similar to: On the "broken" .WAV files issue

Hello, I have an p2p voice chat application using WAVE PCM (winmm). Now i am trying to add opus encoding to it to send it over the TCP/IP and then decoding it back to play - but without success (without opus it works ok) Here is an example of my code. I get message from input device then encode with opus then decode it back to output wave header and play. Doing it i hear only noise in my

Hi all! I've problem with playing ogg files. I'm triing to use tripple buffer method: +-------------+ +-| buffer 1 |-+ | +-------------+ | | | | +-------------+-+ | | buffer 2 | | +-------------+-+ | | | +-------------+-+

Dear all, First, my aim is to achieve VoIP using VBR and DTX under Win32. I face a problem using speex 1.0.4 and need some help. My voice is ok and no background noise when I do NOT normalize 16 bit value to floating value. Normalized means dividing the 16 bit value by 32767. Turning on VBR is also ok but DTX has no effect. However, the speak is has a continous background beep sound AFTER I

Hi! I'm totally new to audio programming. I managed to create a VoIP with waveform (PCM). The problem was the really high traffic, so I desided to use speex for compressing the data. I'm recording with mono, 16bit/sample, 8000 samples per second. I tried to add the speex compression, but I always get a crash when I try to decode the data. Here is my encode and decode function: int

Hi, I am using the speex 1.0.4 library from Windows. I have posted my problem before but didn't get a solution. I am doing an VOIP project in which i am recording sound and streaming it to the peer. I wanted to encode and decode wav files that brought me to this site. I am recording sound in the following format:- m_WaveFormatEx.wFormatTag = WAVE_FORMAT_PCM;

Vinod Vijayan wrote: >Hi, > I am using the speex 1.0.4 library from Windows. > I have posted my problem before but didn't get a solution. I am doing an >VOIP project > in which i am recording sound and streaming it to the peer. I wanted to >encode and decode > wav files that brought me to this site. > > I am recording sound in the following format:- > >

Hi, I am developing a peer-to-peer video conference application which uses speex as a codec for the voice. 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. #define FRAME_SIZE 160 The application is coded in MFC C++. The record

Hi, Can someone please help me with my problem below. Any suggestions is appreciated. thanks, Carine ----- Original Message ---- From: Carine Liang <carineliang@yahoo.com.sg> To: speex-dev@xiph.org; speex-dev@xiph.org Sent: Tuesday, 31 October 2006 1:05:49 PM Subject: [Speex-dev] Integrating speex with VideoNet application: Constant background noise Hi, I am developing a peer-to-peer

Hi John, Thanks for your reply. Yes, my output device is in 8-bit mode, same as my input mode. I'm actually working in char (8 bit) arrays. Is it necessary to work in signed short integers only? I changed all the short arrays in the example to char. Carine ----- Original Message ---- From: John Miles <jmiles@pop.net> To: carineliang@yahoo.com.sg; speex-dev@xiph.org Sent: Thursday,

Hi, I've changed the m_waveFormatEx.wBitsPerSample for both record buffer and playSound buffer, changed my array from char to short and I'm still getting the same constant background noise. I notice that when I talk into the mike, I will get a slightly louder noise, can't hear any clear speech... Any idea what's wrong? Regards, Carine ----- Original Message ---- From: John Miles

Hi, I'm still unable to get rid of the background noise though I've changed the PCM samples to 16 bit. I'm began to wonder if I can just take the PCM samples and encode them. What my application tries to do is to record the voice from a microphone, encode it, send it via the socket to a peer, decode at the peer's side and play the voice. My application works fine without the

--- src/wave_out.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/src/wave_out.c b/src/wave_out.c index 57b5703..0f871b9 100644 --- a/src/wave_out.c +++ b/src/wave_out.c @@ -71,7 +71,7 @@ Box ( const char* msg ) */ static void CALLBACK -wave_callback ( HWAVE hWave, UINT uMsg, DWORD dwInstance, DWORD dwParam1, DWORD dwParam2 ) +wave_callback ( HWAVEOUT hWave, UINT

Hi Ingo, >It is just a WAVEFORMATEX with a GUID in it. For the ACM system it is just a format with the id >WAVE_FORMAT_EXTENSIBLE and since ACM drivers are not registered by wFormatTag it should be working. >For testing purposes I've written a Vorbis ACM driver, that uses a WAVEFORMATEXTENSIBLE and I experienced >a problem when opening files with Media Player. It spend quite some

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

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

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.

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

> 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

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

Hi, While I browsing the source of the Oggenc utility, I saw a comment says: /* A common error is to have an 18-byte format chunk with the last two * bytes 0. This is incorrect, but sufficiently common that we only warn * about it instead of refusing it. * Please, if you have a program that's creating these 18 byte chunks, send * a bug report to whoever makes it */ And it was