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-rw-r--r--src/Classes/UAudioConverter.pas458
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diff --git a/src/Classes/UAudioConverter.pas b/src/Classes/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