{* UltraStar Deluxe - Karaoke Game
*
* UltraStar Deluxe is the legal property of its developers, whose names
* are too numerous to list here. Please refer to the COPYRIGHT
* file distributed with this source distribution.
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* as published by the Free Software Foundation; either version 2
* of the License, or (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; see the file COPYING. If not, write to
* the Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor,
* Boston, MA 02110-1301, USA.
*
* $URL$
* $Id$
*}
unit URecord;
interface
{$IFDEF FPC}
{$MODE Delphi}
{$ENDIF}
{$I switches.inc}
uses
Classes,
Math,
sdl,
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
VoiceStream: TAudioVoiceStream; // stream for voice passthrough
AnalysisBufferLock: PSDL_Mutex;
function GetToneString: string; // converts a tone to its string represenatation;
procedure BoostBuffer(Buffer: PByteArray; Size: integer);
procedure ProcessNewBuffer(Buffer: PByteArray; 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: UTF8String;
end;
// soundcard input-devices information
TAudioInputDevice = class
public
CfgIndex: integer; // index of this device in Ini.InputDeviceConfig
Name: UTF8String; // 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;
TBooleanDynArray = array of boolean;
TAudioInputProcessor = class
public
Sound: array of TCaptureBuffer; // sound-buffers for every player
DeviceList: array of TAudioInputDevice;
constructor Create;
destructor Destroy; override;
procedure UpdateInputDeviceConfig;
{**
* Validates the mic settings.
* If a player was assigned to multiple mics a popup will be displayed
* with the ID of the player.
* The return value is the player number of the first player that is not
* configured correctly or 0 if all players are correct.
*}
function ValidateSettings: integer;
{**
* Checks if players 1 to PlayerCount are configured correctly.
* A player is configured if a device's channel is assigned to him.
* For each player (up to PlayerCount) the state will be in PlayerState.
* If a player's state is true the player is configured, otherwise not.
* The return value is the player number of the first player that is not
* configured correctly or 0 if all players are correct.
* The PlayerState array is zero based (index 0 for player 1).
*}
function CheckPlayersConfig(PlayerCount: cardinal;
var PlayerState: TBooleanDynArray): integer; overload;
{**
* Same as the array version but it does not output a state for each player.
*}
function CheckPlayersConfig(PlayerCount: cardinal): integer; overload;
{**
* Handle microphone input
*}
procedure HandleMicrophoneData(Buffer: PByteArray; Size: integer;
InputDevice: TAudioInputDevice);
end;
TAudioInputBase = class( TInterfacedObject, IAudioInput )
private
Started: boolean;
protected
function UnifyDeviceName(const name: UTF8String; deviceIndex: integer): UTF8String;
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,
UNote;
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: PByteArray; BufferSize: integer);
var
BufferOffset: integer;
SampleCount: integer;
i: integer;
begin
// apply software boost
BoostBuffer(Buffer, BufferSize);
// 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: PByteArray; Size: integer);
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;
// 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;
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);
// assign added channels to no player
for i := channelIndex+1 to High(deviceCfg.ChannelToPlayerMap) do
begin
deviceCfg.ChannelToPlayerMap[i] := CHANNEL_OFF;
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;
deviceCfg.Latency := LATENCY_AUTODETECT;
channelCount := device.AudioFormat.Channels;
SetLength(deviceCfg.ChannelToPlayerMap, channelCount);
for channelIndex := 0 to channelCount-1 do
begin
// Do not set any default on first start of USDX.
// Otherwise most probably the wrong device (internal sound card)
// will be selected.
// It is better to force the user to configure the mics himself.
deviceCfg.ChannelToPlayerMap[channelIndex] := CHANNEL_OFF;
end;
end;
end;
end;
function TAudioInputProcessor.ValidateSettings: integer;
const
MAX_PLAYER_COUNT = 6; // FIXME: there should be a global variable for this
var
I, J: integer;
PlayerID: integer;
PlayerMap: array [0 .. MAX_PLAYER_COUNT - 1] of boolean;
InputDevice: TAudioInputDevice;
InputDeviceCfg: PInputDeviceConfig;
begin
// mark all players as unassigned
for I := 0 to High(PlayerMap) do
PlayerMap[I] := false;
// iterate over all active devices
for I := 0 to High(DeviceList) do
begin
InputDevice := DeviceList[I];
InputDeviceCfg := @Ini.InputDeviceConfig[InputDevice.CfgIndex];
// iterate over all channels of the current devices
for J := 0 to High(InputDeviceCfg.ChannelToPlayerMap) do
begin
// get player that was mapped to the current device channel
PlayerID := InputDeviceCfg.ChannelToPlayerMap[J];
if (PlayerID <> CHANNEL_OFF) then
begin
// check if player is already assigned to another device/channel
if (PlayerMap[PlayerID - 1]) then
begin
Result := PlayerID;
Exit;
end;
// mark player as assigned to a device
PlayerMap[PlayerID - 1] := true;
end;
end;
end;
Result := 0;
end;
function TAudioInputProcessor.CheckPlayersConfig(PlayerCount: cardinal;
var PlayerState: TBooleanDynArray): integer;
var
DeviceIndex: integer;
ChannelIndex: integer;
Device: TAudioInputDevice;
DeviceCfg: PInputDeviceConfig;
PlayerIndex: integer;
I: integer;
begin
SetLength(PlayerState, PlayerCount);
// set all entries to "not configured"
for I := 0 to High(PlayerState) do
begin
PlayerState[I] := false;
end;
// check 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];
// check if device is used
for ChannelIndex := 0 to High(DeviceCfg.ChannelToPlayerMap) do
begin
PlayerIndex := DeviceCfg.ChannelToPlayerMap[ChannelIndex] - 1;
if (PlayerIndex >= 0) and (PlayerIndex < PlayerCount) then
PlayerState[PlayerIndex] := true;
end;
end;
Result := 0;
for I := 0 to High(PlayerState) do
begin
if (PlayerState[I] = false) then
begin
Result := I + 1;
Break;
end;
end;
end;
function TAudioInputProcessor.CheckPlayersConfig(PlayerCount: cardinal): integer;
var
PlayerState: TBooleanDynArray;
begin
CheckPlayersConfig(PlayerCount, PlayerState);
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: PByteArray; Size: integer; InputDevice: TAudioInputDevice);
var
MultiChannelBuffer: PByteArray; // buffer handled as array of bytes (offset relative to channel)
SingleChannelBuffer: PByteArray; // temporary buffer for new samples per channel
SingleChannelBufferSize: integer;
ChannelIndex: integer;
CaptureChannel: TCaptureBuffer;
AudioFormat: TAudioFormatInfo;
SampleSize: integer;
SamplesPerChannel: integer;
i: integer;
begin
AudioFormat := InputDevice.AudioFormat;
SampleSize := AudioSampleSize[AudioFormat.Format];
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];
// separate 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: UTF8String; deviceIndex: integer): UTF8String;
var
count: integer; // count of devices with this name
function IsDuplicate(const name: UTF8String): boolean;
var
i: integer;
begin
Result := false;
// search devices with same description
for i := 0 to deviceIndex-1 do
begin
if (AudioInputProcessor.DeviceList[i] <> nil) then
begin
if (AudioInputProcessor.DeviceList[i].Name = name) then
begin
Result := true;
Break;
end;
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.