If its not relevant here, pls. move to correct place.
I want to find out which all vendors/companies have developed multi-threaded video codecs(decoders , encoders) as commercial products? Not opensource solutions like libavcodec/x264/ffdshow etc... but commercial solutions for which one can obtain licenses/performance numbers of those solutions.
thanks,
-AD.
MainConcept has some excellent codec options. They offer support for multi-threading as well as support for hardware accelerated encoding:
MainConcept Codec SDK
In my opinion they offer the best performance and quality (no I do not work for MainConcept).
Related
Will Realitykit support an ambisonic recording? I am planning on using binaural audio in an AR experience but it looks as if RealityKit doesnt support it. Looking for help.
At the moment RealityKit 2.0 doesn't support Ambisonics. It's hard to say whether it'll support it.
I was wondering where I may find a good C/C++ implementation of the 3DMC mesh coding algorithm which appears in the MPEG4 standard.
Thanks,
Debarati.
http://www.mymultimediaworld.com/
is the place,
SC3DMC is the exact standard, it features faster decoding than the older 3DMC standard.
The implementation works well in windows, in linux some tweeks are required,
Best,
Rufael
I am looking for an audio dsp library for cleaning up some speech (voice) recording. I have not decided which language to use yet.
Here are the feature I am looking for:
Work in Linux and Windows
Importing MP3
Working with multiple channels mixing
Noise Filter
Bandpass filter
Compressor
I love to have these as well, but I can write my own if they are not available:
De-esser
multi-band compressor
Expender
Envelopes
(if you can suggest an application that do these in scripting / one mouse click, I will accept your answer too)
What about something like SoX?? http://sox.sourceforge.net/
Take a look at Juce from Raw Material Software.
It is free for non-commercial use, and very reasonably priced for commercial use. it also has a lot of built in audio capabilities (mixing, file I/O, etc.) and has a nice cross platform GUI toolkit as well.
Audacity does most of those things.
I'm looking for an audio processing language or library which will allow me to experiment with different synthesis techniques. I've looked at Processing which I think is great at what it does, but haven't found any inspiring (and simple) audio libraries.
As a baseline, I want to simply create my own sample buffers and play them back (ideally in realtime). As a plus, the ability to handle MIDI events would be great. I'm an experienced C++ programmer so I could do it natively on but had hoped there was a more DSL (domain specific language) approach.
I have access to Windows, Mac or Linux so not too bothered yet about platform. Other languages I can deal with are C#, Java & Python.
Thanks
James
Depending on how much you want to stay out of the low-level housekeeping details, you may want to look at CSound , or if you want to not actually write code, the patching-based system PureData is great to work with. As #Lou points out, ChucK is interesting (but was too buggy to use the last time I checked it out).
If you really do want to write code, look at the Synthesis Toolkit, a set of C++ classes for audio processing and synthesis.
For an app framework, I recommend JUCE, which has incredibly nice cross-platform handling of audio/midi IO and GUI elements.
Max MSP is an audio production tool that is highly expressive.
I guess you could say it's a high-level tool, and not a low-level programming language. My impression of it is that it's geared towards the technical musician or the artistic engineer, but anyway it kicks ass and you could go low-level with it if you want.
I've always been a big fan of SuperCollider. It's designed for Mac OS X but also works on Linux.
The language is mostly based on SmallTalk, and it's pretty easy to pick up if you understand the basics of functional programming. The quality of the sound output by the SC Server is very good and there is plenty of documentation both built into the app environment and available online.
One interesting point of SuperCollider is the usage on android devices, and it's intercommunication with python trough out other modules.
Here goes an example
I know you didn't say Ruby, but check out Archaeopteryx
https://github.com/gilesbowkett/archaeopteryx/wiki
or ChucK
http://chuck.cs.princeton.edu/
Have a look at NAudio, an open source .NET audio SDK for working with audio files and devices in Windows. Some features include:
http://naudio.codeplex.com/
NAudio Features:
Play back audio using a variety of APIs
Decompress audio from different Wave Formats
Record audio using WaveIn, WASAPI or ASIO
Read and Write standard .WAV files
Mix and manipulate audio streams using a 32 bit floating mixing engine
Extensive support for reading and writing MIDI files
Full MIDI event model
Basic support for Windows Mixer APIs
A collection of useful Windows Forms Controls
Some basic audio effects, including a compressor
We have some raw voice audio that we need to distribute over the internet. We need decent quality, but it doesn't need to be of musical quality. Our main concern is usability by the consumer (i.e. what and where they can play it) and size of the download. My experience has shown that mp3s do not produce the best compression numbers for voice audio, but I am at a loss for what the best alternatives are. Ultimately we would like to automate the conversion process to allow the consumer to choose the quality vs. size level that they would like.
You should give Opus a try. Example compression command line:
ffmpeg -i x.wav -b:a 32k x.opus
Start here.
As you rightly point out, voice compression is different from general audio compression. You'll find many codecs dedicated to telephony applications, ranging from PCM and ADPCM through later packet based encodings such as CELP used on GSM cellular networks.
Still, VOIP voice encoding is slightly different from that due to the medium used. you can find a good, free (unencumbered and open source (BSD)) library for speech encoding/decoding in the Speex software library.
Again, which you choose depends on the speech you're encoding and the medium it's being transmitted over. Also note that many libraries have several algorithms they can use depending on the circumstances, and some will even switch on the fly based on conditions of the sound and network.
To get more help, narrow your question down.
-Adam
The most frequently used compression formats used in live voice audio (like VoIP telephony) are μ-Law (mu-Law/u-Law is used in the US) and a-Law (used in Europe, etc.) which, unlike Uncompressed PCM, don't support as wide of a frequency range (a smaller range of possible values ignores sounds outside of the necessary spectrum and requires less space to store).
For usability sake it is easiest to use mpeg compressions (mp2/3/4) for streaming to standard media players as the algorithms are readily available and typically quite fast and almost all media players should support it, but for voice you might try to specify a lower bitrate or do your conversion from a lower quality file in the first place (WAV can be at several sampling rates and voice requires a much lower sampling rate than music or effects, it's basically like frame-per-second on video). Alternatively you can use Real Media, WMA or other proprietary formats, but this would limit usability since the users would require specific third party software for playback, though WMA has an excellent compression ratio as well as compression options specific to voice audio.
Assuming your users will be running Windows, there is a WMA speech compression codec that you can use with the Windows Media Encoder SDK. Failing that, you can use ACM to use something like G723/G728, ADPCM, mu-law or a-law, some of which are installed as standard on Windows XP & above. These can be packaged inside WAV files. You'll need to experiment a little to find the right bitrate/quality (probably don't bother with mu-law or a-law). With voice data you can get away with quite low sample rates - e.g. 16000 or 8000, as there isn't much above 4Khz in the human spoken voice.
I think AMR is one of the best speech codecs. I was using it about a year ago and I remember that quality was very good and size levels were rather small.
One drawback, especially in your case is that, as far as I know, it isn't supported by wide range of media players. QuickTime and RealPlayer are two which I know to play .amr files.
Try speex ... unencumbered by patents, good performance both sizewise and CPU-wise. I've been having good luck using it on iPhone.