similar to: 【SPEEX】 use speex resample make noise

Displaying 20 results from an estimated 300 matches similar to: "【SPEEX】 use speex resample make noise"

2019 Nov 05
0
【SPEEX】 use speex resample make noise
Be aware that inlen is an in/out parameter. It tells you how many samples the resampler read. That number can be smaller than what you passed -- in which case it means you need to buffer them and give them to the resampler again in the next call. Jean-Marc On 11/5/19 8:27 AM, zhouyuchen at iauto.com wrote: > Hello, > I have encountered some problems. I want to use speex to convert the
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
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
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. >
2004 Aug 06
0
Please 30 second to look a my code
Well, you seem to be using FRAME_SIZE but only defining frame_size. Otherwise, the code looks OK, but it's always hard to tell. I suggest you start from speexenc/speexdec or from the example I wrote in the manual at: http://www.speex.org/manual/node12.html Jean-Marc Le ven 19/12/2003 à 05:22, Fabio a écrit : > Hi > i'm developing a sort of VoIP application > for my
2007 Nov 14
0
Audio glitches/Configuration problem !!?
Hi all, First of all, thanks to Speex developper for the all the job. I am trying to implement my own Speex DirectShow fitlers for VoIP following the documentation and sample code's. I am facing audio glitches when encoding - decoding PCM data. The encoder and decoder procedures are copied below. What about the lookahead size ? how shouw we apply it in encoding stage ? thanks in advance
2009 Mar 18
1
sprintf("%d", integer(0)) aborts
In R's sprintf() if any of the arguments has length 0 the function aborts. E.g., > sprintf("%d", integer(0)) Error in sprintf("%d", integer(0)) : zero-length argument > sprintf(character(), integer(0)) Error in sprintf(character(), integer(0)) : 'fmt' is not a non-empty character vector This comes up in code like x[nchar(x)==0] <-
2004 Aug 06
2
Please 30 second to look a my code
Hi i'm developing a sort of VoIP application for my ipaq using speex... I'm still at beginning and i have many problems encoding and decoding my wav files....output is only noise! Why? I'm using Libspeex 1.1.3, Embedded VisualC++ 3.0, Ipaq 3850(206 MHz Intel® Strong ARM 32-bit RISC Processor) PocketPC 2002 (Windows CE 3.0). Libspeex is complied with the definition of
2007 Nov 04
3
WaveIn/WaveOut and Speex
Hello, I know my question has been asked before because I spent the last week searching the web for how to use Speex in combination with WaveIn/WaveOut and I ran into a few posts, but none of them answer the question. There is still a lot of confusion how to use WaveIn/WaveOut and Speex by junior developers such as myself. Even after examining code for SpeexDec and SpeexEnc, I cannot get clear
2007 Nov 04
0
WaveIn/WaveOut and Speex
I'm not sure what input/output format you're trying to use, but it looks wrong. You're using the float functions that take +-32767 values and you're feeding (or converting) chars. Unless your machine has very special chars (which I doubt), a +-32767 value just isn't going to fit in. This has nothing to do with Speex BTW, it's just handling the audio data properly.
2007 Nov 04
0
WaveIn/WaveOut and Speex
> When I was going from Char to float and back looked very wrong to me it not only *looked* wrong! > as well, but I was just not sure (and still am) how to translate the > Char* audio stream generated by WaveIn to a format that can be > understood by Speex. Would using speex_decode_int and > speex_encode_int instead of speex_decode and speex_encode be the > answer? Using
2007 Dec 11
0
[PATCH] update symbian build
This patch updates the symbian build files to latest svn trunk, and also adds a new makefile for speexdsp. Signed-off-by: Alfred E. Heggestad <aeh@db.org> --- Index: symbian/speexdsp.mmp =================================================================== --- symbian/speexdsp.mmp (revision 0) +++ symbian/speexdsp.mmp (revision 0) @@ -0,0 +1,42 @@ +/* + Copyright (C) 2003 Commonwealth
2007 Nov 04
2
WaveIn/WaveOut and Speex
Thank you for such a quick response. The only reason I started with Char buffers is because WaveIn and WaveOut on Windows XP accept/emit WAVEHDR structures, which store audio data in LPSTR, which is Char*. typedef struct { LPSTR lpData; DWORD dwBufferLength; ... } WAVEHDR; When I was going from Char to float and back looked very wrong to me as well, but I was just not
2007 Nov 05
0
Fw: RE: WaveIn/WaveOut and Speex
Begin forwarded message: Date: Mon, 5 Nov 2007 07:27:21 -0500 From: "Evgueni Tsygankov" <eugenet@rusmex.com> To: "Jean-Marc Valin" <jean-marc.valin@usherbrooke.ca> Cc: speex-dev@xiph.org Subject: RE: [Speex-dev] WaveIn/WaveOut and Speex Again, thank you for helping me. I know this might seem like a trivial matter to you and other experts in the field, but believe
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.
2007 Nov 05
2
WaveIn/WaveOut and Speex
Again, thank you for helping me. I know this might seem like a trivial matter to you and other experts in the field, but believe me, there are a lot of programmers, whose posts I saw on the web, who tried to use WaveIn/WaveOut and Speex and failed. As I understand it, WaveIn just buffers audio data according to the bit rate specified. So, if we use waveFormat.wBitsPerSample = 8, then each Char of
2013 Oct 28
1
how to Build .opus file
That's true I'm trying to build it to use in android I used the -DOUTSIDE_SPEEX. and i get speex_resampler.h:51:2: error: #error "Please define RANDOM_PREFIX (above) to something specific to your project to prevent symbol name clashes" 2013/10/28 Gregory Maxwell <gmaxwell at gmail.com> > On Mon, Oct 28, 2013 at 11:26 AM, Antonio Juan <anquegi at gmail.com>
2013 Oct 28
2
how to Build .opus file
Hi and thanks for your interest! this is the error, i need to add the speex folder and a path to the .h file and it works Compile thumb : opus-proj <= audio-in.c In file included from jni/../../opus-tools-0.1.7/src/audio-in.c:74:0: proj/../../opus-tools-0.1.7/src/speex_resampler.h:90:32: fatal error: speex/speex_types.h: No such file or directory compilation terminated. make: ***
2016 Jul 29
2
SPEEX and OPUS questions and minor issues
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
2008 May 29
2
FFT Resampler
Alexander Chemeris wrote: > Hi, > > Here are some questions from user point of view. :) > > On 5/29/08, Thorvald Natvig <thorvald at natvig.com> wrote: > >> I've done listening tests when converting wb_male.wav to 44.1, 48 and 8khz, >> and there aren't any obvious artifacts. I also did a 16=>16 test, and the >> results are delayed by 10ms