similar to: ACM using TrueSpeech codec (acmStreamOpen())

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, 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

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

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

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

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, 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

Hi (again), I started to work on an ACM (Audio Compression Manager) for FLAC. I currently only have the canvas. The encoding/decoding is not done yet. So it's completely useless. But since the canvas is the harder thing to start, here it is. (just in case I don't have time to continue) http://mukoli.free.fr/flac-ACM.v0.1.0.zip

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

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

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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.

Hi, I am trying to implement a web application that allows the user to playback WAV files inside a web browser. __________ Background __________ - The file server stores audio files as GSM 6.10 or TrueSpeech encoded WAV files. - The web server (most probably ASP.NET on IIS) offers the client the ability to search audio files on the file server (a different machine). - The client will be

I have arranged for an official Speex ACM format tag from Microsoft (see below). Feel free to use this in any ACM codec work. Keith Looney ________________________________________________ Keith Looney Senior Software Engineer On2 Technologies, Inc. The Duck Corporation <p><p>-----Original Message----- From: Bill Schiefelbein [mailto:billsch@windows.microsoft.com] Sent: Wednesday,

> 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