similar to: 22 kHz version of CELT

Problems building celt-0.5.2 for blackfin. Use the following to configure: #!/bin/sh # bfinconf # Set up configuration to cross compile on blackfin ./configure \ --host=bfin \ --enable-fixed-point \ CC=bfin-uclinux-gcc \ CFLAGS='-O2 '\ '-mfast-fp '\ '-ffast-math '\ '-Wall -Dlinux '\ '-D__linux__ '\ '-fno-strict-aliasing '\

Hi I am trying to evaluate the quality of the CELT codec by using the 0.8.0 testcelt tool to encode and decode the input. I want to test different bitrates and selected the below parameters for 64, 96, 128, 196, 256kB: ./celt-0.8.0/libcelt/testcelt.exe 44100 2 256 46 $1.sw $1-64kb.sw ./celt-0.8.0/libcelt/testcelt.exe 44100 2 192 46 $1.sw $1-96kb.sw ./celt-0.8.0/libcelt/testcelt.exe 44100 2 128

Hi, I am a newbe here, so I hope I am not asking something crazy. I was looking on the Celt downloads and I noticed that the latest command line builds were for version 0.6. I know it is still early in development, and that the devs probably don't have time for developing CL versions for everything. since I know nothing about sourcecode and whatnot, I was wondering if there was a fairly simple

Jean-Marc, It appears that since testcelt reads a WAVE file from disck and passes the data directly to celt_encode, so that celt_encode's "in" buffer must be expecting little-endian formatted packets. Is this correct? Is there endiness switch somewhere? Thx MikeH -------------- next part -------------- An HTML attachment was scrubbed... URL:

From: Thorvald Natvig <slicer at users.sourceforge.net> --- celt.pc.in | 2 +- configure.ac | 2 ++ libcelt/Makefile.am | 12 ++++++------ tests/Makefile.am | 2 +- tools/Makefile.am | 4 ++-- 5 files changed, 12 insertions(+), 10 deletions(-) diff --git a/celt.pc.in b/celt.pc.in index 98cc008..67a830b 100644 --- a/celt.pc.in +++ b/celt.pc.in @@ -10,5

Dear all, Iam a student pursuing MSc degree in computer science at the University of Babylon,Iraq. My project is about the CELT. I have chosen version 0.3.2 of the software, lcc compiler, windows xp environment. One of my current problems is that when I execute the?testcelt program tool,?the output audio file can not be payed. Could you help me, please? I have tried some media players ,like

Hi, I've recently successful built and run CELT under Windows using "testcelt.c" example file. Since I'm about to port it on a embedded platform i activated the FIXED_POINT #define. I included fixed_generic.h and without other changes to the code i tried to encode and decode the same file i previously used. The output though is completely saturated i.e. it jumps from -32768 to

? There seems to be a memory violation when calling celt_mode_create(48000, 240). ? The function compute_pulse_cache() calls celt_alloc (the second one, bits=celt_alloc()) in order to allocate 343 byte, but it is using about 872 bytes at that location, which will be deleted by the following allocations. ? In this case the following encoder call crashes at first run. ? Sorry, my time is very

Hello all, after FOMS I decided to take a look at the opus library and I found that I could improve a bit the build system and cleanup the code a little bit. Most of the changes to the code has been suggested by my two tools cowstats and missingstatic (part of the ruby-elf gem if you care). HTH, Diego

Chuck, I'm doing a very similar project. I'm not an audio expert, but here's my take on the subject. 22KHz will take more space, but it is higher quality, and would probably be better if you ever decide to put them on CD (44.1KHz, stereo is the required format for CD Audio). There was a noticeable difference in the two when I tried recording at 11KHz vs 22KHz; I ended up just

I found some posts regarding this issue dating of September 2003, but no real answer. The ADPCM format supported by Asterisk (the .vox files) is 4-bit, 8 kHz. I need 4-bit, 6 kHz, which is also a widespread Dialogic format, to help migration. Is there an existing format/codec for this? If not, can I make myself a shared object in /usr/lib/asterisk/modules? Is this easy??? :-( Thanks, Yves

Could anyone quote a price for the following project. We should be able to generate a specific (say 12Khz) signal at certain intervals (calculated using a price/rate table on a mySQL database) DURING an ongoing conversation. The conversation is to be marked (start and end) with specific signals as well. This is a requirement for special hotel applications where a device counts the signals to

Hi there, I've been playing with a SNOM 360 and 190 trying to get them talk to each other using g722 with 16 kHz. However all I see in the SIP log codec negotiation is "g722/8000" which makes me believe that this is only a 8 kHz link (and that's what it sounds like). Anyone every managed to establish a 16 kHz wideband call between SNOM phones? Cheers, Philipp

Hi Peter, Have you considered resampling the raw 44.1kHz stereo source files using a program such as http://audacity.sourceforge.net/, say to 16kHz mono or 32kHz mono, and then using the wideband or ultrawideband speex modes to encode the result? Alternatively, if you want to programmatically do the resampling yourself, you could try the new resampling API in the svn head of speex, or the GPL

Hi, Does anyone know of a technical reason why FLAC cannot support 88.2 Khz files? I have a reason to uses this rate since it is easy to perform quality conversions from 24 bit 88.1 Khz master files (stored as flac files) to 16 bit 44.1 khz files for CD mastering purposes. I suppose I could Kludge the wav files so that they were half speed wav files at 44.1 khz and then hand the over to Flac, but

hi, is it possible to encode 16 bit, 22 kHz, stereo/mono WAV files to FLAC files or could there be a problem with the low frequency 22 kHz (lower then CD quality)? PS: I'm a FLAC beginner thx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/flac/attachments/20070512/63ed58f6/attachment.html

Hi!, I wondering if It's possible to change 48khz sample rate limit?, I'm Planing to encode with OGG codec a audio signal but I need that OGG Encoder works with 192khz of sample rate. It's Possible? Any Suggestions?

You asked in the Vorbis list, but your text only mentions OGG. The codec commonly used in OGG containers that is limited to 48 khz is Opus. Maybe you are trying to use the wrong codec (i.e. Opus instead of Vorbis)? Using a 44.1 khz wav file, I was able to encode a 192 khz ogg-vorbis file with the following command: $ oggenc --resample 192000 input.wav Of course, if your original material is

One thing that has always bothered me about the ogg format is the distortion of high frequency sounds - even at data rates as high 128 and 160 kbps. I find the best way around this is to resample the wav file to 48 kHz (using SoundForge 6.0) before encoding (using CDex) to ogg. It takes a while, and adds a lot of extra wear and tear on my drive, but what a difference! The result is an 80k ogg file

Hello, Following a discussion initiated on r-devel, that mentions SciViews-R and other GUIs issues for R, I would like to make comments (and would be happy if these comments would initiate interesting initiatives). A big, big problem with SciViews-R is that a part of it is written in Visual Basic 6, a M$$$$ language, not supported any more, buggy, non transposable to other platforms, etc,