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<DIV><SPAN class=828115605-02112006><FONT face=Arial color=#0000ff size=2>At
first glance it looks like you are opening the output device in 8-bit mode <FONT
face="Times New Roman"><FONT color=#000000 size=3>(m_waveFormatEx.wBitsPerSample
=8;). <FONT face=Arial color=#0000ff size=2>You will need to use 16-bit
mode if you expect to write an array of signed short integers to the audio
buffer. 8-bit PCM is an unsigned mode, no longer used much these
days.</FONT></FONT></FONT></FONT></SPAN></DIV>
<DIV><SPAN class=828115605-02112006><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=828115605-02112006><FONT face=Arial color=#0000ff size=2>Also,
a good debugging trick is to include something like this in your output sample
processing loop:</FONT></SPAN></DIV>
<DIV><SPAN class=828115605-02112006><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=828115605-02112006><FONT face=Arial color=#0000ff
size=2> static FILE *hack =
fopen("test.raw","wb");</FONT></SPAN></DIV>
<DIV><SPAN class=828115605-02112006><FONT face=Arial color=#0000ff
size=2> fwrite(output, 2, FRAME_SIZE,
hack);</FONT></SPAN></DIV>
<DIV><SPAN class=828115605-02112006><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=828115605-02112006><FONT face=Arial color=#0000ff
size=2>After your app is closed, there will be a file called "test.raw" in
its working directory, which you can load into Sound Forge or a similar audio
editing package for closer inspection without relying on your output
buffers and playback code to work properly. </FONT></SPAN></DIV>
<DIV><SPAN class=828115605-02112006><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=828115605-02112006><FONT face=Arial color=#0000ff size=2>--
john</FONT></SPAN></DIV>
<DIV><SPAN class=828115605-02112006><FONT face=Arial color=#0000ff size=2>RAD
Game Tools</FONT></SPAN></DIV>
<BLOCKQUOTE
style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px solid">
<DIV class=OutlookMessageHeader dir=ltr align=left><FONT face=Tahoma
size=2>-----Original Message-----<BR><B>From:</B> speex-dev-bounces@xiph.org
[mailto:speex-dev-bounces@xiph.org]<B>On Behalf Of </B>Carine
Liang<BR><B>Sent:</B> Wednesday, November 01, 2006 9:40 PM<BR><B>To:</B>
speex-dev@xiph.org<BR><B>Subject:</B> Re: [Speex-dev] Integrating speex with
VideoNet application: Constantbackground noise<BR><BR></FONT></DIV>
<DIV
style="FONT-SIZE: 12pt; FONT-FAMILY: times new roman, new york, times, serif">
<DIV
style="FONT-SIZE: 12pt; FONT-FAMILY: times new roman,new york,times,serif">Hi,<BR><BR>Can
someone please help me with my problem below. Any suggestions is appreciated.
<BR><BR>thanks,<BR>Carine<BR><BR>
<DIV
style="FONT-SIZE: 12pt; FONT-FAMILY: times new roman,new york,times,serif">-----
Original Message ----<BR>From: Carine Liang
<carineliang@yahoo.com.sg><BR>To: speex-dev@xiph.org;
speex-dev@xiph.org<BR>Sent: Tuesday, 31 October 2006 1:05:49 PM<BR>Subject:
[Speex-dev] Integrating speex with VideoNet application: Constant background
noise<BR><BR>
<DIV
style="FONT-SIZE: 12pt; FONT-FAMILY: times new roman,new york,times,serif">
<DIV>Hi,<BR><BR>I am developing a peer-to-peer video conference application
which uses speex as a codec for the voice. <BR><BR>I am new to speex, so
please bear with me if I asked the obvious. After I added the encode and
decode function to my MFC app, I heard a constant background noise, even when
no one is speaking into the microphone. <BR><BR>#define FRAME_SIZE
160<BR><BR>The application is coded in MFC C++. The record buffer is set to be
the same as the FRAME_SIZE such that my callback function is called everytime
160 bytes of data is recorded.<BR><BR>Here's what I did in the initialisation
(called only once).<BR>m_waveFormatEx.wFormatTag
=WAVE_FORMAT_PCM;<BR>m_waveFormatEx.nChannels
=1;<BR>m_waveFormatEx.wBitsPerSample =8;<BR>m_waveFormatEx.cbSize
=0;<BR>m_waveFormatEx.nSamplesPerSec = 8000;<BR>m_waveFormatEx.nBlockAlign
=1;<BR>m_waveFormatEx.nAvgBytesPerSec = 8000;<BR><BR>int quality
=8;<BR>speex_bits_init(&bits);<BR>enc_state =
speex_encoder_init(&speex_nb_mode);<BR>speex_encoder_ctl(enc_state,
SPEEX_SET_QUALITY, &quality);<BR><BR><BR>In my recording callback
function<BR>LRESULT RecordSound::OnSoundData(WPARAM wParam, LPARAM
lParam)<BR>{<BR> ....<BR> dataPtr = (char *)lpHdr
->lpData;<BR> dataSize = (int)lpHdr
->dwBytesRecorded;<BR><BR> for(int
k=0;k<FRAME_SIZE;k++)<BR> input[k] =
dataPtr[k]; //input is
float[FRAME_SIZE]<BR> <BR>
speex_bits_reset(&bits);<BR> speex_encode(enc_state,
input, &bits);<BR> encByte = speex_bits_write(&bits,
cbits, 200);<BR><BR> //send cbits to peer computer via
sockets<BR>}<BR><BR><BR>In the receiver's initialization function, I've
initialized the speex_decoder.<BR>speex_bits_init(&bits);<BR>dec_state =
speex_decoder_init(&speex_nb_mode);<BR><BR>In the receiver's socket
function, it reads from socket and store the data in cbits (char
array).<BR>{<BR> speex_bits_read_from(&bits, cbits,
retvalue); //retvalue is 38<BR> speex_decode(dec_state,
&bits, output); //where output is float array<BR><BR>
for(int k=0;k<FRAME_SIZE;k++)<BR> out[k] =
output[k]; //out is a char array<BR> <BR>
PostThreadMessage( WM_PLAYSOUND_PLAYBLOCK, FRAME_SIZE, (LPARAM)out
);<BR>}<BR><BR><BR><BR></DIV></DIV><BR>
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