similar to: 【SPEEX】 use speex resample make noise

Look, how about you start from the testresample.c file? Oh, and you seem to be reading 320 samples at a time and processing 640, so that can't be good (and even beyond that your code is wrong for other reasons). Jean-Marc On 11/6/19 1:56 AM, zhouyuchen at iauto.com wrote: > Hello, > I printed the log, in/out len is not truncated, which means that the > input and output are not

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

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

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. >

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

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>

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: ***

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

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

Hi, On Fri, Jul 29, 2016 at 1:50 PM, dos386 <dos386 at gmail.com> wrote: > 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) What do you mean by broken? Those links seem to work here. > - there is some (minor) development (whitespace

On 5/29/08, Thorvald Natvig <thorvald at natvig.com> wrote: > Alexander Chemeris wrote: > > 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

I'm not using Opus at all. I'm just including its resampler in my own sources. It's not even a DLL; it's directly compiled together with the rest of my code. You need these sources from the opus-tools package (http://www.opus-codec.org/downloads/): arch.h resample.c resample_sse.h speex_resampler.h stack_alloc.h In your project file, define these macros: #define

________________________________ Hi David, Thank you very much for your reply. Since I need to resample the audio in the program itself, I decided to try out the resampling API in speex. But now, I have another problem. The resampled sound is very much distorted and clicks appear quite often. (I have attached the source code I used for testing it below). The test data I had was a file sampled

Hi, Here's the (hopefully) final version of the resampler, now always using st->mem as the buffer area. It only allocates buffers on the stack when it's necesarry to convert the output between int and float. -------------- next part -------------- Index: include/speex/speex_resampler.h =================================================================== ---

Dear all: I need the help desparately. The code is attached below. If you guys don't mind take a look at the code below and see how to fit speex's AEC into it. Help me look at the #defines, and give me some suggestions on the AEC parameters, I totally have no idea about them. Feel free to do anything with the code, if it is by any chance valuable. Any ideas or suggestions or sharing

>> Yes, I plan to use it in a VoIP environment if I can get latency reduced to >> an acceptable level :) >> The latency depends directly on the overlap parameter, which also controls >> the quality. Higher quality => higher latency. You could set the overlap to >> 0, but that would give you some nasty artifacts. >> You can also resample with smaller block

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] <-

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.

Dear Nikos, thanks! But you use Opus only for resampling, not for entirely replacing Speex, don't you? Greetings! Hermie Am 07.05.2013 22:53, schrieb Nikos Chantziaras: > The Opus resampler is actually a bugfixed version of the Speex one. Same > interface/API, but with the bugs removed. It's why I recommended it :-) > Otherwise I would have recommended something entirely

Thanks for your help, I will follow that, so in order to sum up I need: - libopus 1.0.3 compiled as static or shared library - libogg 1.3.1 compiled as static or shared library - opus-tools 0.1.7 and then follow the example in opus-tools opusenc.c to get things ready, because if I try to compile opus-tools, this ask for me for the speex library, and other things I think that that will we all.