auto removed a bunch of unused local variables

( removed with a script parsing compiler output )



git-svn-id: svn://svn.code.sf.net/p/ultrastardx/svn/trunk@950 b956fd51-792f-4845-bead-9b4dfca2ff2c

1 files changed, 610 insertions, 610 deletions

diff --git a/Game/Code/Classes/URecord.pas b/Game/Code/Classes/URecord.pas
index bb8e38e6..2f62f441 100644
--- a/Game/Code/Classes/URecord.pas
+++ b/Game/Code/Classes/URecord.pas
@@ -1,610 +1,610 @@
-unit URecord;

-

-interface

-

-{$IFDEF FPC}

-  {$MODE Delphi}

-{$ENDIF}

-

-{$I switches.inc}

-

-uses Classes,

-     Math,

-     SysUtils,

-     UCommon,

-     UMusic,

-     UIni;

-

-const

-  BaseToneFreq = 65.4064; // lowest (half-)tone to analyze (C2 = 65.4064 Hz)

-  NumHalftones = 36; // C2-B4 (for Whitney and my high voice)

-

-type

-  TCaptureBuffer = class

-    private

-      BufferNew:    TMemoryStream; // buffer for newest samples

-

-      function GetToneString: string; // converts a tone to its string represenatation;

-    public

-      BufferArray:  array[0..4095] of smallint; // newest 4096 samples

-      BufferLong:   TMemoryStream;              // full buffer

-      AnalysisBufferSize: integer; // number of samples of BufferArray to analyze

-

-      AudioFormat: TAudioFormatInfo;

-

-      // pitch detection

-      ToneValid:    boolean;    // true if Tone contains a valid value (otherwise it contains noise)

-      Tone:         integer;    // tone relative to one octave (e.g. C2=C3=C4). Range: 0-11

-      ToneAbs:      integer;    // absolute (full range) tone (e.g. C2<>C3). Range: 0..NumHalftones-1

-

-      // methods

-      constructor Create;

-      destructor Destroy; override;

-

-      procedure Clear;

-

-      procedure ProcessNewBuffer;

-      // use to analyze sound from buffers to get new pitch

-      procedure AnalyzeBuffer;

-      // we call it to analyze sound by checking Autocorrelation

-      procedure AnalyzeByAutocorrelation;

-      // use this to check one frequency by Autocorrelation

-      function AnalyzeAutocorrelationFreq(Freq: real): real;

-      function MaxSampleVolume: Single;

-

-      property ToneString: string READ GetToneString;

-  end;

-

-  TAudioInputDeviceSource = record

-    Name:   string;

-  end;

-

-  // soundcard input-devices information

-  TAudioInputDevice = class

-    public

-      CfgIndex:        integer;   // index of this device in Ini.InputDeviceConfig

-      Description:     string;    // soundcard name/description

-      Source:          array of TAudioInputDeviceSource; // soundcard input(-source)s

-      SourceSelected:  integer;  // unused. What is this good for?

-      MicSource:       integer;  // unused. What is this good for?

-

-      AudioFormat:     TAudioFormatInfo; // capture format info (e.g. 44.1kHz SInt16 stereo)

-      CaptureChannel:  array of TCaptureBuffer; // sound-buffer references used for mono or stereo channel's capture data

-

-      destructor Destroy; override;

-

-      procedure LinkCaptureBuffer(ChannelIndex: integer; Sound: TCaptureBuffer);

-      

-      function Start(): boolean; virtual; abstract;

-      procedure Stop();  virtual; abstract;

-  end;

-

-  TAudioInputProcessor = class

-    public

-      Sound:  array of TCaptureBuffer; // sound-buffers for every player

-      Device: array of TAudioInputDevice;

-

-      constructor Create;

-

-      // handle microphone input

-      procedure HandleMicrophoneData(Buffer: Pointer; Size: Cardinal;

-                                     InputDevice: TAudioInputDevice);

-  end;

-

-  TAudioInputBase = class( TInterfacedObject, IAudioInput )

-    private

-      Started: boolean;

-    protected

-      function UnifyDeviceName(const name: string; deviceIndex: integer): string;

-      function UnifyDeviceSourceName(const name: string; const deviceName: string): string;

-    public

-      function GetName: String;           virtual; abstract;

-      function InitializeRecord: boolean; virtual; abstract;

-

-      procedure CaptureStart;

-      procedure CaptureStop;

-  end;

-

-

-  SmallIntArray = array [0..maxInt shr 1-1] of smallInt;

-  PSmallIntArray = ^SmallIntArray;

-

-  function AudioInputProcessor(): TAudioInputProcessor;

-

-implementation

-

-uses

-  ULog,

-  UMain;

-

-var

-  singleton_AudioInputProcessor : TAudioInputProcessor = nil;

-

-

-// FIXME: Race-Conditions between Callback-thread and main-thread

-//        on BufferArray (maybe BufferNew also).

-//        Use SDL-mutexes to solve this problem.

-

-

-{ Global }

-

-function AudioInputProcessor(): TAudioInputProcessor;

-begin

-  if singleton_AudioInputProcessor = nil then

-    singleton_AudioInputProcessor := TAudioInputProcessor.create();

-

-  result := singleton_AudioInputProcessor;

-end;

-

-

-{ TAudioInputDevice }

-

-destructor TAudioInputDevice.Destroy;

-var

-  i: integer;

-begin

-  Stop();

-  Source := nil;

-  CaptureChannel := nil;

-  FreeAndNil(AudioFormat);

-  inherited Destroy;

-end;

-

-procedure TAudioInputDevice.LinkCaptureBuffer(ChannelIndex: integer; Sound: TCaptureBuffer);

-begin

-  // check bounds

-  if ((ChannelIndex < 0) or (ChannelIndex > High(CaptureChannel))) then

-    Exit;

-

-  // reset audio-format of old capture-buffer

-  if (CaptureChannel[ChannelIndex] <> nil) then

-    CaptureChannel[ChannelIndex].AudioFormat := nil;

-

-  // set audio-format of new capture-buffer

-  if (Sound <> nil) then

-    Sound.AudioFormat := AudioFormat;

-

-  // replace old with new buffer

-  CaptureChannel[ChannelIndex] := Sound;

-end;

-

-{ TSound }

-

-constructor TCaptureBuffer.Create;

-begin

-  inherited;

-  BufferNew := TMemoryStream.Create;

-  BufferLong := TMemoryStream.Create;

-  AnalysisBufferSize := Min(4*1024, Length(BufferArray));

-end;

-

-destructor TCaptureBuffer.Destroy;

-begin

-  AudioFormat := nil;

-  FreeAndNil(BufferNew);

-  FreeAndNil(BufferLong);

-  inherited;

-end;

-

-procedure TCaptureBuffer.Clear;

-begin

-  if assigned(BufferNew) then

-    BufferNew.Clear;

-  if assigned(BufferLong) then

-    BufferLong.Clear;

-  FillChar(BufferArray[0], Length(BufferArray) * SizeOf(SmallInt), 0);

-end;

-

-procedure TCaptureBuffer.ProcessNewBuffer;

-var

-  SkipCount:   integer;

-  NumSamples:  integer;

-  SampleIndex: integer;

-begin

-  // process BufferArray

-  SkipCount := 0;

-  NumSamples := BufferNew.Size div 2;

-

-  // check if we have more new samples than we can store

-  if (NumSamples > Length(BufferArray)) then

-  begin

-    // discard the oldest of the new samples

-    SkipCount := NumSamples - Length(BufferArray);

-    NumSamples := Length(BufferArray);

-  end;

-

-  // move old samples to the beginning of the array (if necessary)

-  for SampleIndex := NumSamples to High(BufferArray) do

-    BufferArray[SampleIndex-NumSamples] := BufferArray[SampleIndex];

-

-  // skip samples if necessary

-  BufferNew.Seek(2*SkipCount, soBeginning);

-  // copy samples

-  BufferNew.ReadBuffer(BufferArray[Length(BufferArray)-NumSamples], 2*NumSamples);

-

-  // save capture-data to BufferLong if neccessary

-  if (Ini.SavePlayback = 1) then

-  begin

-    BufferNew.Seek(0, soBeginning);

-    BufferLong.CopyFrom(BufferNew, BufferNew.Size);

-  end;

-end;

-

-procedure TCaptureBuffer.AnalyzeBuffer;

-var

-  Volume:    real;

-  MaxVolume: real;

-  SampleIndex: integer;

-  Threshold: real;

-begin

-  ToneValid := false;

-  ToneAbs := -1;

-  Tone    := -1;

-

-  // find maximum volume of first 1024 samples

-  MaxVolume := 0;

-  for SampleIndex := 0 to 1023 do

-  begin

-    Volume := Abs(BufferArray[SampleIndex]) / -Low(Smallint);

-    if Volume > MaxVolume then

-       MaxVolume := Volume;

-  end;

-

-  case Ini.Threshold of

-    0:   Threshold := 0.05;

-    1:   Threshold := 0.1;

-    2:   Threshold := 0.15;

-    3:   Threshold := 0.2;

-    else Threshold := 0.1;

-  end;

-

-  // check if signal has an acceptable volume (ignore background-noise)

-  if MaxVolume >= Threshold then

-  begin

-    // analyse the current voice pitch

-    AnalyzeByAutocorrelation;

-    ToneValid := true;

-  end;

-end;

-

-procedure TCaptureBuffer.AnalyzeByAutocorrelation;

-var

-  ToneIndex: integer;

-  CurFreq:   real;

-  CurWeight: real;

-  MaxWeight: real;

-  MaxTone:   integer;

-const

-  HalftoneBase = 1.05946309436; // 2^(1/12) -> HalftoneBase^12 = 2 (one octave)

-begin

-  // prepare to analyze

-  MaxWeight := -1;

-

-  // analyze halftones

-  // Note: at the lowest tone (~65Hz) and a buffer-size of 4096

-  // at 44.1 (or 48kHz) only 6 (or 5) samples are compared, this might be

-  // too few samples -> use a bigger buffer-size

-  for ToneIndex := 0 to NumHalftones-1 do

-  begin

-    CurFreq := BaseToneFreq * Power(HalftoneBase, ToneIndex);

-    CurWeight := AnalyzeAutocorrelationFreq(CurFreq);

-

-    // TODO: prefer higher frequencies (use >= or use downto)

-    if (CurWeight > MaxWeight) then

-    begin

-      // this frequency has a higher weight

-      MaxWeight := CurWeight;

-      MaxTone   := ToneIndex;

-    end;

-  end;

-

-  ToneAbs := MaxTone;

-  Tone    := MaxTone mod 12;

-end;

-

-// result medium difference

-function TCaptureBuffer.AnalyzeAutocorrelationFreq(Freq: real): real;

-var

-  Dist:                   real;    // distance (0=equal .. 1=totally different) between correlated samples

-  AccumDist:              real;    // accumulated distances

-  SampleIndex:            integer; // index of sample to analyze

-  CorrelatingSampleIndex: integer; // index of sample one period ahead

-  SamplesPerPeriod:       integer; // samples in one period

-begin

-  SampleIndex := 0;

-  SamplesPerPeriod := Round(AudioFormat.SampleRate/Freq);

-  CorrelatingSampleIndex := SampleIndex + SamplesPerPeriod;

-

-  AccumDist := 0;

-

-  // compare correlating samples

-  while (CorrelatingSampleIndex < AnalysisBufferSize) do

-  begin

-    // calc distance (correlation: 1-dist) to corresponding sample in next period

-    Dist := Abs(BufferArray[SampleIndex] - BufferArray[CorrelatingSampleIndex]) /

-            High(Word);

-    AccumDist := AccumDist + Dist;

-    Inc(SampleIndex);

-    Inc(CorrelatingSampleIndex);

-  end;

-

-  // return "inverse" average distance (=correlation)

-  Result := 1 - AccumDist / AnalysisBufferSize;

-end;

-

-function TCaptureBuffer.MaxSampleVolume: Single;

-var

-  lSampleIndex: Integer;

-  lMaxVol : Longint;

-begin;

-  // FIXME: lock buffer to avoid race-conditions

-  lMaxVol := 0;

-  for lSampleIndex := 0 to High(BufferArray) do

-  begin

-    if Abs(BufferArray[lSampleIndex]) > lMaxVol then

-      lMaxVol := Abs(BufferArray[lSampleIndex]);

-  end;

-

-  result := lMaxVol / -Low(Smallint);

-end;

-

-const

-  ToneStrings: array[0..11] of string = (

-    'C', 'C#', 'D', 'D#', 'E', 'F', 'F#', 'G', 'G#', 'A', 'A#', 'B'

-  );

-

-function TCaptureBuffer.GetToneString: string;

-begin

-  if (ToneValid) then

-    Result := ToneStrings[Tone] + IntToStr(ToneAbs div 12 + 2)

-  else

-    Result := '-';

-end;

-

-

-{ TAudioInputProcessor }

-

-constructor TAudioInputProcessor.Create;

-var

-  i:        integer;

-begin

-  SetLength(Sound, 6 {max players});//Ini.Players+1);

-  for i := 0 to High(Sound) do

-  begin

-    Sound[i] := TCaptureBuffer.Create;

-  end;

-end;

-

-{*

- * Handle captured microphone input data.

- * Params:

- *   Buffer - buffer of signed 16bit interleaved stereo PCM-samples.

- *     Interleaved means that a right-channel sample follows a left-

- *     channel sample and vice versa (0:left[0],1:right[0],2:left[1],...).

- *   Length - number of bytes in Buffer

- *   Input - Soundcard-Input used for capture

- *}

-procedure TAudioInputProcessor.HandleMicrophoneData(Buffer: Pointer; Size: Cardinal; InputDevice: TAudioInputDevice);

-var

-  Value:        integer;

-  ChannelBuffer: PChar;         // buffer handled as array of bytes (offset relative to channel)

-  SampleBuffer: PSmallIntArray; // buffer handled as array of samples

-  Boost:  byte;

-  ChannelCount: integer;

-  ChannelIndex: integer;

-  ChannelOffset: integer;

-  CaptureChannel: TCaptureBuffer;

-  AudioFormat: TAudioFormatInfo;

-  FrameSize: integer;

-  NumSamples: integer;

-  NumFrames: integer; // number of frames (stereo: 2xsamples)

-  i: integer;

-begin

-  // set boost

-  case Ini.MicBoost of

-    0:   Boost := 1;

-    1:   Boost := 2;

-    2:   Boost := 4;

-    3:   Boost := 8;

-    else Boost := 1;

-  end;

-

-  AudioFormat := InputDevice.AudioFormat;

-

-  // FIXME: At the moment we assume a SInt16 format

-  // TODO:  use SDL_AudioConvert to convert to SInt16 but do NOT change the

-  // samplerate (SDL does not convert 44.1kHz to 48kHz so we might get wrong

-  // results in the analysis phase otherwise)

-  if (AudioFormat.Format <> asfS16) then

-  begin

-    // this only occurs if a developer choosed a wrong input sample-format

-    Log.CriticalError('TAudioInputProcessor.HandleMicrophoneData: Wrong sample-format');

-    Exit;

-  end;

-

-  // interpret buffer as buffer of bytes

-  SampleBuffer := Buffer;

-

-  NumSamples := Size div SizeOf(Smallint);

-

-  // boost buffer

-  // TODO: remove this senseless stuff - adjust the threshold instead

-  for i := 0 to NumSamples-1 do

-  begin

-    Value := SampleBuffer^[i] * Boost;

-

-    // TODO :  JB -  This will clip the audio... cant we reduce the "Boost" if the data clips ??

-    if Value > High(Smallint) then

-      Value := High(Smallint);

-

-    if Value < Low(Smallint) then

-      Value := Low(Smallint);

-

-    SampleBuffer^[i] := Value;

-  end;

-

-  // samples per channel

-  FrameSize := AudioFormat.Channels * SizeOf(SmallInt);

-  NumFrames := Size div FrameSize;

-

-  // process channels

-  for ChannelIndex := 0 to High(InputDevice.CaptureChannel) do

-  begin

-    CaptureChannel := InputDevice.CaptureChannel[ChannelIndex];

-    if (CaptureChannel <> nil) then

-    begin

-      // set offset according to channel index

-      ChannelBuffer := @PChar(Buffer)[ChannelIndex * SizeOf(SmallInt)];

-

-      // TODO: remove BufferNew and write to BufferArray directly

-

-      CaptureChannel.BufferNew.Clear;

-      for i := 0 to NumFrames-1 do

-      begin

-        CaptureChannel.BufferNew.Write(ChannelBuffer[i*FrameSize], SizeOf(SmallInt));

-      end;

-      CaptureChannel.ProcessNewBuffer();

-    end;

-  end;

-end;

-

-

-{ TAudioInputBase }

-

-{*

- * Start capturing on all used input-device.

- *}

-procedure TAudioInputBase.CaptureStart;

-var

-  S:  integer;

-  DeviceIndex: integer;

-  ChannelIndex: integer;

-  Device: TAudioInputDevice;

-  DeviceCfg: PInputDeviceConfig;

-  DeviceUsed: boolean;

-  Player: integer;

-begin

-  if (Started) then

-    CaptureStop();

-

-  // reset buffers

-  for S := 0 to High(AudioInputProcessor.Sound) do

-    AudioInputProcessor.Sound[S].Clear;

-

-  // start capturing on each used device

-  for DeviceIndex := 0 to High(AudioInputProcessor.Device) do

-  begin

-    Device := AudioInputProcessor.Device[DeviceIndex];

-    if not assigned(Device) then

-      continue;

-    DeviceCfg := @Ini.InputDeviceConfig[Device.CfgIndex];

-

-    DeviceUsed := false;

-

-    // check if device is used

-    for ChannelIndex := 0 to High(DeviceCfg.ChannelToPlayerMap) do

-    begin

-      Player := DeviceCfg.ChannelToPlayerMap[ChannelIndex]-1;

-      if (Player < 0) or (Player >= PlayersPlay) then

-      begin

-        Device.LinkCaptureBuffer(ChannelIndex, nil);

-      end

-      else

-      begin

-        Device.LinkCaptureBuffer(ChannelIndex, AudioInputProcessor.Sound[Player]);

-        DeviceUsed := true;

-      end;

-    end;

-

-    // start device if used

-    if (DeviceUsed) then

-    begin

-      //Log.BenchmarkStart(2);

-      Device.Start();

-      //Log.BenchmarkEnd(2);

-      //Log.LogBenchmark('Device.Start', 2) ;

-    end;

-  end;

-

-  Started := true;

-end;

-

-{*

- * Stop input-capturing on all soundcards.

- *}

-procedure TAudioInputBase.CaptureStop;

-var

-  DeviceIndex: integer;

-  Player:  integer;

-  Device: TAudioInputDevice;

-  DeviceCfg: PInputDeviceConfig;

-begin

-  for DeviceIndex := 0 to High(AudioInputProcessor.Device) do

-  begin

-    Device := AudioInputProcessor.Device[DeviceIndex];

-    if not assigned(Device) then

-      continue;

-    Device.Stop();

-  end;

-

-  Started := false;

-end;

-

-function TAudioInputBase.UnifyDeviceName(const name: string; deviceIndex: integer): string;

-var

-  count: integer; // count of devices with this name

-

-  function IsDuplicate(const name: string): boolean;

-  var

-    i: integer;

-  begin

-    Result := False;

-    // search devices with same description

-    For i := 0 to deviceIndex-1 do

-    begin

-      if (AudioInputProcessor.Device[i].Description = name) then

-      begin

-        Result := True;

-        Break;

-      end;

-    end;

-  end;

-begin

-  count := 1;

-  result := name;

-

-  // if there is another device with the same ID, search for an available name

-  while (IsDuplicate(result)) do

-  begin

-    Inc(count);

-    // set description

-    result := name + ' ('+IntToStr(count)+')';

-  end;

-end;

-

-{*

- * Unifies an input-device's source name.

- * Note: the description member of the device must already be set when

- * calling this function.

- *}

-function TAudioInputBase.UnifyDeviceSourceName(const name: string; const deviceName: string): string;

-var

-  Descr: string;

-begin

-  result := name;

-

-  {$IFDEF DARWIN}

-    // Under MacOSX the SingStar Mics have an empty

-    // InputName. So, we have to add a hard coded

-    // Workaround for this problem

-    if (name = '') and (Pos( 'USBMIC Serial#', deviceName) > 0) then

-    begin

-      result := 'Microphone';

-    end;

-  {$ENDIF}

-end;

-

-end.

-

-

-

+unit URecord;
+
+interface
+
+{$IFDEF FPC}
+  {$MODE Delphi}
+{$ENDIF}
+
+{$I switches.inc}
+
+uses Classes,
+     Math,
+     SysUtils,
+     UCommon,
+     UMusic,
+     UIni;
+
+const
+  BaseToneFreq = 65.4064; // lowest (half-)tone to analyze (C2 = 65.4064 Hz)
+  NumHalftones = 36; // C2-B4 (for Whitney and my high voice)
+
+type
+  TCaptureBuffer = class
+    private
+      BufferNew:    TMemoryStream; // buffer for newest samples
+
+      function GetToneString: string; // converts a tone to its string represenatation;
+    public
+      BufferArray:  array[0..4095] of smallint; // newest 4096 samples
+      BufferLong:   TMemoryStream;              // full buffer
+      AnalysisBufferSize: integer; // number of samples of BufferArray to analyze
+
+      AudioFormat: TAudioFormatInfo;
+
+      // pitch detection
+      ToneValid:    boolean;    // true if Tone contains a valid value (otherwise it contains noise)
+      Tone:         integer;    // tone relative to one octave (e.g. C2=C3=C4). Range: 0-11
+      ToneAbs:      integer;    // absolute (full range) tone (e.g. C2<>C3). Range: 0..NumHalftones-1
+
+      // methods
+      constructor Create;
+      destructor Destroy; override;
+
+      procedure Clear;
+
+      procedure ProcessNewBuffer;
+      // use to analyze sound from buffers to get new pitch
+      procedure AnalyzeBuffer;
+      // we call it to analyze sound by checking Autocorrelation
+      procedure AnalyzeByAutocorrelation;
+      // use this to check one frequency by Autocorrelation
+      function AnalyzeAutocorrelationFreq(Freq: real): real;
+      function MaxSampleVolume: Single;
+
+      property ToneString: string READ GetToneString;
+  end;
+
+  TAudioInputDeviceSource = record
+    Name:   string;
+  end;
+
+  // soundcard input-devices information
+  TAudioInputDevice = class
+    public
+      CfgIndex:        integer;   // index of this device in Ini.InputDeviceConfig
+      Description:     string;    // soundcard name/description
+      Source:          array of TAudioInputDeviceSource; // soundcard input(-source)s
+      SourceSelected:  integer;  // unused. What is this good for?
+      MicSource:       integer;  // unused. What is this good for?
+
+      AudioFormat:     TAudioFormatInfo; // capture format info (e.g. 44.1kHz SInt16 stereo)
+      CaptureChannel:  array of TCaptureBuffer; // sound-buffer references used for mono or stereo channel's capture data
+
+      destructor Destroy; override;
+
+      procedure LinkCaptureBuffer(ChannelIndex: integer; Sound: TCaptureBuffer);
+      
+      function Start(): boolean; virtual; abstract;
+      procedure Stop();  virtual; abstract;
+  end;
+
+  TAudioInputProcessor = class
+    public
+      Sound:  array of TCaptureBuffer; // sound-buffers for every player
+      Device: array of TAudioInputDevice;
+
+      constructor Create;
+
+      // handle microphone input
+      procedure HandleMicrophoneData(Buffer: Pointer; Size: Cardinal;
+                                     InputDevice: TAudioInputDevice);
+  end;
+
+  TAudioInputBase = class( TInterfacedObject, IAudioInput )
+    private
+      Started: boolean;
+    protected
+      function UnifyDeviceName(const name: string; deviceIndex: integer): string;
+      function UnifyDeviceSourceName(const name: string; const deviceName: string): string;
+    public
+      function GetName: String;           virtual; abstract;
+      function InitializeRecord: boolean; virtual; abstract;
+
+      procedure CaptureStart;
+      procedure CaptureStop;
+  end;
+
+
+  SmallIntArray = array [0..maxInt shr 1-1] of smallInt;
+  PSmallIntArray = ^SmallIntArray;
+
+  function AudioInputProcessor(): TAudioInputProcessor;
+
+implementation
+
+uses
+  ULog,
+  UMain;
+
+var
+  singleton_AudioInputProcessor : TAudioInputProcessor = nil;
+
+
+// FIXME: Race-Conditions between Callback-thread and main-thread
+//        on BufferArray (maybe BufferNew also).
+//        Use SDL-mutexes to solve this problem.
+
+
+{ Global }
+
+function AudioInputProcessor(): TAudioInputProcessor;
+begin
+  if singleton_AudioInputProcessor = nil then
+    singleton_AudioInputProcessor := TAudioInputProcessor.create();
+
+  result := singleton_AudioInputProcessor;
+end;
+
+
+{ TAudioInputDevice }
+
+destructor TAudioInputDevice.Destroy;
+//var
+// i: integer; // Auto Removed, Unused Variable
+begin
+  Stop();
+  Source := nil;
+  CaptureChannel := nil;
+  FreeAndNil(AudioFormat);
+  inherited Destroy;
+end;
+
+procedure TAudioInputDevice.LinkCaptureBuffer(ChannelIndex: integer; Sound: TCaptureBuffer);
+begin
+  // check bounds
+  if ((ChannelIndex < 0) or (ChannelIndex > High(CaptureChannel))) then
+    Exit;
+
+  // reset audio-format of old capture-buffer
+  if (CaptureChannel[ChannelIndex] <> nil) then
+    CaptureChannel[ChannelIndex].AudioFormat := nil;
+
+  // set audio-format of new capture-buffer
+  if (Sound <> nil) then
+    Sound.AudioFormat := AudioFormat;
+
+  // replace old with new buffer
+  CaptureChannel[ChannelIndex] := Sound;
+end;
+
+{ TSound }
+
+constructor TCaptureBuffer.Create;
+begin
+  inherited;
+  BufferNew := TMemoryStream.Create;
+  BufferLong := TMemoryStream.Create;
+  AnalysisBufferSize := Min(4*1024, Length(BufferArray));
+end;
+
+destructor TCaptureBuffer.Destroy;
+begin
+  AudioFormat := nil;
+  FreeAndNil(BufferNew);
+  FreeAndNil(BufferLong);
+  inherited;
+end;
+
+procedure TCaptureBuffer.Clear;
+begin
+  if assigned(BufferNew) then
+    BufferNew.Clear;
+  if assigned(BufferLong) then
+    BufferLong.Clear;
+  FillChar(BufferArray[0], Length(BufferArray) * SizeOf(SmallInt), 0);
+end;
+
+procedure TCaptureBuffer.ProcessNewBuffer;
+var
+  SkipCount:   integer;
+  NumSamples:  integer;
+  SampleIndex: integer;
+begin
+  // process BufferArray
+  SkipCount := 0;
+  NumSamples := BufferNew.Size div 2;
+
+  // check if we have more new samples than we can store
+  if (NumSamples > Length(BufferArray)) then
+  begin
+    // discard the oldest of the new samples
+    SkipCount := NumSamples - Length(BufferArray);
+    NumSamples := Length(BufferArray);
+  end;
+
+  // move old samples to the beginning of the array (if necessary)
+  for SampleIndex := NumSamples to High(BufferArray) do
+    BufferArray[SampleIndex-NumSamples] := BufferArray[SampleIndex];
+
+  // skip samples if necessary
+  BufferNew.Seek(2*SkipCount, soBeginning);
+  // copy samples
+  BufferNew.ReadBuffer(BufferArray[Length(BufferArray)-NumSamples], 2*NumSamples);
+
+  // save capture-data to BufferLong if neccessary
+  if (Ini.SavePlayback = 1) then
+  begin
+    BufferNew.Seek(0, soBeginning);
+    BufferLong.CopyFrom(BufferNew, BufferNew.Size);
+  end;
+end;
+
+procedure TCaptureBuffer.AnalyzeBuffer;
+var
+  Volume:    real;
+  MaxVolume: real;
+  SampleIndex: integer;
+  Threshold: real;
+begin
+  ToneValid := false;
+  ToneAbs := -1;
+  Tone    := -1;
+
+  // find maximum volume of first 1024 samples
+  MaxVolume := 0;
+  for SampleIndex := 0 to 1023 do
+  begin
+    Volume := Abs(BufferArray[SampleIndex]) / -Low(Smallint);
+    if Volume > MaxVolume then
+       MaxVolume := Volume;
+  end;
+
+  case Ini.Threshold of
+    0:   Threshold := 0.05;
+    1:   Threshold := 0.1;
+    2:   Threshold := 0.15;
+    3:   Threshold := 0.2;
+    else Threshold := 0.1;
+  end;
+
+  // check if signal has an acceptable volume (ignore background-noise)
+  if MaxVolume >= Threshold then
+  begin
+    // analyse the current voice pitch
+    AnalyzeByAutocorrelation;
+    ToneValid := true;
+  end;
+end;
+
+procedure TCaptureBuffer.AnalyzeByAutocorrelation;
+var
+  ToneIndex: integer;
+  CurFreq:   real;
+  CurWeight: real;
+  MaxWeight: real;
+  MaxTone:   integer;
+const
+  HalftoneBase = 1.05946309436; // 2^(1/12) -> HalftoneBase^12 = 2 (one octave)
+begin
+  // prepare to analyze
+  MaxWeight := -1;
+
+  // analyze halftones
+  // Note: at the lowest tone (~65Hz) and a buffer-size of 4096
+  // at 44.1 (or 48kHz) only 6 (or 5) samples are compared, this might be
+  // too few samples -> use a bigger buffer-size
+  for ToneIndex := 0 to NumHalftones-1 do
+  begin
+    CurFreq := BaseToneFreq * Power(HalftoneBase, ToneIndex);
+    CurWeight := AnalyzeAutocorrelationFreq(CurFreq);
+
+    // TODO: prefer higher frequencies (use >= or use downto)
+    if (CurWeight > MaxWeight) then
+    begin
+      // this frequency has a higher weight
+      MaxWeight := CurWeight;
+      MaxTone   := ToneIndex;
+    end;
+  end;
+
+  ToneAbs := MaxTone;
+  Tone    := MaxTone mod 12;
+end;
+
+// result medium difference
+function TCaptureBuffer.AnalyzeAutocorrelationFreq(Freq: real): real;
+var
+  Dist:                   real;    // distance (0=equal .. 1=totally different) between correlated samples
+  AccumDist:              real;    // accumulated distances
+  SampleIndex:            integer; // index of sample to analyze
+  CorrelatingSampleIndex: integer; // index of sample one period ahead
+  SamplesPerPeriod:       integer; // samples in one period
+begin
+  SampleIndex := 0;
+  SamplesPerPeriod := Round(AudioFormat.SampleRate/Freq);
+  CorrelatingSampleIndex := SampleIndex + SamplesPerPeriod;
+
+  AccumDist := 0;
+
+  // compare correlating samples
+  while (CorrelatingSampleIndex < AnalysisBufferSize) do
+  begin
+    // calc distance (correlation: 1-dist) to corresponding sample in next period
+    Dist := Abs(BufferArray[SampleIndex] - BufferArray[CorrelatingSampleIndex]) /
+            High(Word);
+    AccumDist := AccumDist + Dist;
+    Inc(SampleIndex);
+    Inc(CorrelatingSampleIndex);
+  end;
+
+  // return "inverse" average distance (=correlation)
+  Result := 1 - AccumDist / AnalysisBufferSize;
+end;
+
+function TCaptureBuffer.MaxSampleVolume: Single;
+var
+  lSampleIndex: Integer;
+  lMaxVol : Longint;
+begin;
+  // FIXME: lock buffer to avoid race-conditions
+  lMaxVol := 0;
+  for lSampleIndex := 0 to High(BufferArray) do
+  begin
+    if Abs(BufferArray[lSampleIndex]) > lMaxVol then
+      lMaxVol := Abs(BufferArray[lSampleIndex]);
+  end;
+
+  result := lMaxVol / -Low(Smallint);
+end;
+
+const
+  ToneStrings: array[0..11] of string = (
+    'C', 'C#', 'D', 'D#', 'E', 'F', 'F#', 'G', 'G#', 'A', 'A#', 'B'
+  );
+
+function TCaptureBuffer.GetToneString: string;
+begin
+  if (ToneValid) then
+    Result := ToneStrings[Tone] + IntToStr(ToneAbs div 12 + 2)
+  else
+    Result := '-';
+end;
+
+
+{ TAudioInputProcessor }
+
+constructor TAudioInputProcessor.Create;
+var
+  i:        integer;
+begin
+  SetLength(Sound, 6 {max players});//Ini.Players+1);
+  for i := 0 to High(Sound) do
+  begin
+    Sound[i] := TCaptureBuffer.Create;
+  end;
+end;
+
+{*
+ * Handle captured microphone input data.
+ * Params:
+ *   Buffer - buffer of signed 16bit interleaved stereo PCM-samples.
+ *     Interleaved means that a right-channel sample follows a left-
+ *     channel sample and vice versa (0:left[0],1:right[0],2:left[1],...).
+ *   Length - number of bytes in Buffer
+ *   Input - Soundcard-Input used for capture
+ *}
+procedure TAudioInputProcessor.HandleMicrophoneData(Buffer: Pointer; Size: Cardinal; InputDevice: TAudioInputDevice);
+var
+  Value:        integer;
+  ChannelBuffer: PChar;         // buffer handled as array of bytes (offset relative to channel)
+  SampleBuffer: PSmallIntArray; // buffer handled as array of samples
+  Boost:  byte;
+// ChannelCount: integer; // Auto Removed, Unused Variable
+  ChannelIndex: integer;
+// ChannelOffset: integer; // Auto Removed, Unused Variable
+  CaptureChannel: TCaptureBuffer;
+  AudioFormat: TAudioFormatInfo;
+  FrameSize: integer;
+  NumSamples: integer;
+  NumFrames: integer; // number of frames (stereo: 2xsamples)
+  i: integer;
+begin
+  // set boost
+  case Ini.MicBoost of
+    0:   Boost := 1;
+    1:   Boost := 2;
+    2:   Boost := 4;
+    3:   Boost := 8;
+    else Boost := 1;
+  end;
+
+  AudioFormat := InputDevice.AudioFormat;
+
+  // FIXME: At the moment we assume a SInt16 format
+  // TODO:  use SDL_AudioConvert to convert to SInt16 but do NOT change the
+  // samplerate (SDL does not convert 44.1kHz to 48kHz so we might get wrong
+  // results in the analysis phase otherwise)
+  if (AudioFormat.Format <> asfS16) then
+  begin
+    // this only occurs if a developer choosed a wrong input sample-format
+    Log.CriticalError('TAudioInputProcessor.HandleMicrophoneData: Wrong sample-format');
+    Exit;
+  end;
+
+  // interpret buffer as buffer of bytes
+  SampleBuffer := Buffer;
+
+  NumSamples := Size div SizeOf(Smallint);
+
+  // boost buffer
+  // TODO: remove this senseless stuff - adjust the threshold instead
+  for i := 0 to NumSamples-1 do
+  begin
+    Value := SampleBuffer^[i] * Boost;
+
+    // TODO :  JB -  This will clip the audio... cant we reduce the "Boost" if the data clips ??
+    if Value > High(Smallint) then
+      Value := High(Smallint);
+
+    if Value < Low(Smallint) then
+      Value := Low(Smallint);
+
+    SampleBuffer^[i] := Value;
+  end;
+
+  // samples per channel
+  FrameSize := AudioFormat.Channels * SizeOf(SmallInt);
+  NumFrames := Size div FrameSize;
+
+  // process channels
+  for ChannelIndex := 0 to High(InputDevice.CaptureChannel) do
+  begin
+    CaptureChannel := InputDevice.CaptureChannel[ChannelIndex];
+    if (CaptureChannel <> nil) then
+    begin
+      // set offset according to channel index
+      ChannelBuffer := @PChar(Buffer)[ChannelIndex * SizeOf(SmallInt)];
+
+      // TODO: remove BufferNew and write to BufferArray directly
+
+      CaptureChannel.BufferNew.Clear;
+      for i := 0 to NumFrames-1 do
+      begin
+        CaptureChannel.BufferNew.Write(ChannelBuffer[i*FrameSize], SizeOf(SmallInt));
+      end;
+      CaptureChannel.ProcessNewBuffer();
+    end;
+  end;
+end;
+
+
+{ TAudioInputBase }
+
+{*
+ * Start capturing on all used input-device.
+ *}
+procedure TAudioInputBase.CaptureStart;
+var
+  S:  integer;
+  DeviceIndex: integer;
+  ChannelIndex: integer;
+  Device: TAudioInputDevice;
+  DeviceCfg: PInputDeviceConfig;
+  DeviceUsed: boolean;
+  Player: integer;
+begin
+  if (Started) then
+    CaptureStop();
+
+  // reset buffers
+  for S := 0 to High(AudioInputProcessor.Sound) do
+    AudioInputProcessor.Sound[S].Clear;
+
+  // start capturing on each used device
+  for DeviceIndex := 0 to High(AudioInputProcessor.Device) do
+  begin
+    Device := AudioInputProcessor.Device[DeviceIndex];
+    if not assigned(Device) then
+      continue;
+    DeviceCfg := @Ini.InputDeviceConfig[Device.CfgIndex];
+
+    DeviceUsed := false;
+
+    // check if device is used
+    for ChannelIndex := 0 to High(DeviceCfg.ChannelToPlayerMap) do
+    begin
+      Player := DeviceCfg.ChannelToPlayerMap[ChannelIndex]-1;
+      if (Player < 0) or (Player >= PlayersPlay) then
+      begin
+        Device.LinkCaptureBuffer(ChannelIndex, nil);
+      end
+      else
+      begin
+        Device.LinkCaptureBuffer(ChannelIndex, AudioInputProcessor.Sound[Player]);
+        DeviceUsed := true;
+      end;
+    end;
+
+    // start device if used
+    if (DeviceUsed) then
+    begin
+      //Log.BenchmarkStart(2);
+      Device.Start();
+      //Log.BenchmarkEnd(2);
+      //Log.LogBenchmark('Device.Start', 2) ;
+    end;
+  end;
+
+  Started := true;
+end;
+
+{*
+ * Stop input-capturing on all soundcards.
+ *}
+procedure TAudioInputBase.CaptureStop;
+var
+  DeviceIndex: integer;
+// Player:  integer; // Auto Removed, Unused Variable
+  Device: TAudioInputDevice;
+// DeviceCfg: PInputDeviceConfig; // Auto Removed, Unused Variable
+begin
+  for DeviceIndex := 0 to High(AudioInputProcessor.Device) do
+  begin
+    Device := AudioInputProcessor.Device[DeviceIndex];
+    if not assigned(Device) then
+      continue;
+    Device.Stop();
+  end;
+
+  Started := false;
+end;
+
+function TAudioInputBase.UnifyDeviceName(const name: string; deviceIndex: integer): string;
+var
+  count: integer; // count of devices with this name
+
+  function IsDuplicate(const name: string): boolean;
+  var
+    i: integer;
+  begin
+    Result := False;
+    // search devices with same description
+    For i := 0 to deviceIndex-1 do
+    begin
+      if (AudioInputProcessor.Device[i].Description = name) then
+      begin
+        Result := True;
+        Break;
+      end;
+    end;
+  end;
+begin
+  count := 1;
+  result := name;
+
+  // if there is another device with the same ID, search for an available name
+  while (IsDuplicate(result)) do
+  begin
+    Inc(count);
+    // set description
+    result := name + ' ('+IntToStr(count)+')';
+  end;
+end;
+
+{*
+ * Unifies an input-device's source name.
+ * Note: the description member of the device must already be set when
+ * calling this function.
+ *}
+function TAudioInputBase.UnifyDeviceSourceName(const name: string; const deviceName: string): string;
+//var
+// Descr: string; // Auto Removed, Unused Variable
+begin
+  result := name;
+
+  {$IFDEF DARWIN}
+    // Under MacOSX the SingStar Mics have an empty
+    // InputName. So, we have to add a hard coded
+    // Workaround for this problem
+    if (name = '') and (Pos( 'USBMIC Serial#', deviceName) > 0) then
+    begin
+      result := 'Microphone';
+    end;
+  {$ENDIF}
+end;
+
+end.
+
+
+