How to Play MKV Files

Posted by Mohsin | 8:24 AM | | 0 comments »

A MKV or Matroska file is a format that can support a huge number of video, audio, picture or subtitle tracks inside a single file. The menu options on MKV files are quite similar to a regular DVD menu. An encoder is present within this file along with compressed video and audio streams, therefore you will need a codec to read it. Theses codecs can easily be downloaded from the Internet. Its main purpose is to create a format with multimedia storage along with being completely open source. In many ways, MKV is similar to other containers such as MP4, AVI, ASF etc, but it is overtaking other formats such as MP4 and MP3, because of its various advantages. The main advantage and feature of an MKV file is its ability to hold unlimited open-source video, audio and subtitling tracks within the single file format.

MKV files usually have three extensions:

.mkv – Include subtitles and audio
.mka – Include audio-only files
.mks – Include subtitles only

How to play an MKV file?

The first step would be to download an appropriate codec. After downloading and saving, you need to install it on your PC. After installing it, right click on the file you want to open and open with your preferred application. If any error is encountered while opening the file, it is likely that you have not downloaded the correct codec. You can repeat the process again. 

You Can Download ac3filter and VLC Player to Play mkv files with out any problem 

Real Player

Posted by Mohsin | 11:08 AM | 0 comments »

Real Player is the Best of all the players and the point is that it can play any format of the video/audio file by downloading its codec... So Guyz what are you waiting for download and install it into your Computer :)

AC3 Filter

Posted by Mohsin | 8:48 AM | , | 0 comments »

AC3Filter is high quality freeware DirectShow audio decoder and processor filter used to decode audio tracks in movies (DVD, MPEG4 and others). It has a priority on wide functionality and convenient settings

AC3 Filter can decode following audio formats: AC3 / DTS / MPEG Audio. It also supports multi-channel and digital (SPDIF) outputs.

Features of AC3 Filter 1.63 :

- Decoding of AC3/DTS/MPEG1/2 Audio Layer I/II formats 
- Support of DVD, AVI/AC3, AVI/DTS, WAV/AC3 and WAV/DTS 
- Audio processing for any source 
- Decomposition of any source to 6 channels 
- Full information about audio track format 
- Support of SPDIF passthrough mode 
- Multi-channel audio output from all sources on SPDIF (on-the-fly AC3 encoding) 
- Per-channel amplification for all input/output channels 
- Per-channel delays (for compensation of distance difference to speakers) 
- Automatic gain control 
- Clipping 
- Dynamic Range Compression, DRC 
- Level indication for input and output channels 
- Matrix mixer and ability of direct modification of mixing matrix 
- Dolby Surround / Pro Logic / Pro Logic II downmix

Tags :

- Audio Codec Tag 2000 may be requested when AC3 Filter is missing.
- Audio Codec Tag 8192 may be requested when AC3 Filter is missing.
- It also solves AC3 DVM codec missing, reported by AVIcodec utility.

Changes in AC3 Filter 1.63b :

- AC3 decoding bug fixed (Issue 1)
- Channel order bug fixed (issue 43)
- Remote control using messages
- Write version number to the registry on install (issue 17)
- Command line interface for AC3Config utility (run ac3config /? for help)
- Swedish translation added (thanks to Niclas Burgren)
- French language updated (thanks to Philippe AGUESSE)

Changes in AC3 Filter 1.46 :

- AC3 encoder bitrate is now adjustable
- Korean translation added (thanks to starcodec)
- Upmixing of Dolby Surround files works now (thanks to Stefan Schott)
- MPEG Audio Joint stereo playback fixed (thanks to Daniel Bechter)
- Problem with decimal point in mixing matrix on some non-english Windows (thanks to Marcin Hencz)

Changes in AC3 Filter 1.11 :

- Now filter saves output settings to preset (so now you can create presets with different output configurations)
- Tray icon and some other options display fixed

FLAC 1.2.1b

Posted by Mohsin | 6:57 AM | , , | 0 comments »

FLAC stands for Free Lossless Audio Codec. Grossly oversimplified, FLAC is similar to MP3, but lossless, meaning that audio is compressed in FLAC without any loss in quality. This is similar to how Zip works, except with FLAC you will get much better compression because it is designed specifically for audio, and you can play back compressed FLAC files in your favorite player (or your car or home stereo, see links to the right for supported devices) just like you would an MP3 file

The FLAC project consists of : 

- the stream format 
- reference encoders and decoders in library form 
- flac, a command-line program to encode and decode FLAC files 
- metaflac, a command-line metadata editor for FLAC files
- input plugins for various music players

Features of FLAC 1.2.1 :

• Lossless: The encoding of audio (PCM) data incurs no loss of information, and the decoded audio is bit-for-bit identical to what went into the encoder. Each frame contains a 16-bit CRC of the frame data for detecting transmission errors. The integrity of the audio data is further insured by storing an MD5 signature of the original unencoded audio data in the file header, which can be compared against later during decoding or testing. 
• Fast: FLAC is asymmetric in favor of decode speed. Decoding requires only integer arithmetic, and is much less compute-intensive than for most perceptual codecs. Real-time decode performance is easily achievable on even modest hardware. 
• Hardware support: Because of FLAC's free reference implementation and low decoding complexity, FLAC is currently the only lossless codec that has any kind of hardware support. 
• Streamable: Each FLAC frame contains enough data to decode that frame. FLAC does not even rely on previous or following frames. FLAC uses sync codes and CRCs (similar to MPEG and other formats), which, along with framing, allow decoders to pick up in the middle of a stream with a minimum of delay. 
• Seekable: FLAC supports fast sample-accurate seeking. Not only is this useful for playback, it makes FLAC files suitable for use in editing applications. 
• Flexible metadata: New metadata blocks can be defined and implemented in future versions of FLAC without breaking older streams or decoders. Currently there are metadata types for tags, cue sheets, and seek tables. Applications can write their own APPLICATION metadata once they register an ID 
• Suitable for archiving: FLAC is an open format, and there is no generation loss if you need to convert your data to another format in the future. In addition to the frame CRCs and MD5 signature, flac has a verify option that decodes the encoded stream in parallel with the encoding process and compares the result to the original, aborting with an error if there is a mismatch. 
• Convenient CD archiving: FLAC has a "cue sheet" metadata block for storing a CD table of contents and all track and index points. For instance, you can rip a CD to a single file, then import the CD's extracted cue sheet while encoding to yield a single file representation of the entire CD. If your original CD is damaged, the cue sheet can be exported later in order to burn an exact copy. 
• Error resistant: Because of FLAC's framing, stream errors limit the damage to the frame in which the error occurred, typically a small fraction of a second worth of data. Contrast this with some other lossless codecs, in which a single error destroys the remainder of the stream.

Changes from FLAC 1.2.0 to FLAC 1.2.1b :

• General:
- With the new --keep-foreign-metadata in flac, non-audio RIFF and AIFF chunks can be stored in FLAC files and recreated when decoding. This allows, among other, things support for archiving BWF files and other WAVE files from editing tools that preserves all the metadata.
• FLAC format:
- Specified 2 new APPLICATION metadata blocks for storing WAVE and AIFF chunks (for use with --keep-foreign-metadata in flac).
- The lead-out track number for non-CDDA cuesheets now must be 255.
• Ogg FLAC format:
- This is not a format change, but changed default extension for Ogg FLAC from .ogg to .oga, according to new Xiph specification (SF #1762492).
• flac:
- Added a new option --no-utf8-convert which works like it does in metaflac (SF #973740).
- Added a new option --keep-foreign-metadata which can save/restore RIFF and AIFF chunks to/from FLAC files (SF #363478).
- Changed default extension for Ogg FLAC from .ogg to .oga, according to new Xiph specification (SF #1762492).
- Fixed bug where encoding from stdin on Windows could fail if WAVE/AIFF contained unknown chunks (SF #1776803).
- Fixed bug where importing non-CDDA cuesheets would cause an invalid lead-out track number (SF #1764105).
• metaflac:
- Changed default extension for Ogg FLAC from .ogg to .oga, according to new Xiph specification (SF #1762492).
- Fixed bug where importing non-CDDA cuesheets would cause an invalid lead-out track number (SF #1764105).
• build system:
- New configure option --disable-cpplibs to prevent building libFLAC++ (SF #1723295).
- Fixed bug compiling flac without Ogg support (SF #1760786).
- Fixed bug where sometimes an existing installation of flac could interfere with the build process (SF #1763690).
- OS X fixes (SF #1786225).
- MinGW fixes (SF #1684879).
- Solaris 10 fixes (SF #1783225 SF #1783630).
- OS/2 fixes (SF #1771378 SF #1229495).
- automake-1.10 fixes (SF #1791361 SF #1792179).
• documentation:
- Added new tutorial section for flac.
- Added example code section for using libFLAC/libFLAC++.
• libraries:
- libFLAC: Fixed very rare seek bug (SF #1684049).
- libFLAC: Fixed seek bug with Ogg FLAC and small streams (SF #1792172).
- libFLAC: 64-bit fixes (SF #1790872).
• Interface changes (see also the porting guide for specific instructions on porting to FLAC 1.2.1):
- libFLAC:
+ Added FLAC__metadata_simple_iterator_is_last()
+ Added FLAC__metadata_simple_iterator_get_block_offset()
+ Added FLAC__metadata_simple_iterator_get_block_length()
+ Added FLAC__metadata_simple_iterator_get_application_id()
- libFLAC++:
+ Added FLAC::Metadata::SimpleIterator::is_last()
+ Added FLAC::Metadata::SimpleIterator::get_block_offset()
+ Added FLAC::Metadata::SimpleIterator::get_block_length()
+ Added FLAC::Metadata::SimpleIterator::get_application_id()

Elecard MPEG Layer III Audio Encoder provides software-only MPEG Layer III audio encoding solution. It is implemented as a Microsoft DirectShow filter and could be easily incorporated into your capture and network streaming applications.

Features of Elecard MPEG Layer III Audio Encoder:

- DirectShowR interface 
- Property pages 
- Open Source (GNU GPL)

Note: This is a development release, it can crash without warning, has memory leaks and can spoil your display (with movies you never wanted to see), and many things more - but it will not work as expected.

The code uses LAME 3.88 beta MPEG Layer III encoding engine. The original LAME code and filter code could be found at

DTS/AC3 Source Filter reads and passes DTS and AC3 frames to decoders from files...

How to install DTS/AC3 Source Filter :

If you are unable to install the filter, please read this small guide: How to install/uninstall DLL and AX codec files from Guides section.

1. Unarchive the package and use the files from Release folder if you're running Windows 9x/Me or Release Unicode if you're running Windows 2000/XP.

2. Copy DLL or AX files in system32 folder [ C:WINDOWSsystem32 ]

3. Click on Start > Run...

• To install the files, type: regsvr32 filename.dll or regsvr32
• To uninstall the files, use: regsvr32 -u filename.dll or regsvr32 -u

4. In the end, you should receive a message saying that your file was succesfully installed/uninstalled.

An easy way to register/unregister [install/uninstall] the files is to use one of these small tools:
• RadLight Filter Manager - can be used to list all registered codecs and to register or unregister codecs.
• DirectShow Filter Manager: helps to list, sort, find, add or remove DirectShow filters.
• RegShell - a handy utility that will help you avoid the inconvenience of running regsvr32 from the start menu all the time.
• DllRegSvr - allows to register/unregister DLL and OCX codec files using RegSvr32 tool.
• Emsa DLL Register Tool - register dll/ocx filters, codecs... and get extended file informations

DS-MP3 Source Filter 1.30

Posted by Mohsin | 6:48 AM | | 0 comments »

The MP3 Source Filter is a Filesource Filter that uses bass.dll to decode MPEG Layer 3 Files. With it you're able to play MP3 Files from any DirectShow based player: Windows Media Player, Zoom Player, MPC, TCMP and so on.

The reason I build this Filter is because of the buggy MPEG-1 Splitter by Microsoft. Some MP3 files that contains Pictures within Id3v2 Tags, or wrong MPEG Headers caused by bugged Taggers caused that the AsyncSource Filter didn't connect to the MPEG1-Splitter. The MP3 Source Filter is able to play these files and since 1 Filter replaces 3, the Graphbuilding is much faster. 

The Filter renders only Files with the .mp3 Extension. MP3 Streams within other Containers (like AVI, MKV) are still decoded by the FhG Decoder.

If you are using a DivX codec 4, 5 or later for playing DivX 3 contents and you do not have the DivX 3.11 Alpha codec installed this is a tool that you might need.

Some movies are created with WMA sound format which is unrecognizable by the mp3 codecs of DivX versions higher than 3. In such cases your player starts searching the net for the appropriate codec and never finds it because there is no such available for download. 

The only available appropriate DivX audio codec comes with the DivX 3.11 alpha codec installation pack (other repacks also) - DivXa32.acm. So now you've got an unique DivX 3.11 Alpha repack that will install only the missing audio decompressor, will add an item in your audio compression manager (for win 9x - control panel / multimedia / devices / audio compression codecs) and also an option for uninstallation from control panel (add/remove programs). 
This is an advantage over installing the whole of the DivX 3.11 pack because it installs its older video codecs which you do not need and it overwrites the registry and uninstall information which makes it impossible to uninstall DivX 3.11 Alpha and the other later DivX codec version you might have separately. This may cause some undesired behaviour of your codecs and players. All you need to do for hearing the WMA sound of your movie is to install this pack.
I hope you will enjoy it!

Tags :

- Audio Codec Tag 161 may be requested when DivX MPEG-4 Audio Compressor is missing.
- Audio Codec Tag 353 may be requested when DivX MPEG-4 Audio Compressor is missing.

Note :

- GSpot gives the following information with this type of audio codec: divx (same as wma) (0x0161) DivX Networks.

Installing this package you will be able to play Ogg Vorbis, Ogg Speex, Ogg Theora and Ogg FLAC with Windows Media Player or any other DirectShow application, like Media Player Classic, foobar2000, etc.

Known issues :

- If you want to install both 32bit and 64 bit installers on ax64 system you will need to change the path of the second run installer. For example if you run 32bit installer first, you need to change for the 64 bit installer the path from "C:\Program Files (x86)\Xiph.Org\Ogg Codecs\" "to C:\Program Files\Xiph.Org\Ogg Codecs\". This is due to a bug in the installer (Ticket #1474) 
- Can't play file with names that use extended character sets (ie Japanese) under certain locales of windows (Thanks Liisachan)
- Make the installer properly detect if it's already installed. If you don't uninstall the old version you will get multiple identical file masks in WMP
- If you are getting codec problems, there appears to be a conflict with something in Media Player Classic... it tries to enforce the use of another ogg demultiplexer, either by design or accident.If you see in MPC it complaining about a media subtype with a GUID {CDDCA2D5-6D75-4F98-840E737BEDD5C63B}. This is the guid of the old tobias vorbis filter, and this one {8D2FD10B-5841-4a6b-8905-588FEC1ADED9} is CoreVorbis. This is most likely due to the fact that these filters don't clean up the registry when they uninstall. I am working on something to "clean up" all these remnants from the registry (don't expect it too soon though... it's not that high priority). You can also apparently an option to disable the inbuilt demuxer (Thanks Bond)
- Cannot handle any chained multiplexed files or streams.

Changes in Directshow Filters for Ogg Vorbis, Speex, Theora and FLAC 0.83.17220 :

- Updated libvorbis to version 1.3.1
- Updated libogg to version 1.2.0
- Ogg Demux filter has been changed from a source + parser filter to just parser filter. This step allows usage of the following DirectShow filters:
o File Source (Async.) filter which can handle files bigger than 2GB
o File Source (URL) filter which gives better networking support (proxy, ipv6, etc.)

x86:                                                     X64:

DFX Audio Enhancer 9.300

Posted by Mohsin | 4:25 AM | , | 0 comments »

DFX enhances your music listening experience by improving the sound quality of MP3, Windows Media, Internet radio and other music files. With DFX you can transform the sound of your PC into that of an expensive stereosystem placed in a perfectly designed listening environment

Features of DFX Audio Enhancer 9 :

• Harmonic Fidelity Restoration

Adding Fidelity to your audio eliminates the "muffled" sound that is an artifact of the data compression algorithms used in Internet audio formats. DFX compensates for this loss of high frequency fidelity by carefully regenerating the missing high frequency harmonics in the audio. This harmonic restoration is performed using patent-pending technology that carefully synthesizes high frequency harmonics to replace the harmonics lost during the encoding process.
• Ambience, Stereo Imaging
Adding Ambience to your audio compensates for the lost or diminished stereo depth that is the result of closely located speakers, poor listening environments, and sonic losses due to data compression of Internet audio formats. DFX compensates for the reduced stereo imaging and depth of PC audio by carefully regenerating the ambience and stereo depth. This same ambience processing has been used by Grammy winning engineers of artists such as Alanis Morissette and Ella Fitzgerald.
• 3D Surround Sound
Using the 3D Surround Sound component adds amazing depth and body to your audio, compensating for monitoring limitations and sonic losses due to data compression. With 3D Surround processing the sounds will surround you, virtually putting you inside the music! 3D Surround enhances the sound played on conventional 2-speaker systems and makes even small PC speaker systems sound larger and richer. It is also fully compatible with Surround Sound playback systems.
• HyperBass
Using the HyperBass component adds deep, rich bass sound to your audio by compensating for the bass limitations of almost all PC-based audio systems and data-compressed audio formats. DFX does this by carefully regenerating the low frequency harmonics, thus increasing the perceived bass, but without exceeding the speakers physical limits, adding or changing speakers or amplifiers, and without increased power consumption. HyperBass greatly improves the bass performance of any sound system.
• Dynamic Gain Boosting
Adding Dynamic Boost increases the perceived loudness of your audio while minimizing distortion levels, particularly when using Internet audio systems. By adding Dynamic Gain Boosting, your playback system will sound twice as loud without losing any punch on loud passages.
Typical PC-based multimedia playback systems and even home stereo playback systems suffer from limited dynamic range and headroom. Dynamic Boost compensates for this limited headroom by carefully processing the audio to increase the perceived loudness of the audio without altering the perceived dynamic range.

Changes in DFX Audio Enhancer 8.501 :

- Enhanced stylish interface 
- New vertical mini-mode which nicely integrates with media players 
- Better sound enhancement processing 
- More informative audio status messaging within the user interface

Important Note :

DFX Audio Enhancer installer is bundled with "Ask Toolbar". However, you may continue the installation process without installing it.

DC-Graphic EQ Filter 1.10

Posted by Mohsin | 4:17 AM | , | 0 comments »

The DC-Graphic EQ Filter is a 31-Band Graphic Equalizer for MS DirectShow. The Frequencytable is a slightly modified ISO Table to prevent Overlapping.

The Filter will work on 8, 16, 24, 32 Bit Integer as well as on 32 Bit Float. The Internal DSP is done in 32 Bit Floatingpoint and is optimized using SSE and SSE3 instructions.

The Filter is also setup to connect to SPDIF and DTS Types in case you are playing around with AC3 Filter. This will prevent the typical "cannot connect Pins" Error. However, in DTS/SPDIF Mode, the Audiodata will be bypassed and not modified.

DC-DSP Filter 1.03

Posted by Mohsin | 4:16 AM | , | 0 comments »

DC-DSP Filter is a Microsoft DirectShow 9 Audio Filter which is able to Process (decoded) 8/16/24/32 Bit Integer PCM and 32 Bit IEEE Float Audio Data.

It works on every Win32 Platform (except NT) with DirectX 9 or above installed.

Features of DC-DSP Filter 1.03 :

- A Dynamic Filter list that allows to use as much DSP Filters as needed.
- Multiple Instances are possible.
- Can export Visualazation and PCM Data to any Application (see Sample Apps in the source\samples Directory).
- Supports up to 10 Channels. 
- Most Filters can be configured to do seperate DSP on each Channel.
- Each combination of Filters can be saved into a Preset.
- Winamp2 DSP and Visual Plugin Support (Note that not every Plugin is working!).
- Audiotrack delaying through Timestamps changing.
- Bitrate Conversion.
- Stream Switching through IAMStreamSelect with Matroska, OGM and AVI Language Detection. 
- Supports different MediaTypes on each Pin. 
- Works in Windows Media Player 9 and 10. 
- Trayicon for fast access to the Property Page.

DSP Filters :

Amplification; Band Pass; Channel Reordering; Compressor; Down Mix; Dynamic Amplification; Echo/Delay; 10 Band Equalizer with Presets (Load/Save); Flanger; High Pass; Low Pass; Notch; Parametric Equalizer; Phase Invert; Phaser; Pitch Scale; Pitch Shift; Sound 3D (2 Channels only); Tempo; Treble Enhancer; True Bass 

MS DMO Filters :

DMO Chorus; DMO Compressor; DMO Distortion; DMO Echo; DMO Flanger; DMO Gargle; DMO I3DL2 Reverb; DMO Param EQ; DMO Waves Reverb

How to use :

When loading a File (Movie or Audio) in your MediaPlayer you should see "DC-DSP Filter" in the Filtergraph´s List. You can control the Filter through the Propertypage.

Changes in DC-DSP Filter 1.03 :

The new Version has a better check for previous Instances, integrates a faster Memory Manager and, thanks to Jason Huang, an improved Compressor Filter. A few Bugs has been fixed too ...

DC-Bass Source Filter 1.1.1

Posted by Mohsin | 4:14 AM | , | 0 comments »

The DC-Bass Source Filter is a BASS based DirectShow Audio Decoder.

Currently supported Formats are :

• Shoutcast: MP3, AAC, OGG
• Tracker: MO3, IT, XM, S3M, MTM, MOD, UMX

Shoutcast Streams can be saved from within the PropertyPage or by using the IDCBassSource Interface.

Shoutcast Tags can be read using the IAMMediaContant or IDCBassSource Interface.

CoreWavPack 1.1.1

Posted by Mohsin | 4:10 AM | | 0 comments »

CoreWavPack is a set of DirectShow filters - & - created to support the WavePcak Audio Codec

CoreWavPack allows playing the WavPack files (.wv .wvc) in any DirectShow based media player Windows Media Player, Zoom Player, MPC and TCMP.

This filter also allows WavPack embedded in the Matroskacontainer.

Changes in CoreWavPack 1.1.1 :

- fix bug in WMP extension registration / un-registration (reported by Nicholi)


Posted by Mohsin | 4:07 AM | | 0 comments »

CoreVorbis is a DirectShow-Decoder for the Vorbis audio codec It allows you to play files containing Vorbis-audio in many players, for example Windows Media Player, Media Player Classic or The Core Media Player.

Vorbis is a fully open, non-proprietary, patent-and-royalty-free, general-purpose compressed audio format for mid to high quality (6kHz-96kHz, 16+ bit, polyphonic) audio and music at fixed and variable bitrates from 2 to 256 kbps/channel. This places Vorbis in the same competitive class as audio representations such as MPEG-4(AAC), and similar to, but higher performance than MPEG-1/2 audio layer 3 (MP3), TwinVQ (VQF), WMA and PAC.

Changes from CoreVorbis 1.0 to CoreVorbis 1.1

• Notes:
- Better mediatype change handling.
- Compiled with last version of the vorbis library.
• Changes:
- Better mediatype change handling.
- Compiled with last version of the vorbis library

CoreFlac is a DirectShow-Decoder for the Flac audio codec, it allows you to play files containing Flac audio in many players, for example Windows Media Player, Media Player Classic or The Core Media Player.

All DirectShow-based players. This should be 95%.

Changes in CoreFLAC Decoder 0.4

- Fix problem when reaching end of file, and restarting.

Encoder:                                                                Decoder:


CoreAAC is an AAC DirectShow filter decoder based on FAAD2.

AAC means "Advanced Audio Coding", and in the beginning it was also called MPEG-2 NBC for "Non-Backwards Compatible" as opposed to the MPEG-1 and MPEG-2 BC (with 5.1 channels) standards.

It is now considered to be the actual "state of the art" in general audio coding and the natural successor of MPEG-1/2 Layer III / MP3 in the new multimedia standard MPEG-4 that uses MP4 as the container format for all kinds of content.

AAC is able to include 48 full-bandwidth (up to 96 kHz) audio channels in one stream plus 15 low frequency enhancement (LFE, limited to 120 Hz) channels and up to 15 data streams. Besides it has further multi-language capacities.

MPEG formal listening tests demonstrated that AAC provides an audio quality at 96 kbps which is slightly better than MP3 at 128 kbps and MP2 at 192 kbps.

Convolver 4.4

Posted by Mohsin | 3:58 AM | | 0 comments »

Convolver is a DSP Audio convolution plug-in for Windows Media Player or any application that accepts DMOs or DirectShow filters. Passes what is being played back through a user-provided FIR (Finite Impulse Response) filter (to provide room-corrected sound, for example).

It will take a set of single-channel impulse response files and convolve them with the input stream.

Why would you want to do this? With a suitable impulse response generated by a tool such as DRC you will be able to play sound corrected for your room response. Read morehere.

You will also be able to generate cross-overs, equalizers and other tools that require the source signal to be filtered and redirected to different output channels.

How to install Convolver 4.4 :

1. Run Setup.exe to install the installation package version.
2. In Windows Media Player (10, it wasn't tested with earlier versions), select Tools > Plug-ins > Options. On the Plug-ins tab, select Audio DSP.
3. Select Properties to configure the plug-in.
4. Select Get config ... and pick up your configuration file (config.txt). [The configuration file will be replaced by a user interface after further testing.]

AudioCD Reader 1.0

Posted by Mohsin | 3:56 AM | | 0 comments »

AudioCD Reader enables AudioCD playback for every DirectShow based player.

Aud-X surround codec 1.24

Posted by Mohsin | 3:54 AM | | 0 comments »

Aud-X consists of 3 independent parts:

- Aud-X executable file encoder decoder

- Aud-X DirectShow decoder filter

- Aud-X ACM encoder (for VirtualDubMod and other Audio Compression Manager compatible software).

DirectShow Filter ensures compatibility with the most popular multitedia software players utilizing Directx (this is around 90% of all). It is responsible for reading Aud-X content present in movies or separate Aud-X soundtracks.

ACM Encoder has been implemented to make movie encoding possible with the use of Audio Compression Manager compatible software (eg. VirtualDubMod)

Executable Encoder/Decoder allows for: 

- Independent sound compression (demuxed movie soundtracks, or multichannel recordings), 

- Conversion from Aud-X to 6 channel PCM or AC3 format.

Changes in Aud-X surround codec 1.24 :

- [1.24] - MP3 48 kHz problems fixed 

- [1.24] - PseudoSurround artifacts for noisy signals eliminated

ATSurround Processor 1.02

Posted by Mohsin | 3:47 AM | | 0 comments »

Many songs today found on CDs, MP3s, etc. have surround information encoded. Even when played on a 5.1 PC system, the surround channels may not be extracted from the audio as it is processed from start to finish as a 2 channel source. What you get as a result is simply many speakers playing exactly the same sound, which does not enhance the audio. A surround decoder such as the ATSurround Processor is required to extract the encoded surround information for playback.

The process of extracting surround channels and routing it to the appropriate speaker is known as matrix decoding. Hardware and software decoders are available for this purpose. The ATSurround Processor is a software decoder designed to work as a plugin in the foobar2000 and Winamp players. Multichannel output capable soundcard and speakers are required.

The Headphone mode allows the listening of audio sources which sound like they're really coming from a live 5 point surround system. Music no longer sounds like it's "inside" your head, listening fatigue is thus reduced. Perfect when you don't want to disturb anybody around you by using headphones, but don't want to sacrifice the surround decoding capability of ATSurround. 

ATSurround Processor is also capable of performing matrix encoding. The Processor receives multichannel audio and matrixes it into stereo . This stereo output may be distributed via any 2 channel medium, such as CD-Audio and MP3 or direct to an external audio receiver which performs audio surround decoding, useful when the receiver is connected using a stereo cable. 

Features of ATSurround Processor :

- Reproduces surround sound information present in many stereo audio material 

- Headphone mode allows listening of audio in virtual surround 

- Compatible with Dolby® or similar processors 

- Makes full use of your 5.1 speaker systems 

- Bass redirection for discrete 5.1 satellite systems 

- Surround channel delay 

- Channel amplitude trim 

- Converts multichannel audio into surround encoded stereo audio

Changes in ATSurround Processor 1.01 :

- Fixed: LFE redirection, channel trims and channel delays now do not affect ATSurround Encoder and Bypass output

- Fixed: In some situations, Winamp crashes when quitting after using ATSurround with Headphone mode

- Fixed: The 'X' button on the configuration dialog now acts like 'Cancel'

- Fixed: Channel trim settings above 6dB or below -6dB did not save properly

Changes in ATSurround Processor 0.1.6a :

• ATSurround Processor 0.1.6a for foobar2000

- Fixes bug causing foobar2000 to remain in memory after closing.

• ATSurround Processor 0.1.6 for Winamp and foobar2000

- Added system tray icon to change processing modes.