Basic Dynamic Range Compressor

My last post described the need for dynamic range compression in hearing aids: if you amplify enough to hear quiet sounds, loud sounds will become too loud.  One solution is to include a Dynamic Range Compressor (DRC), which changes the gain depending upon the loudness of the signal.  Because of the DRC, loud sounds will be amplified less than quiet sounds.  Perfect!  In this post, I describe how I implemented a DRC and I show its effect on some simple signals.

Algorithm Overview:  In the figure above, I illustrate the basic signal flow through my DRC algorithm.  It's a feed-forward design with a side-chain that computes the time-varying amount of gain that should be applied.  The function of each block is described below:

• Pre-Process Signal:  In this block, I apply the "pre-gain", which is the amount of gain that will be applied when the compressor is in its linear regime.  Also in this block, I use a high-pass filter to remove any DC offset in the audio signal.
• Level Estimator:  For the compressor to vary the gain based on the loudness of the signal, it needs to first estimate the loudness of the signal.  That's what this block does.
• Gain Calculator: Using the estimated loudness, this block calculates how much to reduce the gain of the system.  It calculates the desired gain reduction knowing the "compression ratio" and "compression threshold" that have been supplied by the user.  It also smooths the compressed gain value through time via "attack" and "release" time constants that have been supplied by the user.
• Apply Gain: Once the desired compressed gain value has been calculated by the side-chain, this block applies the gain to the audio signal.

Arduino/Teensy Implementation:  I implemented this algorithm as an C++ class that can be called from any Arduino or Teensy program (see AudioEffectCompressor_F32 in my GitHub library here). It is built upon my F32-extension of the Teensy Audio Library.  Because it is so reliant on floating-point operations, it is probably only appropriate for use on the Teensy 3.5 or 3.6 (which have floating-point hardware support) and not the older 3.0, 3.1, or 3.2.

Algorithm Parameters:  In the signal flow diagram below, I show the algorithm parameters that are available for the user to set.  The main tricky part is how to set the time constant for the Level Estimator.  By default, I have this value always scale itself to be 20% of the time constants that the user has set for the Gain Calculator block.  If you don't like this default behavior, you can specify your own value using the method setLevelTimeConst_sec().

Example Sketch:  As part of including this DRC algorithm in my OpenAudio_ArduinoLibrary (GitHub page here), I made sure to include an example sketch called BasicCompressor_Float.  This example code assumes that you have a Teensy Audio Board along with a Teensy 3.5/3.6.  Within the code, there are a couple of options:

• USB Audio:  For my testing, I chose to send and receive audio over the USB link, instead of via analog audio cables.  See this post, if you want more info on how to use the Teensy's USB Audio link.  To enable USB audio in this sketch, set DO_USB to a value of one.
• Fast or Slow:  Another choice is how to set the values for the compressor's parameters.  In my example, I have two sets of parameters.  One set gives a "fast" compressor response that is appropriate for quick limiting of very loud sounds.  The other set of values gives a "slow" response that can be used as a automatic volume control.  For my testing, I chose to use the "fast" response, as shown below.

Test Setup:  For my testing, I'm using my breadboard prototype of my Teensy Hearing Aid, which is just a Teensy 3.6 with the Teensy Audio Board (plus microphones, battery, and Bluetooth module, which are not relevant to this test).   I've got it plugged into my laptop via USB, which is also used to send and receive audio to the Teensy via the Teensy's ability to do USB Audio.  On my laptop, I'm using the Arduino IDE to set the compressor's parameters (again, I'm using the "fast" configuration) and I'm using Audacity to send the audio to the Teensy and to record the audio returned by the Teensy.

Testing, Amplitude Sweep:  For my first test, I generated a steady 2kHz tone whose amplitude increases from quiet to loud at a rate of about 6 dB/sec (my test signals are shared here).  This is the signal that I'm sending to the Teensy (also shown in blue in the figure below).  When sending this signal, the compression algorithm on the Teensy returns the signal shown in orange.  If you look at the loudest portion of the audio (starting around time = 8 seconds) the amplitude of the orange output signal looks to be squashed compared to the blue input signal.  This is the effect of the compressor!  It successfully reduced the dynamic range of the audio, as intended.

Quantifying the Response:  One important question is whether the compressor kicked in at the correct amplitude.  To answer this question, I assessed the instantaneous amplitude of each signal, as shown in the bottom plot in the figure above.  It clearly shows that the compression of the output signal begins at an amplitude that is 15 dB below full-scale.  This is exactly the value that was specified in the example code by the call to setThresh_dBFS().  It is pleasing to see that the algorithm works as intended.

Testing, Step Changes in Amplitude:  Another important question is whether the compressor is responding at the correct speed to changes in signal amplitude.  To answer this question, I generated a test signal with instantaneous step changes in amplitude, as shown in blue in the figure below.  The compressor's response is shown in orange.

Quantifying the Attack and Release:  When I extract the amplitude of each signal versus time (bottom plot), you can see that the compressor lags in its response to the step changes.  When the input signal gets louder (say, at time = 2 sec), the compressor's 5 msec "attack" time constant allows the compressor to response very quickly.  But, when the input signal gets quieter (say, at time = 3 sec), the compressor's 200 msec "release" time constant makes it respond more slowly.  In my code, these time constants are used in the traditional sense of "time constant", meaning that 63% of the change (in dB) is achieved in one time constant.  Based on the figure above, it appears that my compressor is responding at the correct speed.  I'm pleased.

Next Steps:  With the dynamic range compressor working correctly (and sounding pretty good when used with my Teensy Hearing Aid's microphones and my headphones), I'm well on my way to completing the signal processing hardware and software for a very basic hearing aid.  The main component that I'm still missing is some sort of frequency compensation to increase the gain on the frequencies that the listener needs to hear better, while reducing gain on those frequencies that the listener already hears well enough.  Stay tuned!

Follow-Up:  This algorithm converts to dB and back.  When doing the log10(x) and pow(10,x), be sure to call the correct versions and they'll run *much* faster.  Check it out here.

Received My First Pull Request!

After my post about my floating-point extension of the Teensy Audio Library, I received my first Pull Request!  Someone besides me is working on the library, and that is such a complement.  Thanks, @patrick-radius!

Converting Existing Blocks to F32:  For his first contributions, Patrick created AudioMixer4_F32 and AudioMultiply_F32.  His approach was to take blocks from the existing Teensy Audio Library (their GitHub is here) and create F32 versions.  Since many people are already familiar with the Teensy Audio Library, this is a great approach for quickly adding more floating-point capability to the library (and to your own project!).

Suggestions:  If you want to make your own contributions to this floating-point library, I'd be very happy to receive them.  Like Patrick, following the model of the Teensy Audio Library is a great approach.  You could follow both the conventions as well as the style of the Teensy Audio Library.  The primary convention that is different with my F32 library is that:

• Full-scale for the floating-point algorithms is +/-1.0 instead of +/-32768.  So, if you are copying algorithms from the Teensy Audio Library, you don't need to include scale factors or gain limits with values such as 32768 or 255 or 127.

Furthermore, if you want to contribute to my library, may I suggest that you be sure to:

• In your *.h file, include a comment block at the top that identifies the name of the class and you as the creator.  For an example of such a header, see AudioEffectGain_F32.h.
• Once your class works, be sure to put an #include for your *,.h file into the library's overall header file: OpenAudio_ArduinoLibrary.h.  Also, add your class as a keyword in keywords.txt.

Example Sketch:  When contributing audio blocks, consider making an example sketch so that I (and others) can see how it works.  Your example sketch should go in the examples directory of this library.  

Stereo to Mono Example:  Patrick did not include an example sketch with his contributions, so I made one.  I made an example using his   My example,  MixStereoToMono_Float, uses his new AudioMixer4_F32 class to take a two-channel (ie, stereo) input and mix them into a single-channel (ie, mono) output.  By including such an example, a new person can see how to use your new audio processing block.

Testing Using Audacity:  To test my example sketch, I used Audacity to generate some test audio and to send the audio to my Teensy 3.6 via the Teeny's USB audio connection.  For my test audio, I created an upward sweeping chirp for the left channel and I created a steady 2 kHz tone for the right channel.  In the screenshot below, the yellow arrows point to the panning controls that I used to force my audio into the left and right channels.

Result:  Hitting the record button, Audacity played the stereo audio out to the Teensy while simultaneously recording the audio being output by the Teensy.  As you can see, the audio produced by the Teensy was the mixture of the left and right inputs.  Success!

Many Thanks:  I'm so appreciative for Patrick's high-quality contributions to this library.  Many, many thanks.  And, from him or from anyone elese, I greatly look forward to additional Pull Requests!

Extending Teensy Audio Library for Floating-Point

For my Teensy-based hearing aid, it will be much easier to program new audio processing algorithms if I can use floating-point aido data instead of fixed-point audio data.  Unfortunately, the Teensy Audio Library assumes the use fixed-point data types (ie, Int16) for all of its processing blocks.  So, I decided to extend that library to enable the floating-point processing that I want.  Here's an overview of how I made it work.  My "OpenAudio" library is available on my GitHub here.  I'd love any feedback (or GitHub pull requests!) on how it could be done better.

Teensy Audio Library Assumes Int16:  My goal has been to maintain as much of the Teensy Audio Library's structure as possible.  But, many of the core elements of the Teensy Audio Library are deeply entwined with the assumption of fixed-point Int16 audio data.  Shown in red in the figure above are several of the foundational elements of the Teensy Audio Library that are tied to Int16 audio data.  These are the elements that I extended.

OpenAudio F32 Versions:  To enable Float32 operations, I wrote new Float32 versions of these elements.  I used inheritance where possible to reduce the duplication of functionality, particularly for the AudioStream to AudioStream_F32 conversion.  To maintain the structure and conventions of the Teensy Audio Library, I made a one-for-one replacement so that one only need to substitute the "_F32" version for the standard version.

Conversion Routines:  To interface between the Int16 data of the Teensy Audio Library and the Float32 data used my extended library, I also wrote two new "AudioConvert" classes.  In addition to converting between the Int16 and Float32 data types, these routines also re-scale the data.  The Teensy Audio Library assumes "full scale" is ±32768 whereas my floating-point objects assume that ±1.0 is full scale.  My conversion routines automatically account for this difference.

OpenAudio F32 Example:  If you download the library and unzip it into your Arduino Libraries directory (see "Installation" notes here), you can start the Arduino IDE and load an example sketch that comes with the library.  Go under the "File" menu and select "Examples", then, "OpenAudio_ArduinoLibrary", then "BasicGain_Float".

Add a New #include:  Inside this sketch, you'll see many features that are similar to every sketch that uses the Teensy Audio Library.  For example, the screenshot below shows the #include statements at the beginning of the sketch.  Note that I added an #include for the new OpenAudio library.  Once this line is added, any of the new OpenAudio classes can be used.

Instantiate the New Classes:  In a typical sketch using the Teensy Audio Library, the next block of code instantiates the audio-related objects.  The screenshot below illustrates this by invoking the standard AudioControlSGTL5000, AudioInputI2S, and AudioOutputI2S classes.   After these standard lines, I added three new lines that instantiate blocks for floating-point processing.

In this case, AudioConvert_I16toF32 will convert the standard Int16 audio data to float32.  AudioEffectGain_F32 will apply gain to the float32 audio data.  AudioConvert_F32toI16 will convert the audio data back to Int16 so that it can be output by the usual functions of the Teensy Audio Library.

New Audio Connections:  After instantiating the objects, the next step is to make the audio connections between the objects.  The screenshot below shows that a mix of standard (Int16) connections and new (Float32) connections are used to make the full processing chain.  The standard connections (AudioConnection) are used whenever the data being passed is Int16 data.  The new connections (AudioConnection_F32) are used whenever the data is float32 data.

Allocate Float32 Memory:  The last step is shown in the screenshot below.  In the setup() function is where memory is allocated for the Teensy Audio Library via the AudioMemory() statement.  Following this same pattern, I added the AudioMemory_F32() statement to allocate the float32 memory that is needed for the floating-point processing.

Complie and Run:  If you have a Teensy 3.5 or 3.6 and a Teensy Audio Board, you can compile and run this example sketch.  If you have a potentiometer attached to the Teensy Audio Board's volume control, you can use the pot to adjust the volume of the sound (well, it actually adjusts the gain applied by the new floating-point gain block).  On my hardware, it works great!  Hopefully it works well on yours, too.

Next Steps:  With the floating-point audio processing structure in-place, I can move forward with adding more "F32" processing blocks for the different functions that I need.  My next block will be a dynamic range compressor.  Then I'll probably add a filtering block.  Eventually, I'll be looking to add frequency-domain processing, which will likely involve extending my library again for a "complex_float32" data type.  That'll be even more fun!

Update:  I had my first Pull Request with a user contribution.  So awesome!
Update:  I've added an algorithms: a basic Dynamic Range Compressor.