similar to: Resampler set_rate improvements

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 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

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

06.02.2016 01:12, Tristan Matthews ?????: > On Thu, Feb 4, 2016 at 12:31 PM, Alexander E. Patrakov > <patrakov at gmail.com> wrote: >> 04.02.2016 22:05, Jean-Marc Valin ?????: >>> I can see how that would cause multiple rate changes. At the same time, >>> since this would be using the interpolated code (unless you're really >>> lucky with the rate),

04.02.2016 22:05, Jean-Marc Valin ?????: > I can see how that would cause multiple rate changes. At the same time, > since this would be using the interpolated code (unless you're really > lucky with the rate), the cost shouldn't be too high. BTW, do you know > how often the rate gets updated? The rate is, by default, updated every 10 seconds. And sometimes (especially with

I recently stumbled upon <speex/speex_resampler.h> and made a chain of discoveries: - "http://speex.org/downloads" some links are broken ("rarewares" and git) - there is some (minor) development (whitespace and more) of the "dead" Speex codec ... will there be a release? (I don't really need such myself) - there is some (more useful?) development of the

Hi, another question on the same topic Speex resampler at 44.1kHz seems to be very CPU intensive on Android (even more than the Opus encoder) While Speex at 48kHz is just fine. I wonder any alternate solutions or ideas ? Improve it, look for alternate solution ... I am guessing the NEON optimization are still used for both, etc. On Thu, Apr 2, 2015 at 4:46 PM, Jean-Marc Valin <jmvalin at

Thanks ... > Those links seem to work here. 2 links "http://rarewares.hydrogenaudio.org" redirect to "https://hydrogenaud.io" ... no "rarewares" subdomain "http://git.xiph.org/speex.git" maybe better "http://git.xiph.org/?p=speex.git;a=summary" "http://git.xiph.org/speexdsp.git" maybe better

Hi, is there any way to tell the decoder the output sampling Fz we want ? opus_decoder_create = Sampling rate of input signal (Hz) Considering this example (VoIP-out from WebRTC/RTP) MICROPHONE(44.1/48kHz) >> [encoder created at 48kHz but with internalSampleRate set to 8kHz]>> INTERNET >> [decoder(created with 48kHz)] >> 48kHz(?) >> G.711(8kHz) This leaves us with

Hi: I'm using speex to perform echo cancellation in Windows. I'm aware of the problem about out of sync clocks in record and play sample rates in usual sound cards . In order to have an idea of how good is my echo cancelation working I would like to know if there is any #define thing i can pass to speex_echo_ctl to get the actual level of echo cancellation. If not, how can i extract that

Hi FLAC hackers, I just thought you might want to know that we are making FLAC decode and encode plugins for GStreamer. (http://www.gstreamer.net). The decoder plugin is already in CVS and the encoder plugin is underway, the plugins are made by one of our lead developers Wim Taymans. FLAC decoder CVS entry: (http://www.geocrawler.com/lists/3/SourceForge/1535/0/5982915/) Thanks for the great

Hello, in case somebody wants to test Theora without compilation, feel free to use konvalo.org's setup. It includes mplayer with theora from svn 2004-09-21 with Wim Taymans' mmx-patches from August 2004 applied (there are mplayer and mplayer.theora-mmx, the latter with decoder mmx-patches from VP3HoSwiYO) and a theora-encoder script (a bit slower than ffmpeg2theora as it runs several

Hello, in case somebody wants to test Theora without compilation, feel free to use konvalo.org's setup. It includes mplayer with theora from svn 2004-09-21 with Wim Taymans' mmx-patches from August 2004 applied (there are mplayer and mplayer.theora-mmx, the latter with decoder mmx-patches from VP3HoSwiYO) and a theora-encoder script (a bit slower than ffmpeg2theora as it runs several

----- Original Message ----- From: "Alexander Chemeris" <Alexander.Chemeris at sipez.com> To: "Vincent Burel" <vincent.burel at vb-audio.com> Cc: "Conrad Parker" <conrad at metadecks.org>; <speex-dev at xiph.org>; "Jean-Marc Valin" <jean-marc.valin at usherbrooke.ca> Sent: Thursday, November 13, 2008 11:31 PM Subject: Re:

Hi I am using speex for cancelling echo.but i am not succeeded. I need echo cancellation in urgent.Please help m. Thanks in advance Devi

Hi, When doing a live ogg+theora stream on our server we ran into the following problem: When a client connects we send out the theora header pages followed by the actual live stream pages. The problem is that the first ogg data page does not have the right page number (2), which causes libogg to not want to decode the first _packet_ of that page. Since that first packet happens to contain the

I've filed a bug for aarch64 https://github.com/xiph/speexdsp/issues/7 and provided the port in a fork with a pull request. We need someone to review/merge in the pull request? It provides the source code, but my testing was under Android builds, so there would be some configure changes needed to build it stand alone. On Tue, Apr 19, 2016 at 4:32 PM, Frank Barchard <fbarchard at

Hi, 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. I find the quality of Speex resampler to be decreasing when I increase the quality q of the resampler init function. Can some one give me pointers regarding this?? As per the API, if the

Hi all, I build successfully with speex-1.2rc2. And with speexdsp 1.2rc3, I build with i386, X86_64, armv7 and armv7s all passed. But when I build for ARM64 (for iPhone 6), it failed with: /Applications/Xcode.app/Contents/Developer/usr/bin/make all-recursive Making all in libspeexdsp CC preprocess.lo CC jitter.lo CC mdf.lo CC fftwrap.lo CC

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 ,