From 5ac0d54bb595538740610feee26acbe7181984c8 Mon Sep 17 00:00:00 2001 From: tobigun Date: Wed, 27 Aug 2008 09:53:18 +0000 Subject: spelling corrected FFMpeg -> FFmpeg git-svn-id: svn://svn.code.sf.net/p/ultrastardx/svn/trunk@1295 b956fd51-792f-4845-bead-9b4dfca2ff2c --- Game/Code/Classes/UAudioConverter.pas | 32 ++++++++++++++++---------------- 1 file changed, 16 insertions(+), 16 deletions(-) (limited to 'Game/Code/Classes/UAudioConverter.pas') diff --git a/Game/Code/Classes/UAudioConverter.pas b/Game/Code/Classes/UAudioConverter.pas index 74ae88df..5647f27b 100644 --- a/Game/Code/Classes/UAudioConverter.pas +++ b/Game/Code/Classes/UAudioConverter.pas @@ -15,7 +15,7 @@ uses {$IFDEF UseSRCResample} samplerate, {$ENDIF} - {$IFDEF UseFFMpegResample} + {$IFDEF UseFFmpegResample} avcodec, {$ENDIF} UMediaCore_SDL, @@ -35,7 +35,7 @@ type * with the video or the lyrics timer. * - float<->int16 conversion is not supported (will be part of 1.3) and * SDL (<1.3) is not capable of handling floats at all. - * -> Using FFMpeg or libsamplerate for resampling is preferred. + * -> Using FFmpeg or libsamplerate for resampling is preferred. * Use SDL for channel and format conversion only. *} TAudioConverter_SDL = class(TAudioConverter) @@ -50,10 +50,10 @@ type function GetRatio(): double; override; end; - {$IFDEF UseFFMpegResample} - // Note: FFMpeg seems to be using "kaiser windowed sinc" for resampling, so + {$IFDEF UseFFmpegResample} + // Note: FFmpeg seems to be using "kaiser windowed sinc" for resampling, so // the quality should be good. - TAudioConverter_FFMpeg = class(TAudioConverter) + TAudioConverter_FFmpeg = class(TAudioConverter) private // TODO: use SDL for multi-channel->stereo and format conversion ResampleContext: PReSampleContext; @@ -92,7 +92,7 @@ type // because it interpolates the samples. Normal "non-audiophile" users should not // be able to hear a difference between the SINC_* ones and LINEAR. Especially // if people sing along with the song. - // But FFMpeg might offer a better quality/speed ratio than SRC_LINEAR. + // But FFmpeg might offer a better quality/speed ratio than SRC_LINEAR. const SRC_CONVERTER_TYPE = SRC_LINEAR; {$ENDIF} @@ -171,9 +171,9 @@ begin end; -{$IFDEF UseFFMpegResample} +{$IFDEF UseFFmpegResample} -function TAudioConverter_FFMpeg.Init(SrcFormatInfo: TAudioFormatInfo; DstFormatInfo: TAudioFormatInfo): boolean; +function TAudioConverter_FFmpeg.Init(SrcFormatInfo: TAudioFormatInfo; DstFormatInfo: TAudioFormatInfo): boolean; begin inherited Init(SrcFormatInfo, DstFormatInfo); @@ -183,14 +183,14 @@ begin if (srcFormatInfo.Format <> asfS16) then begin - Log.LogError('Unsupported format', 'TAudioConverter_FFMpeg.Init'); + 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'); + Log.LogError('Incompatible formats', 'TAudioConverter_FFmpeg.Init'); Exit; end; @@ -199,7 +199,7 @@ begin Round(dstFormatInfo.SampleRate), Round(srcFormatInfo.SampleRate)); if (ResampleContext = nil) then begin - Log.LogError('audio_resample_init() failed', 'TAudioConverter_FFMpeg.Init'); + Log.LogError('audio_resample_init() failed', 'TAudioConverter_FFmpeg.Init'); Exit; end; @@ -210,14 +210,14 @@ begin Result := true; end; -destructor TAudioConverter_FFMpeg.Destroy(); +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; +function TAudioConverter_FFmpeg.Convert(InputBuffer: PChar; OutputBuffer: PChar; var InputSize: integer): integer; var InputSampleCount: integer; OutputSampleCount: integer; @@ -238,18 +238,18 @@ begin InputSampleCount); if (OutputSampleCount = -1) then begin - Log.LogError('audio_resample() failed', 'TAudioConverter_FFMpeg.Convert'); + Log.LogError('audio_resample() failed', 'TAudioConverter_FFmpeg.Convert'); Exit; end; Result := OutputSampleCount * DstFormatInfo.FrameSize; end; -function TAudioConverter_FFMpeg.GetOutputBufferSize(InputSize: integer): integer; +function TAudioConverter_FFmpeg.GetOutputBufferSize(InputSize: integer): integer; begin Result := Ceil(InputSize * GetRatio()); end; -function TAudioConverter_FFMpeg.GetRatio(): double; +function TAudioConverter_FFmpeg.GetRatio(): double; begin Result := Ratio; end; -- cgit v1.2.3