unit URecord;
interface
{$IFDEF FPC}
{$MODE Delphi}
{$ENDIF}
{$I switches.inc}
uses Classes,
Math,
SysUtils,
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
BufferNew: TMemoryStream; // buffer for newest samples
function GetToneString: string; // converts a tone to its string represenatation;
public
BufferArray: array[0..4095] of smallint; // newest 4096 samples
BufferLong: TMemoryStream; // full buffer
AnalysisBufferSize: integer; // number of samples of BufferArray to analyze
AudioFormat: TAudioFormatInfo;
// pitch detection
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;
procedure ProcessNewBuffer;
// use to analyze sound from buffers to get new pitch
procedure AnalyzeBuffer;
// we call it to analyze sound by checking Autocorrelation
procedure AnalyzeByAutocorrelation;
// use this to check one frequency by Autocorrelation
function AnalyzeAutocorrelationFreq(Freq: real): real;
function MaxSampleVolume: Single;
property ToneString: string READ GetToneString;
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?
MicSource: integer; // unused. What is this good for?
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);
function Start(): boolean; virtual; abstract;
procedure Stop(); virtual; abstract;
end;
TAudioInputProcessor = class
public
Sound: array of TCaptureBuffer; // sound-buffers for every player
Device: array of TAudioInputDevice;
constructor Create;
// handle microphone input
procedure HandleMicrophoneData(Buffer: Pointer; Size: Cardinal;
InputDevice: TAudioInputDevice);
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;
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;
CaptureChannel := nil;
FreeAndNil(AudioFormat);
inherited Destroy;
end;
procedure TAudioInputDevice.LinkCaptureBuffer(ChannelIndex: integer; Sound: TCaptureBuffer);
begin
// check bounds
if ((ChannelIndex < 0) or (ChannelIndex > High(CaptureChannel))) then
Exit;
// reset audio-format of old capture-buffer
if (CaptureChannel[ChannelIndex] <> nil) then
CaptureChannel[ChannelIndex].AudioFormat := nil;
// set audio-format of new capture-buffer
if (Sound <> nil) then
Sound.AudioFormat := AudioFormat;
// replace old with new buffer
CaptureChannel[ChannelIndex] := Sound;
end;
{ TSound }
constructor TCaptureBuffer.Create;
begin
inherited;
BufferNew := TMemoryStream.Create;
BufferLong := TMemoryStream.Create;
AnalysisBufferSize := Min(4*1024, Length(BufferArray));
end;
destructor TCaptureBuffer.Destroy;
begin
AudioFormat := nil;
FreeAndNil(BufferNew);
FreeAndNil(BufferLong);
inherited;
end;
procedure TCaptureBuffer.Clear;
begin
if assigned(BufferNew) then
BufferNew.Clear;
if assigned(BufferLong) then
BufferLong.Clear;
FillChar(BufferArray[0], Length(BufferArray) * SizeOf(SmallInt), 0);
end;
procedure TCaptureBuffer.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.CopyFrom(BufferNew, BufferNew.Size);
end;
end;
procedure TCaptureBuffer.AnalyzeBuffer;
var
Volume: real;
MaxVolume: real;
SampleIndex: integer;
Threshold: real;
begin
ToneValid := false;
ToneAbs := -1;
Tone := -1;
// find maximum volume of first 1024 samples
MaxVolume := 0;
for SampleIndex := 0 to 1023 do
begin
Volume := Abs(BufferArray[SampleIndex]) / -Low(Smallint);
if Volume > MaxVolume then
MaxVolume := Volume;
end;
case Ini.Threshold of
0: Threshold := 0.05;
1: Threshold := 0.1;
2: Threshold := 0.15;
3: Threshold := 0.2;
else Threshold := 0.1;
end;
// check if signal has an acceptable volume (ignore background-noise)
if MaxVolume >= Threshold then
begin
// analyse the current voice pitch
AnalyzeByAutocorrelation;
ToneValid := true;
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;
// 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(BufferArray[SampleIndex] - BufferArray[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;
// FIXME: lock buffer to avoid race-conditions
lMaxVol := 0;
for lSampleIndex := 0 to High(BufferArray) do
begin
if Abs(BufferArray[lSampleIndex]) > lMaxVol then
lMaxVol := Abs(BufferArray[lSampleIndex]);
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;
{ TAudioInputProcessor }
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] := TCaptureBuffer.Create;
end;
end;
{*
* 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
Value: integer;
ChannelBuffer: PChar; // buffer handled as array of bytes (offset relative to channel)
SampleBuffer: PSmallIntArray; // buffer handled as array of samples
Boost: byte;
ChannelCount: integer;
ChannelIndex: integer;
ChannelOffset: integer;
CaptureChannel: TCaptureBuffer;
AudioFormat: TAudioFormatInfo;
FrameSize: integer;
NumSamples: integer;
NumFrames: integer; // number of frames (stereo: 2xsamples)
i: integer;
begin
// set boost
case Ini.MicBoost of
0: Boost := 1;
1: Boost := 2;
2: Boost := 4;
3: Boost := 8;
else Boost := 1;
end;
AudioFormat := InputDevice.AudioFormat;
// FIXME: At the moment we assume a SInt16 format
// TODO: use SDL_AudioConvert to convert to SInt16 but do NOT change the
// samplerate (SDL does not convert 44.1kHz to 48kHz so we might get wrong
// results in the analysis phase otherwise)
if (AudioFormat.Format <> asfS16) then
begin
// this only occurs if a developer choosed a wrong input sample-format
Log.CriticalError('TAudioInputProcessor.HandleMicrophoneData: Wrong sample-format');
Exit;
end;
// interpret buffer as buffer of bytes
SampleBuffer := Buffer;
NumSamples := Size div SizeOf(Smallint);
// boost buffer
// TODO: remove this senseless stuff - adjust the threshold instead
for i := 0 to NumSamples-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;
// samples per channel
FrameSize := AudioFormat.Channels * SizeOf(SmallInt);
NumFrames := Size div FrameSize;
// process channels
for ChannelIndex := 0 to High(InputDevice.CaptureChannel) do
begin
CaptureChannel := InputDevice.CaptureChannel[ChannelIndex];
if (CaptureChannel <> nil) then
begin
// set offset according to channel index
ChannelBuffer := @PChar(Buffer)[ChannelIndex * SizeOf(SmallInt)];
// TODO: remove BufferNew and write to BufferArray directly
CaptureChannel.BufferNew.Clear;
for i := 0 to NumFrames-1 do
begin
CaptureChannel.BufferNew.Write(ChannelBuffer[i*FrameSize], SizeOf(SmallInt));
end;
CaptureChannel.ProcessNewBuffer();
end;
end;
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();
// 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.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.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;
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.