diff options
Diffstat (limited to '')
| -rw-r--r-- | Game/Code/Classes/URecord.pas | 535 |
1 files changed, 0 insertions, 535 deletions
diff --git a/Game/Code/Classes/URecord.pas b/Game/Code/Classes/URecord.pas deleted file mode 100644 index 8ae0978a..00000000 --- a/Game/Code/Classes/URecord.pas +++ /dev/null @@ -1,535 +0,0 @@ -unit URecord;
-
-interface
-
-{$IFDEF FPC}
- {$MODE Delphi}
-{$ENDIF}
-
-{$I switches.inc}
-
-uses Classes,
- Math,
- SysUtils,
- UCommon,
- UMusic,
- UIni;
-
-type
- TSound = class
- private
- BufferNew: TMemoryStream; // buffer for newest samples
- public
- BufferArray: array[0..4095] of smallint; // newest 4096 samples
- BufferLong: array of TMemoryStream; // full buffer
-
- Index: integer; // index in TAudioInputProcessor.Sound[] (TODO: Remove if not used)
-
- AnalysisBufferSize: integer; // number of samples to analyze
-
- // pitch detection
- ToneValid: boolean; // true if Tone contains a valid value (otherwise it contains noise)
- //Peak: integer; // position of peak on horizontal pivot (TODO: Remove if not used)
- //ToneAccuracy: real; // tone accuracy (TODO: Remove if not used)
- Tone: integer; // TODO: should be a non-unified full range tone (e.g. C2<>C3). Range: 0..NumHalftones-1
- // Note: at the moment it is the same as ToneUnified
- ToneUnified: integer; // tone unified to one octave (e.g. C2=C3=C4). Range: 0-11
- //Scale: real; // FFT scale (TODO: Remove if not used)
-
- // procedures
- procedure ProcessNewBuffer;
- procedure AnalyzeBuffer; // use to analyze sound from buffers to get new pitch
- procedure AnalyzeByAutocorrelation; // we call it to analyze sound by checking Autocorrelation
- function AnalyzeAutocorrelationFreq(Freq: real): real; // use this to check one frequency by Autocorrelation
- 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?
- MicInput: integer; // unused. What is this good for?
- SampleRate: integer; // capture sample-rate (e.g. 44.1kHz -> 44100)
- CaptureChannel: array[0..1] of TSound; // sound(-buffers) used for left/right channel's capture data
-
- procedure Start(); virtual; abstract;
- procedure Stop(); virtual; abstract;
-
- destructor Destroy; override;
- end;
-
- TAudioInputProcessor = class
- Sound: array of TSound;
- Device: array of TAudioInputDevice;
-
- constructor Create;
-
- // handle microphone input
- procedure HandleMicrophoneData(Buffer: Pointer; Size: Cardinal;
- InputDevice: TAudioInputDevice);
-
- function Volume( aChannel : byte ): byte;
- 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;
-
-const
- CaptureFreq = 44100;
- BaseToneFreq = 65.4064; // lowest (half-)tone to analyze (C2 = 65.4064 Hz)
- NumHalftones = 36; // C2-B4 (for Whitney and my high voice)
-
-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;
- for i := 0 to High(CaptureChannel) do
- CaptureChannel[i] := nil;
- inherited Destroy;
-end;
-
-
-{ TSound }
-
-procedure TSound.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[0].CopyFrom(BufferNew, BufferNew.Size);
- end;
-end;
-
-procedure TSound.AnalyzeBuffer;
-begin
- AnalyzeByAutocorrelation;
-end;
-
-procedure TSound.AnalyzeByAutocorrelation;
-var
- ToneIndex: integer;
- Freq: real;
- Wages: array[0..NumHalftones-1] of real;
- MaxTone: integer;
- MaxWage: real;
- Volume: real;
- MaxVolume: real;
- SampleIndex: integer;
- Threshold: real;
-const
- HalftoneBase = 1.05946309436; // 2^(1/12) -> HalftoneBase^12 = 2 (one octave)
-begin
- ToneValid := false;
-
- // find maximum volume of first 1024 samples
- MaxVolume := 0;
- for SampleIndex := 0 to 1023 do
- begin
- Volume := Abs(BufferArray[SampleIndex]) /
- -Low(Smallint); // was $10000 (65536) before but must be 32768
-
- if Volume > MaxVolume then
- MaxVolume := Volume;
- end;
-
- // prepare to analyze
- MaxWage := 0;
-
- // analyze halftones
- for ToneIndex := 0 to NumHalftones-1 do
- begin
- Freq := BaseToneFreq * Power(HalftoneBase, ToneIndex);
- Wages[ToneIndex] := AnalyzeAutocorrelationFreq(Freq);
-
- if Wages[ToneIndex] > MaxWage then
- begin
- // this frequency has better wage
- MaxWage := Wages[ToneIndex];
- MaxTone := ToneIndex;
- end;
- end;
-
- Threshold := 0.2;
- case Ini.Threshold of
- 0: Threshold := 0.1;
- 1: Threshold := 0.2;
- 2: Threshold := 0.3;
- 3: Threshold := 0.4;
- end;
-
- // check if signal has an acceptable volume (ignore background-noise)
- if MaxVolume >= Threshold then
- begin
- ToneValid := true;
- ToneUnified := MaxTone mod 12;
- Tone := MaxTone mod 12;
- end;
-
-end;
-
-function TSound.AnalyzeAutocorrelationFreq(Freq: real): real; // result medium difference
-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(CaptureFreq/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); // was $10000 (65536) before but must be 65535
- AccumDist := AccumDist + Dist;
- Inc(SampleIndex);
- Inc(CorrelatingSampleIndex);
- end;
-
- // return "inverse" average distance (=correlation)
- Result := 1 - AccumDist / AnalysisBufferSize;
-end;
-
-
-{ TAudioInputProcessor }
-
-{*
- * 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
- NumSamples: integer; // number of samples
- SampleIndex: integer;
- Value: integer;
- ByteBuffer: PByteArray; // buffer handled as array of bytes
- SampleBuffer: PSmallIntArray; // buffer handled as array of samples
- Offset: integer;
- Boost: byte;
- ChannelCount: integer;
- ChannelIndex: integer;
- CaptureChannel: TSound;
- SampleSize: integer;
-begin
- // set boost
- case Ini.MicBoost of
- 0: Boost := 1;
- 1: Boost := 2;
- 2: Boost := 4;
- 3: Boost := 8;
- end;
-
- // boost buffer
- NumSamples := Size div 2;
- SampleBuffer := Buffer;
- for SampleIndex := 0 to NumSamples-1 do
- begin
- Value := SampleBuffer^[SampleIndex] * 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^[SampleIndex] := Value;
- end;
-
- // number of channels
- ChannelCount := Length(InputDevice.CaptureChannel);
- // size of one sample
- SampleSize := ChannelCount * SizeOf(SmallInt);
- // samples per channel
- NumSamples := Size div SampleSize;
-
- // interpret buffer as buffer of bytes
- ByteBuffer := Buffer;
-
- // process channels
- for ChannelIndex := 0 to High(InputDevice.CaptureChannel) do
- begin
- CaptureChannel := InputDevice.CaptureChannel[ChannelIndex];
- if (CaptureChannel <> nil) then
- begin
- Offset := ChannelIndex * SizeOf(SmallInt);
-
- // TODO: remove BufferNew and write to BufferArray directly
-
- CaptureChannel.BufferNew.Clear;
- for SampleIndex := 0 to NumSamples-1 do
- begin
- CaptureChannel.BufferNew.Write(ByteBuffer^[Offset + SampleIndex*SampleSize],
- SizeOf(SmallInt));
- end;
- CaptureChannel.ProcessNewBuffer();
- end;
- end;
-end;
-
-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] := TSound.Create;
- Sound[i].Index := i;
- Sound[i].BufferNew := TMemoryStream.Create;
- SetLength(Sound[i].BufferLong, 1);
- Sound[i].BufferLong[0] := TMemoryStream.Create;
- Sound[i].AnalysisBufferSize := Min(4*1024, Length(Sound[i].BufferArray));
- end;
-end;
-
-function TAudioInputProcessor.Volume( aChannel : byte ): byte;
-var
- lSampleIndex: Integer;
- lMaxVol : Word;
-begin;
- with AudioInputProcessor.Sound[aChannel] do
- begin
- lMaxVol := BufferArray[0];
- for lSampleIndex := 1 to High(BufferArray) do
- begin
- if Abs(BufferArray[lSampleIndex]) > lMaxVol then
- lMaxVol := Abs(BufferArray[lSampleIndex]);
- end;
- end;
-
- result := trunc( ( 255 / -Low(Smallint) ) * lMaxVol );
-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();
-
- Log.BenchmarkStart(1);
-
- // reset buffers
- for S := 0 to High(AudioInputProcessor.Sound) do
- AudioInputProcessor.Sound[S].BufferLong[0].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.CaptureChannel[ChannelIndex] := nil;
- end
- else
- begin
- Device.CaptureChannel[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;
-
- Log.BenchmarkEnd(1);
- Log.LogBenchmark('CaptureStart', 1) ; -
- 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.
-
-
-
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