similar to: Speex Resampler quality

Hi, *"That's in general not very reliable. You can get PEAQ to say all sorts of silly things." Can you provide me links for any more effective tools other than PEAQ? Which is more reliable for Speex resampler? * *"strongly suspect that it's just not compensating for the delay introduced by the resampler. Because higher quality means higher delay, you'd find that PEAQ

Premkiran Mannava a ?crit : > I just built a sample application with speex resampler in linux and I > tried to resample 8K sine wave tone mono to 48k using > speex_resample_process_int. I am using a tool called EAQUAL for audio > quality. That's in general not very reliable. You can get PEAQ to say all sorts of silly things. > I find the quality of Speex resampler to be

On 2/18/08, Premkiran Mannava <loverays@gmail.com> wrote: > > Hi, > > *"That's in general not very reliable. You can get PEAQ to say all sorts > of silly things." > > Can you provide me links for any more effective tools other than PEAQ? > Which is more reliable for Speex resampler?* I can already tell you what Jean-Marc will say -- use your ears :)

Hi Jean-Marc, I use the resampler to convert various sampling frequencies to 48 kHz on my Blackfin platform (fixed-point) 48K -> 16K speex -> 48K chain does not sound very good compared to plain 16K. But the main issue is when processing loud signals, I have truncation (and not clipping/saturation) I could hear it and see it with various music and speech messages. See example.png. I also

Well, this question comes once and again. I have RC3 and Garf's RC2 tuned versions (160 and 350) in a nice graph. Nobody should be surprised that RC3 is better, but in some cases GRC2 shows better EAQUAL/bitrate results. There seems to be room for improvement in RC3 with some tweaking. See it by yourself at http://audio.sinderman.com/ Cheers, AGS.

Hi, I've noticed that the opus-tools is using a really old version of Speex's resampler code - a version that I've seen fail in the wild first-hand under low resource circumstances. I've actually submitted patches for some issues in the Speex resampler a while ago (and IIRC they were accepted): http://lists.xiph.org/pipermail/speex-dev/2009-November/007541.html ,

For those who are interested, I've run 3 EAQUAL tests on both the LAME and OGG encoders: LAME with the --alt-standard preset (with and without --nspsytune) and OGGENC (RC3) with -q 6 (my personal "sweet spot") I am somewhat unsure though about the validity of these tests, since the original and decoded files were not the same size (2 KB difference). The LAME encoded files were also

Colin, If you're using stereo audio with the resampler, there is a bug that can occur under certain situations (and maybe that's what's been happening to you). Check out: http://lists.xiph.org/pipermail/speex-dev/2009-August/007406.html With that patch, I've used the resampler extensively in many situations without any problems. John Ridges On 12/3/2010 1:00 PM,

Attached is a snapshot of work-in-progress of a FFT based resampler. At the moment it works in floating point only, and only basic quality inspection has been done. Some benchmarks comparing the filter-based resampler at Q3 with the FFT resampler with overlap = in_len / 2, using 20ms chunks of data. (-O3 -ffast-math, FFTW3, gcc 4.3.0 on x86_64) 16=>48: 59us vs 19us 16=>44.1: 204us vs

Hi, I just started to examine the DIV32_16 function (Blackfin ASM version), and wondered why the return value of the function inside 'fixed_bfin.h' is of type 'spx_word16_t', but the local variable 'res' which is returned by this function is of type 'spx_word32_t'. Is this a trick of optimization or a bug? (Same question for PDIV32_16 and MAX16, too!) best

Hi all, I have a couple of questions about the speex resampler: 1. Can a resampler state be used to process frames belonging to several audio streams? I would guess that there is information about the audio streams stored in the state so the answer is no, but it would be good to have some confirmation. 2. What is the best way to reset a resampler state? speex_resampler_reset_mem appears

Alexander Chemeris wrote: > Hi, > > Here are some questions from user point of view. :) > > On 5/29/08, Thorvald Natvig <thorvald at natvig.com> wrote: > >> I've done listening tests when converting wb_male.wav to 44.1, 48 and 8khz, >> and there aren't any obvious artifacts. I also did a 16=>16 test, and the >> results are delayed by 10ms

Hi, I tried to figure out what the problem is -- but it seems to be totally different from what I expected. My status at the moment is: - computing results for "generic" and "Blackfin ASM" versions of the DIV32_16 function are the same, there is no "algorithmic bug" - Instead, there seems some sort of memory corruption: When I comment out the DIV32_16 function

Hi Jean-Marc, Thanks for taking a look. I've added an example program in the patches that changes the rate frequently. You can run test-resample2 >test.raw and open in audacity or so to look at the spectrum etc. I've attached a before/after screenshot. In theory, depending on the current phase and the rate changes that are applied, the error can be audible as a pop when changing

________________________________ Hi David, Thank you very much for your reply. Since I need to resample the audio in the program itself, I decided to try out the resampling API in speex. But now, I have another problem. The resampled sound is very much distorted and clicks appear quite often. (I have attached the source code I used for testing it below). The test data I had was a file sampled

Hello Jean-Marc, I did some experiments with the fixed-point version of the resampler in SVN. Basically it works very well, great work! But I experienced a problem when upsampling audio data with slight clipping. This seems to cause an overflow somewhere, resulting in "cracks" in the output. I'm aware that the resampler hasn't been released yet, but I wanted to mention it. :-)

Hi, Here's an updated resampler patch against current SVN. It includes SSE and SSE2 optimizations (the latter if included by _USE_SSE2). Best regards, Thorvald -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: speex-resampler-update.diff Url: http://lists.xiph.org/pipermail/speex-dev/attachments/20080426/e055077f/attachment-0001.txt

Hi Jean-Marc, didn't get a reply to my last post (see below) -- do you have no idea what happens here? After some more tests, I disabled the DIV32_16 Blackfin optimizations and now get good quality on the Blackfin. But when I have overdrive on the input, things become very bad -- I'm not sure if this is really a filter stability issue like I wrote some weeks ago. I use the speex

Ok. I did some quality tests. First off; never do quality tests with ints. I had serious problems interpreting my results until it dawned on me that the signal differences were just 0 or 1. So, after a lot of scratching my head, these are done comparing the result from the _float versions (which is how both resamplers work internally anyway). What I did was this: Load speex_wb.wav as one

Hi all, I am using Speex in a VoIP application and everything is working great. Now I am trying to integrate the Resampler in order to convert data input, especially in the Wide Band mode (16 Khz). I have seen in the doc that for mono the ChannelID parameter should be 0, but How one should call the resampler in the case of PCM stereo data ? It is also stated that "It is also possible to