. --- src/flac/decode.c | 24 +++++++++--------------- 1 file changed, 9 insertions(+), 15 deletions(-) diff --git a/src/flac/decode.c b/src/flac/decode.c index fa82c04..98fc430 100644 --- a/src/flac/decode.c +++ b/src/flac/decode.c @@ -333,32 +333,26 @@ FLAC__bool DecoderSession_process(DecoderSession *d) return false; /* set channel

Dan Nelson (dnelson@allantgroup.com) wrote: > I guess the problem might be oggenc's option parser, then. Given a > stereo 22050hz input file, I can't seem to get oggenc to encode less > than 22kbits. The lowest bitrate it will allow on the commandline (for > 22050hz/2ch input) is -b 30, but if you also add -M 1, it will generate > a file with an average bitrate of 22. $

Hello list, after the last time that mpg123 wen ballistic on our production system, we decided to skip mp3 playback altogether and to go for raw files. After half an hour playing with mpg123 and sox parameters in order to translate a mp3 file to a wav file that can be streamed back through * with no need for an mp3 decoder, I thought I'd post the result to the list to avoid wasting

Hey there, I'm trying to convert some call recordings from asterisk we have in .gsm format to something I can pipe through ffmpeg - wav would be good, mp3 would be amazing! I've been trying playing with sox but I don't seem to be getting too far with 1239101491.30.gsm -ql -r 64000 -t wav 1239101491.30.conv.wav resample as ffmpeg borks at it: tim at freee-meee:~/dmc/call

Juha Heinanen writes: > any idea why size of wav file doubles when it is encoded to speex and > back to wav: > > $ ls -ls testi.wav > 40 -rw-r--r-- 1 foo foo 40674 May 4 14:38 testi.wav > > $ speexenc --denoise --agc --quality 10 testi.wav testi.spx > Encoding 8000 Hz audio using narrowband mode (mono) > > $ ls -ls testi.spx > 20 -rw-r--r-- 1 foo foo 16405

lvqcl wrote: > Martin Leese wrote: >> Note that the channel order may not be defined. > > IMHO it doesn't matter in this place of documentation (which describes > default channel assignments for FLAC). Your proposed wording was: 0000-0111 : (number of independent channels)-1. The channel order follows SMPTE/ITU-R recommendations. The assignments are as follows: The