rename Classes part2

git-svn-id: svn://svn.code.sf.net/p/ultrastardx/svn/trunk@1308 b956fd51-792f-4845-bead-9b4dfca2ff2c

1 files changed, 0 insertions, 766 deletions

diff --git a/src/classes0/URecord.pas b/src/classes0/URecord.pas
deleted file mode 100644
index 8a537dc9..00000000
--- a/src/classes0/URecord.pas
+++ /dev/null
@@ -1,766 +0,0 @@
-unit URecord;
-
-interface
-
-{$IFDEF FPC}
-  {$MODE Delphi}
-{$ENDIF}
-
-{$I switches.inc}
-
-uses Classes,
-     Math,
-     SysUtils,
-     sdl,
-     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
-      VoiceStream:  TAudioVoiceStream; // stream for voice passthrough
-      AnalysisBufferLock: PSDL_Mutex;
-
-      function GetToneString: string; // converts a tone to its string represenatation;
-
-      procedure BoostBuffer(Buffer: PChar; Size: Cardinal);
-      procedure ProcessNewBuffer(Buffer: PChar; BufferSize: integer);
-
-      // we call it to analyze sound by checking Autocorrelation
-      procedure AnalyzeByAutocorrelation;
-      // use this to check one frequency by Autocorrelation
-      function AnalyzeAutocorrelationFreq(Freq: real): real;
-    public
-      AnalysisBuffer:  array[0..4095] of smallint; // newest 4096 samples
-      AnalysisBufferSize: integer; // number of samples of BufferArray to analyze
-
-      LogBuffer:   TMemoryStream;              // full buffer
-
-      AudioFormat: TAudioFormatInfo;
-
-      // pitch detection
-      // TODO: remove ToneValid, set Tone/ToneAbs=-1 if invalid instead
-      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;
-
-      // use to analyze sound from buffers to get new pitch
-      procedure AnalyzeBuffer;
-      procedure LockAnalysisBuffer();   {$IFDEF HasInline}inline;{$ENDIF}
-      procedure UnlockAnalysisBuffer(); {$IFDEF HasInline}inline;{$ENDIF}
-
-      function MaxSampleVolume: Single;
-      property ToneString: string READ GetToneString;
-  end;
-
-const
-  DEFAULT_SOURCE_NAME = '[Default]';
-
-type
-  TAudioInputSource = record
-    Name:   string;
-  end;
-
-  // soundcard input-devices information
-  TAudioInputDevice = class
-    public
-      CfgIndex:        integer;   // index of this device in Ini.InputDeviceConfig
-      Name:            string;    // soundcard name
-      Source:          array of TAudioInputSource; // soundcard input-sources
-      SourceRestore:   integer;  // source-index that will be selected after capturing (-1: not detected)
-      MicSource:       integer;  // source-index of mic (-1: none detected)
-
-      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);
-
-      // TODO: add Open/Close functions so Start/Stop becomes faster
-      //function Open(): boolean;  virtual; abstract;
-      //function Close(): boolean; virtual; abstract;
-      function Start(): boolean; virtual; abstract;
-      function Stop(): boolean;  virtual; abstract;
-
-      function GetVolume(): single;        virtual; abstract;
-      procedure SetVolume(Volume: single); virtual; abstract;
-  end;
-
-  TAudioInputProcessor = class
-    public
-      Sound:  array of TCaptureBuffer; // sound-buffers for every player
-      DeviceList: array of TAudioInputDevice;
-
-      constructor Create;
-      destructor Destroy; override;
-
-      procedure UpdateInputDeviceConfig;
-
-      // handle microphone input
-      procedure HandleMicrophoneData(Buffer: PChar; Size: Cardinal;
-                                     InputDevice: TAudioInputDevice);
-  end;
-
-  TAudioInputBase = class( TInterfacedObject, IAudioInput )
-    private
-      Started: boolean;
-    protected
-      function UnifyDeviceName(const name: string; deviceIndex: integer): string;
-    public
-      function GetName: String;           virtual; abstract;
-      function InitializeRecord: boolean; virtual; abstract;
-      function FinalizeRecord: boolean;   virtual;
-
-      procedure CaptureStart;
-      procedure CaptureStop;
-  end;
-
-
-  TSmallIntArray = array [0..(MaxInt div SizeOf(SmallInt))-1] of SmallInt;
-  PSmallIntArray = ^TSmallIntArray;
-
-  function AudioInputProcessor(): TAudioInputProcessor;
-
-implementation
-
-uses
-  ULog,
-  UMain;
-
-var
-  singleton_AudioInputProcessor : TAudioInputProcessor = nil;
-
-
-{ Global }
-
-function AudioInputProcessor(): TAudioInputProcessor;
-begin
-  if singleton_AudioInputProcessor = nil then
-    singleton_AudioInputProcessor := TAudioInputProcessor.create();
-
-  result := singleton_AudioInputProcessor;
-end;
-
-
-{ TAudioInputDevice }
-
-destructor TAudioInputDevice.Destroy;
-begin
-  Stop();
-  Source := nil;
-  CaptureChannel := nil;
-  FreeAndNil(AudioFormat);
-  inherited Destroy;
-end;
-
-procedure TAudioInputDevice.LinkCaptureBuffer(ChannelIndex: integer; Sound: TCaptureBuffer);
-var
-  DeviceCfg: PInputDeviceConfig;
-  OldSound: TCaptureBuffer;
-begin
-  // check bounds
-  if ((ChannelIndex < 0) or (ChannelIndex > High(CaptureChannel))) then
-    Exit;
-
-  // reset previously assigned (old) capture-buffer
-  OldSound := CaptureChannel[ChannelIndex];
-  if (OldSound <> nil) then
-  begin
-    // close voice stream
-    FreeAndNil(OldSound.VoiceStream);
-    // free old audio-format info
-    FreeAndNil(OldSound.AudioFormat);
-  end;
-
-  // set audio-format of new capture-buffer
-  if (Sound <> nil) then
-  begin
-    // copy the input-device audio-format ...
-    Sound.AudioFormat := AudioFormat.Copy;
-    // and adjust it because capture buffers are always mono
-    Sound.AudioFormat.Channels := 1;
-    DeviceCfg := @Ini.InputDeviceConfig[CfgIndex];
-
-    if (Ini.VoicePassthrough = 1) then
-    begin
-      // TODO: map odd players to the left and even players to the right speaker
-      Sound.VoiceStream := AudioPlayback.CreateVoiceStream(CHANNELMAP_FRONT, AudioFormat);
-    end;
-  end;
-
-  // replace old with new buffer (Note: Sound might be nil)
-  CaptureChannel[ChannelIndex] := Sound;
-end;
-
-{ TSound }
-
-constructor TCaptureBuffer.Create;
-begin
-  inherited;
-  LogBuffer := TMemoryStream.Create;
-  AnalysisBufferLock := SDL_CreateMutex();
-  AnalysisBufferSize := Length(AnalysisBuffer);
-end;
-
-destructor TCaptureBuffer.Destroy;
-begin
-  FreeAndNil(LogBuffer);
-  FreeAndNil(VoiceStream);
-  FreeAndNil(AudioFormat);
-  SDL_DestroyMutex(AnalysisBufferLock);
-  inherited;
-end;
-
-procedure TCaptureBuffer.LockAnalysisBuffer();
-begin
-  SDL_mutexP(AnalysisBufferLock);
-end;
-
-procedure TCaptureBuffer.UnlockAnalysisBuffer();
-begin
-  SDL_mutexV(AnalysisBufferLock);
-end;
-
-procedure TCaptureBuffer.Clear;
-begin
-  if assigned(LogBuffer) then
-    LogBuffer.Clear;
-  LockAnalysisBuffer();
-  FillChar(AnalysisBuffer[0], Length(AnalysisBuffer) * SizeOf(SmallInt), 0);
-  UnlockAnalysisBuffer();
-end;
-
-procedure TCaptureBuffer.ProcessNewBuffer(Buffer: PChar; BufferSize: integer);
-var
-  BufferOffset: integer;
-  SampleCount: integer;
-  i: integer;
-begin
-  // apply software boost
-  //BoostBuffer(Buffer, Size);
-
-  // voice passthrough (send data to playback-device)
-  if (assigned(VoiceStream)) then
-    VoiceStream.WriteData(Buffer, BufferSize);
-
-  // we assume that samples are in S16Int format
-  // TODO: support float too
-  if (AudioFormat.Format <> asfS16) then
-    Exit;
-
-  // process BufferArray
-  BufferOffset := 0;
-
-  SampleCount := BufferSize div SizeOf(SmallInt);
-
-  // check if we have more new samples than we can store
-  if (SampleCount > Length(AnalysisBuffer)) then
-  begin
-    // discard the oldest of the new samples
-    BufferOffset := (SampleCount - Length(AnalysisBuffer)) * SizeOf(SmallInt);
-    SampleCount := Length(AnalysisBuffer);
-  end;
-
-
-  LockAnalysisBuffer();
-  try
-
-    // move old samples to the beginning of the array (if necessary)
-    for i := 0 to High(AnalysisBuffer)-SampleCount do
-      AnalysisBuffer[i] := AnalysisBuffer[i+SampleCount];
-
-    // copy new samples to analysis buffer
-    Move(Buffer[BufferOffset], AnalysisBuffer[Length(AnalysisBuffer)-SampleCount],
-         SampleCount * SizeOf(SmallInt));
-
-  finally
-    UnlockAnalysisBuffer();
-  end;
-
-
-  // save capture-data to BufferLong if enabled
-  if (Ini.SavePlayback = 1) then
-  begin
-    // this is just for debugging (approx 15MB per player for a 3min song!!!)
-    // For an in-game replay-mode we need to compress data so we do not
-    // waste that much memory. Maybe ogg-vorbis with voice-preset in fast-mode?
-    // Or we could use a faster but not that efficient lossless compression.
-    LogBuffer.WriteBuffer(Buffer, BufferSize);
-  end;
-end;
-
-procedure TCaptureBuffer.AnalyzeBuffer;
-var
-  Volume:    single;
-  MaxVolume: single;
-  SampleIndex: integer;
-  Threshold: single;
-begin
-  ToneValid := false;
-  ToneAbs := -1;
-  Tone    := -1;
-
-  LockAnalysisBuffer();
-  try
-
-    // find maximum volume of first 1024 samples
-    MaxVolume := 0;
-    for SampleIndex := 0 to 1023 do
-    begin
-      Volume := Abs(AnalysisBuffer[SampleIndex]) / -Low(Smallint);
-      if Volume > MaxVolume then
-         MaxVolume := Volume;
-    end;
-
-    Threshold := IThresholdVals[Ini.ThresholdIndex];
-
-    // check if signal has an acceptable volume (ignore background-noise)
-    if MaxVolume >= Threshold then
-    begin
-      // analyse the current voice pitch
-      AnalyzeByAutocorrelation;
-      ToneValid := true;
-    end;
-
-  finally
-    UnlockAnalysisBuffer();
-  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;
-  MaxTone := 0; // this is not needed, but it satifies the compiler
-
-  // 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(AnalysisBuffer[SampleIndex] - AnalysisBuffer[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;
-  LockAnalysisBuffer();
-  try
-    lMaxVol := 0;
-    for lSampleIndex := 0 to High(AnalysisBuffer) do
-    begin
-      if Abs(AnalysisBuffer[lSampleIndex]) > lMaxVol then
-        lMaxVol := Abs(AnalysisBuffer[lSampleIndex]);
-    end;
-  finally
-    UnlockAnalysisBuffer();
-  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;
-
-procedure TCaptureBuffer.BoostBuffer(Buffer: PChar; Size: Cardinal);
-var
-  i: integer;
-  Value: Longint;
-  SampleCount: integer;
-  SampleBuffer: PSmallIntArray; // buffer handled as array of samples
-  Boost:  byte;
-begin
-  // TODO: set boost per device
-  {
-  case Ini.MicBoost of
-    0:   Boost := 1;
-    1:   Boost := 2;
-    2:   Boost := 4;
-    3:   Boost := 8;
-    else Boost := 1;
-  end;
-  }
-  Boost := 1;
-
-  // at the moment we will boost SInt16 data only
-  if (AudioFormat.Format = asfS16) then
-  begin
-    // interpret buffer as buffer of bytes
-    SampleBuffer := PSmallIntArray(Buffer);
-    SampleCount := Size div AudioFormat.FrameSize;
-
-    // boost buffer
-    for i := 0 to SampleCount-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;
-  end;
-end;
-
-
-{ TAudioInputProcessor }
-
-constructor TAudioInputProcessor.Create;
-var
-  i: integer;
-begin
-  inherited;
-  SetLength(Sound, 6 {max players});//Ini.Players+1);
-  for i := 0 to High(Sound) do
-    Sound[i] := TCaptureBuffer.Create;
-end;
-
-destructor TAudioInputProcessor.Destroy;
-var
-  i: integer;
-begin
-  for i := 0 to High(Sound) do
-    Sound[i].Free;
-  SetLength(Sound, 0);
-  inherited;
-end;
-
-// updates InputDeviceConfig with current input-device information
-// See: TIni.LoadInputDeviceCfg()
-procedure TAudioInputProcessor.UpdateInputDeviceConfig;
-var
-  deviceIndex: integer;
-  newDevice: boolean;
-  deviceIniIndex: integer;
-  deviceCfg: PInputDeviceConfig;
-  device: TAudioInputDevice;
-  channelCount: integer;
-  channelIndex: integer;
-  i: integer;
-begin
-  // Input devices - append detected soundcards
-  for deviceIndex := 0 to High(DeviceList) do
-  begin
-    newDevice := true;
-    //Search for Card in List
-    for deviceIniIndex := 0 to High(Ini.InputDeviceConfig) do
-    begin
-      deviceCfg := @Ini.InputDeviceConfig[deviceIniIndex];
-      device := DeviceList[deviceIndex];
-
-      if (deviceCfg.Name = Trim(device.Name)) then
-      begin
-        newDevice := false;
-
-        // store highest channel index as an offset for the new channels
-        channelIndex := High(deviceCfg.ChannelToPlayerMap);
-        // add missing channels or remove non-existing ones
-        SetLength(deviceCfg.ChannelToPlayerMap, device.AudioFormat.Channels);
-        // initialize added channels to 0
-        for i := channelIndex+1 to High(deviceCfg.ChannelToPlayerMap) do
-        begin
-          deviceCfg.ChannelToPlayerMap[i] := 0;
-        end;
-
-        // associate ini-index with device
-        device.CfgIndex := deviceIniIndex;
-        break;
-      end;
-    end;
-
-    //If not in List -> Add
-    if newDevice then
-    begin
-      // resize list
-      SetLength(Ini.InputDeviceConfig, Length(Ini.InputDeviceConfig)+1);
-      deviceCfg := @Ini.InputDeviceConfig[High(Ini.InputDeviceConfig)];
-      device := DeviceList[deviceIndex];
-
-      // associate ini-index with device
-      device.CfgIndex := High(Ini.InputDeviceConfig);
-
-      deviceCfg.Name := Trim(device.Name);
-      deviceCfg.Input := 0;
-
-      channelCount := device.AudioFormat.Channels;
-      SetLength(deviceCfg.ChannelToPlayerMap, channelCount);
-
-      for channelIndex := 0 to channelCount-1 do
-      begin
-        // set default at first start of USDX (1st device, 1st channel -> player1)
-        if ((channelIndex = 0) and (device.CfgIndex = 0)) then
-          deviceCfg.ChannelToPlayerMap[0] := 1
-        else
-          deviceCfg.ChannelToPlayerMap[channelIndex] := 0;
-      end;
-    end;
-  end;
-end;
-
-{*
- * Handles 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: PChar; Size: Cardinal; InputDevice: TAudioInputDevice);
-var
-  MultiChannelBuffer: PChar;  // buffer handled as array of bytes (offset relative to channel)
-  SingleChannelBuffer: PChar; // temporary buffer for new samples per channel
-  SingleChannelBufferSize: integer;
-  ChannelIndex: integer;
-  CaptureChannel: TCaptureBuffer;
-  AudioFormat: TAudioFormatInfo;
-  SampleSize: integer;
-  SampleCount: integer;
-  SamplesPerChannel: integer;
-  i: integer;
-begin
-  AudioFormat := InputDevice.AudioFormat;
-  SampleSize := AudioSampleSize[AudioFormat.Format];
-  SampleCount := Size div SampleSize;
-  SamplesPerChannel := Size div AudioFormat.FrameSize;
-
-  SingleChannelBufferSize := SamplesPerChannel * SampleSize;
-  GetMem(SingleChannelBuffer, SingleChannelBufferSize);
-
-  // process channels
-  for ChannelIndex := 0 to High(InputDevice.CaptureChannel) do
-  begin
-    CaptureChannel := InputDevice.CaptureChannel[ChannelIndex];
-    // check if a capture buffer was assigned, otherwise there is nothing to do
-    if (CaptureChannel <> nil) then
-    begin
-      // set offset according to channel index
-      MultiChannelBuffer := @Buffer[ChannelIndex * SampleSize];
-      // seperate channel-data from interleaved multi-channel (e.g. stereo) data
-      for i := 0 to SamplesPerChannel-1 do
-      begin
-        Move(MultiChannelBuffer[i*AudioFormat.FrameSize],
-             SingleChannelBuffer[i*SampleSize],
-             SampleSize);
-      end;
-      CaptureChannel.ProcessNewBuffer(SingleChannelBuffer, SingleChannelBufferSize);
-    end;
-  end;
-
-  FreeMem(SingleChannelBuffer);
-end;
-
-
-{ TAudioInputBase }
-
-function TAudioInputBase.FinalizeRecord: boolean;
-var
-  i: integer;
-begin
-  for i := 0 to High(AudioInputProcessor.DeviceList) do
-    AudioInputProcessor.DeviceList[i].Free();
-  AudioInputProcessor.DeviceList := nil;
-  Result := true;
-end;
-
-{*
- * 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.DeviceList) do
-  begin
-    Device := AudioInputProcessor.DeviceList[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;
-  ChannelIndex: integer;
-  Device: TAudioInputDevice;
-  DeviceCfg: PInputDeviceConfig;
-begin
-  for DeviceIndex := 0 to High(AudioInputProcessor.DeviceList) do
-  begin
-    Device := AudioInputProcessor.DeviceList[DeviceIndex];
-    if not assigned(Device) then
-      continue;
-
-    Device.Stop();
-
-    // disconnect capture buffers
-    DeviceCfg := @Ini.InputDeviceConfig[Device.CfgIndex];
-    for ChannelIndex := 0 to High(DeviceCfg.ChannelToPlayerMap) do
-      Device.LinkCaptureBuffer(ChannelIndex, nil);
-  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.DeviceList[i].Name = 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;
-
-end.
-
-
-