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-rw-r--r--src/base/URecord.pas766
1 files changed, 766 insertions, 0 deletions
diff --git a/src/base/URecord.pas b/src/base/URecord.pas
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+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.
+
+
+
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