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Diffstat (limited to 'src/classes0/URecord.pas')
-rw-r--r-- | src/classes0/URecord.pas | 766 |
1 files changed, 766 insertions, 0 deletions
diff --git a/src/classes0/URecord.pas b/src/classes0/URecord.pas new file mode 100644 index 00000000..8a537dc9 --- /dev/null +++ b/src/classes0/URecord.pas @@ -0,0 +1,766 @@ +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. + + + |