similar to: Fw: new message

Displaying 20 results from an estimated 30000 matches similar to: "Fw: new message"

2010 Apr 15
0
Decoded output buffer size
On 15 April 2010 17:55, Daniele Barzotti <daniele.barzotti at eurocomtel.com> wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Il 15/04/2010 01:30, Conrad Parker wrote: >> >>> But how can I know the size of each speex frame into a multiframe payload? >> >> use speex_bit_read_from() just once on the packet, then call >> speex_decode()
2010 Apr 14
0
Decoded output buffer size
On 14 April 2010 23:50, Daniele Barzotti <daniele.barzotti at eurocomtel.com> wrote: > Il 14/04/2010 14:37, Randy Yates wrote: >> >> Usually a buffer is one frame of data, and a frame is 20 milliseconds. >> Since the sample rate is typically 8 kHz in narrowband mode, this >> corresponds to a buffer size of 160 samples. > > Hi Randy, thanks for the reply. >
2010 Apr 14
3
Decoded output buffer size
Il 14/04/2010 14:37, Randy Yates wrote: > > Usually a buffer is one frame of data, and a frame is 20 milliseconds. > Since the sample rate is typically 8 kHz in narrowband mode, this > corresponds to a buffer size of 160 samples. Hi Randy, thanks for the reply. So, suppose I encode an audio buffer (8000 kHz, MONO, float) of 640 PCM frames. In output I have 4 speex frame of 20 byte
2010 Apr 14
2
Decoded output buffer size
Hi, in a VoIP application, the endpoint A send speex payload to B. B doesn't know how A acquire audio, it only know that the channel is narrowband so, how can B know the size of the output buffer to pass to the speex_decode()? Thanks, Daniele.
2010 Apr 15
2
Decoded output buffer size
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Il 15/04/2010 01:30, Conrad Parker wrote: > >> But how can I know the size of each speex frame into a multiframe payload? > > use speex_bit_read_from() just once on the packet, then call > speex_decode() once for each frame. > > Conrad. Thanks for the reply, Conrad. What is not clear for me (and I didn't found it on the
2016 Aug 26
2
Using opus on ATMEL 32-bit RISC microcontroller
Hello Daniele It would be worthwhile to attach an external serial flash or USB thumb drive, if the intent is store data. This allows for far more flexibility in storage Regards Amit On Fri, Aug 26, 2016 at 11:02 AM, Jean-Marc Valin <jmvalin at jmvalin.ca> wrote: > On 26/08/16 11:40 AM, Daniele Barzotti wrote: > > You're right! I forgot to say that I need only the encoder
2007 May 14
1
FW: Message to Speex-dev
Dear All, I would like to have following clarifications regarding the Speex. 1. I have downloaded Speex-ver1.1.5.tar.gz from the site "http://svn.xiph.org/releases/speex/". I tried to open the speexenc and speexdec workspace. I get the message window displaying ..\..\speexenc.dsp "This make file was not generated by developer studio" Continuing will create
2016 Aug 26
0
Using opus on ATMEL 32-bit RISC microcontroller
Hi all, thanks a lot for your replies! Sorry for my typo, the flash size is 8MB (Byte) :-) Unfortunately I cannot use another flash because I'm working on a proprietary board. Jean-Marc, thanks for your suggestions. I thought to use fixed point for convenience, but I can work on floating point too, so I will take in account the codec2 (I didn't know it). Moreover, if you all have
2016 Aug 26
3
Using opus on ATMEL 32-bit RISC microcontroller
Hi Jean-Marc, thanks a lot for your reply. > Well, the first question is whether you want encoding, decoding, or > both. If there's one you don't need then you can remove that > (unfortunately, there's no easy way right now). You're right! I forgot to say that I need only the encoder side (and only for voice). My application have to acquire a 16bit 8KHz PCM stream and
2016 Aug 26
0
Using opus on ATMEL 32-bit RISC microcontroller
Hi Daniele, On 26/08/16 10:39 AM, Daniele Barzotti wrote: > Have I any way to reduce the source footprint? > (Eg. throw away unecessary files, if there are any..) Well, the first question is whether you want encoding, decoding, or both. If there's one you don't need then you can remove that (unfortunately, there's no easy way right now). Even if you need both, then you can
2010 Apr 13
1
Another newbie question on encoding
Hi, I'm very sorry if those questions are repeated over and over, but I cannot find a solution on the net. I try to use speex to encode/decode voice to send over the network. My doubts are: 1. The Bits_Per_Sample I use, are independent from the speex encoding/decoding? (So...can I use 8, 16, 24..and so on?) 2. If I have this situation: SAMPLE RATE.....: 8000 BITS PER SAMPLE.: 16
2009 Dec 01
0
FW: FW: Using speex for gsm-efr coding
Hello, I would like to ask please to remove all of my questions from the forum. (see at the bottom of this email) I was asking it by email and didn't know that it will be published in the forum. Thank you, Ilan Borenshtein ________________________________ From: Ilan Borenshtein Sent: Tuesday, December 01, 2009 2:50 PM To: 'webmaster at xiph.org' Subject: FW: FW: [Speex-dev]
2016 Aug 26
0
Using opus on ATMEL 32-bit RISC microcontroller
On 26/08/16 11:40 AM, Daniele Barzotti wrote: > You're right! I forgot to say that I need only the encoder side (and > only for voice). Then you can remove all of the decoder. As for the encoder, it depends on the bitrate and sampling rate you want (more below). > My application have to acquire a 16bit 8KHz PCM stream and save a > compressed audio into a flash. Sounds like a job
2006 Mar 03
0
Fw: Voice Activation Level (speex 1.1.11.1)
I done it speex_preprocess_ctl(sppPreprocess, SPEEX_PREPROCESS_GET_PROB_START, &g.s.VADstart); speex_preprocess_ctl(sppPreprocess, SPEEX_PREPROCESS_GET_PROB_CONTINUE, &g.s.VADcontin); /*char *lisDebugCh = (char*) malloc(20); int decimalFcvt, signFcvt; static int firstDebug = 1; lisDebugCh = gcvt(g.s.VADstart, 20, lisDebugCh); if(firstDebug == 1) { ::MessageBoxA(NULL,
2008 Aug 29
0
Fw: Voice Activation Level (speex 1.1.11.1)
Manisha, I'm still here. :-) Here's the function: // Returns the average power level in the given signal float getPower(signed short int *signal, int numSamples) { int i; float amp; float powerSum = 0.0f; for (i = 0; i < numSamples; i++) { amp = (float) abs(signal[i]); powerSum += amp * amp; } return powerSum / (32768.0f * 32768.0f *
2005 Sep 22
1
Fw: Results of Automated Batch Tests
The results are at www.rational.co.za/speex.csv Each of the 11 quality settings is tested 3 times (narrow, wide and ultra wide band). Strangely narrow band quality 11 outperforms all wide band tests, but it can be due to my subsampling or some other inaudible effect like delaying. You can import it into Excel and sort it by SNR or other value. Divide the bits by 24 to get the bps. The
2005 Aug 17
1
Fw: Winamp
Hi Shane and all on the list, Thanks - I already have the Ogg Vorbis encoder (oggenc.exe). I am looking to send voice messages to a friend in India and they have terrible connections. I just need to communicate with him quickly and easily from time to time which is why I'm hoping to compress voice with Speex. There's an encoder called Speexenc.exe which I already have, now I only need
2009 Nov 03
0
FW: Handling allocation failure in resampler init - patch
From: Sherief Farouk [mailto:sherief.farouk at twisted-works.com] Sent: Tuesday, November 03, 2009 5:34 PM To: 'speex-dev at xiph.org' Subject: Handling allocation failure in resampler init - patch Hi, Here's a patch that stops allocation failures during resampler init from segfaulting, now they fail gracefully and report the error through the error code parameter
2004 Aug 06
0
Fw: failure notice
I can't unsubscribe. ----- Original Message ----- From: <MAILER-DAEMON@server268.com> To: <gameprogrammer@rakkar.org> Sent: Thursday, April 01, 2004 9:13 AM Subject: failure notice <p>> Hi. This is the qmail-send program at server268.com. > I'm afraid I wasn't able to deliver your message to the following addresses. > This is a permanent error; I've
2008 Jan 24
0
FW: Re: Problem with Blackfin assembly optimizations -- bug in fixed_bfin.h?
Hi Jean-Marc, I did some further checks with 1.2 beta 3: The problematic function is DIV32_16 inside fixed_bfin.h. When I comment it out (i.e. replace it by the generic version of the routine), the quality improves significantly. Nevertheless, there is still a horrible distortion when there is some overdrive/saturation in the input signal. For me, it looks like there's an instable filter