similar to: VAD always return 1

Dear Ogg Vorbis, I'm a senior in Information Technology, and my thesis is on Audio Watermarking. As a new starter in audio, I really need your help, and suggestions. I think in order to persue something on Audio I have to know much details about audio formats. So could you please show me where I can find the documents about the information I need ? Finally,could you please answer a confused

Hello I am a university student and I have just started to usespeex newly, I am implementing this codec on STM32F4 and I use speex library. Ihave a problem with VAD, can anybody help me? I want to know which parameters Ishould tune in my code (using speex library)? Which function should be called?I want to record a voice on SD card and I want to use this algorithm to detectsilent periods to save

https://bugzilla.samba.org/show_bug.cgi?id=11414 Bug ID: 11414 Summary: rsync: chgrp "/.filename.5afK5X" (in dirdir) failed: Operation not permitted (1) Product: rsync Version: 3.1.1 Hardware: x64 OS: Linux Status: NEW Severity: normal Priority: P5

Hi all, I have been trying to impelement echo cancellation on STM32F4 Discovery board (having ARM based uC).In my project, I simply initialized the echo_state by speex_echo_state_init() by using frame_size 160, filter_length 800 (at 8kHz Sampling rate.) Then, I simply used speex_echo_cancellation() function in where i am getting sound. However, I have recently realised that output_frame has been

Hi all, I have been trying to implement speex echo cancellation algorithm on STM32F4 Discovery board; however, I could not make it work properly. I tried synchronous, asynchronous way of to do that and the only thing I heard was "cazzsdaos?eo?hodg etc" :) I try to use empty array for all parameters in the cancellation function, but the result is always the same

Hello again, I want to ask a question in my mind by reminding the topic below. Can the reason of failing be that Codec,which plays sound, and Mic are connected the different clock source? If it is because of that, what can I do to cancel the echo? Isn't there any solution for this? Is asynchronus speex functions for this? Best Regards, KAHYA PS: Sorry for my bad english. From: ?evket K?r?ad

I'd like to compress a 32 kHz 16-bit mono using an STM32F4 microcontroller. My plan is to capture from 500-1000 ms of data, compress, and send the result over a UART. I have the arm-none-eabi tools installed on MacOS. I used homebrew to install autoconf, automake, and libtool. autogen.sh succeeded and I was able to create my makefiles using the following: *./configure

I have been trying to understand how to get the VAD algorithm working. I sent an input stream of all zeros into the preprocessor but still got a return value of 1 indicating that speech was detected. Is this feature not available with the latest release? I thought at the very least it would detect this as silence and return 0 but that does not seem to be the case. Does anyone have any

The reset control API has two modes: exclusive access, where the driver expects to have full and immediate control over the state of the reset line, and shared (clock-like) access, where drivers only request reset deassertion while active, but don't care about the state of the reset line while inactive. Commit a53e35db70d1 ("reset: Ensure drivers are explicit when requesting reset

We have found that it is possible to achieve a 30 to 50% reduction in MHz requirement for implementation of OPUS on Cortex M4 compared to the public version (v1.3 beta/v1.2.1). For the CELT configuration you mention (complexity 0, 16kHz, mono, 20ms) we are measuring a 4ms encode time and a 3ms decode time for that platform (32kbit/s). An important issue that I haven't seen much discussion

Okay, Not exactly the answer I was looking for. This sounds like a big change. I don't mind re-writing the LSP quantizer but re-training code books and breaking compatibility is not what I want to do. I am working on an optimization for an ARM cortex-A8. It is desirable to process things in 4 element blocks. Is there a simpler approach you could recommend? Vasant Shridhar

Thanks Jean-Marc, I was able to get both encode and decode working the CC3220 device! But for bi-directional communication, I need decode and encode to occur in less time than the frame size I’m sending (20 ms). Currently decode takes 16~22 ms and encode is ~13 ms. What is the best way to try to reduce this time? Also, unsure why encode is taking less time than decode... I've also

On Wed, Jul 19, 2017 at 05:25:04PM +0200, Philipp Zabel wrote: > The reset control API has two modes: exclusive access, where the driver > expects to have full and immediate control over the state of the reset > line, and shared (clock-like) access, where drivers only request reset > deassertion while active, but don't care about the state of the reset line > while inactive.

Hi all, We all know that mismatch between clocks of ADCs of far-end voice and near-end voice is not allowed in a time-domain or frequency-domain LMS based AEC system. It means that capture and render audio streams must be synchronized to a same sample rate. However, I found that this restriction is removed in microsoft AEC from Windows XP SP1. Anyone knows how microsoft AEC do it? This technology

I would like to make lpcSize a multiple of 4. In the current library for narrow band mode lpcSize is currently set to 10. I would like to increase this 12 for optimization reasons. Is it a simple matter of just changing lpcSize in the structures in mode.c or are there other implications to doing this? I did try and change the value but I seem to be getting some runtime errors indicating a

On mer, 2011-08-10 at 09:41 -0400, Rony Nandy wrote: > Hi All, > I have downloaded libogg-1.3.0 along with speex.But,during > build speex is exiting with ogg.h missing.Any suggestions will be highly > appreciated. IIRC, speexenc encodes your data into a speex stream which is encapsulated into an OGG container, so you need to libogg to compile it. Though, it has been ages

From 1.2rc1 source code: preprocess.c: /* FIXME: This VAD is a kludge */ st->speech_prob = Pframe; if (st->vad_enabled) { if (st->speech_prob > st->speech_prob_start || (st->was_speech && st->speech_prob > st->speech_prob_continue)) { st->was_speech=1; return 1; } else { st->was_speech=0; return 0; } } else { return 1; } AND case

Hi Shridhar, Sample rate conversion is not enough to solve this problem. I have tried this method several months ago. The first step is to measure the difference between sample rate of capturing and rendering. Then resampling (by what you said "sinc interpolation") one signal to eliminate the difference. The frequency step in my experiment is less than 0.1Hz. I have tried speex AEC

https://bugs.freedesktop.org/show_bug.cgi?id=80627 Priority: medium Bug ID: 80627 Assignee: nouveau at lists.freedesktop.org Summary: [NVE6] 'HUB_INIT timed out' 'Watchdog detected hard LOCKUP on cpu 7' QA Contact: xorg-team at lists.x.org Severity: normal Classification: Unclassified

On 04/13/2011 02:58 AM, Shridhar, Vasant wrote: > I am doing this right now with no problem. I am not using speex for this at the moment though. Group delay is the biggest problem. I implemented a version where the input and output sample rates are known up front. The routine than interpolates between the jitter. This should solve the problem. The crystals used to clock the input and