similar to: Winamp Wave Out resampler (was- make lo-fi sound as good as RealAudio?

Can someone who really knows the Ogg command-line encoder, help recommend the best setting for 33.6k modem stereo music streaming? (56k doesn't count cuz many people's 56k modems don't work at a full 56k, and I want them to be able to surf CD Baby at the same time as listening. 2 minutes / 120 seconds of audio should be about 400k.) I'm at my wit's end: tried everything I

Hi, (this is my first post here) A previous thread, starting Date: Tue 19 Nov 2002 - 06:09:56 EST "[vorbis] need speech and music in one" http://www.xiph.org/archives/vorbis/200211/0142.html expressed needs similar to mine, to encode a lengthy speech at low bitrate. I did some tests initially in September then concluded in December, and I was surprised to find Vorbis to be the best

Hello, how good is the quality with very low bitrates like 16kbps mono. 16 kbps files compressed with lame sound awful, realaudio is much better. Will vorbis sound as good as realaudio with low bitrates? How fast is vorbis (in comparison to lame). Smoerk --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list,

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

Hello. As you probably know, PulseAudio uses the resampler from libspeexdsp by default. As a PulseAudio contributor, I have a feature request. As you can see from old publications [1,2] by Lennart Poettering, PulseAudio has a "timer-based scheduling" feature which is now active by default. PulseAudio attempts to use as high latency as possible (sometimes up to 2 seconds) in order

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

On 5/29/08, Thorvald Natvig <thorvald at natvig.com> wrote: > Alexander Chemeris wrote: > > 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

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, 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 and within +/- 1 (basically, rounding errors > from the FFT). Do

First of all, bufout_len = 320 is correct since that value is the number of stereo samples. Using 640 would be wrong (and no wonder that it crashes). Secondly, "speex_resampler_process_interleaved_float" has a bug that keeps it from working when it is output limited, so if you ever set "buf_len" to any value greater than 1764 the resampler will stop functioning properly

Hi Wim, On 02/04/2016 10:14 AM, Wim Taymans wrote: > 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. I'll have a closer look at your test program. > In theory, depending on the current phase and the

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 ,

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

Hi, The attached patch includes a fully working patch for the resampler, including manual SSE optimizations for the single target. I've tested up and downsampling, changing the filter quality on the fly, changing sampling speed on the fly, multichannel resampling, _float and _int versions in both floating and fixed point and resetting the resampler. They all give the same results as the

Using the following MATLAB snippet: fnames={'chirp','perfect','block','filter'} for k=1:length(fnames) fn=fnames{k}; myfile=fopen([fn, '.fl'], 'r', 'ieee-le'); x=fread(myfile, Inf, 'float32', 0, 'ieee-le'); fclose(myfile); X=specgram(x,2048,1,kaiser(2048,14)); spectrogram_floor=-96;

Frank Lorenz a ?crit : > And yes, the same "overflow" happens even when I disable Blackfin ASM > optimizations. Indeed, that shouldn't happen. Just to make sure I understand, so far there's two problems: 1) DIV32_16() in Blackfin assembly causes problems 2) The resampler overflows When you fix/workaround those two, is the encoder/decoder working correctly or are there

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 Wim, I had a quick look at your patch and I wanted to check what issue you were trying to solve. Do you have an example where changing the rate really causes bad artefacts in practice? If so, does that happen on a single change or only on continuous changes? The reason I ask is that one of the fundamental design assumptions of this resampler is that rate changes are relatively infrequent. For

Hello Jean-Marc, thank you for your answer! > I'll look into this. There's basically no overflow prevention for now. > I'll think about how to add that without affecting performance too much > (on CPUs that don't have hardware saturation). I'm open to suggestions :-) I'm not sure if I can really help, but I did a few more tests. Reducing the volume of the input