FFT support for SDL/Portaudio playback

git-svn-id: svn://svn.code.sf.net/p/ultrastardx/svn/trunk@909 b956fd51-792f-4845-bead-9b4dfca2ff2c

3 files changed, 100 insertions, 38 deletions

diff --git a/Game/Code/Classes/UAudioInput_Portaudio.pas b/Game/Code/Classes/UAudioInput_Portaudio.pas
index 74c8d98f..077598a9 100644
--- a/Game/Code/Classes/UAudioInput_Portaudio.pas
+++ b/Game/Code/Classes/UAudioInput_Portaudio.pas
@@ -149,6 +149,7 @@ var
   stream:      PPaStream;
   streamInfo:  PPaStreamInfo;
   sampleRate:  integer;
+  latency:     TPaTime;
   {$IFDEF UsePortmixer}
   mixer:       PPxMixer;
   {$ENDIF}
@@ -210,13 +211,17 @@ begin
     else
       sampleRate := 44100;
 
+    // on vista and xp the defaultLowInputLatency may be set to 0 but it works.
+    // TODO: correct too low latencies (what is a too low latency, maybe < 10ms?)
+    latency := deviceInfo^.defaultLowInputLatency;
+
     // setup desired input parameters
     with inputParams do
     begin
       device := deviceIndex;
       channelCount := channelCnt;
       sampleFormat := paInt16;
-      suggestedLatency := deviceInfo^.defaultLowInputLatency;
+      suggestedLatency := latency;
       hostApiSpecificStreamInfo := nil;
     end;
 
diff --git a/Game/Code/Classes/UAudioPlayback_SoftMixer.pas b/Game/Code/Classes/UAudioPlayback_SoftMixer.pas
index e145587c..5beff682 100644
--- a/Game/Code/Classes/UAudioPlayback_SoftMixer.pas
+++ b/Game/Code/Classes/UAudioPlayback_SoftMixer.pas
@@ -24,6 +24,7 @@ type
 

       DecodeStream: TAudioDecodeStream;

       SampleBuffer    : PChar;

+      SampleBufferCount: integer; // number of available bytes in SampleBuffer

       SampleBufferPos : cardinal;

       BytesAvail: integer;

       cvt: TSDL_AudioCVT;

@@ -65,6 +66,8 @@ type
       function GetPosition: real;

       procedure SetPosition(Time: real);

       function ReadData(Buffer: PChar; BufSize: integer): integer;

+

+      procedure GetFFTData(var data: TFFTData);

   end;

 

   TAudioMixerStream = class

@@ -507,43 +510,50 @@ begin
   if (remFrameBytes > 0) then

     decodeBufSize := decodeBufSize + (frameSize - remFrameBytes);

 

-  // calc buffer size

-  sampleBufSize := decodeBufSize * cvt.len_mult;

-

-  // resize buffer if necessary.

-  // The required buffer-size will be smaller than the result-buffer

-  // in most cases (if the decoded signal is mono or has a lesser bitrate).

-  // If the output-rate is 44.1kHz and the decode-rate is 48kHz or 96kHz it

-  // will be ~1.09 or ~2.18 times bigger. Those extra memory consumption

-  // should be reasonable. If not we should call TDecodeStream.ReadData()

-  // multiple times.

-  // Note: we do not decrease the buffer by the count of bytes used from

-  //   the previous call of this function (bytesAvail). Otherwise the

-  //   buffer will be reallocated each time this function is called just to

-  //   add or remove a few bytes from the buffer.

-  //   By not doing this the buffer's size should be rather stable and it

-  //   will not be reallocated/resized at all if the BufSize params does not

-  //   change in consecutive calls.

-  ReallocMem(SampleBuffer, sampleBufSize);

-  if not assigned(SampleBuffer) then

-    Exit;

+  Lock();

+  try

+    // calc buffer size

+    sampleBufSize := decodeBufSize * cvt.len_mult;

+

+    // resize buffer if necessary.

+    // The required buffer-size will be smaller than the result-buffer

+    // in most cases (if the decoded signal is mono or has a lesser bitrate).

+    // If the output-rate is 44.1kHz and the decode-rate is 48kHz or 96kHz it

+    // will be ~1.09 or ~2.18 times bigger. Those extra memory consumption

+    // should be reasonable. If not we should call TDecodeStream.ReadData()

+    // multiple times.

+    // Note: we do not decrease the buffer by the count of bytes used from

+    //   the previous call of this function (bytesAvail). Otherwise the

+    //   buffer will be reallocated each time this function is called just to

+    //   add or remove a few bytes from the buffer.

+    //   By not doing this the buffer's size should be rather stable and it

+    //   will not be reallocated/resized at all if the BufSize params does not

+    //   change in consecutive calls.

+    ReallocMem(SampleBuffer, sampleBufSize);

+    if not assigned(SampleBuffer) then

+      Exit;

 

-  // decode data

-  nBytesDecoded := DecodeStream.ReadData(SampleBuffer, decodeBufSize);

-  if (nBytesDecoded = -1) then

-    Exit;

+    // decode data

+    nBytesDecoded := DecodeStream.ReadData(SampleBuffer, decodeBufSize);

+    if (nBytesDecoded = -1) then

+      Exit;

 

-  // end-of-file reached -> stop playback

-  if (DecodeStream.EOF) then

-    Stop();

+    // end-of-file reached -> stop playback

+    if (DecodeStream.EOF) then

+      Stop();

 

-  // resample decoded data

-  cvt.buf := PUint8(SampleBuffer);

-  cvt.len := nBytesDecoded;

-  if (SDL_ConvertAudio(@cvt) = -1) then

-    Exit;

+    // resample decoded data

+    cvt.buf := PUint8(SampleBuffer);

+    cvt.len := nBytesDecoded;

+    if (SDL_ConvertAudio(@cvt) = -1) then

+      Exit;

+

+    SampleBufferCount := cvt.len_cvt;

+  finally

+    Unlock();

+  end;

 

-  BytesAvail := cvt.len_cvt;

+  BytesAvail := SampleBufferCount;

   SampleBufferPos := 0;

 

   // copy data to result buffer

@@ -556,6 +566,50 @@ begin
   Result := BufSize - BytesNeeded;

 end;

 

+procedure TSoftMixerPlaybackStream.GetFFTData(var data: TFFTData);

+var

+  i: integer;

+  Frames: integer;

+  DataIn: PSingleArray;

+  AudioFormat: TAudioFormatInfo;

+begin

+  // only works with SInt16 and Float values at the moment

+  AudioFormat := Engine.GetAudioFormatInfo();

+

+  GetMem(DataIn, FFTSize * SizeOf(Single));

+

+  Lock();

+  // FIXME: We just use the first Frames frames, the others are ignored.

+  // This is OK for the equalizer display but not if we want to use

+  // this function for voice-analysis.

+  Frames := Min(FFTSize, SampleBufferCount div AudioFormat.FrameSize);

+  // use only first channel and convert data to float-values

+  case AudioFormat.Format of

+    asfS16:

+    begin

+      for i := 0 to Frames-1 do

+        DataIn[i] := PSmallInt(@SampleBuffer[i*AudioFormat.FrameSize])^ / -Low(SmallInt);

+    end;

+    asfFloat:

+    begin

+      for i := 0 to Frames-1 do

+        DataIn[i] := PSingle(@SampleBuffer[i*AudioFormat.FrameSize])^;

+    end;

+  end;

+  Unlock();

+

+  WindowFunc(FFTSize, DataIn);

+  PowerSpectrum(FFTSize, DataIn, @data);

+  FreeMem(DataIn);

+

+  // resize data to a 0..1 range

+  for i := 0 to High(TFFTData) do

+  begin

+    // TODO: this might need some work

+    data[i] := Sqrt(data[i]) / 100;

+  end;

+end;

+

 (* TODO: libsamplerate support

 function TSoftMixerPlaybackStream.ReadData(Buffer: PChar; BufSize: integer): integer;

 var

@@ -778,8 +832,8 @@ end;
 //Equalizer

 procedure TAudioPlayback_SoftMixer.GetFFTData(var data: TFFTData);

 begin

-  //Get Channel Data Mono and 256 Values

-//  BASS_ChannelGetData(Bass, @Result, BASS_DATA_FFT512);

+  if assigned(MusicStream) then

+    MusicStream.GetFFTData(data);

 end;

 

 // Interface for Visualizer

diff --git a/Game/Code/Classes/UMusic.pas b/Game/Code/Classes/UMusic.pas
index 44ae2a61..f0bf3a9c 100644
--- a/Game/Code/Classes/UMusic.pas
+++ b/Game/Code/Classes/UMusic.pas
@@ -85,10 +85,13 @@ type
     Razem:        real;       // (TODO: Razem = total time)

   end;

 

-  

+

+const

+  FFTSize = 512; // size of FFT data (output: FFTSize/2 values)

 type

-  TFFTData  = array[0..255] of Single;

+  TFFTData  = array[0..(FFTSize div 2-1)] of Single;

 

+type

   TPCMStereoSample = array[0..1] of Smallint;

   TPCMData  = array[0..511] of TPCMStereoSample;