Displaying 19 results from an estimated 19 matches for "yueweng".
2005 Apr 18
3
speex voice seems to be bit breaking over long distance.
...ket. Hence, all the breaking
portion is filled with previous packet. This trick
seems to work so far. I am not sure what is the side
effect.
I think jitter buffering is more correct way to solve
this problem but I think this is not so easy. Is there
got any source code to reference from?
Regards,
YueWeng
--- "Kleijn, J.S. de" <J.S.d.Kleijn@student.tue.nl>
wrote:
> Hi,
>
> You should implement a jitter buffer, this buffers
> the audio to prevent the 'breaking'.
>
> My VoIP application called VoIPerized has a jitter
> buffer. You can try it to see if...
2004 Dec 28
5
Sound distorted after normalized.
> 16 bit ints have a range of -32768 to 32767. If you divide
> -32768 by 32767.0 you end up with -1.00003051850948 which
> is a bad thing.
>
> Try normalizing with a value of 32768.0.
No. Speex expects values in the +-32767 range, not +-1.0. Just
converting from int16 to float *is* the right thing to do.
Jean-Marc
--
Jean-Marc Valin <Jean-Marc.Valin@USherbrooke.ca>
2005 Apr 26
2
100% CPU usage
...(decoder_state, NULL, shortData);
//check if shortData[] array is zero bytes.
m_IsZeroData = IsZeroData(shortData);
}
m_IsZeroData will be set to false when new voice
packet arrive.
It seems to me that speex_decode_int() can not be
execute too many times after shortData[] become zero.
Regards,
YueWeng
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2004 Dec 28
1
How to convert from Microsft PCM 16bit to float
...d speex_decode takes float * as an
arguement to encode and decode the sound. However,
when I get the PCM data from the sound card under
win32, it is a just 16 bit array. May I know how do I
convert this 16 bit value to speex float format and to
convert back? Is there got any routine to do this?
YueWeng
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2004 Dec 30
2
Speex sound a little artificial?
...oy speex 1.1.6 in my application. With no
option set, I can hear that the voice sounds a little
bit artificial like robot. Any idea what causes this?
I use openh323 with speex, but it seems ok. Is it
neccessary for me to use more other filter prior to
encode the sound or after decode my sound?
yueweng
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2004 Dec 28
2
Sound distorted after normalized.
...is my setting from sound card using
mm_system sdk.
WAVEFORMATEX *wfx = (WAVEFORMATEX *)input;
wfx->nSamplesPerSec = 8000;
wfx->nChannels = 1;
wfx->wFormatTag = WAVE_FORMAT_PCM;
wfx->wBitsPerSample = 16;
wfx->nBlockAlign = 2;
wfx->nAvgBytesPerSec = 16000;
Thanks in advance,
YueWeng
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2005 Feb 17
1
Hellolololo... effect in long distance?
...ot;hellololololololo....". This does not always
happened. The chance of happen is quite random. This
effect does not happen in my LAN. It only happen when
ping time is about 500 milisecond or more.
Is it a speex problem? Any suggestion how to resolve
this issue?
Thanks in advance!
Regards,
YueWeng
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2005 Apr 26
1
100% CPU usage
...e and seems that (not confirmed) the
speaker callback WaveOutCallback() function which call
speex_decode_int(decoder_state, NULL, shortData)
(when no data arrived for PLC purpose) seems to
consume a lot of CPU slice. Is this normal or I am
making a mistake other portion of the program.
Regards,
YueWeng
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2005 May 04
0
Speex over 56.6K modem
...entirely smooth,
enough audio is buffered to avoid dropouts.
In my program using Speex, I've been able to have a dialup
modem user send audio and video to two people at the same time
and not have any breakups in the audio. (It was *very* low
bitrate video.) :)
Hope this helps,
Tom
Tay YueWeng <yueweng@yahoo.com> wrote:
>
> Dear all,
>
> Have anybody uses speex codec over 56.6K bit/s line
> that hear clearly in both direction before? I perform
> a testing with one side using 56.6K dialup modem and
> the other side using broadband. The user that uses
> 56.6...
2005 Apr 18
0
speex voice seems to be bit breaking over long distance.
...just try decoding by
passing NULL instead of the SpeexBits struct.
> I think jitter buffering is more correct way to solve
> this problem but I think this is not so easy. Is there
> got any source code to reference from?
Look for the speex_jitter.h header.
Jean-Marc
> Regards,
> YueWeng
>
> --- "Kleijn, J.S. de" <J.S.d.Kleijn@student.tue.nl>
> wrote:
> > Hi,
> >
> > You should implement a jitter buffer, this buffers
> > the audio to prevent the 'breaking'.
> >
> > My VoIP application called VoIPerized has a...
2004 Dec 28
0
Sound distorted after normalized.
Hmmm... sorry i mislead you Tay... i didn't realise my encoder
(libfishsound) was again shifting the data i gave it back to being 32767
based internally.
Zen.
----- Original Message -----
From: "Jean-Marc Valin" <Jean-Marc.Valin@USherbrooke.ca>
To: "Tay YueWeng" <yueweng@yahoo.com>
Cc: "speex" <speex-dev@xiph.org>
Sent: Wednesday, December 29, 2004 1:04 PM
Subject: Re: [Speex-dev] Sound distorted after normalized.
> 16 bit ints have a range of -32768 to 32767. If you divide
> -32768 by 32767.0 you end up with -1.00003051...
2004 Dec 30
0
Speex sound a little artificial?
...ith your
application, it's likely buggy. If the result isn't OK, then it can be:
1) Normal given the bandwidth/bit-rate used
2) A conditioning problem with your audio (i.e. DC not removed)
3) A bug in Speex (quite rare at this point)
Jean-Marc
Le jeudi 30 d?cembre 2004 ? 17:17 -0800, Tay YueWeng a ?crit :
> Hi all,
>
> I have deploy speex 1.1.6 in my application. With no
> option set, I can hear that the voice sounds a little
> bit artificial like robot. Any idea what causes this?
> I use openh323 with speex, but it seems ok. Is it
> neccessary for me to use more...
2004 Dec 31
2
Speex sound a little artificial?
Hi,
> 1) Normal given the bandwidth/bit-rate used
Do you mean the bit-rate that I should set in the
speex codec?
> 2) A conditioning problem with your audio (i.e. DC
> not removed)
What is DC?
YueWeng
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2005 Jan 05
4
Encoding and decoding problem in speex 1.0.4
...ed==0 || lpHdr==NULL)
return ERROR_SUCCESS;
::waveInUnprepareHeader(m_hRecord, lpHdr, sizeof(WAVEHDR));
Here lpHdr->lpData contains the audio data in a character array.
Now here I want to use Speex codec for encoding the data so the encoding
function is
called (I am thankful to Tay YueWeng for the function).
char *encode(char *buffer, int &encodeSize)
{
char *encodedBuffer = new char[RECBUFFER/2]; /*
RECBUFFER = 2000 */
short speexShort;
float speexFloat[RECBUFFER/2];
void *mEncode = speex_encoder_init(&speex_nb_mode);...
2005 Jan 04
1
Conversion problem in encoding and decoding
Hi,
I am new to this newsgroup. I have a problem using the Speex library.
I have recorded sound data using a character buffer like
char recMessg[BUFFERSIZE]. I have problem converting this to the
float array for encoding. Also help me in converting the decoded
information in float array back to character array.
Can anybody give me function codes for this conversion.
Thanks in
2005 Jan 04
0
Conversion problem in encoding and decoding
> short temp_short;
> float temp_float;
> float floatData[number_of_sample];
>
> for (int i=0; i < number_of_sample; i++)
> {
> memcpy(&temp_short, &src[i*2], 2);
> temp_float = temp_short;
> floatData[i] = temp_float;
> }
>
> yueweng
>
Thanks for the encoding code can yu plz send me the decoding code
for conversion from float array to character array.
2005 Apr 18
0
speex voice seems to be bit breaking over long distance.
...cket lost is less than 10%. I have
tried 8KHz, 16KHz, 32KHz. The bigger it gets, the more
lousy the quality. I believe I have enough bandwidth
since the packet lost is less than 10%. The only
factor that could affect the quality is may be
delayed.
Any one have any idea of what is wrong?
Regards,
YueWeng
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2005 Apr 19
1
speex voice seems to be bit breaking over long distance.
...the third
parameter of speex_jitter_put(), should I use the
GetTickCount() value at the point when I first receive
a voice packet from the microphone?
Finally, what is the use of the third parameter
(timestamp) of speex_jitter_get()?
Thanks very much. Hopes my question is not too many.
Regards,
YueWeng
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2004 Dec 30
2
echo cancellation
I'm going to test the echo cancellation...
but before doing something wrong, i want to know what is the right way
to use it...
-------------------------------------------------------------------------------------
short * speakerSample; //will contain what the speakers are going to play
short * microphoneSample; //what the microphone is going to register
short* cancelledSample; //the ouput