search for: wavs

Hi, I'm using the FLAC command line tool to encode my WAV files in some directory to FLAC files. Now I noticed sometimes the output of the tool is mostly like this: flac -V --best "14 The Hose - Time Warp (Old Skull Mix).wav" flac 1.1.4, Copyright (C) 2000,2001,2002,2003,2004,2005,2006,2007 Josh Coalson flac comes with ABSOLUTELY NO WARRANTY. This is free software, and you are

i hope you can help me. The game's site is http://withinadeepforest.ni2.se/index.php? here's the terminal output from the install (from install binary) and the run. the install: $ wine install_wadf_11.exe fixme:advapi:CheckTokenMembership ((nil) 0x7ff18088 0x7fc8fe1c) stub! fixme:sfc:SfcIsFileProtected ((nil), L"C:\\Program Files\\Within a Deep Forest\\unins000.exe") stub

Hello. I just added 24 bit support to CRAM and noticed that 24 bit data isn't compressing much, if at all. I saw a recent thread on this list concerning this same issue. http://lists.xiph.org/pipermail/flac-dev/2006-July/001907.html I ran some tests using the command line 'flac' program and also experienced rather poor results. When doing a 'flac -a' many of the files have

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

Hello All, I am using monitor with soxmix, however the quality seems somewhat low after sox converts to mp3. Does anyone know a way to get a higher quality file? Some of my lines are coming in on isdn. Regards, Greg

using iTunes i've noticed that AAC is very good at re-encoding own lossy sound. let's test Opus! neroaacenc.exe -q 0.75 -if 000.wav -of 001.m4a neroaacdec.exe -if 001.m4a -of aac001.wav wavdiff.exe 000.wav aac001.wav Comparing 000.wav - aac001.wav... Max diff: -17.3867dB RMS diff: -33.0851dB Mean diff: -32.4582dB opusenc.exe --bitrate 512 "000.wav" 001.opus opusdec.exe 001.opus

Hey Users, I am using record application to record MeetMe conf. but look like its creating individual files for every channel. What applucation is best to record MeetMe conf ? ~ # ls -l /var/spool/asterisk/monitor/ total 489220 -rw-r--r-- 1 asterisk asterisk 44 Feb 16 08:42 8881-conf-20110216-084224.wav -rw-r--r-- 1 asterisk asterisk 1858284 Feb 16 13:05 8881-conf-20110216-130321.wav

A friend of mine made the following comment in a discussion I had with him that on a website we adminster we should offer a) WAV or maybe shorten files b) Ogg as a decent reference lossy encoded version He's been trying to convince me that we should offer MP3 (in lieu of WAV) and possibly Ogg. The audio files are primarily vocals I am not a physics guy but his statements don't

Hi All, Need Help ! I am interested testing opus codec encoding decoding qaulity. for this have complied opus code codec from souce. After compiling i got opus_demo app. for Encoding i followed below steps: 1) Reference file used music_orig.wav (http://www.opus-codec.org/examples/samples/music_orig.wav) Number of samples : 4358219 (90.8 s) 2015-04-13 10:40:07 UTC Sampling

Hi all, I have to play a alaw file with .wav ext. How can I do this? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100721/de46328f/attachment.htm

Dear sir, I've a problem that the .wav file couldn't play after calling "sampleenc male.wav|sampledec male_speex_15.wav".I found that the new file male_speex_15.wav is smaller than the original file in size.I implemented the test on Linux system.The original file male.wav is 96044 bytes,while the new file male_speex_15.wav is 96000 bytes.I'm eager to know the reason.Thankyou!

Hi, I am trying to implement a web application that allows the user to playback WAV files inside a web browser. __________ Background __________ - The file server stores audio files as GSM 6.10 or TrueSpeech encoded WAV files. - The web server (most probably ASP.NET on IIS) offers the client the ability to search audio files on the file server (a different machine). - The client will be

...-keep-foreign-metadata to keep) 10_A.wav: WARNING: skipping unknown sub-chunk 'umid' (use --keep-foreign-metadata to keep) 10_A.wav: Verify OK, wrote 168060055 bytes, ratio=0.624 As a final random test, I attempted to split one of the wav files (my ultimate goal is split flac files) using wavsplit. That resulted in the following output/error. Channels: 1 Samplerate: 96000Hz Samplebits: 24 Databytes: 269503836 Split Hours Mins Seconds Bytes % Bad file format Thank you in advance for any help anyone can provide me! I've never used a mailing list before....

Hello Guys, I am trying to convert files that are .wac to mp3 after mixmonitor command is called but it doesnt execute the command, I tried the command in terminal it worked, any help please ... below is my dial plan exten=6500,n,Set(MIXMONITOR_EXEC=&& nice -n 19 /usr/local/bin/lame -b 8 -t -F -m m --bitwidth 8 --quiet "/var/spool/asterisk/monitor/${CALLFILENAME}.wav"

Please compare samples: http://www.ivo.pl/ivo2/sound/krwawa_org.wav http://www.ivo.pl/ivo2/sound/krwawa_fixed.wav http://www.ivo.pl/ivo2/sound/krwawa_float.wav krwawa_org.wav is original file. Both output files was created using same program by compressing from wav to speex, then decompressing from speex to wav (using speex_encode_int and speex_decode_int). Only difference was used speex

...#39;ve managed to convert using both of those programs. I'm only running into difficulty when it comes to these large WAV files. By "Does not work" I mean that they do not play, and instead I receive the errors I mentioned in my original post. I wasn't actually intending to use wavsplit to split the flac files, it was just something I had during some testing for how to split these files. I'm actually going to be splitting during the conversion since the flac tool supports specifying time marks. Is there something that could be wrong with the WAV files that prevent them f...

Hi, I am running a service where I play full songs but MP3 files kept on crashing my server. I resorted to wav but the quality is really poor after converting..or even sometimes not audible at all! Do you guys know of a better way I can convert mp3 to wav and restore quality? Below is the script I am using, I also tried the steps at

this is not really a development question, but i didn't find another mailing list to ask it. 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

Hi there. First of all, I'd like to congratulate to developers for making a free specification and LGPLed decoder/encoder. :-). I did some test, ripping 17 tracks of my CD and converting them to both MP3 and OGG format. MP3 encoder/player was Lame 3.70 OGG encoder/player was from a Saturday nightly tgz package. ------------------------------------ MP3: LAME-3.70 mpg123-0.59r-4 MP3: lame -S

Hi there. First of all, I'd like to congratulate to developers for making a free specification and LGPLed decoder/encoder. :-). I did some test, ripping 17 tracks of my CD and converting them to both MP3 and OGG format. MP3 encoder/player was Lame 3.70 OGG encoder/player was from a Saturday nightly tgz package. ------------------------------------ MP3: LAME-3.70 mpg123-0.59r-4 MP3: lame -S