similar to: Some queries

Hi, I am currently working on a thin-client "low-cpu" usage telephone application. The current setup is that you can make phonecalls using an alsa supported usb headset and a smartlink based usb modem (a sweex usb modem, which -in large quantities- goes for 17.95 euro's) for analog lines. Unfortunately, it is really made as a modem and not for audio purposes, so, there is a pretty

Hello, I am testing speex 1.1.6's echo canceller. I am using testecho.c, with a few modifications to get it to run on Windows. My problem is that I am unable to get the echo cancellation to work correctly. I am working on an audio conferencing software, and one issue we have is sometimes the microphone picks up what is being played through the headset, resulting in an echo of the other

Hi, The structure kvm_segment is defined as follows: struct kvm_segment { __u64 base; __u32 limit; __u16 selector; __u8 type; __u8 present, dpl, db, s, l, g, avl; __u8 unusable; __u8 padding; }; Since the VMX spec defines base is 64 bits, limit is 32bits, selector is 16bits, understandably, __u64, __u32, __u16 is used for them

Hi, The structure kvm_segment is defined as follows: struct kvm_segment { __u64 base; __u32 limit; __u16 selector; __u8 type; __u8 present, dpl, db, s, l, g, avl; __u8 unusable; __u8 padding; }; Since the VMX spec defines base is 64 bits, limit is 32bits, selector is 16bits, understandably, __u64, __u32, __u16 is used for them

Probably the level of your signal is too low and/or you're just not letting it time to adapt. Jean-Marc Le mercredi 19 juillet 2006 ? 19:00 -0400, ac2491@columbia.edu a ?crit : > On closer looks and debugging I always end up in > > speex_echo_cancel function with comment > /* Temporary adaption rate if filter is not adapted correctly */ > > > Does this give any clue

Hi, I'm trying to use FLAC (v1.1.0) on a Pocket PC (Toshiba e740). It compiles fine without any warning. When I try to compresse 16bits waves, everything is fine. When I try to compress 8bits waves, I get a audio mismatch error (FLAC__STREAM_ENCODER_VERIFY_MISMATCH_IN_AUDIO_DATA). Does anyone have had this problem? I can send the project if someone can look at it. Thanks, Jehan

>> you are better off using the vogg orbis codec. speex is meant >> specifically for telephonic voice. it takes a single human voice and >> compresses it well. it cannot handle muliple voices or music very well. >That part is true, so of course it depends on the application. I guess I >should have added that for most applications, 16 kHz is recommended >instead of 44.1

Flash player encodes speex at 16 kHz, mono, 16 bit. Fields in the format byte should be ignored if the format is speex. You can set the quality by Microphone.encodeQuality (default 6). You can also set the number of speex frames per tc message using Microphone.framesPerPacket. Flash player can only decode speex at 16 kHz, so make sure you have the proper sample rate. Jozsef > > Message:

Hi ppl, I have audio file recorded at 8kHz, 8bits per sample. I am facing problem with encoding and decoding. I have tried with default programs which comes with Speex. can you please suggest with options or any kind of help. I am trying VOIP application so sending packets on RTP. any help is welcomed... Thanks -- Rahul Ruikar Systems Software Engineer Rahul@Phonologies.COM

Hi, I am very fresh man at CODEC part.. I wanna know SPEEX's compression rate.. to say , I tested wav file at 8000 Hz, 8bits to spx file.. I hoped 1/16 compression rate more.. but,, the result was not good for me.. to say example.. I made wav file of 254Kb at 8000Hz , 8bits ...and then encode that file.. the result was that spx file of 33.7Kb is made... so, I though that compression rate

Hi, If I pass the same ref and the echo data to the echo cancellation API, I am expecting silence as output. I get back the original audio data. Is this correct? Thanks -Anurag Quoting ac2491@columbia.edu: > Hi Jean, > > I got the earlier problem fied with correct NN and tail values. > But > I dont see any echo being cancelled. To the echo cancel API I am > giving, audio

Josh Coalson wrote: > yeah, we'll need more info than that. Here is my test project then. To keep the zip small, I didn't put the flac files. Just copy the flac-1.1.0 directory into flactest (so that the path flac-1.1.0\include and flac-1.1.0\src exists in flactest). The project should then compile as is. All the code is in flactest.cpp. It's pretty basic. You'll have to

>> Actually, computing the "power spectrum" for each frame of W shows >> how large an ammount of the original signal at time offset j the >> echo canceller thinks should be removed from the current input frame. > > Careful when looking at W because of how the real and imaginary parts > are packed in the array. Err. Ok, as I got it, 'bin 0' has it's

I found the details: 44100hz, 16bits, stereo I am looking at rewriting the program to record at 8Khz/16Khz/32Khz mono only should I record at 44100hz (and convert down) or record at the required hz level? ----- Original Message ----- From: <speex-dev-request@xiph.org> To: <speex-dev@xiph.org> Sent: Thursday, June 14, 2007 3:44 AM Subject: Speex-dev Digest, Vol 37, Issue 19 >

Hi, Somebody has tested succesfully the speex echo cancellator? Sometimes ago I post the same question, and it was very untested. Best regards, and thank you very much for your work. G.

Hi all, I'm new to LLVM dev mailling list and I'm starting to discover some aspects of LLVM. Actually I'm looking for a solution to create a tool chain for my own chip (a 8bit micro controller processor) that include a compiler/linker/assembler toolset and a simulator/debugger. >From what I've read, LLVM is a good tool to implement a compiler for this proprietary platform,

Hello Guillaume, > - Is there estimation (from your experiences) of the work required to > implement a backend for a simple 8bits micro controller architecture (1 > men-month, 10 or 100 ?) What is the instruction set of your microcontroller? How rich it is? What is the architecture? Is it RISC-y? -- With best regards, Anton Korobeynikov Faculty of Mathematics and Mechanics, Saint

>> Generate a test signal (10+x sine waves per frame), where x increases by >> one for each iteration, and wraps around at 100. > > Testing with sine waves is usually not a good idea. If you intend on > cancelling speech, then test with speech. Ok, I tested more extensively with both music and two-way speech. More on this below. >> However, when peeking at the

Hi Jose, Am 11.10.2017 um 13:13 schrieb Jose Celestino: >> Before I go down the route of upgrading our main Solr instance from 6 to 7, >> just checking if anyone is using Solr 7 with Dovecot. > SOLR 7 works ok with dovecot. We did not upgrade but indexed from scratch with > 7.0.0. > > From our experience: > > Make sure that you set the default queryResponseWriter to

On Thu, Oct 20, 2016 at 12:05 PM, Mehdi Amini <mehdi.amini at apple.com> wrote: > >> On Oct 20, 2016, at 8:54 AM, Cameron McInally via llvm-dev <llvm-dev at lists.llvm.org> wrote: >> >> Hey guys, >> >> I've hit a pretty nasty issue on SKX with ANDs of masks <= 4 bits. >> >> In the IR, we represent a 4b vector mask as <4 x i1>.