similar to: Wrong warning in encoder for 24bits WAV

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

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;

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

Hi Josh, Here are all the patches that are currently applied to the Debian packages for flac. You may disagree with some of them but as the current maintainer, I'm submitting them all for your consideration. (I should've submitted them earlier, but I've been quite busy. Sorry!) Here is a listing of the patches that I'll be sending: Patch 1 fixes a simple typo in metaflac

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 am attempting to use flac to encode 96k/24-bit broadcast wav (BWF) files. BWFs are wav files with some extra meta-data chunks, and is the favoured archival format for many institutions around the world. These files are encoded successfully by flac, however the resulting flac file is not playable on all flac players - it plays successfully in foobar2000 but is silent in winamp, and when

A friend of mine recently had a problem with a "broken" .WAV file (as you call them) because oggenc first printed out a warning and then didn't accept the file because of a "unexpected EOF error". Because I was interested in the issue, I decided to take a look at the oggenc source, and in fact, it is your .WAV reader that's wrong. More precisely: there are two version

Hi, >That's what I wanted to say a month ago. Mr. Blum, I'm really sorry for misunderstanding you. My mistake. I choose an ACM codec because it's the only option that works on every Windows Platform, and since the DirectShow has a generic wrapper that allows it to use ACM codecs, it appeared a good idea. I didn't note that the file format was going to be a problem. But I think

Can you please wrap the setvbuf in _WIN32 IFDEFs too? Currently memory usage of FLAC decoding is about 1MB, so this patch is increasing memory usage tenfold, also for platforms that do not need this. It is a non-problem on my system anyway. Op 26-09-14 om 10:36 schreef Janne Hyv?rinen: > I made some changes to the previous patch. I don't know why I > originally didn't put the

Decoding flac files is also prone to producing fragmented files. NTFS has the ability to completely avoid fragmentation if it is told the file size before hand, but that would require using special Windows-only functions. Increasing the write buffer from the default 512 bytes to 10 MB already reduces the problem tremendously. -------------- next part -------------- diff --git

Hello, I've tried encoding a FLAC file from a WAVE file saved with 24-bits-per-sample, using the flac executable. I get this warning: WARNING: legacy WAVE file has format type 1 but bits-per-sample=24 The code is currently here on line 373: src/flac/encode.c where the comment says let these slide with a warning since they're unambiguous I do not really understand the warning

Hi all, I recently managed to install "multimedia German dictionary", which I get with the local newspaper , under wine and it works OK. The problem it is when I try to use the most important feature for me - the pronunciation. When I try to listen to any word , no sound is played and wine outputs the following to the console: wodOpen unimplemented format: WAVE_FORMAT_ADPCM I rewieved

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, Is there a defined format tag for Opus that should be used with the WAVEFORMATEX structure? See wFormatTag (http://msdn.microsoft.com/en-us/library/windows/desktop/dd390970.aspx). /John

I made some changes to the previous patch. I don't know why I originally didn't put the output buffering to piped output too but that is now moved to cover both file and pipe output. Additionally this patch informs the Windows filesystem in advance about the decoded size to eliminate NTFS fragmentation. On 25.9.2014 23:01, Janne Hyv?rinen wrote: > Decoding flac files is also prone

On Fri, Jun 20, 2014 at 01:21:03PM +0400, lvqcl wrote: > Miroslav Lichvar ?????: > > > +/* > > + * This is used to avoid overflow with unusual signals in 32-bit > > + * accumulator in the *precompute_partition_info_sums_* functions. > > + */ > > +#define FLAC__MAX_EXTRA_RESIDUAL_BPS 4 > > > + /* WATCHOUT: "+ bps +

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

>> Is there a defined format tag for Opus that should be used with the >> WAVEFORMATEX structure? See wFormatTag >> (http://msdn.microsoft.com/en-us/library/windows/desktop/dd390970.aspx). > > There isn't one. Putting Opus into RIFF is complicated by the fact that > it requires both variable-duration and variable-size frames, though I > suppose something could be

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