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

@@ -1,4 +1,4 @@
-# Delay and Sum Beamformer
+git # Delay and Sum Beamformer
 This model implements a delay and sum beamformer, intended for use with the [Audio Logic Microphone Array](https://fpga-open-speech-tools.github.io/ab_mic_array.html). In audio applications, beamforming is a spatial filtering technique in which an array of microphones is used to "focus" on signals coming from a particular direction. 
 
 This model was developed using the Frost Autogen Framework, and can be manually deployed to an Audio Research or Audio Blade. To set up Frost Autogen, please review the [Getting Started Guides](https://github.com/fpga-open-speech-tools/docs/tree/master/getting_started).

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;