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Diffstat (limited to 'src/classes/URecord.pas')
| -rw-r--r-- | src/classes/URecord.pas | 766 |
1 files changed, 0 insertions, 766 deletions
diff --git a/src/classes/URecord.pas b/src/classes/URecord.pas deleted file mode 100644 index 8a537dc9..00000000 --- a/src/classes/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. - - - |