similar to: WAVE PCM to OPUS and back

Hi, I have a piece of R code that calls mgcv::gam. The code runs fine in the interactive mode, but terminates R w/o a single message when run non-interactively. Though I think I should be able to locate the problem by brute force I'd appreciate an advise how to do it more intelligently using R debugging tools. At this time I only know that it has something to do with me loading my custom

Ronald, I suggest we take this discussion off the list after this post. I don't see any problems with your code. However, I can make a few suggestions: 1) If you replace the lines: pSpeexHdr->lpData = (char*)pSound; pSpeexHdr->dwBufferLength = decodedSamples * sizeof(short); with: pSpeexHdr->lpData = lpHdr->lpData; pSpeexHdr->dwBufferLength = lpHdr->dwBufferLength;

hi Tom, i downloaded your speex wrappers and they're really what i'm looking for! i really cannot worry about container support at the moment because i'm working with real-time audio, and hopefully toward VoIP when everything else is in order. im using waveform to capture data from the microphone. the raw PCM sample is stored in the WAVEHDR structure's member lpData, and this

Hi, I?ve strange problem when I?m making a call from SIP (Cisco 7960) to capi (Fritz PCI). When I call a national number, I?m hearing the ringtone when the called party is ringing but when I call an international number, I don?t hear the ringtone and I?ve a silence until the called party answers. Both call are going through the same extension. Here?s 2 log files, one with a national number and

Hi Rowland, -<| Quoting Rowland Penny via samba <rpenny at samba.org>, on Wednesday, 2019-02-13 05:01:19 PM |>- > On Wed, 13 Feb 2019 17:16:21 +0100 > Philipp Gesang via samba <samba at lists.samba.org> wrote: > > -<| Quoting L.P.H. van Belle via samba <belle at bazuin.nl>, on > > Wednesday, 2019-02-13 04:59:55 PM |>- > > >

Hi! I'm totally new to audio programming. I managed to create a VoIP with waveform (PCM). The problem was the really high traffic, so I desided to use speex for compressing the data. I'm recording with mono, 16bit/sample, 8000 samples per second. I tried to add the speex compression, but I always get a crash when I try to decode the data. Here is my encode and decode function: int

hi sirs, i am only recently studying speex and i'm looking for a C++ wrapper for speex that has higer-level implementations for the Encode and Decode functions, to sort of base my initial understaning of how the speex API work. i have in fact created my own, following on the instructions in the speex codec manual, but i havn't really made any positive progress. (a lot of people have

Ronald, I recently wrote some wrapper classes for the Speex encoder and decoder for use with my VoIP program. I think they're pretty high level and easy to use. They handle all buffer allocation internally to make life easy. Here's a simple little example that happens to use VAD: // buf is float[] or short int[], buflen is multiple of frame size // (there is a getFrameSize() method

The attached patch resolves a buffer reuse bug in the efi/tcp.c code. The bug can be triggered when multiple TCP connections are opened when using the EFI bootloader. Signed-off-by: Joshua Weage <joshua.weage at dell.com> --- --- master/core/fs/pxe/pxe.h 2017-07-22 14:45:46.561674329 -0500 +++ jw/core/fs/pxe/pxe.h 2017-07-23 19:44:46.466498950 -0500 @@ -144,6 +144,7 @@ struct

Hi Louis, thanks for your reply. -<| Quoting L.P.H. van Belle via samba <belle at bazuin.nl>, on Wednesday, 2019-02-13 04:59:55 PM |>- > > DOM.AIN\foobar's password: > ^^^^^^^^ > > No dot is allowed in the NTDOM > Fix that first, then try again. That’s the output when logon succeeds though nor does the value seem to matter anywhere else. This is just Samba

Irritating quirkyness: we have a bunch of videocams. To use, we use gspca. Usually, on an upgrade, I just go into the gspca directory (which appears, from their website, to have not been updated since '07), make clean, make, make install. Having gone up to 5.5, did the same. What's happening now is that it works, delivers the mpgs... but dumps errors in the logs: <snip> kernel:

Hi, Can someone please help me with my problem below. Any suggestions is appreciated. thanks, Carine ----- Original Message ---- From: Carine Liang <carineliang@yahoo.com.sg> To: speex-dev@xiph.org; speex-dev@xiph.org Sent: Tuesday, 31 October 2006 1:05:49 PM Subject: [Speex-dev] Integrating speex with VideoNet application: Constant background noise Hi, I am developing a peer-to-peer

Hi, I am using the speex 1.0.4 library from Windows. I have posted my problem before but didn't get a solution. I am doing an VOIP project in which i am recording sound and streaming it to the peer. I wanted to encode and decode wav files that brought me to this site. I am recording sound in the following format:- m_WaveFormatEx.wFormatTag = WAVE_FORMAT_PCM;

Hi John, Thanks for your reply. Yes, my output device is in 8-bit mode, same as my input mode. I'm actually working in char (8 bit) arrays. Is it necessary to work in signed short integers only? I changed all the short arrays in the example to char. Carine ----- Original Message ---- From: John Miles <jmiles@pop.net> To: carineliang@yahoo.com.sg; speex-dev@xiph.org Sent: Thursday,

I've looked everywhere and tinkered for three days now, so I figure asking might be good. So here's a general rundown of what I am trying to get my code to do I am giving you the whole rundown because I need a solution that retain certain ways of doing things because they give me the information i need. I want to examine the effect of increasing my sampling interval on my data. Example:

Hi, I am developing a peer-to-peer video conference application which uses speex as a codec for the voice. I am new to speex, so please bear with me if I asked the obvious. After I added the encode and decode function to my MFC app, I heard a constant background noise, even when no one is speaking into the microphone. #define FRAME_SIZE 160 The application is coded in MFC C++. The record

I'm trying to use the speex API, and I think I'm missing something. I'm running on Mac OSX Tiger, and the attached microphone is a iSight. If I call the "speex_encode_int" function, and then immediately call the "speex_decode_int" function, shouldn't I get back the same data? i.e. in my example code, shouldn't the two arrays contain at least similar

I''m seeing a "marshal data too short" error with an ActiveRecord store for my session data. Other posts say that this happens when the size of the session data exceeds the size of the "data" column in the sessions table. But my "data" column is a TEXT field so it seems unlikely that I could have blown it out. Has anyone else seen "marshal data too

The variable 'ok' is never used and generates a warning. Remove it. Also ntfssect.c is designed to be compiled in non Unicode mode when using MSVC compilers, so remove all ambiguity about it (LPCTSTR -> LPCSTR, use of 'A' API calls) so that it doesn't break when compiled in Unicode mode, which is what Rufus uses with MSVC. -------------- next part --------------

Hi, I've changed the m_waveFormatEx.wBitsPerSample for both record buffer and playSound buffer, changed my array from char to short and I'm still getting the same constant background noise. I notice that when I talk into the mike, I will get a slightly louder noise, can't hear any clear speech... Any idea what's wrong? Regards, Carine ----- Original Message ---- From: John Miles