This is mainly a bug fix release. A nasty bug on increasing processing sizes would corrupt the input data and thus change the results. It is advised to upgrade to this release as soon as possible.
This is the first stable release of the Audio Toolkit, after more than a year of development. In addition to the serial pipeline, there is now an option to use TBB to render each chunk in parallel. The pipeline can also return the maximum latency the pipeline possesses if all latency information is given during the build of the pipeline.
Additional filters were also added to complement the current set of filters.
There are several different low pass filters, and as many high pass, band pass, band stop… filters. In Audio toolkit, there are different usual implementation available:
- Chebyshev type 1
- Chebyshev type 2
- Second order
and it is possible to implement other, different orders as well…
Continue reading Audio Toolkit: Different low pass filters
Focus on this release was on performance. As such the core functions were optimized, as well as some tools and EQ.
A new filter dedicated to fast convolution (using a fixed-size partition with a mix of FFT convolution and explicit FIR filter) with 0 latency was added.
Convolution is an algorithm that is often used for reverberations. If the equation is easy to understand and to implement, the implementation is costly. The other way of doing it is to use Fast Fourier Transform (FFT), but the direct/crude implementation requires latency. If it is possible to optimize the basic convolution code, it is sometimes more interesting to use a different algorithm, as it is the case here.
When I first read about transient shaper, I was like “what’s the difference with a compressor? Is there one?”. And I tried to see how to get these transient without relying on the transient energy, with a relative power (ratio between the instant power and the mean power) filter, or its derivative, but nothing worked. Until someone explained that the gain was driven by the difference between a fast attack filtered power and a slower one. So here it goes.
The main changes for this release are first trials at modulated filters, C++11 usage (nullptr, override and final), and some API changes (the main process_impl function is now const).
Sometimes images are worth a thousand words, so let’s look at some pictures of a middle-side compressor behavior.
Continue reading Audio Toolkit: Anatomy of a middle-side compressor
I’m happy to announce the release of a stereo compressor based on the Audio Toolkit. It is available on Windows and OS X (min. 10.8) in different formats. This stereo compressor can work on two channels, left/right or middle/side, possibly in linked mode (only one set of parameters), and can be set up to mix the input signal with the compressed signal (serial/parallel compression).
Nice title, surfing on the many core hype, and with a practical approach! What more could one expect from a book on such an interesting subject?