Updated DSBF and added more convience to clean up the models.

autogen/AudioSource.m

@@ -19,6 +19,7 @@
     properties
         audio
         sampleRateHz
+        dataType
     end
 
     properties(SetAccess = private)
@@ -28,13 +29,18 @@
     end
 
     methods
-        function obj = AudioSource(audio, sampleRateHz)
+        function obj = AudioSource(audio, sampleRateHz, dataType)
         %AudioSource Initializes AudioSource with m x n audio input and
         % sample rate in hz, where m is the the number of samples and n is
         % the number of channels.
 
             obj.sampleRateHz = sampleRateHz;
             obj.audio = audio;
+            if exist('dataType','var')
+                obj.dataType = dataType;
+            else
+                obj.dataType = fixdt(1,24,23);
+            end
         end
 
         function obj = set.audio(obj, audio)
@@ -71,12 +77,13 @@
             % requires the input to be time series. Call this method
             % to use an AudioSource in Simulink simulations.
             time = [0:(obj.nSamples - 1)] / obj.sampleRateHz;
-            ts = timeseries(obj.audio, time);
+            fix_audio = fi(obj.audio, obj.dataType);
+            ts = timeseries(fix_audio, time);
         end
     end
 
     methods(Static)
-        function audioSource = fromFile(filepath, sampleRate, nSamples)
+        function audioSource = fromFile(filepath, sampleRate, nSamples, dataType)
             info = audioinfo(filepath);
             fileFs = info.SampleRate;
 
@@ -91,7 +98,7 @@
             end     
             yResampled = resample(y,sampleRate, fileFs);
 
-            audioSource = AudioSource(yResampled, sampleRate);
+            audioSource = AudioSource(yResampled, sampleRate, dataType);
         end
     end
 

autogen/configureModel.m

@@ -31,6 +31,7 @@
 mp.signed = config.system.audioIn.signed;
 mp.W_bits = config.system.audioIn.wordLength;  % Word length
 mp.F_bits = config.system.audioIn.fractionLength;  % Number of fractional bits in word
+mp.audio_dt = fixdt(mp.signed, mp.W_bits, mp.F_bits);
 mp.nChannels = config.system.audioIn.numberOfChannels;
 mp.nOutChannels = config.system.audioOut.numberOfChannels;
 
@@ -76,6 +77,18 @@
 mp.Ts_sim = 1/(mp.Fs * mp.nChannels);         % System clock period
 mp.rate_change = mp.Fs_system/mp.Fs;   % how much faster the system clock is to the sample clock
 
+regSize = size(mp.register);
+nRegs = regSize(1);
+tempRegs = containers.Map();
+for idx=1:nRegs
+    tempRegs(mp.register{idx}.name) = mp.register{idx};
+end
+
+
+mp.getReg = tempRegs;
+clear tempRegs;
+clear nRegs;
+clear regSize;
 
 %% Helper functions
 function regs = parseregisters(config)
@@ -100,14 +113,15 @@
             reg = device.registers{idx};
         end
         tempReg.name = reg.name;
-        tempReg.value = reg.defaultValue;
 
         if reg.dataType.type == "boolean"
             tempReg.dataType = fixdt('boolean');
         else
             tempReg.dataType = fixdt(reg.dataType.signed, ...
                 reg.dataType.wordLength, reg.dataType.fractionLength);
         end
+
+        tempReg.value = fi(reg.defaultValue, tempReg.dataType);
         tempReg.timeseries = timeseries(tempReg.value, 0);
 
         regs{idx} = tempReg; 

autogen/initCallback.m

@@ -31,7 +31,7 @@
 
 configureModel;
 
-testSignal = AudioSource.fromFile(mp.testFile, mp.Fs, mp.nSamples);
+testSignal = AudioSource.fromFile(mp.testFile, mp.Fs, mp.nSamples, mp.audio_dt);
 stopTime = testSignal.duration;
 
 nSamples = testSignal.nSamples;

models/delay_and_sum_beamformer/README.md

@@ -11,7 +11,7 @@ The basic idea behind [beamforming](https://en.wikipedia.org/wiki/Beamforming) i
 
 ## Usage
 - `Azimuth`: The horizontal steering angle in degrees (audio coming from this direction is reinforced); this ranges between +/-90 degrees.
-- `Elevation`: The vertical steering angle in degrees (audio coming from this direction is reincofreced); this ranges between +/-90 degrees.
+- `Elevation`: The vertical steering angle in degrees (audio coming from this direction is reinforced); this ranges between +/-90 degrees.
 
 ## Implementation
 The following block diagram shows a high-level view of the implementation.

models/delay_and_sum_beamformer/model.json

@@ -5,14 +5,14 @@
         "systemClockFrequency": 98304000,
         "processing": "sample",
         "audioIn": {
-            "wordLength": 32,
-            "fractionLength": 28,
+            "wordLength": 24,
+            "fractionLength": 23,
             "signed": true,
             "numberOfChannels": 16
         },
         "audioOut": {
-            "wordLength": 32,
-            "fractionLength": 28,
+            "wordLength": 24,
+            "fractionLength": 23,
             "signed": true,
             "numberOfChannels": 2
         }

models/delay_and_sum_beamformer/modelparameters.m

@@ -3,42 +3,31 @@
 
 mp.testFile = [mp.test_signals_path filesep 'beamformer_test.wav'];
 
-% TODO: use booleans instead of 0 and 1
 mp.sim_prompts = 0;
-mp.sim_verify = 0;
+mp.sim_verify = 1;
 mp.simDuration = .03;
 mp.nSamples = config.system.sampleClockFrequency * mp.simDuration;
-
-%% Exponential moving average setup
-
-% % 10 ms window
-% mp.windowSize = 10e-3 * mp.Fs;
-% 
-% % define the exponential moving average weight to be roughly equivalent to
-% % a simple moving average of length mp.windowSize
-% % https://en.wikipedia.org/wiki/Moving_average#Relationship_between_SMA_and_EMA
-% mp.exponentialMovingAverageWeight = fi(2/(mp.windowSize + 1), 0, 32, 31);
 
 mp.speedOfSound = 343;
 
-% TODO: it'd sure be nice to have access to mp.Fs right here, but we don't....
-mp.samplingRate = 48e3;
+mp.samplingRate = mp.Fs;
 mp.arraySpacing = 25e-3;
 mp.arraySize = [4,4];
 arraySize = mp.arraySize;
 arraySpacing = mp.arraySpacing;
 samplingRate = mp.samplingRate;
 speedOfSound = mp.speedOfSound;
 
-% TODO: document this equation... this is across the entire array, 
-% but delays are relative to the array center, so delays are +/- maxDelay/2
+% this is across the entire array,  but delays are relative to the array
+% center, so delays are +/- maxDelay/2
 mp.maxDelay = sqrt(((mp.arraySize(1) - 1)*mp.arraySpacing).^2 + ((mp.arraySize(2) - 1)*mp.arraySpacing).^2)*mp.samplingRate/mp.speedOfSound;
-% buffer size to accomodate max delay; buffer size is a power of 2
 
+% buffer size to accomodate max delay; buffer size is a power of 2
 mp.integerDelayAddrSize = ceil(log2(floor(mp.maxDelay))) + 2;
 mp.integerDelayBufferSize = 2^mp.integerDelayAddrSize;
 mp.upsampleFactor = 16;
 upsampleFactor = mp.upsampleFactor;
+
 % number of required bits is two more than that required to represent upsampleFactor
 % because we need to be able to represent +/- upsample factor inclusive
 mp.fractionalDelayAddrSize = ceil(log2(floor(mp.upsampleFactor))) + 2;