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Diffstat (limited to 'src/base/UAudioConverter.pas')
| -rw-r--r-- | src/base/UAudioConverter.pas | 458 |
1 files changed, 0 insertions, 458 deletions
diff --git a/src/base/UAudioConverter.pas b/src/base/UAudioConverter.pas deleted file mode 100644 index 5647f27b..00000000 --- a/src/base/UAudioConverter.pas +++ /dev/null @@ -1,458 +0,0 @@ -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.
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