similar to: Speex 1.1.2 - Try it on ARM

Hello developers, I have a question about supported bitrates for SPEEX. Regardless of whether I specify a 10kbps or a 8kbps bitrate, the encoded .spx file is the same size and have the same PESQ value. This makes me wonder if SPEEX supports arbitrary bitrates, or only a certain set. A few more details about my configuration: I am using SPEEX encoder version 1.1.12 I am encoding a narrowband

Hello all, I'm testing the Speex codec for my diploma thesis on a BF-533 Blackfin under uCLinux (2005R3 RC3 release). I successfully compiled the Speex (1.1.11-svn) and I can encode/decode wav-files on my STAMP-board using the speexenc/speexdec sample apps. But I encountered that the decoded file sounds strange/noisy, when compiling with "--enable-blackfin-asm" +

Please consider using 16-bit 16kHz (wideband) instead. It's a huge increase in audio quality and the bitrate is still very low, especially if you take advantage of Speex features such as VBR. 8kHz seems totally inappropriate to me for desktop streaming audio, let alone 8-bit samples. Or perhaps your recording equipment is an original Sound Blaster from 1989? (Even that could record at

Hi all, I've got a 400MHz XScale-PXA255 board, and I want to stream voice from it over a network connection at 28.8baud. This calls for a capable voice encoder which can encode at about 24kbps. I was damn happy when I found Speex and said goodbye to MP3 :) However, i'm still a long way from realtime encoding using speexenc, is this possible? Using the fixed point math option in

Hello, I do test runs of speex using speexenc -n [name].wav [name].spx The wav-files (6 different speech files) hereby are sampled at 44100Hz. Most interestingly, all Frames have SubModeID 5. I haven't tracked the Assignment of the SubModeID through the source code, so I just wanted to ask, if I can assume "speexenc -n" to use SubModeID 5 always? Ciao, Bj?rn -- You will

Hello, I just signed up to this mailing-list (note: my first mailing list at all), because I'm having some problems related to speex. Let me just introduce you to what I'm doing. I am writing a short (really short) paper about VoIP techniques, especially audio codecs for speech. I pointed out basic technologies behind audio codecs; vector quantization, lpc, long-term prediction and some

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

Jean-Marc Valin wrote: > Hi, > > I just released unstable version 1.1.2 that contains more fixed-point > work. Though it's still not 100% complete, enough have been done to make > it run in real-time on ARM. In order to do that, compile with > --enable-fixed-point --enable-arm-asm. All narrowband modes work in > real-time with complexity 1 (some work with higher

Hi, I am working with libspeex api. I would like compare the size of speex file encoded with libspeex and the size of speex file encoded with speexenc. For example, I encode with speexenc (narrowband, quality 1) a wav file of 1.43 MB (17 seconds) output a spx file weight 21KB, for other hand I encode with libspeex the same wav file with same characteristics in arm processor, pocket pc and the

This is a really good point, and definitely a recurring theme on this mailing list. :) I wonder, what are some better options for handling this issue, other than to keep saying "just use 8/16/32kHz"? - Extend Speex to support other sample rates (seems unlikely..?) - Integrate a resampling algorithm into libspeex - Maintain a list of recommended resampling libraries that work well

Hi i'm developing a sort of VoIP application for my ipaq using speex... I'm still at beginning and i have many problems encoding and decoding my wav files....output is only noise! Why? I'm using Libspeex 1.1.3, Embedded VisualC++ 3.0, Ipaq 3850(206 MHz Intel® Strong ARM 32-bit RISC Processor) PocketPC 2002 (Windows CE 3.0). Libspeex is complied with the definition of

I'm a new Asterisk user. I've managed to set it up to do everything I want except sound good. Currently, Asterisk sounds considerably worse than my cell phone. I know VOIP can be _better_ than my cell phone, because I've heard Skype do it. (Using 32k iLBC, I believe.) I did an experiment with audio quality: 1) I made a recording which was pretty good. I used an iSight

Hi, Jean-Marc Valin wrote: >> I haven't tracked the Assignment of the SubModeID through the source >> code, so I just wanted to ask, if I can assume "speexenc -n" to use >> SubModeID 5 always? > > Only if you never change the bit-rate. But speexenc -n was a dumb idea > in the first place... I know ... Thats why I downsampled the WAV-File to 8kHz (I got

I'm trying to get speex to encode a bit faster, mainly by rewriting a few functions in SSE and translating the GCC __asm__ to VC __asm. There's 2 functions I'm targeting, first is vq_nbest which consumes 40% of the time at high complexity and split_cb_search_shape_sign. Which consumes just over 30%. I've split out two functions from: cb_search_precompute_energy - loop at the

Hi All, In sampleenc.c and sampledec.c, if I change the FRAME_SIZE to any other value, I get very garbled speech. Can anyone tell me if I need to set something else if I would like to change the frame size ? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20070502/35ea524b/attachment.htm

Interestingly enough.. I started playing around with preprocessing options in 1.1.6 and happened upon the denoise filter (SPEEX_PREPROCESS_SET_DENOISE). When i run the test tone using that option it is completely filtered out and I just get (complete) silence. When the test tone is intermixed with regular voice I only get the voice. So while i still don't quite understand why the test tone

Hi -- I'm looking at adding Speex decoder and encoder support to the Miles Sound System, a commercial library used by game developers on various platforms including PCs, the Mac, and game consoles. As part of my initial evaluation, I've gone ahead and downloaded the 1.0.4 and 1.1.1 Win32 binary distributions and played around encoding and decoding various files with the speexenc.exe and

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <meta content="text/html;charset=ISO-8859-1" http-equiv="Content-Type"> <title></title> </head> <body bgcolor="#ffffff" text="#000000"> <b></b>Hi,<br> <br> I also tried to build speex

Hi I have just written a small c programm which reads in speex files and outputs them as asci text numbers. This is necessary as for writing a hw testbench for decoding tests, since thy only read ascii files :-(.I can't read the bitrate properly from the header file. It's allways -1. Please see example below and attached c programm. I wan't to check for a fixed bitrate (8000).

hi, The problem associated with Line in Encoding (as discussed in the mail dated Mon May 2 05:09:11 PDT 2005 -SUB:Line in Encoding) is still persisting. I tried encoding raw input (speech) at sampling rate of 16kHz (The encoding was done on the Cirrus ARM EDB9302 in linux platform)My command line was ./speexenc --rate 16000 /dev/audio /filename.spx The encoding was fine