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-rw-r--r--src/base/UAudioConverter.pas458
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diff --git a/src/base/UAudioConverter.pas b/src/base/UAudioConverter.pas
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--- a/src/base/UAudioConverter.pas
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-unit UAudioConverter;
-
-interface
-
-{$IFDEF FPC}
- {$MODE Delphi}
-{$ENDIF}
-
-{$I switches.inc}
-
-uses
- UMusic,
- ULog,
- ctypes,
- {$IFDEF UseSRCResample}
- samplerate,
- {$ENDIF}
- {$IFDEF UseFFmpegResample}
- avcodec,
- {$ENDIF}
- UMediaCore_SDL,
- sdl,
- SysUtils,
- Math;
-
-type
- {*
- * Notes:
- * - 44.1kHz to 48kHz conversion or vice versa is not supported
- * by SDL 1.2 (will be introduced in 1.3).
- * No conversion takes place in this cases.
- * This is because SDL just converts differences in powers of 2.
- * So the result might not be that accurate.
- * This IS audible (voice to high/low) and it needs good synchronization
- * with the video or the lyrics timer.
- * - float<->int16 conversion is not supported (will be part of 1.3) and
- * SDL (<1.3) is not capable of handling floats at all.
- * -> Using FFmpeg or libsamplerate for resampling is preferred.
- * Use SDL for channel and format conversion only.
- *}
- TAudioConverter_SDL = class(TAudioConverter)
- private
- cvt: TSDL_AudioCVT;
- public
- function Init(SrcFormatInfo: TAudioFormatInfo; DstFormatInfo: TAudioFormatInfo): boolean; override;
- destructor Destroy(); override;
-
- function Convert(InputBuffer: PChar; OutputBuffer: PChar; var InputSize: integer): integer; override;
- function GetOutputBufferSize(InputSize: integer): integer; override;
- function GetRatio(): double; override;
- end;
-
- {$IFDEF UseFFmpegResample}
- // Note: FFmpeg seems to be using "kaiser windowed sinc" for resampling, so
- // the quality should be good.
- TAudioConverter_FFmpeg = class(TAudioConverter)
- private
- // TODO: use SDL for multi-channel->stereo and format conversion
- ResampleContext: PReSampleContext;
- Ratio: double;
- public
- function Init(SrcFormatInfo: TAudioFormatInfo; DstFormatInfo: TAudioFormatInfo): boolean; override;
- destructor Destroy(); override;
-
- function Convert(InputBuffer: PChar; OutputBuffer: PChar; var InputSize: integer): integer; override;
- function GetOutputBufferSize(InputSize: integer): integer; override;
- function GetRatio(): double; override;
- end;
- {$ENDIF}
-
- {$IFDEF UseSRCResample}
- TAudioConverter_SRC = class(TAudioConverter)
- private
- ConverterState: PSRC_STATE;
- ConversionData: SRC_DATA;
- FormatConverter: TAudioConverter;
- public
- function Init(SrcFormatInfo: TAudioFormatInfo; DstFormatInfo: TAudioFormatInfo): boolean; override;
- destructor Destroy(); override;
-
- function Convert(InputBuffer: PChar; OutputBuffer: PChar; var InputSize: integer): integer; override;
- function GetOutputBufferSize(InputSize: integer): integer; override;
- function GetRatio(): double; override;
- end;
-
- // Note: SRC (=libsamplerate) provides several converters with different quality
- // speed trade-offs. The SINC-types are slow but offer best quality.
- // The SRC_SINC_* converters are too slow for realtime conversion,
- // (SRC_SINC_FASTEST is approx. ten times slower than SRC_LINEAR) resulting
- // in audible clicks and pops.
- // SRC_LINEAR is very fast and should have a better quality than SRC_ZERO_ORDER_HOLD
- // because it interpolates the samples. Normal "non-audiophile" users should not
- // be able to hear a difference between the SINC_* ones and LINEAR. Especially
- // if people sing along with the song.
- // But FFmpeg might offer a better quality/speed ratio than SRC_LINEAR.
- const
- SRC_CONVERTER_TYPE = SRC_LINEAR;
- {$ENDIF}
-
-implementation
-
-function TAudioConverter_SDL.Init(srcFormatInfo: TAudioFormatInfo; dstFormatInfo: TAudioFormatInfo): boolean;
-var
- srcFormat: UInt16;
- dstFormat: UInt16;
-begin
- inherited Init(SrcFormatInfo, DstFormatInfo);
-
- Result := false;
-
- if (not ConvertAudioFormatToSDL(srcFormatInfo.Format, srcFormat) or
- not ConvertAudioFormatToSDL(dstFormatInfo.Format, dstFormat)) then
- begin
- Log.LogError('Audio-format not supported by SDL', 'TSoftMixerPlaybackStream.InitFormatConversion');
- Exit;
- end;
-
- if (SDL_BuildAudioCVT(@cvt,
- srcFormat, srcFormatInfo.Channels, Round(srcFormatInfo.SampleRate),
- dstFormat, dstFormatInfo.Channels, Round(dstFormatInfo.SampleRate)) = -1) then
- begin
- Log.LogError(SDL_GetError(), 'TSoftMixerPlaybackStream.InitFormatConversion');
- Exit;
- end;
-
- Result := true;
-end;
-
-destructor TAudioConverter_SDL.Destroy();
-begin
- // nothing to be done here
- inherited;
-end;
-
-(*
- * Returns the size of the output buffer. This might be bigger than the actual
- * size of resampled audio data.
- *)
-function TAudioConverter_SDL.GetOutputBufferSize(InputSize: integer): integer;
-begin
- // Note: len_ratio must not be used here. Even if the len_ratio is 1.0, len_mult might be 2.
- // Example: 44.1kHz/mono to 22.05kHz/stereo -> len_ratio=1, len_mult=2
- Result := InputSize * cvt.len_mult;
-end;
-
-function TAudioConverter_SDL.GetRatio(): double;
-begin
- Result := cvt.len_ratio;
-end;
-
-function TAudioConverter_SDL.Convert(InputBuffer: PChar; OutputBuffer: PChar; var InputSize: integer): integer;
-begin
- Result := -1;
-
- if (InputSize <= 0) then
- begin
- // avoid div-by-zero problems
- if (InputSize = 0) then
- Result := 0;
- Exit;
- end;
-
- // OutputBuffer is always bigger than or equal to InputBuffer
- Move(InputBuffer[0], OutputBuffer[0], InputSize);
- cvt.buf := PUint8(OutputBuffer);
- cvt.len := InputSize;
- if (SDL_ConvertAudio(@cvt) = -1) then
- Exit;
-
- Result := cvt.len_cvt;
-end;
-
-
-{$IFDEF UseFFmpegResample}
-
-function TAudioConverter_FFmpeg.Init(SrcFormatInfo: TAudioFormatInfo; DstFormatInfo: TAudioFormatInfo): boolean;
-begin
- inherited Init(SrcFormatInfo, DstFormatInfo);
-
- Result := false;
-
- // Note: ffmpeg does not support resampling for more than 2 input channels
-
- if (srcFormatInfo.Format <> asfS16) then
- begin
- Log.LogError('Unsupported format', 'TAudioConverter_FFmpeg.Init');
- Exit;
- end;
-
- // TODO: use SDL here
- if (srcFormatInfo.Format <> dstFormatInfo.Format) then
- begin
- Log.LogError('Incompatible formats', 'TAudioConverter_FFmpeg.Init');
- Exit;
- end;
-
- ResampleContext := audio_resample_init(
- dstFormatInfo.Channels, srcFormatInfo.Channels,
- Round(dstFormatInfo.SampleRate), Round(srcFormatInfo.SampleRate));
- if (ResampleContext = nil) then
- begin
- Log.LogError('audio_resample_init() failed', 'TAudioConverter_FFmpeg.Init');
- Exit;
- end;
-
- // calculate ratio
- Ratio := (dstFormatInfo.Channels / srcFormatInfo.Channels) *
- (dstFormatInfo.SampleRate / srcFormatInfo.SampleRate);
-
- Result := true;
-end;
-
-destructor TAudioConverter_FFmpeg.Destroy();
-begin
- if (ResampleContext <> nil) then
- audio_resample_close(ResampleContext);
- inherited;
-end;
-
-function TAudioConverter_FFmpeg.Convert(InputBuffer: PChar; OutputBuffer: PChar; var InputSize: integer): integer;
-var
- InputSampleCount: integer;
- OutputSampleCount: integer;
-begin
- Result := -1;
-
- if (InputSize <= 0) then
- begin
- // avoid div-by-zero in audio_resample()
- if (InputSize = 0) then
- Result := 0;
- Exit;
- end;
-
- InputSampleCount := InputSize div SrcFormatInfo.FrameSize;
- OutputSampleCount := audio_resample(
- ResampleContext, PSmallInt(OutputBuffer), PSmallInt(InputBuffer),
- InputSampleCount);
- if (OutputSampleCount = -1) then
- begin
- Log.LogError('audio_resample() failed', 'TAudioConverter_FFmpeg.Convert');
- Exit;
- end;
- Result := OutputSampleCount * DstFormatInfo.FrameSize;
-end;
-
-function TAudioConverter_FFmpeg.GetOutputBufferSize(InputSize: integer): integer;
-begin
- Result := Ceil(InputSize * GetRatio());
-end;
-
-function TAudioConverter_FFmpeg.GetRatio(): double;
-begin
- Result := Ratio;
-end;
-
-{$ENDIF}
-
-
-{$IFDEF UseSRCResample}
-
-function TAudioConverter_SRC.Init(SrcFormatInfo: TAudioFormatInfo; DstFormatInfo: TAudioFormatInfo): boolean;
-var
- error: integer;
- TempSrcFormatInfo: TAudioFormatInfo;
- TempDstFormatInfo: TAudioFormatInfo;
-begin
- inherited Init(SrcFormatInfo, DstFormatInfo);
-
- Result := false;
-
- FormatConverter := nil;
-
- // SRC does not handle channel or format conversion
- if ((SrcFormatInfo.Channels <> DstFormatInfo.Channels) or
- not (SrcFormatInfo.Format in [asfS16, asfFloat])) then
- begin
- // SDL can not convert to float, so we have to convert to SInt16 first
- TempSrcFormatInfo := TAudioFormatInfo.Create(
- SrcFormatInfo.Channels, SrcFormatInfo.SampleRate, SrcFormatInfo.Format);
- TempDstFormatInfo := TAudioFormatInfo.Create(
- DstFormatInfo.Channels, SrcFormatInfo.SampleRate, asfS16);
-
- // init format/channel conversion
- FormatConverter := TAudioConverter_SDL.Create();
- if (not FormatConverter.Init(TempSrcFormatInfo, TempDstFormatInfo)) then
- begin
- Log.LogError('Unsupported input format', 'TAudioConverter_SRC.Init');
- FormatConverter.Free;
- // exit after the format-info is freed
- end;
-
- // this info was copied so we do not need it anymore
- TempSrcFormatInfo.Free;
- TempDstFormatInfo.Free;
-
- // leave if the format is not supported
- if (not assigned(FormatConverter)) then
- Exit;
-
- // adjust our copy of the input audio-format for SRC conversion
- Self.SrcFormatInfo.Channels := DstFormatInfo.Channels;
- Self.SrcFormatInfo.Format := asfS16;
- end;
-
- if ((DstFormatInfo.Format <> asfS16) and
- (DstFormatInfo.Format <> asfFloat)) then
- begin
- Log.LogError('Unsupported output format', 'TAudioConverter_SRC.Init');
- Exit;
- end;
-
- ConversionData.src_ratio := DstFormatInfo.SampleRate / SrcFormatInfo.SampleRate;
- if (src_is_valid_ratio(ConversionData.src_ratio) = 0) then
- begin
- Log.LogError('Invalid samplerate ratio', 'TAudioConverter_SRC.Init');
- Exit;
- end;
-
- ConverterState := src_new(SRC_CONVERTER_TYPE, DstFormatInfo.Channels, @error);
- if (ConverterState = nil) then
- begin
- Log.LogError('src_new() failed: ' + src_strerror(error), 'TAudioConverter_SRC.Init');
- Exit;
- end;
-
- Result := true;
-end;
-
-destructor TAudioConverter_SRC.Destroy();
-begin
- if (ConverterState <> nil) then
- src_delete(ConverterState);
- FormatConverter.Free;
- inherited;
-end;
-
-function TAudioConverter_SRC.Convert(InputBuffer: PChar; OutputBuffer: PChar; var InputSize: integer): integer;
-var
- FloatInputBuffer: PSingle;
- FloatOutputBuffer: PSingle;
- TempBuffer: PChar;
- TempSize: integer;
- NumSamples: integer;
- OutputSize: integer;
- error: integer;
-begin
- Result := -1;
-
- TempBuffer := nil;
-
- // format conversion with external converter (to correct number of channels and format)
- if (assigned(FormatConverter)) then
- begin
- TempSize := FormatConverter.GetOutputBufferSize(InputSize);
- GetMem(TempBuffer, TempSize);
- InputSize := FormatConverter.Convert(InputBuffer, TempBuffer, InputSize);
- InputBuffer := TempBuffer;
- end;
-
- if (InputSize <= 0) then
- begin
- // avoid div-by-zero problems
- if (InputSize = 0) then
- Result := 0;
- if (TempBuffer <> nil) then
- FreeMem(TempBuffer);
- Exit;
- end;
-
- if (SrcFormatInfo.Format = asfFloat) then
- begin
- FloatInputBuffer := PSingle(InputBuffer);
- end else begin
- NumSamples := InputSize div AudioSampleSize[SrcFormatInfo.Format];
- GetMem(FloatInputBuffer, NumSamples * SizeOf(Single));
- src_short_to_float_array(PCshort(InputBuffer), PCfloat(FloatInputBuffer), NumSamples);
- end;
-
- // calculate approx. output size
- OutputSize := Ceil(InputSize * ConversionData.src_ratio);
-
- if (DstFormatInfo.Format = asfFloat) then
- begin
- FloatOutputBuffer := PSingle(OutputBuffer);
- end else begin
- NumSamples := OutputSize div AudioSampleSize[DstFormatInfo.Format];
- GetMem(FloatOutputBuffer, NumSamples * SizeOf(Single));
- end;
-
- with ConversionData do
- begin
- data_in := PCFloat(FloatInputBuffer);
- input_frames := InputSize div SrcFormatInfo.FrameSize;
- data_out := PCFloat(FloatOutputBuffer);
- output_frames := OutputSize div DstFormatInfo.FrameSize;
- // TODO: set this to 1 at end of file-playback
- end_of_input := 0;
- end;
-
- error := src_process(ConverterState, @ConversionData);
- if (error <> 0) then
- begin
- Log.LogError(src_strerror(error), 'TAudioConverter_SRC.Convert');
- if (SrcFormatInfo.Format <> asfFloat) then
- FreeMem(FloatInputBuffer);
- if (DstFormatInfo.Format <> asfFloat) then
- FreeMem(FloatOutputBuffer);
- if (TempBuffer <> nil) then
- FreeMem(TempBuffer);
- Exit;
- end;
-
- if (SrcFormatInfo.Format <> asfFloat) then
- FreeMem(FloatInputBuffer);
-
- if (DstFormatInfo.Format <> asfFloat) then
- begin
- NumSamples := ConversionData.output_frames_gen * DstFormatInfo.Channels;
- src_float_to_short_array(PCfloat(FloatOutputBuffer), PCshort(OutputBuffer), NumSamples);
- FreeMem(FloatOutputBuffer);
- end;
-
- // free format conversion buffer if used
- if (TempBuffer <> nil) then
- FreeMem(TempBuffer);
-
- if (assigned(FormatConverter)) then
- InputSize := ConversionData.input_frames_used * FormatConverter.SrcFormatInfo.FrameSize
- else
- InputSize := ConversionData.input_frames_used * SrcFormatInfo.FrameSize;
-
- // set result to output size according to SRC
- Result := ConversionData.output_frames_gen * DstFormatInfo.FrameSize;
-end;
-
-function TAudioConverter_SRC.GetOutputBufferSize(InputSize: integer): integer;
-begin
- Result := Ceil(InputSize * GetRatio());
-end;
-
-function TAudioConverter_SRC.GetRatio(): double;
-begin
- // if we need additional channel/format conversion, use this ratio
- if (assigned(FormatConverter)) then
- Result := FormatConverter.GetRatio()
- else
- Result := 1.0;
-
- // now the SRC ratio (Note: the format might change from SInt16 to float)
- Result := Result *
- ConversionData.src_ratio *
- (DstFormatInfo.FrameSize / SrcFormatInfo.FrameSize);
-end;
-
-{$ENDIF}
-
-end. \ No newline at end of file