similar to: A broken WAV header?

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

Hey I got a server setup where the Postfix and dovecot is on separated servers. In the postfis log is goes like this on mail in: Jul 26 10:23:13 edimailsl2 postfix/smtpd[12812]: 41blTs18Bzz40nl: client=unknown[172.0.0.12] Jul 26 10:23:21 edimailsl2 postfix/cleanup[12825]: 41blTs18Bzz40nl: message-id=<> Jul 26 10:23:21 edimailsl2 postfix/qmgr[10666]: 41blTs18Bzz40nl: from=<tk at

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

Hey I am trying to setup a dovecot cluster with 2 servers using replication /dsync. In front of it I got a Fortinet ADC (Load balance) and I think that I messing up the dsync. I see mails duplicated in the sync progress. If I disable one of the servers in the ADC, it seems to work and the sync if working without a problem. But if I use both servers with a round robin on the ADC, I see mailed

Hi, As a potential user of the Speex ACMs and DirectShow filters to come I am also curious about how many ACM codecs for Speex there are under development right now. Also, the WAVEFORMATEX structure must be planned so that it is extensible, with backward-compatibility, without having to ring back Microsoft for further WAVE_FORMAT tags. For instance, it is a good idea to put a flexible sub-format

Yep, that's the bad news, you need an ID from Microsoft to make sure this ID is only used for FLAC. But since it's a 16 bits value, you can probably put anything randomly (only few are reserved). And I don't think it's illegal to use a number like that. Ingo Ralf Blum wrote: >>WAVEFORMATEX >>yes, that's the way to know what the codec does. >>But no

Prior to having your ACM codec, I was manually compressing Speex audio and writing it into one stream of an AVI file. To be able to use your codec for playback, I simply appended your WAVEFORMATEX extra data (the extra fields in SPEEXWAVEFORMAT) to the audio format. In looking at your code trying to determine how to fill in that data, it looked like the fmtnotice was optional. Is this truly the

My arrangement for a Speex format tag was actually accidental. I was asking what would be involved and instead of an answer, I got a format tag. Since there is only one functional Speex ACM codec available as far as I know, it should use the official format tag. I will be glad to modify our applications to conform to this de-facto standard. Others who may develop ACM codecs in the future

If you want to use the Speex ACM codec in Netmeeting, you need to register the codec with Netmeeting. Microsoft offers an API to accomplish this task. http://msdn.microsoft.com/library/default.asp?url=/library/en-us/netmeet/nm3 codec_1xk3.asp The function is called AddACMFormat in the IInstallAudioCodecs COM interface. It takes a fully initialized WAVEFORMATEX structure as well as an extra

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

Hello, All: What references exist on how to link to C? I'm familiar with sections 5.2 and 5.6 of the "Writing R Extension" manual plus chapter 6 of Venables and Ripley (2000) S Programming (Springer). From these, I get the following: R storage mode C type logical int * integer int * double double * complex Rcomplex *

On Dec 14, 2014, at 10:02 AM, lvqcl <lvqcl.mail at gmail.com> wrote: > Currently the header of a decoded WAV file can be different to the > original WAV file because FLAC doesn't preserve 'fmt ' chunk. > > For example: create a 24-bit stereo .wav file with WAVEFORMATEXTENSIBLE > header with channel mask == 3. Encode it to .flac then decode back to .wav. > FLAC

On Sun, 14 Dec 2014 01:14:16 -0800 Erik de Castro Lopo <mle+la at mega-nerd.com> wrote: > > -------------------------- > > Also, I have a question. > > > > Currently flac complains about 24-bit .wav files if they have > > old WAVEFORMATEX header and not 'proper' WAVEFORMATEXTENSIBLE > > header. However it writes such files itself. > > >

Hello, On the Windows platform, I would like to compile a version of the codec files ogg.dll, vorbis.dll, and vorbisfile.dll that are small. I notice that the provided Windows projects are very large...the release Vorbis.dll is > 1 mb. To me, that is huge. I am only using decoding features. I looked in the Zinf project, and they seem to have a version with the encoding functions

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

The commit http://git.xiph.org/?p=flac.git;a=commitdiff;h=99257e177eac96fa41a107b423080232f59ebe45 also requires some changes in write_iff_headers() function: currently flac don't write WAVEFORMATEXTENSIBLE header if decoder_session->channel_mask is equal to 0, 1 or 3. After the commit 99257e17 flac should do this for channel_mask equal to 0, 4 or 3. The patch fixes this.

Christian, I have successfully integrated your ACM codec in our applications, but I have run into one minor issue. In a code comment you imply that fmtnotice is an optional part of the WAVEFORMATEX extra data. I tried to play a file that had no fmtnotice but Windows did not recognize it as valid and did not play the audio. Your work is very much appreciated here and we are anxiously awaiting

> If I understand what you said correctly, there is such a call: > speex_bits_read_whole_bytes, which adds a couple byte to the SpeexBits > struct while removing the ones that have already been read. Oh, how could I not have seen that one? That is exactly what I need. Thanks for the pointer. >> It is unusual to require a different sequence of API calls >> depending on

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 and a few others are trying to implement Yahoo voice chat under Linux. Since Yahoo uses the TrueSpeech codec, we are trying to load the TrueSpeech DLL directly under Linux. mplayer can play TrueSpeech files already, and it appears that it contains portions of Wine code to do so. But mplayer cannot create TrueSpeech files as far as I know. We have ripped the relevant code out