diff options
Diffstat (limited to 'src/media')
| -rw-r--r-- | src/media/UAudioConverter.pas | 458 | ||||
| -rw-r--r-- | src/media/UAudioCore_Bass.pas | 123 | ||||
| -rw-r--r-- | src/media/UAudioCore_Portaudio.pas | 257 | ||||
| -rw-r--r-- | src/media/UAudioDecoder_Bass.pas | 242 | ||||
| -rw-r--r-- | src/media/UAudioDecoder_FFmpeg.pas | 1114 | ||||
| -rw-r--r-- | src/media/UAudioInput_Bass.pas | 481 | ||||
| -rw-r--r-- | src/media/UAudioInput_Portaudio.pas | 474 | ||||
| -rw-r--r-- | src/media/UAudioPlaybackBase.pas | 292 | ||||
| -rw-r--r-- | src/media/UAudioPlayback_Bass.pas | 731 | ||||
| -rw-r--r-- | src/media/UAudioPlayback_Portaudio.pas | 361 | ||||
| -rw-r--r-- | src/media/UAudioPlayback_SDL.pas | 160 | ||||
| -rw-r--r-- | src/media/UAudioPlayback_SoftMixer.pas | 1132 | ||||
| -rw-r--r-- | src/media/UMediaCore_FFmpeg.pas | 405 | ||||
| -rw-r--r-- | src/media/UMediaCore_SDL.pas | 38 | ||||
| -rw-r--r-- | src/media/UMedia_dummy.pas | 243 | ||||
| -rw-r--r-- | src/media/UVideo.pas | 828 | ||||
| -rw-r--r-- | src/media/UVisualizer.pas | 442 |
17 files changed, 7781 insertions, 0 deletions
diff --git a/src/media/UAudioConverter.pas b/src/media/UAudioConverter.pas new file mode 100644 index 00000000..5647f27b --- /dev/null +++ b/src/media/UAudioConverter.pas @@ -0,0 +1,458 @@ +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 diff --git a/src/media/UAudioCore_Bass.pas b/src/media/UAudioCore_Bass.pas new file mode 100644 index 00000000..beb2db16 --- /dev/null +++ b/src/media/UAudioCore_Bass.pas @@ -0,0 +1,123 @@ +unit UAudioCore_Bass; + +interface + +{$IFDEF FPC} + {$MODE Delphi} +{$ENDIF} + +{$I switches.inc} + +uses + Classes, + SysUtils, + UMusic, + bass; // (Note: DWORD is defined here) + +type + TAudioCore_Bass = class + public + constructor Create(); + class function GetInstance(): TAudioCore_Bass; + function ErrorGetString(): string; overload; + function ErrorGetString(errCode: integer): string; overload; + function ConvertAudioFormatToBASSFlags(Format: TAudioSampleFormat; out Flags: DWORD): boolean; + function ConvertBASSFlagsToAudioFormat(Flags: DWORD; out Format: TAudioSampleFormat): boolean; + end; + +implementation + +uses + UMain, + ULog; + +var + Instance: TAudioCore_Bass; + +constructor TAudioCore_Bass.Create(); +begin + inherited; +end; + +class function TAudioCore_Bass.GetInstance(): TAudioCore_Bass; +begin + if (not Assigned(Instance)) then + Instance := TAudioCore_Bass.Create(); + Result := Instance; +end; + +function TAudioCore_Bass.ErrorGetString(): string; +begin + Result := ErrorGetString(BASS_ErrorGetCode()); +end; + +function TAudioCore_Bass.ErrorGetString(errCode: integer): string; +begin + case errCode of + BASS_OK: result := 'No error'; + BASS_ERROR_MEM: result := 'Insufficient memory'; + BASS_ERROR_FILEOPEN: result := 'File could not be opened'; + BASS_ERROR_DRIVER: result := 'Device driver not available'; + BASS_ERROR_BUFLOST: result := 'Buffer lost'; + BASS_ERROR_HANDLE: result := 'Invalid Handle'; + BASS_ERROR_FORMAT: result := 'Sample-Format not supported'; + BASS_ERROR_POSITION: result := 'Illegal position'; + BASS_ERROR_INIT: result := 'BASS_Init has not been successfully called'; + BASS_ERROR_START: result := 'Paused/stopped'; + BASS_ERROR_ALREADY: result := 'Already created/used'; + BASS_ERROR_NOCHAN: result := 'No free channels'; + BASS_ERROR_ILLTYPE: result := 'Type is invalid'; + BASS_ERROR_ILLPARAM: result := 'Illegal parameter'; + BASS_ERROR_NO3D: result := 'No 3D support'; + BASS_ERROR_NOEAX: result := 'No EAX support'; + BASS_ERROR_DEVICE: result := 'Invalid device number'; + BASS_ERROR_NOPLAY: result := 'Channel not playing'; + BASS_ERROR_FREQ: result := 'Freq out of range'; + BASS_ERROR_NOTFILE: result := 'Not a file stream'; + BASS_ERROR_NOHW: result := 'No hardware support'; + BASS_ERROR_EMPTY: result := 'Is empty'; + BASS_ERROR_NONET: result := 'Network unavailable'; + BASS_ERROR_CREATE: result := 'Creation error'; + BASS_ERROR_NOFX: result := 'DX8 effects unavailable'; + BASS_ERROR_NOTAVAIL: result := 'Not available'; + BASS_ERROR_DECODE: result := 'Is a decoding channel'; + BASS_ERROR_DX: result := 'Insufficient version of DirectX'; + BASS_ERROR_TIMEOUT: result := 'Timeout'; + BASS_ERROR_FILEFORM: result := 'File-Format not recognised/supported'; + BASS_ERROR_SPEAKER: result := 'Requested speaker(s) not support'; + BASS_ERROR_VERSION: result := 'Version error'; + BASS_ERROR_CODEC: result := 'Codec not available/supported'; + BASS_ERROR_ENDED: result := 'The channel/file has ended'; + BASS_ERROR_UNKNOWN: result := 'Unknown error'; + else result := 'Unknown error'; + end; +end; + +function TAudioCore_Bass.ConvertAudioFormatToBASSFlags(Format: TAudioSampleFormat; out Flags: DWORD): boolean; +begin + case Format of + asfS16: Flags := 0; + asfFloat: Flags := BASS_SAMPLE_FLOAT; + asfU8: Flags := BASS_SAMPLE_8BITS; + else begin + Result := false; + Exit; + end; + end; + + Result := true; +end; + +function TAudioCore_Bass.ConvertBASSFlagsToAudioFormat(Flags: DWORD; out Format: TAudioSampleFormat): boolean; +begin + if ((Flags and BASS_SAMPLE_FLOAT) <> 0) then + Format := asfFloat + else if ((Flags and BASS_SAMPLE_8BITS) <> 0) then + Format := asfU8 + else + Format := asfS16; + + Result := true; +end; + +end. diff --git a/src/media/UAudioCore_Portaudio.pas b/src/media/UAudioCore_Portaudio.pas new file mode 100644 index 00000000..cd279d99 --- /dev/null +++ b/src/media/UAudioCore_Portaudio.pas @@ -0,0 +1,257 @@ +unit UAudioCore_Portaudio; + +interface + +{$IFDEF FPC} + {$MODE Delphi} +{$ENDIF} + +{$I ../switches.inc} + + +uses + Classes, + SysUtils, + portaudio; + +type + TAudioCore_Portaudio = class + public + constructor Create(); + class function GetInstance(): TAudioCore_Portaudio; + function GetPreferredApiIndex(): TPaHostApiIndex; + function TestDevice(inParams, outParams: PPaStreamParameters; var sampleRate: Double): boolean; + end; + +implementation + +uses + ULog; + +{* + * The default API used by Portaudio is the least common denominator + * and might lack efficiency. In addition it might not even work. + * We use an array named ApiPreferenceOrder with which we define the order of + * preferred APIs to use. The first API-type in the list is tried first. + * If it is not available the next one is tried and so on ... + * If none of the preferred APIs was found the default API (detected by + * portaudio) is used. + * + * Pascal does not permit zero-length static arrays, so you must use paDefaultApi + * as an array's only member if you do not have any preferences. + * You can also append paDefaultApi to a non-zero length preferences array but + * this is optional because the default API is always used as a fallback. + *} +const + paDefaultApi = -1; +const + ApiPreferenceOrder: +{$IF Defined(MSWINDOWS)} + // Note1: Portmixer has no mixer support for paASIO and paWASAPI at the moment + // Note2: Windows Default-API is MME, but DirectSound is faster + array[0..0] of TPaHostApiTypeId = ( paDirectSound ); +{$ELSEIF Defined(DARWIN)} + array[0..0] of TPaHostApiTypeId = ( paDefaultApi ); // paCoreAudio +{$ELSEIF Defined(UNIX)} + // Note: Portmixer has no mixer support for JACK at the moment + array[0..2] of TPaHostApiTypeId = ( paALSA, paJACK, paOSS ); +{$ELSE} + array[0..0] of TPaHostApiTypeId = ( paDefaultApi ); +{$IFEND} + + +{ TAudioInput_Portaudio } + +var + Instance: TAudioCore_Portaudio; + +constructor TAudioCore_Portaudio.Create(); +begin + inherited; +end; + +class function TAudioCore_Portaudio.GetInstance(): TAudioCore_Portaudio; +begin + if not assigned(Instance) then + Instance := TAudioCore_Portaudio.Create(); + Result := Instance; +end; + +function TAudioCore_Portaudio.GetPreferredApiIndex(): TPaHostApiIndex; +var + i: integer; + apiIndex: TPaHostApiIndex; + apiInfo: PPaHostApiInfo; +begin + result := -1; + + // select preferred sound-API + for i:= 0 to High(ApiPreferenceOrder) do + begin + if(ApiPreferenceOrder[i] <> paDefaultApi) then + begin + // check if API is available + apiIndex := Pa_HostApiTypeIdToHostApiIndex(ApiPreferenceOrder[i]); + if(apiIndex >= 0) then + begin + // we found an API but we must check if it works + // (on linux portaudio might detect OSS but does not provide + // any devices if ALSA is enabled) + apiInfo := Pa_GetHostApiInfo(apiIndex); + if (apiInfo^.deviceCount > 0) then + begin + Result := apiIndex; + break; + end; + end; + end; + end; + + // None of the preferred APIs is available -> use default + if(result < 0) then + begin + result := Pa_GetDefaultHostApi(); + end; +end; + +{* + * Portaudio test callback used by TestDevice(). + *} +function TestCallback(input: Pointer; output: Pointer; frameCount: Longword; + timeInfo: PPaStreamCallbackTimeInfo; statusFlags: TPaStreamCallbackFlags; + inputDevice: Pointer): Integer; cdecl; +begin + // this callback is called only once + result := paAbort; +end; + +(* + * Tests if the callback works. Some devices can be opened without + * an error but the callback is never called. Calling Pa_StopStream() on such + * a stream freezes USDX then. Probably because the callback-thread is deadlocked + * due to some bug in portaudio. The blocking Pa_ReadStream() and Pa_WriteStream() + * block forever too and though can't be used for testing. + * + * To avoid freezing Pa_AbortStream (or Pa_CloseStream which calls Pa_AbortStream) + * can be used to force the stream to stop. But for some reason this stops debugging + * in gdb with a "no process found" message. + * + * Because freezing devices are non-working devices we test the devices here to + * be able to exclude them from the device-selection list. + * + * Portaudio does not provide any test to check this error case (probably because + * it should not even occur). So we have to open the device, start the stream and + * check if the callback is called (the stream is stopped if the callback is called + * for the first time, so we can poll until the stream is stopped). + * + * Another error that occurs is that some devices (even the default device) might + * work at the beginning but stop after a few calls (maybe 50) of the callback. + * For me this problem occurs with the default output-device. The "dmix" or "front" + * device must be selected instead. Another problem is that (due to a bug in + * portaudio or ALSA) the "front" device is not detected every time portaudio + * is started. Sometimes it needs two or more restarts. + * + * There is no reasonable way to test for these errors. For the first error-case + * we could test if the callback is called 50 times but this can take a second + * for each device and it can fail in the 51st or even 100th callback call then. + * + * The second error-case cannot be tested at all. How should we now that one + * device is missing if portaudio is not even able to detect it. + * We could start and terminate Portaudio for several times and see if the device + * count changes but this is ugly. + * + * Conclusion: We are not able to autodetect a working device with + * portaudio (at least not with the newest v19_20071207) at the moment. + * So we have to provide the possibility to manually select an output device + * in the UltraStar options if we want to use portaudio instead of SDL. + *) +function TAudioCore_Portaudio.TestDevice(inParams, outParams: PPaStreamParameters; var sampleRate: Double): boolean; +var + stream: PPaStream; + err: TPaError; + cbWorks: boolean; + cbPolls: integer; + i: integer; +const + altSampleRates: array[0..1] of Double = (44100, 48000); // alternative sample-rates +begin + Result := false; + + if (sampleRate <= 0) then + sampleRate := 44100; + + // check if device supports our input-format + err := Pa_IsFormatSupported(inParams, outParams, sampleRate); + if(err <> paNoError) then + begin + // we cannot fix the error -> exit + if (err <> paInvalidSampleRate) then + Exit; + + // try alternative sample-rates to the detected one + sampleRate := 0; + for i := 0 to High(altSampleRates) do + begin + // do not check the detected sample-rate twice + if (altSampleRates[i] = sampleRate) then + continue; + // check alternative + err := Pa_IsFormatSupported(inParams, outParams, altSampleRates[i]); + if (err = paNoError) then + begin + // sample-rate works + sampleRate := altSampleRates[i]; + break; + end; + end; + // no working sample-rate found + if (sampleRate = 0) then + Exit; + end; + + // FIXME: for some reason gdb stops after a call of Pa_AbortStream() + // which is implicitely called by Pa_CloseStream(). + // gdb's stops with the message: "ptrace: no process found". + // Probably because the callback-thread is killed what confuses gdb. + {$IF Defined(Debug) and Defined(Linux)} + cbWorks := true; + {$ELSE} + // open device for testing + err := Pa_OpenStream(stream, inParams, outParams, sampleRate, + paFramesPerBufferUnspecified, + paNoFlag, @TestCallback, nil); + if(err <> paNoError) then + begin + exit; + end; + + // start the callback + err := Pa_StartStream(stream); + if(err <> paNoError) then + begin + Pa_CloseStream(stream); + exit; + end; + + cbWorks := false; + // check if the callback was called (poll for max. 200ms) + for cbPolls := 1 to 20 do + begin + // if the test-callback was called it should be aborted now + if (Pa_IsStreamActive(stream) = 0) then + begin + cbWorks := true; + break; + end; + // not yet aborted, wait and try (poll) again + Pa_Sleep(10); + end; + + // finally abort the stream + Pa_CloseStream(stream); + {$IFEND} + + Result := cbWorks; +end; + +end. diff --git a/src/media/UAudioDecoder_Bass.pas b/src/media/UAudioDecoder_Bass.pas new file mode 100644 index 00000000..dba1fde4 --- /dev/null +++ b/src/media/UAudioDecoder_Bass.pas @@ -0,0 +1,242 @@ +unit UAudioDecoder_Bass; + +interface + +{$IFDEF FPC} + {$MODE Delphi} +{$ENDIF} + +{$I switches.inc} + +implementation + +uses + Classes, + SysUtils, + UMain, + UMusic, + UAudioCore_Bass, + ULog, + bass; + +type + TBassDecodeStream = class(TAudioDecodeStream) + private + Handle: HSTREAM; + FormatInfo : TAudioFormatInfo; + Error: boolean; + public + constructor Create(Handle: HSTREAM); + destructor Destroy(); override; + + procedure Close(); override; + + function GetLength(): real; override; + function GetAudioFormatInfo(): TAudioFormatInfo; override; + function GetPosition: real; override; + procedure SetPosition(Time: real); override; + function GetLoop(): boolean; override; + procedure SetLoop(Enabled: boolean); override; + function IsEOF(): boolean; override; + function IsError(): boolean; override; + + function ReadData(Buffer: PChar; BufSize: integer): integer; override; + end; + +type + TAudioDecoder_Bass = class( TInterfacedObject, IAudioDecoder ) + public + function GetName: string; + + function InitializeDecoder(): boolean; + function FinalizeDecoder(): boolean; + function Open(const Filename: string): TAudioDecodeStream; + end; + +var + BassCore: TAudioCore_Bass; + + +{ TBassDecodeStream } + +constructor TBassDecodeStream.Create(Handle: HSTREAM); +var + ChannelInfo: BASS_CHANNELINFO; + Format: TAudioSampleFormat; +begin + inherited Create(); + Self.Handle := Handle; + + // setup format info + if (not BASS_ChannelGetInfo(Handle, ChannelInfo)) then + begin + raise Exception.Create('Failed to open decode-stream'); + end; + BassCore.ConvertBASSFlagsToAudioFormat(ChannelInfo.flags, Format); + FormatInfo := TAudioFormatInfo.Create(ChannelInfo.chans, ChannelInfo.freq, format); + + Error := false; +end; + +destructor TBassDecodeStream.Destroy(); +begin + Close(); + inherited; +end; + +procedure TBassDecodeStream.Close(); +begin + if (Handle <> 0) then + begin + BASS_StreamFree(Handle); + Handle := 0; + end; + PerformOnClose(); + FreeAndNil(FormatInfo); + Error := false; +end; + +function TBassDecodeStream.GetAudioFormatInfo(): TAudioFormatInfo; +begin + Result := FormatInfo; +end; + +function TBassDecodeStream.GetLength(): real; +var + bytes: QWORD; +begin + bytes := BASS_ChannelGetLength(Handle, BASS_POS_BYTE); + Result := BASS_ChannelBytes2Seconds(Handle, bytes); +end; + +function TBassDecodeStream.GetPosition: real; +var + bytes: QWORD; +begin + bytes := BASS_ChannelGetPosition(Handle, BASS_POS_BYTE); + Result := BASS_ChannelBytes2Seconds(Handle, bytes); +end; + +procedure TBassDecodeStream.SetPosition(Time: real); +var + bytes: QWORD; +begin + bytes := BASS_ChannelSeconds2Bytes(Handle, Time); + BASS_ChannelSetPosition(Handle, bytes, BASS_POS_BYTE); +end; + +function TBassDecodeStream.GetLoop(): boolean; +var + flags: DWORD; +begin + // retrieve channel flags + flags := BASS_ChannelFlags(Handle, 0, 0); + if (flags = DWORD(-1)) then + begin + Log.LogError('BASS_ChannelFlags: ' + BassCore.ErrorGetString(), 'TBassDecodeStream.GetLoop'); + Result := false; + Exit; + end; + Result := (flags and BASS_SAMPLE_LOOP) <> 0; +end; + +procedure TBassDecodeStream.SetLoop(Enabled: boolean); +var + flags: DWORD; +begin + // set/unset loop-flag + if (Enabled) then + flags := BASS_SAMPLE_LOOP + else + flags := 0; + + // set new flag-bits + if (BASS_ChannelFlags(Handle, flags, BASS_SAMPLE_LOOP) = DWORD(-1)) then + begin + Log.LogError('BASS_ChannelFlags: ' + BassCore.ErrorGetString(), 'TBassDecodeStream.SetLoop'); + Exit; + end; +end; + +function TBassDecodeStream.IsEOF(): boolean; +begin + Result := (BASS_ChannelIsActive(Handle) = BASS_ACTIVE_STOPPED); +end; + +function TBassDecodeStream.IsError(): boolean; +begin + Result := Error; +end; + +function TBassDecodeStream.ReadData(Buffer: PChar; BufSize: integer): integer; +begin + Result := BASS_ChannelGetData(Handle, Buffer, BufSize); + // check error state (do not handle EOF as error) + if ((Result = -1) and (BASS_ErrorGetCode() <> BASS_ERROR_ENDED)) then + Error := true + else + Error := false; +end; + + +{ TAudioDecoder_Bass } + +function TAudioDecoder_Bass.GetName: String; +begin + result := 'BASS_Decoder'; +end; + +function TAudioDecoder_Bass.InitializeDecoder(): boolean; +begin + BassCore := TAudioCore_Bass.GetInstance(); + Result := true; +end; + +function TAudioDecoder_Bass.FinalizeDecoder(): boolean; +begin + Result := true; +end; + +function TAudioDecoder_Bass.Open(const Filename: string): TAudioDecodeStream; +var + Stream: HSTREAM; + ChannelInfo: BASS_CHANNELINFO; + FileExt: string; +begin + Result := nil; + + // check if BASS was initialized + // in case the decoder is not used with BASS playback, init the NO_SOUND device + if ((integer(BASS_GetDevice) = -1) and (BASS_ErrorGetCode() = BASS_ERROR_INIT)) then + BASS_Init(0, 44100, 0, 0, nil); + + // TODO: use BASS_STREAM_PRESCAN for accurate seeking in VBR-files? + // disadvantage: seeking will slow down. + Stream := BASS_StreamCreateFile(False, PChar(Filename), 0, 0, BASS_STREAM_DECODE); + if (Stream = 0) then + begin + //Log.LogError(BassCore.ErrorGetString(), 'TAudioDecoder_Bass.Open'); + Exit; + end; + + // check if BASS opened some erroneously recognized file-formats + if BASS_ChannelGetInfo(Stream, channelInfo) then + begin + fileExt := ExtractFileExt(Filename); + // BASS opens FLV-files (maybe others too) although it cannot handle them. + // Setting BASS_CONFIG_VERIFY to the max. value (100000) does not help. + if ((fileExt = '.flv') and (channelInfo.ctype = BASS_CTYPE_STREAM_MP1)) then + begin + BASS_StreamFree(Stream); + Exit; + end; + end; + + Result := TBassDecodeStream.Create(Stream); +end; + + +initialization + MediaManager.Add(TAudioDecoder_Bass.Create); + +end. diff --git a/src/media/UAudioDecoder_FFmpeg.pas b/src/media/UAudioDecoder_FFmpeg.pas new file mode 100644 index 00000000..d9b4c93c --- /dev/null +++ b/src/media/UAudioDecoder_FFmpeg.pas @@ -0,0 +1,1114 @@ +unit UAudioDecoder_FFmpeg; + +(******************************************************************************* + * + * This unit is primarily based upon - + * http://www.dranger.com/ffmpeg/ffmpegtutorial_all.html + * + * and tutorial03.c + * + * http://www.inb.uni-luebeck.de/~boehme/using_libavcodec.html + * + *******************************************************************************) + +interface + +{$IFDEF FPC} + {$MODE Delphi} +{$ENDIF} + +{$I switches.inc} + +// show FFmpeg specific debug output +{.$DEFINE DebugFFmpegDecode} + +// FFmpeg is very verbose and shows a bunch of errors. +// Those errors (they can be considered as warnings by us) can be ignored +// as they do not give any useful information. +// There is no solution to fix this except for turning them off. +{.$DEFINE EnableFFmpegErrorOutput} + +implementation + +uses + Classes, + SysUtils, + Math, + UMusic, + UIni, + UMain, + avcodec, + avformat, + avutil, + avio, + mathematics, // used for av_rescale_q + rational, + UMediaCore_FFmpeg, + SDL, + ULog, + UCommon, + UConfig; + +const + MAX_AUDIOQ_SIZE = (5 * 16 * 1024); + +const + // TODO: The factor 3/2 might not be necessary as we do not need extra + // space for synchronizing as in the tutorial. + AUDIO_BUFFER_SIZE = (AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) div 2; + +type + TFFmpegDecodeStream = class(TAudioDecodeStream) + private + StateLock: PSDL_Mutex; + + EOFState: boolean; // end-of-stream flag (locked by StateLock) + ErrorState: boolean; // error flag (locked by StateLock) + + QuitRequest: boolean; // (locked by StateLock) + ParserIdleCond: PSDL_Cond; + + // parser pause/resume data + ParserLocked: boolean; + ParserPauseRequestCount: integer; + ParserUnlockedCond: PSDL_Cond; + ParserResumeCond: PSDL_Cond; + + SeekRequest: boolean; // (locked by StateLock) + SeekFlags: integer; // (locked by StateLock) + SeekPos: double; // stream position to seek for (in secs) (locked by StateLock) + SeekFlush: boolean; // true if the buffers should be flushed after seeking (locked by StateLock) + SeekFinishedCond: PSDL_Cond; + + Loop: boolean; // (locked by StateLock) + + ParseThread: PSDL_Thread; + PacketQueue: TPacketQueue; + + FormatInfo: TAudioFormatInfo; + + // FFmpeg specific data + FormatCtx: PAVFormatContext; + CodecCtx: PAVCodecContext; + Codec: PAVCodec; + + AudioStreamIndex: integer; + AudioStream: PAVStream; + AudioStreamPos: double; // stream position in seconds (locked by DecoderLock) + + // decoder pause/resume data + DecoderLocked: boolean; + DecoderPauseRequestCount: integer; + DecoderUnlockedCond: PSDL_Cond; + DecoderResumeCond: PSDL_Cond; + + // state-vars for DecodeFrame (locked by DecoderLock) + AudioPaket: TAVPacket; + AudioPaketData: PChar; + AudioPaketSize: integer; + AudioPaketSilence: integer; // number of bytes of silence to return + + // state-vars for AudioCallback (locked by DecoderLock) + AudioBufferPos: integer; + AudioBufferSize: integer; + AudioBuffer: PChar; + + Filename: string; + + procedure SetPositionIntern(Time: real; Flush: boolean; Blocking: boolean); + procedure SetEOF(State: boolean); {$IFDEF HasInline}inline;{$ENDIF} + procedure SetError(State: boolean); {$IFDEF HasInline}inline;{$ENDIF} + function IsSeeking(): boolean; + function IsQuit(): boolean; + + procedure Reset(); + + procedure Parse(); + function ParseLoop(): boolean; + procedure PauseParser(); + procedure ResumeParser(); + + function DecodeFrame(Buffer: PChar; BufferSize: integer): integer; + procedure FlushCodecBuffers(); + procedure PauseDecoder(); + procedure ResumeDecoder(); + public + constructor Create(); + destructor Destroy(); override; + + function Open(const Filename: string): boolean; + procedure Close(); override; + + function GetLength(): real; override; + function GetAudioFormatInfo(): TAudioFormatInfo; override; + function GetPosition: real; override; + procedure SetPosition(Time: real); override; + function GetLoop(): boolean; override; + procedure SetLoop(Enabled: boolean); override; + function IsEOF(): boolean; override; + function IsError(): boolean; override; + + function ReadData(Buffer: PChar; BufferSize: integer): integer; override; + end; + +type + TAudioDecoder_FFmpeg = class( TInterfacedObject, IAudioDecoder ) + public + function GetName: string; + + function InitializeDecoder(): boolean; + function FinalizeDecoder(): boolean; + function Open(const Filename: string): TAudioDecodeStream; + end; + +var + FFmpegCore: TMediaCore_FFmpeg; + +function ParseThreadMain(Data: Pointer): integer; cdecl; forward; + + +{ TFFmpegDecodeStream } + +constructor TFFmpegDecodeStream.Create(); +begin + inherited Create(); + + StateLock := SDL_CreateMutex(); + ParserUnlockedCond := SDL_CreateCond(); + ParserResumeCond := SDL_CreateCond(); + ParserIdleCond := SDL_CreateCond(); + SeekFinishedCond := SDL_CreateCond(); + DecoderUnlockedCond := SDL_CreateCond(); + DecoderResumeCond := SDL_CreateCond(); + + // according to the documentation of avcodec_decode_audio(2), sample-data + // should be aligned on a 16 byte boundary. Otherwise internal calls + // (e.g. to SSE or Altivec operations) might fail or lack performance on some + // CPUs. Although GetMem() in Delphi and FPC seems to use a 16 byte or higher + // alignment for buffers of this size (alignment depends on the size of the + // requested buffer), we will set the alignment explicitly as the minimum + // alignment used by Delphi and FPC is on an 8 byte boundary. + // + // Note: AudioBuffer was previously defined as a field of type TAudioBuffer + // (array[0..AUDIO_BUFFER_SIZE-1] of byte) and hence statically allocated. + // Fields of records are aligned different to memory allocated with GetMem(), + // aligning depending on the type but will be at least 2 bytes. + // AudioBuffer was not aligned to a 16 byte boundary. The {$ALIGN x} directive + // was not applicable as Delphi in contrast to FPC provides at most 8 byte + // alignment ({$ALIGN 16} is not supported) by this directive. + AudioBuffer := GetAlignedMem(AUDIO_BUFFER_SIZE, 16); + + Reset(); +end; + +procedure TFFmpegDecodeStream.Reset(); +begin + ParseThread := nil; + + EOFState := false; + ErrorState := false; + Loop := false; + QuitRequest := false; + + AudioPaketData := nil; + AudioPaketSize := 0; + AudioPaketSilence := 0; + + AudioBufferPos := 0; + AudioBufferSize := 0; + + ParserLocked := false; + ParserPauseRequestCount := 0; + DecoderLocked := false; + DecoderPauseRequestCount := 0; + + FillChar(AudioPaket, SizeOf(TAVPacket), 0); +end; + +{* + * Frees the decode-stream data. + *} +destructor TFFmpegDecodeStream.Destroy(); +begin + Close(); + + SDL_DestroyMutex(StateLock); + SDL_DestroyCond(ParserUnlockedCond); + SDL_DestroyCond(ParserResumeCond); + SDL_DestroyCond(ParserIdleCond); + SDL_DestroyCond(SeekFinishedCond); + SDL_DestroyCond(DecoderUnlockedCond); + SDL_DestroyCond(DecoderResumeCond); + + FreeAlignedMem(AudioBuffer); + + inherited; +end; + +function TFFmpegDecodeStream.Open(const Filename: string): boolean; +var + SampleFormat: TAudioSampleFormat; + AVResult: integer; +begin + Result := false; + + Close(); + Reset(); + + if (not FileExists(Filename)) then + begin + Log.LogError('Audio-file does not exist: "' + Filename + '"', 'UAudio_FFmpeg'); + Exit; + end; + + Self.Filename := Filename; + + // open audio file + if (av_open_input_file(FormatCtx, PChar(Filename), nil, 0, nil) <> 0) then + begin + Log.LogError('av_open_input_file failed: "' + Filename + '"', 'UAudio_FFmpeg'); + Exit; + end; + + // generate PTS values if they do not exist + FormatCtx^.flags := FormatCtx^.flags or AVFMT_FLAG_GENPTS; + + // retrieve stream information + if (av_find_stream_info(FormatCtx) < 0) then + begin + Log.LogError('av_find_stream_info failed: "' + Filename + '"', 'UAudio_FFmpeg'); + Close(); + Exit; + end; + + // FIXME: hack used by ffplay. Maybe should not use url_feof() to test for the end + FormatCtx^.pb.eof_reached := 0; + + {$IFDEF DebugFFmpegDecode} + dump_format(FormatCtx, 0, pchar(Filename), 0); + {$ENDIF} + + AudioStreamIndex := FFmpegCore.FindAudioStreamIndex(FormatCtx); + if (AudioStreamIndex < 0) then + begin + Log.LogError('FindAudioStreamIndex: No Audio-stream found "' + Filename + '"', 'UAudio_FFmpeg'); + Close(); + Exit; + end; + + //Log.LogStatus('AudioStreamIndex is: '+ inttostr(ffmpegStreamID), 'UAudio_FFmpeg'); + + AudioStream := FormatCtx.streams[AudioStreamIndex]; + CodecCtx := AudioStream^.codec; + + // TODO: should we use this or not? Should we allow 5.1 channel audio? + (* + {$IF LIBAVCODEC_VERSION >= 51042000} + if (CodecCtx^.channels > 0) then + CodecCtx^.request_channels := Min(2, CodecCtx^.channels) + else + CodecCtx^.request_channels := 2; + {$IFEND} + *) + + Codec := avcodec_find_decoder(CodecCtx^.codec_id); + if (Codec = nil) then + begin + Log.LogError('Unsupported codec!', 'UAudio_FFmpeg'); + CodecCtx := nil; + Close(); + Exit; + end; + + // set debug options + CodecCtx^.debug_mv := 0; + CodecCtx^.debug := 0; + + // detect bug-workarounds automatically + CodecCtx^.workaround_bugs := FF_BUG_AUTODETECT; + // error resilience strategy (careful/compliant/agressive/very_aggressive) + //CodecCtx^.error_resilience := FF_ER_CAREFUL; //FF_ER_COMPLIANT; + // allow non spec compliant speedup tricks. + //CodecCtx^.flags2 := CodecCtx^.flags2 or CODEC_FLAG2_FAST; + + // Note: avcodec_open() and avcodec_close() are not thread-safe and will + // fail if called concurrently by different threads. + FFmpegCore.LockAVCodec(); + try + AVResult := avcodec_open(CodecCtx, Codec); + finally + FFmpegCore.UnlockAVCodec(); + end; + if (AVResult < 0) then + begin + Log.LogError('avcodec_open failed!', 'UAudio_FFmpeg'); + Close(); + Exit; + end; + + // now initialize the audio-format + + if (not FFmpegCore.ConvertFFmpegToAudioFormat(CodecCtx^.sample_fmt, SampleFormat)) then + begin + // try standard format + SampleFormat := asfS16; + end; + + FormatInfo := TAudioFormatInfo.Create( + CodecCtx^.channels, + CodecCtx^.sample_rate, + SampleFormat + ); + + + PacketQueue := TPacketQueue.Create(); + + // finally start the decode thread + ParseThread := SDL_CreateThread(@ParseThreadMain, Self); + + Result := true; +end; + +procedure TFFmpegDecodeStream.Close(); +var + ThreadResult: integer; +begin + // wake threads waiting for packet-queue data + // Note: normally, there are no waiting threads. If there were waiting + // ones, they would block the audio-callback thread. + if (assigned(PacketQueue)) then + PacketQueue.Abort(); + + // send quit request (to parse-thread etc) + SDL_mutexP(StateLock); + QuitRequest := true; + SDL_CondBroadcast(ParserIdleCond); + SDL_mutexV(StateLock); + + // abort parse-thread + if (ParseThread <> nil) then + begin + // and wait until it terminates + SDL_WaitThread(ParseThread, ThreadResult); + ParseThread := nil; + end; + + // Close the codec + if (CodecCtx <> nil) then + begin + // avcodec_close() is not thread-safe + FFmpegCore.LockAVCodec(); + try + avcodec_close(CodecCtx); + finally + FFmpegCore.UnlockAVCodec(); + end; + CodecCtx := nil; + end; + + // Close the video file + if (FormatCtx <> nil) then + begin + av_close_input_file(FormatCtx); + FormatCtx := nil; + end; + + PerformOnClose(); + + FreeAndNil(PacketQueue); + FreeAndNil(FormatInfo); +end; + +function TFFmpegDecodeStream.GetLength(): real; +begin + // do not forget to consider the start_time value here + Result := (FormatCtx^.start_time + FormatCtx^.duration) / AV_TIME_BASE; +end; + +function TFFmpegDecodeStream.GetAudioFormatInfo(): TAudioFormatInfo; +begin + Result := FormatInfo; +end; + +function TFFmpegDecodeStream.IsEOF(): boolean; +begin + SDL_mutexP(StateLock); + Result := EOFState; + SDL_mutexV(StateLock); +end; + +procedure TFFmpegDecodeStream.SetEOF(State: boolean); +begin + SDL_mutexP(StateLock); + EOFState := State; + SDL_mutexV(StateLock); +end; + +function TFFmpegDecodeStream.IsError(): boolean; +begin + SDL_mutexP(StateLock); + Result := ErrorState; + SDL_mutexV(StateLock); +end; + +procedure TFFmpegDecodeStream.SetError(State: boolean); +begin + SDL_mutexP(StateLock); + ErrorState := State; + SDL_mutexV(StateLock); +end; + +function TFFmpegDecodeStream.IsSeeking(): boolean; +begin + SDL_mutexP(StateLock); + Result := SeekRequest; + SDL_mutexV(StateLock); +end; + +function TFFmpegDecodeStream.IsQuit(): boolean; +begin + SDL_mutexP(StateLock); + Result := QuitRequest; + SDL_mutexV(StateLock); +end; + +function TFFmpegDecodeStream.GetPosition(): real; +var + BufferSizeSec: double; +begin + PauseDecoder(); + + // ReadData() does not return all of the buffer retrieved by DecodeFrame(). + // Determine the size of the unused part of the decode-buffer. + BufferSizeSec := (AudioBufferSize - AudioBufferPos) / + FormatInfo.BytesPerSec; + + // subtract the size of unused buffer-data from the audio clock. + Result := AudioStreamPos - BufferSizeSec; + + ResumeDecoder(); +end; + +procedure TFFmpegDecodeStream.SetPosition(Time: real); +begin + SetPositionIntern(Time, true, true); +end; + +function TFFmpegDecodeStream.GetLoop(): boolean; +begin + SDL_mutexP(StateLock); + Result := Loop; + SDL_mutexV(StateLock); +end; + +procedure TFFmpegDecodeStream.SetLoop(Enabled: boolean); +begin + SDL_mutexP(StateLock); + Loop := Enabled; + SDL_mutexV(StateLock); +end; + + +(******************************************** + * Parser section + ********************************************) + +procedure TFFmpegDecodeStream.PauseParser(); +begin + if (SDL_ThreadID() = ParseThread.threadid) then + Exit; + + SDL_mutexP(StateLock); + Inc(ParserPauseRequestCount); + while (ParserLocked) do + SDL_CondWait(ParserUnlockedCond, StateLock); + SDL_mutexV(StateLock); +end; + +procedure TFFmpegDecodeStream.ResumeParser(); +begin + if (SDL_ThreadID() = ParseThread.threadid) then + Exit; + + SDL_mutexP(StateLock); + Dec(ParserPauseRequestCount); + SDL_CondSignal(ParserResumeCond); + SDL_mutexV(StateLock); +end; + +procedure TFFmpegDecodeStream.SetPositionIntern(Time: real; Flush: boolean; Blocking: boolean); +begin + // - Pause the parser first to prevent it from putting obsolete packages + // into the queue after the queue was flushed and before seeking is done. + // Otherwise we will hear fragments of old data, if the stream was seeked + // in stopped mode and resumed afterwards (applies to non-blocking mode only). + // - Pause the decoder to avoid race-condition that might occur otherwise. + // - Last lock the state lock because we are manipulating some shared state-vars. + PauseParser(); + PauseDecoder(); + SDL_mutexP(StateLock); + + // configure seek parameters + SeekPos := Time; + SeekFlush := Flush; + SeekFlags := AVSEEK_FLAG_ANY; + SeekRequest := true; + + // Note: the BACKWARD-flag seeks to the first position <= the position + // searched for. Otherwise e.g. position 0 might not be seeked correct. + // For some reason ffmpeg sometimes doesn't use position 0 but the key-frame + // following. In streams with few key-frames (like many flv-files) the next + // key-frame after 0 might be 5secs ahead. + if (Time < AudioStreamPos) then + SeekFlags := SeekFlags or AVSEEK_FLAG_BACKWARD; + + EOFState := false; + ErrorState := false; + + // send a reuse signal in case the parser was stopped (e.g. because of an EOF) + SDL_CondSignal(ParserIdleCond); + + SDL_mutexV(StateLock); + ResumeDecoder(); + ResumeParser(); + + // in blocking mode, wait until seeking is done + if (Blocking) then + begin + SDL_mutexP(StateLock); + while (SeekRequest) do + SDL_CondWait(SeekFinishedCond, StateLock); + SDL_mutexV(StateLock); + end; +end; + +function ParseThreadMain(Data: Pointer): integer; cdecl; +var + Stream: TFFmpegDecodeStream; +begin + Stream := TFFmpegDecodeStream(Data); + if (Stream <> nil) then + Stream.Parse(); + Result := 0; +end; + +procedure TFFmpegDecodeStream.Parse(); +begin + // reuse thread as long as the stream is not terminated + while (ParseLoop()) do + begin + // wait for reuse or destruction of stream + SDL_mutexP(StateLock); + while (not (SeekRequest or QuitRequest)) do + SDL_CondWait(ParserIdleCond, StateLock); + SDL_mutexV(StateLock); + end; +end; + +(** + * Parser main loop. + * Will not return until parsing of the stream is finished. + * Reasons for the parser to return are: + * - the end-of-file is reached + * - an error occured + * - the stream was quited (received a quit-request) + * Returns true if the stream can be resumed or false if the stream has to + * be terminated. + *) +function TFFmpegDecodeStream.ParseLoop(): boolean; +var + Packet: TAVPacket; + StatusPacket: PAVPacket; + SeekTarget: int64; + ByteIOCtx: PByteIOContext; + ErrorCode: integer; + StartSilence: double; // duration of silence at start of stream + StartSilencePtr: PDouble; // pointer for the EMPTY status packet + + // Note: pthreads wakes threads waiting on a mutex in the order of their + // priority and not in FIFO order. SDL does not provide any option to + // control priorities. This might (and already did) starve threads waiting + // on the mutex (e.g. SetPosition) making usdx look like it was froozen. + // Instead of simply locking the critical section we set a ParserLocked flag + // instead and give priority to the threads requesting the parser to pause. + procedure LockParser(); + begin + SDL_mutexP(StateLock); + while (ParserPauseRequestCount > 0) do + SDL_CondWait(ParserResumeCond, StateLock); + ParserLocked := true; + SDL_mutexV(StateLock); + end; + + procedure UnlockParser(); + begin + SDL_mutexP(StateLock); + ParserLocked := false; + SDL_CondBroadcast(ParserUnlockedCond); + SDL_mutexV(StateLock); + end; + +begin + Result := true; + + while (true) do + begin + LockParser(); + try + + if (IsQuit()) then + begin + Result := false; + Exit; + end; + + // handle seek-request (Note: no need to lock SeekRequest here) + if (SeekRequest) then + begin + // first try: seek on the audio stream + SeekTarget := Round(SeekPos / av_q2d(AudioStream^.time_base)); + StartSilence := 0; + if (SeekTarget < AudioStream^.start_time) then + StartSilence := (AudioStream^.start_time - SeekTarget) * av_q2d(AudioStream^.time_base); + ErrorCode := av_seek_frame(FormatCtx, AudioStreamIndex, SeekTarget, SeekFlags); + + if (ErrorCode < 0) then + begin + // second try: seek on the default stream (necessary for flv-videos and some ogg-files) + SeekTarget := Round(SeekPos * AV_TIME_BASE); + StartSilence := 0; + if (SeekTarget < FormatCtx^.start_time) then + StartSilence := (FormatCtx^.start_time - SeekTarget) / AV_TIME_BASE; + ErrorCode := av_seek_frame(FormatCtx, -1, SeekTarget, SeekFlags); + end; + + // pause decoder and lock state (keep the lock-order to avoid deadlocks). + // Note that the decoder does not block in the packet-queue in seeking state, + // so locking the decoder here does not cause a dead-lock. + PauseDecoder(); + SDL_mutexP(StateLock); + try + if (ErrorCode < 0) then + begin + // seeking failed + ErrorState := true; + Log.LogStatus('Seek Error in "'+FormatCtx^.filename+'"', 'UAudioDecoder_FFmpeg'); + end + else + begin + if (SeekFlush) then + begin + // flush queue (we will send a Flush-Packet when seeking is finished) + PacketQueue.Flush(); + + // flush the decode buffers + AudioBufferSize := 0; + AudioBufferPos := 0; + AudioPaketSize := 0; + AudioPaketSilence := 0; + FlushCodecBuffers(); + + // Set preliminary stream position. The position will be set to + // the correct value as soon as the first packet is decoded. + AudioStreamPos := SeekPos; + end + else + begin + // request avcodec buffer flush + PacketQueue.PutStatus(PKT_STATUS_FLAG_FLUSH, nil); + end; + + // fill the gap between position 0 and start_time with silence + // but not if we are in loop mode + if ((StartSilence > 0) and (not Loop)) then + begin + GetMem(StartSilencePtr, SizeOf(StartSilence)); + StartSilencePtr^ := StartSilence; + PacketQueue.PutStatus(PKT_STATUS_FLAG_EMPTY, StartSilencePtr); + end; + end; + + SeekRequest := false; + SDL_CondBroadcast(SeekFinishedCond); + finally + SDL_mutexV(StateLock); + ResumeDecoder(); + end; + end; + + if (PacketQueue.GetSize() > MAX_AUDIOQ_SIZE) then + begin + SDL_Delay(10); + Continue; + end; + + if (av_read_frame(FormatCtx, Packet) < 0) then + begin + // failed to read a frame, check reason + {$IF (LIBAVFORMAT_VERSION_MAJOR >= 52)} + ByteIOCtx := FormatCtx^.pb; + {$ELSE} + ByteIOCtx := @FormatCtx^.pb; + {$IFEND} + + // check for end-of-file (eof is not an error) + if (url_feof(ByteIOCtx) <> 0) then + begin + if (GetLoop()) then + begin + // rewind stream (but do not flush) + SetPositionIntern(0, false, false); + Continue; + end + else + begin + // signal end-of-file + PacketQueue.PutStatus(PKT_STATUS_FLAG_EOF, nil); + Exit; + end; + end; + + // check for errors + if (url_ferror(ByteIOCtx) <> 0) then + begin + // an error occured -> abort and wait for repositioning or termination + PacketQueue.PutStatus(PKT_STATUS_FLAG_ERROR, nil); + Exit; + end; + + // no error -> wait for user input + SDL_Delay(100); + Continue; + end; + + if (Packet.stream_index = AudioStreamIndex) then + PacketQueue.Put(@Packet) + else + av_free_packet(@Packet); + + finally + UnlockParser(); + end; + end; +end; + + +(******************************************** + * Decoder section + ********************************************) + +procedure TFFmpegDecodeStream.PauseDecoder(); +begin + SDL_mutexP(StateLock); + Inc(DecoderPauseRequestCount); + while (DecoderLocked) do + SDL_CondWait(DecoderUnlockedCond, StateLock); + SDL_mutexV(StateLock); +end; + +procedure TFFmpegDecodeStream.ResumeDecoder(); +begin + SDL_mutexP(StateLock); + Dec(DecoderPauseRequestCount); + SDL_CondSignal(DecoderResumeCond); + SDL_mutexV(StateLock); +end; + +procedure TFFmpegDecodeStream.FlushCodecBuffers(); +begin + // if no flush operation is specified, avcodec_flush_buffers will not do anything. + if (@CodecCtx.codec.flush <> nil) then + begin + // flush buffers used by avcodec_decode_audio, etc. + avcodec_flush_buffers(CodecCtx); + end + else + begin + // we need a Workaround to avoid plopping noise with ogg-vorbis and + // mp3 (in older versions of FFmpeg). + // We will just reopen the codec. + FFmpegCore.LockAVCodec(); + try + avcodec_close(CodecCtx); + avcodec_open(CodecCtx, Codec); + finally + FFmpegCore.UnlockAVCodec(); + end; + end; +end; + +function TFFmpegDecodeStream.DecodeFrame(Buffer: PChar; BufferSize: integer): integer; +var + PaketDecodedSize: integer; // size of packet data used for decoding + DataSize: integer; // size of output data decoded by FFmpeg + BlockQueue: boolean; + SilenceDuration: double; + {$IFDEF DebugFFmpegDecode} + TmpPos: double; + {$ENDIF} +begin + Result := -1; + + if (EOF) then + Exit; + + while(true) do + begin + // for titles with start_time > 0 we have to generate silence + // until we reach the pts of the first data packet. + if (AudioPaketSilence > 0) then + begin + DataSize := Min(AudioPaketSilence, BufferSize); + FillChar(Buffer[0], DataSize, 0); + Dec(AudioPaketSilence, DataSize); + AudioStreamPos := AudioStreamPos + DataSize / FormatInfo.BytesPerSec; + Result := DataSize; + Exit; + end; + + // read packet data + while (AudioPaketSize > 0) do + begin + DataSize := BufferSize; + + {$IF LIBAVCODEC_VERSION >= 51030000} // 51.30.0 + PaketDecodedSize := avcodec_decode_audio2(CodecCtx, PSmallint(Buffer), + DataSize, AudioPaketData, AudioPaketSize); + {$ELSE} + PaketDecodedSize := avcodec_decode_audio(CodecCtx, PSmallint(Buffer), + DataSize, AudioPaketData, AudioPaketSize); + {$IFEND} + + if(PaketDecodedSize < 0) then + begin + // if error, skip frame + {$IFDEF DebugFFmpegDecode} + DebugWriteln('Skip audio frame'); + {$ENDIF} + AudioPaketSize := 0; + Break; + end; + + Inc(AudioPaketData, PaketDecodedSize); + Dec(AudioPaketSize, PaketDecodedSize); + + // check if avcodec_decode_audio returned data, otherwise fetch more frames + if (DataSize <= 0) then + Continue; + + // update stream position by the amount of fetched data + AudioStreamPos := AudioStreamPos + DataSize / FormatInfo.BytesPerSec; + + // we have data, return it and come back for more later + Result := DataSize; + Exit; + end; + + // free old packet data + if (AudioPaket.data <> nil) then + av_free_packet(@AudioPaket); + + // do not block queue on seeking (to avoid deadlocks on the DecoderLock) + if (IsSeeking()) then + BlockQueue := false + else + BlockQueue := true; + + // request a new packet and block if none available. + // If this fails, the queue was aborted. + if (PacketQueue.Get(AudioPaket, BlockQueue) <= 0) then + Exit; + + // handle Status-packet + if (PChar(AudioPaket.data) = STATUS_PACKET) then + begin + AudioPaket.data := nil; + AudioPaketData := nil; + AudioPaketSize := 0; + + case (AudioPaket.flags) of + PKT_STATUS_FLAG_FLUSH: + begin + // just used if SetPositionIntern was called without the flush flag. + FlushCodecBuffers; + end; + PKT_STATUS_FLAG_EOF: // end-of-file + begin + // ignore EOF while seeking + if (not IsSeeking()) then + SetEOF(true); + // buffer contains no data -> result = -1 + Exit; + end; + PKT_STATUS_FLAG_ERROR: + begin + SetError(true); + Log.LogStatus('I/O Error', 'TFFmpegDecodeStream.DecodeFrame'); + Exit; + end; + PKT_STATUS_FLAG_EMPTY: + begin + SilenceDuration := PDouble(PacketQueue.GetStatusInfo(AudioPaket))^; + AudioPaketSilence := Round(SilenceDuration * FormatInfo.SampleRate) * FormatInfo.FrameSize; + PacketQueue.FreeStatusInfo(AudioPaket); + end + else + begin + Log.LogStatus('Unknown status', 'TFFmpegDecodeStream.DecodeFrame'); + end; + end; + + Continue; + end; + + AudioPaketData := PChar(AudioPaket.data); + AudioPaketSize := AudioPaket.size; + + // if available, update the stream position to the presentation time of this package + if(AudioPaket.pts <> AV_NOPTS_VALUE) then + begin + {$IFDEF DebugFFmpegDecode} + TmpPos := AudioStreamPos; + {$ENDIF} + AudioStreamPos := av_q2d(AudioStream^.time_base) * AudioPaket.pts; + {$IFDEF DebugFFmpegDecode} + DebugWriteln('Timestamp: ' + floattostrf(AudioStreamPos, ffFixed, 15, 3) + ' ' + + '(Calc: ' + floattostrf(TmpPos, ffFixed, 15, 3) + '), ' + + 'Diff: ' + floattostrf(AudioStreamPos-TmpPos, ffFixed, 15, 3)); + {$ENDIF} + end; + end; +end; + +function TFFmpegDecodeStream.ReadData(Buffer: PChar; BufferSize: integer): integer; +var + CopyByteCount: integer; // number of bytes to copy + RemainByteCount: integer; // number of bytes left (remain) to read + BufferPos: integer; + + // prioritize pause requests + procedure LockDecoder(); + begin + SDL_mutexP(StateLock); + while (DecoderPauseRequestCount > 0) do + SDL_CondWait(DecoderResumeCond, StateLock); + DecoderLocked := true; + SDL_mutexV(StateLock); + end; + + procedure UnlockDecoder(); + begin + SDL_mutexP(StateLock); + DecoderLocked := false; + SDL_CondBroadcast(DecoderUnlockedCond); + SDL_mutexV(StateLock); + end; + +begin + Result := -1; + + // set number of bytes to copy to the output buffer + BufferPos := 0; + + LockDecoder(); + try + // leave if end-of-file is reached + if (EOF) then + Exit; + + // copy data to output buffer + while (BufferPos < BufferSize) do + begin + // check if we need more data + if (AudioBufferPos >= AudioBufferSize) then + begin + AudioBufferPos := 0; + + // we have already sent all our data; get more + AudioBufferSize := DecodeFrame(AudioBuffer, AUDIO_BUFFER_SIZE); + + // check for errors or EOF + if(AudioBufferSize < 0) then + begin + Result := BufferPos; + Exit; + end; + end; + + // calc number of new bytes in the decode-buffer + CopyByteCount := AudioBufferSize - AudioBufferPos; + // resize copy-count if more bytes available than needed (remaining bytes are used the next time) + RemainByteCount := BufferSize - BufferPos; + if (CopyByteCount > RemainByteCount) then + CopyByteCount := RemainByteCount; + + Move(AudioBuffer[AudioBufferPos], Buffer[BufferPos], CopyByteCount); + + Inc(BufferPos, CopyByteCount); + Inc(AudioBufferPos, CopyByteCount); + end; + finally + UnlockDecoder(); + end; + + Result := BufferSize; +end; + + +{ TAudioDecoder_FFmpeg } + +function TAudioDecoder_FFmpeg.GetName: String; +begin + Result := 'FFmpeg_Decoder'; +end; + +function TAudioDecoder_FFmpeg.InitializeDecoder: boolean; +begin + //Log.LogStatus('InitializeDecoder', 'UAudioDecoder_FFmpeg'); + FFmpegCore := TMediaCore_FFmpeg.GetInstance(); + av_register_all(); + + // Do not show uninformative error messages by default. + // FFmpeg prints all error-infos on the console by default what + // is very confusing as the playback of the files is correct. + // We consider these errors to be internal to FFMpeg. They can be fixed + // by the FFmpeg guys only and do not provide any useful information in + // respect to USDX. + {$IFNDEF EnableFFmpegErrorOutput} + {$IF LIBAVUTIL_VERSION_MAJOR >= 50} + av_log_set_level(AV_LOG_FATAL); + {$ELSE} + // FATAL and ERROR share one log-level, so we have to use QUIET + av_log_set_level(AV_LOG_QUIET); + {$IFEND} + {$ENDIF} + + Result := true; +end; + +function TAudioDecoder_FFmpeg.FinalizeDecoder(): boolean; +begin + Result := true; +end; + +function TAudioDecoder_FFmpeg.Open(const Filename: string): TAudioDecodeStream; +var + Stream: TFFmpegDecodeStream; +begin + Result := nil; + + Stream := TFFmpegDecodeStream.Create(); + if (not Stream.Open(Filename)) then + begin + Stream.Free; + Exit; + end; + + Result := Stream; +end; + + +initialization + MediaManager.Add(TAudioDecoder_FFmpeg.Create); + +end. diff --git a/src/media/UAudioInput_Bass.pas b/src/media/UAudioInput_Bass.pas new file mode 100644 index 00000000..65a4704d --- /dev/null +++ b/src/media/UAudioInput_Bass.pas @@ -0,0 +1,481 @@ +unit UAudioInput_Bass; + +interface + +{$IFDEF FPC} + {$MODE Delphi} +{$ENDIF} + +{$I switches.inc} + + +uses + Classes, + SysUtils, + URecord, + UMusic; + +implementation + +uses + UMain, + UIni, + ULog, + UAudioCore_Bass, + UCommon, // (Note: for MakeLong on non-windows platforms) + {$IFDEF MSWINDOWS} + Windows, // (Note: for MakeLong) + {$ENDIF} + bass; // (Note: DWORD is redefined here -> insert after Windows-unit) + +type + TAudioInput_Bass = class(TAudioInputBase) + private + function EnumDevices(): boolean; + public + function GetName: String; override; + function InitializeRecord: boolean; override; + function FinalizeRecord: boolean; override; + end; + + TBassInputDevice = class(TAudioInputDevice) + private + RecordStream: HSTREAM; + BassDeviceID: DWORD; // DeviceID used by BASS + SingleIn: boolean; + + function SetInputSource(SourceIndex: integer): boolean; + function GetInputSource(): integer; + public + function Open(): boolean; + function Close(): boolean; + function Start(): boolean; override; + function Stop(): boolean; override; + + function GetVolume(): single; override; + procedure SetVolume(Volume: single); override; + end; + +var + BassCore: TAudioCore_Bass; + + +{ Global } + +{* + * Bass input capture callback. + * Params: + * stream - BASS input stream + * buffer - buffer of captured samples + * len - size of buffer in bytes + * user - players associated with left/right channels + *} +function MicrophoneCallback(stream: HSTREAM; buffer: Pointer; + len: Cardinal; inputDevice: Pointer): boolean; {$IFDEF MSWINDOWS}stdcall;{$ELSE}cdecl;{$ENDIF} +begin + AudioInputProcessor.HandleMicrophoneData(buffer, len, inputDevice); + Result := true; +end; + + +{ TBassInputDevice } + +function TBassInputDevice.GetInputSource(): integer; +var + SourceCnt: integer; + i: integer; + flags: DWORD; +begin + // get input-source config (subtract virtual device to get BASS indices) + SourceCnt := Length(Source)-1; + + // find source + Result := -1; + for i := 0 to SourceCnt-1 do + begin + // get input settings + flags := BASS_RecordGetInput(i, PSingle(nil)^); + if (flags = DWORD(-1)) then + begin + Log.LogError('BASS_RecordGetInput: ' + BassCore.ErrorGetString(), 'TBassInputDevice.GetInputSource'); + Exit; + end; + + // check if current source is selected + if ((flags and BASS_INPUT_OFF) = 0) then + begin + // selected source found + Result := i; + Exit; + end; + end; +end; + +function TBassInputDevice.SetInputSource(SourceIndex: integer): boolean; +var + SourceCnt: integer; + i: integer; + flags: DWORD; +begin + Result := false; + + // check for invalid source index + if (SourceIndex < 0) then + Exit; + + // get input-source config (subtract virtual device to get BASS indices) + SourceCnt := Length(Source)-1; + + // turn on selected source (turns off the others for single-in devices) + if (not BASS_RecordSetInput(SourceIndex, BASS_INPUT_ON, -1)) then + begin + Log.LogError('BASS_RecordSetInput: ' + BassCore.ErrorGetString(), 'TBassInputDevice.Start'); + Exit; + end; + + // turn off all other sources (not needed for single-in devices) + if (not SingleIn) then + begin + for i := 0 to SourceCnt-1 do + begin + if (i = SourceIndex) then + continue; + // get input settings + flags := BASS_RecordGetInput(i, PSingle(nil)^); + if (flags = DWORD(-1)) then + begin + Log.LogError('BASS_RecordGetInput: ' + BassCore.ErrorGetString(), 'TBassInputDevice.GetInputSource'); + Exit; + end; + // deselect source if selected + if ((flags and BASS_INPUT_OFF) = 0) then + BASS_RecordSetInput(i, BASS_INPUT_OFF, -1); + end; + end; + + Result := true; +end; + +function TBassInputDevice.Open(): boolean; +var + FormatFlags: DWORD; + SourceIndex: integer; +const + latency = 20; // 20ms callback period (= latency) +begin + Result := false; + + if (not BASS_RecordInit(BassDeviceID)) then + begin + Log.LogError('BASS_RecordInit['+Name+']: ' + + BassCore.ErrorGetString(), 'TBassInputDevice.Open'); + Exit; + end; + + if (not BassCore.ConvertAudioFormatToBASSFlags(AudioFormat.Format, FormatFlags)) then + begin + Log.LogError('Unhandled sample-format', 'TBassInputDevice.Open'); + Exit; + end; + + // start capturing in paused state + RecordStream := BASS_RecordStart(Round(AudioFormat.SampleRate), AudioFormat.Channels, + MakeLong(FormatFlags or BASS_RECORD_PAUSE, latency), + @MicrophoneCallback, Self); + if (RecordStream = 0) then + begin + Log.LogError('BASS_RecordStart: ' + BassCore.ErrorGetString(), 'TBassInputDevice.Open'); + BASS_RecordFree; + Exit; + end; + + // save current source selection and select new source + SourceIndex := Ini.InputDeviceConfig[CfgIndex].Input-1; + if (SourceIndex = -1) then + begin + // nothing to do if default source is used + SourceRestore := -1; + end + else + begin + // store current source-index and select new source + SourceRestore := GetInputSource(); + SetInputSource(SourceIndex); + end; + + Result := true; +end; + +{* Start input-capturing on this device. *} +function TBassInputDevice.Start(): boolean; +begin + Result := false; + + // recording already started -> stop first + if (RecordStream <> 0) then + Stop(); + + // TODO: Do not open the device here (takes too much time). + if not Open() then + Exit; + + if (not BASS_ChannelPlay(RecordStream, true)) then + begin + Log.LogError('BASS_ChannelPlay: ' + BassCore.ErrorGetString(), 'TBassInputDevice.Start'); + Exit; + end; + + Result := true; +end; + +{* Stop input-capturing on this device. *} +function TBassInputDevice.Stop(): boolean; +begin + Result := false; + + if (RecordStream = 0) then + Exit; + if (not BASS_RecordSetDevice(BassDeviceID)) then + Exit; + + if (not BASS_ChannelStop(RecordStream)) then + begin + Log.LogError('BASS_ChannelStop: ' + BassCore.ErrorGetString(), 'TBassInputDevice.Stop'); + end; + + // TODO: Do not close the device here (takes too much time). + Result := Close(); +end; + +function TBassInputDevice.Close(): boolean; +begin + // restore source selection + if (SourceRestore >= 0) then + begin + SetInputSource(SourceRestore); + end; + + // free data + if (not BASS_RecordFree()) then + begin + Log.LogError('BASS_RecordFree: ' + BassCore.ErrorGetString(), 'TBassInputDevice.Close'); + Result := false; + end + else + begin + Result := true; + end; + + RecordStream := 0; +end; + +function TBassInputDevice.GetVolume(): single; +var + SourceIndex: integer; + lVolume: Single; +begin + Result := 0; + + SourceIndex := Ini.InputDeviceConfig[CfgIndex].Input-1; + if (SourceIndex = -1) then + begin + // if default source used find selected source + SourceIndex := GetInputSource(); + if (SourceIndex = -1) then + Exit; + end; + + if (BASS_RecordGetInput(SourceIndex, lVolume) = DWORD(-1)) then + begin + Log.LogError('BASS_RecordGetInput: ' + BassCore.ErrorGetString() , 'TBassInputDevice.GetVolume'); + Exit; + end; + Result := lVolume; +end; + +procedure TBassInputDevice.SetVolume(Volume: single); +var + SourceIndex: integer; +begin + SourceIndex := Ini.InputDeviceConfig[CfgIndex].Input-1; + if (SourceIndex = -1) then + begin + // if default source used find selected source + SourceIndex := GetInputSource(); + if (SourceIndex = -1) then + Exit; + end; + + // clip volume to valid range + if (Volume > 1.0) then + Volume := 1.0 + else if (Volume < 0) then + Volume := 0; + + if (not BASS_RecordSetInput(SourceIndex, 0, Volume)) then + begin + Log.LogError('BASS_RecordSetInput: ' + BassCore.ErrorGetString() , 'TBassInputDevice.SetVolume'); + end; +end; + + +{ TAudioInput_Bass } + +function TAudioInput_Bass.GetName: String; +begin + result := 'BASS_Input'; +end; + +function TAudioInput_Bass.EnumDevices(): boolean; +var + Descr: PChar; + SourceName: PChar; + Flags: integer; + BassDeviceID: integer; + BassDevice: TBassInputDevice; + DeviceIndex: integer; + DeviceInfo: BASS_DEVICEINFO; + SourceIndex: integer; + RecordInfo: BASS_RECORDINFO; + SelectedSourceIndex: integer; +begin + result := false; + + DeviceIndex := 0; + BassDeviceID := 0; + SetLength(AudioInputProcessor.DeviceList, 0); + + // checks for recording devices and puts them into an array + while true do + begin + if (not BASS_RecordGetDeviceInfo(BassDeviceID, DeviceInfo)) then + break; + + // try to initialize the device + if not BASS_RecordInit(BassDeviceID) then + begin + Log.LogStatus('Failed to initialize BASS Capture-Device['+inttostr(BassDeviceID)+']', + 'TAudioInput_Bass.InitializeRecord'); + end + else + begin + SetLength(AudioInputProcessor.DeviceList, DeviceIndex+1); + + // TODO: free object on termination + BassDevice := TBassInputDevice.Create(); + AudioInputProcessor.DeviceList[DeviceIndex] := BassDevice; + + Descr := DeviceInfo.name; + + BassDevice.BassDeviceID := BassDeviceID; + BassDevice.Name := UnifyDeviceName(Descr, DeviceIndex); + + // zero info-struct as some fields might not be set (e.g. freq is just set on Vista and MacOSX) + FillChar(RecordInfo, SizeOf(RecordInfo), 0); + // retrieve recording device info + BASS_RecordGetInfo(RecordInfo); + + // check if BASS has capture-freq. info + if (RecordInfo.freq > 0) then + begin + // use current input sample rate (available only on Windows Vista and OSX). + // Recording at this rate will give the best quality and performance, as no resampling is required. + // FIXME: does BASS use LSB/MSB or system integer values for 16bit? + BassDevice.AudioFormat := TAudioFormatInfo.Create(2, RecordInfo.freq, asfS16) + end + else + begin + // BASS does not provide an explizit input channel count (except BASS_RECORDINFO.formats) + // but it doesn't fail if we use stereo input on a mono device + // -> use stereo by default + BassDevice.AudioFormat := TAudioFormatInfo.Create(2, 44100, asfS16) + end; + + // get info if multiple input-sources can be selected at once + BassDevice.SingleIn := RecordInfo.singlein; + + // init list for capture buffers per channel + SetLength(BassDevice.CaptureChannel, BassDevice.AudioFormat.Channels); + + BassDevice.MicSource := -1; + BassDevice.SourceRestore := -1; + + // add a virtual default source (will not change mixer-settings) + SetLength(BassDevice.Source, 1); + BassDevice.Source[0].Name := DEFAULT_SOURCE_NAME; + + // add real input sources + SourceIndex := 1; + + // process each input + while true do + begin + SourceName := BASS_RecordGetInputName(SourceIndex-1); + + {$IFDEF DARWIN} + // Under MacOSX the SingStar Mics have an empty InputName. + // So, we have to add a hard coded Workaround for this problem + // FIXME: - Do we need this anymore? Doesn't the (new) default source already solve this problem? + // - Normally a nil return value of BASS_RecordGetInputName() means end-of-list, so maybe + // BASS is not able to detect any mic-sources (the default source will work then). + // - Does BASS_RecordGetInfo() return true or false? If it returns true in this case + // we could use this value to check if the device exists. + // Please check that, eddie. + // If it returns false, then the source is not detected and it does not make sense to add a second + // fake device here. + // What about BASS_RecordGetInput()? Does it return a value <> -1? + // - Does it even work at all with this fake source-index, now that input switching works? + // This info was not used before (sources were never switched), so it did not matter what source-index was used. + // But now BASS_RecordSetInput() will probably fail. + if ((SourceName = nil) and (SourceIndex = 1) and (Pos('USBMIC Serial#', Descr) > 0)) then + SourceName := 'Microphone' + {$ENDIF} + + if (SourceName = nil) then + break; + + SetLength(BassDevice.Source, Length(BassDevice.Source)+1); + BassDevice.Source[SourceIndex].Name := SourceName; + + // get input-source info + Flags := BASS_RecordGetInput(SourceIndex, PSingle(nil)^); + if (Flags <> -1) then + begin + // is the current source a mic-source? + if ((Flags and BASS_INPUT_TYPE_MIC) <> 0) then + BassDevice.MicSource := SourceIndex; + end; + + Inc(SourceIndex); + end; + + // FIXME: this call hangs in FPC (windows) every 2nd time USDX is called. + // Maybe because the sound-device was not released properly? + BASS_RecordFree; + + Inc(DeviceIndex); + end; + + Inc(BassDeviceID); + end; + + result := true; +end; + +function TAudioInput_Bass.InitializeRecord(): boolean; +begin + BassCore := TAudioCore_Bass.GetInstance(); + Result := EnumDevices(); +end; + +function TAudioInput_Bass.FinalizeRecord(): boolean; +begin + CaptureStop; + Result := inherited FinalizeRecord; +end; + + +initialization + MediaManager.Add(TAudioInput_Bass.Create); + +end. diff --git a/src/media/UAudioInput_Portaudio.pas b/src/media/UAudioInput_Portaudio.pas new file mode 100644 index 00000000..9a1c3e99 --- /dev/null +++ b/src/media/UAudioInput_Portaudio.pas @@ -0,0 +1,474 @@ +unit UAudioInput_Portaudio; + +interface + +{$IFDEF FPC} + {$MODE Delphi} +{$ENDIF} + +{$I ../switches.inc} + + +uses + Classes, + SysUtils, + UMusic; + +implementation + +uses + {$IFDEF UsePortmixer} + portmixer, + {$ENDIF} + portaudio, + UAudioCore_Portaudio, + URecord, + UIni, + ULog, + UMain; + +type + TAudioInput_Portaudio = class(TAudioInputBase) + private + AudioCore: TAudioCore_Portaudio; + function EnumDevices(): boolean; + public + function GetName: String; override; + function InitializeRecord: boolean; override; + function FinalizeRecord: boolean; override; + end; + + TPortaudioInputDevice = class(TAudioInputDevice) + private + RecordStream: PPaStream; + {$IFDEF UsePortmixer} + Mixer: PPxMixer; + {$ENDIF} + PaDeviceIndex: TPaDeviceIndex; + public + function Open(): boolean; + function Close(): boolean; + function Start(): boolean; override; + function Stop(): boolean; override; + + function GetVolume(): single; override; + procedure SetVolume(Volume: single); override; + end; + +function MicrophoneCallback(input: Pointer; output: Pointer; frameCount: Longword; + timeInfo: PPaStreamCallbackTimeInfo; statusFlags: TPaStreamCallbackFlags; + inputDevice: Pointer): Integer; cdecl; forward; + +function MicrophoneTestCallback(input: Pointer; output: Pointer; frameCount: Longword; + timeInfo: PPaStreamCallbackTimeInfo; statusFlags: TPaStreamCallbackFlags; + inputDevice: Pointer): Integer; cdecl; forward; + + +{ TPortaudioInputDevice } + +function TPortaudioInputDevice.Open(): boolean; +var + Error: TPaError; + inputParams: TPaStreamParameters; + deviceInfo: PPaDeviceInfo; + SourceIndex: integer; +begin + Result := false; + + // get input latency info + deviceInfo := Pa_GetDeviceInfo(PaDeviceIndex); + + // set input stream parameters + with inputParams do + begin + device := PaDeviceIndex; + channelCount := AudioFormat.Channels; + sampleFormat := paInt16; + suggestedLatency := deviceInfo^.defaultLowInputLatency; + hostApiSpecificStreamInfo := nil; + end; + + //Log.LogStatus(deviceInfo^.name, 'Portaudio'); + //Log.LogStatus(floattostr(deviceInfo^.defaultLowInputLatency), 'Portaudio'); + + // open input stream + Error := Pa_OpenStream(RecordStream, @inputParams, nil, + AudioFormat.SampleRate, + paFramesPerBufferUnspecified, paNoFlag, + @MicrophoneCallback, Pointer(Self)); + if(Error <> paNoError) then + begin + Log.LogError('Error opening stream: ' + Pa_GetErrorText(Error), 'TPortaudioInputDevice.Open'); + Exit; + end; + + {$IFDEF UsePortmixer} + // open default mixer + Mixer := Px_OpenMixer(RecordStream, 0); + if (Mixer = nil) then + begin + Log.LogError('Error opening mixer: ' + Pa_GetErrorText(Error), 'TPortaudioInputDevice.Open'); + end + else + begin + // save current source selection and select new source + SourceIndex := Ini.InputDeviceConfig[CfgIndex].Input-1; + if (SourceIndex = -1) then + begin + // nothing to do if default source is used + SourceRestore := -1; + end + else + begin + // store current source-index and select new source + SourceRestore := Px_GetCurrentInputSource(Mixer); // -1 in error case + Px_SetCurrentInputSource(Mixer, SourceIndex); + end; + end; + {$ENDIF} + + Result := true; +end; + +function TPortaudioInputDevice.Start(): boolean; +var + Error: TPaError; +begin + Result := false; + + // recording already started -> stop first + if (RecordStream <> nil) then + Stop(); + + // TODO: Do not open the device here (takes too much time). + if (not Open()) then + Exit; + + // start capture + Error := Pa_StartStream(RecordStream); + if(Error <> paNoError) then + begin + Log.LogError('Error starting stream: ' + Pa_GetErrorText(Error), 'TPortaudioInputDevice.Start'); + Close(); + RecordStream := nil; + Exit; + end; + + Result := true; +end; + +function TPortaudioInputDevice.Stop(): boolean; +var + Error: TPaError; +begin + Result := false; + + if (RecordStream = nil) then + Exit; + + // Note: do NOT call Pa_StopStream here! + // It gets stuck on devices with non-working callback as Pa_StopStream + // waits until all buffers have been handled (which never occurs in that case). + Error := Pa_AbortStream(RecordStream); + if (Error <> paNoError) then + begin + Log.LogError('Pa_AbortStream: ' + Pa_GetErrorText(Error), 'TPortaudioInputDevice.Stop'); + end; + + Result := Close(); +end; + +function TPortaudioInputDevice.Close(): boolean; +var + Error: TPaError; +begin + {$IFDEF UsePortmixer} + if (Mixer <> nil) then + begin + // restore source selection + if (SourceRestore >= 0) then + begin + Px_SetCurrentInputSource(Mixer, SourceRestore); + end; + + // close mixer + Px_CloseMixer(Mixer); + Mixer := nil; + end; + {$ENDIF} + + Error := Pa_CloseStream(RecordStream); + if (Error <> paNoError) then + begin + Log.LogError('Pa_CloseStream: ' + Pa_GetErrorText(Error), 'TPortaudioInputDevice.Close'); + Result := false; + end + else + begin + Result := true; + end; + + RecordStream := nil; +end; + +function TPortaudioInputDevice.GetVolume(): single; +begin + Result := 0; + {$IFDEF UsePortmixer} + if (Mixer <> nil) then + Result := Px_GetInputVolume(Mixer); + {$ENDIF} +end; + +procedure TPortaudioInputDevice.SetVolume(Volume: single); +begin + {$IFDEF UsePortmixer} + if (Mixer <> nil) then + begin + // clip to valid range + if (Volume > 1.0) then + Volume := 1.0 + else if (Volume < 0) then + Volume := 0; + Px_SetInputVolume(Mixer, Volume); + end; + {$ENDIF} +end; + + +{ TAudioInput_Portaudio } + +function TAudioInput_Portaudio.GetName: String; +begin + result := 'Portaudio'; +end; + +function TAudioInput_Portaudio.EnumDevices(): boolean; +var + i: integer; + paApiIndex: TPaHostApiIndex; + paApiInfo: PPaHostApiInfo; + deviceName: string; + deviceIndex: TPaDeviceIndex; + deviceInfo: PPaDeviceInfo; + channelCnt: integer; + SC: integer; // soundcard + err: TPaError; + errMsg: string; + paDevice: TPortaudioInputDevice; + inputParams: TPaStreamParameters; + stream: PPaStream; + streamInfo: PPaStreamInfo; + sampleRate: double; + latency: TPaTime; + {$IFDEF UsePortmixer} + mixer: PPxMixer; + sourceCnt: integer; + sourceIndex: integer; + sourceName: string; + {$ENDIF} + cbPolls: integer; + cbWorks: boolean; +begin + Result := false; + + // choose the best available Audio-API + paApiIndex := AudioCore.GetPreferredApiIndex(); + if(paApiIndex = -1) then + begin + Log.LogError('No working Audio-API found', 'TAudioInput_Portaudio.EnumDevices'); + Exit; + end; + + paApiInfo := Pa_GetHostApiInfo(paApiIndex); + + SC := 0; + + // init array-size to max. input-devices count + SetLength(AudioInputProcessor.DeviceList, paApiInfo^.deviceCount); + for i:= 0 to High(AudioInputProcessor.DeviceList) do + begin + // convert API-specific device-index to global index + deviceIndex := Pa_HostApiDeviceIndexToDeviceIndex(paApiIndex, i); + deviceInfo := Pa_GetDeviceInfo(deviceIndex); + + channelCnt := deviceInfo^.maxInputChannels; + + // current device is no input device -> skip + if (channelCnt <= 0) then + continue; + + // portaudio returns a channel-count of 128 for some devices + // (e.g. the "default"-device), so we have to detect those + // fantasy channel counts. + if (channelCnt > 8) then + channelCnt := 2; + + paDevice := TPortaudioInputDevice.Create(); + AudioInputProcessor.DeviceList[SC] := paDevice; + + // retrieve device-name + deviceName := deviceInfo^.name; + paDevice.Name := deviceName; + paDevice.PaDeviceIndex := deviceIndex; + + sampleRate := deviceInfo^.defaultSampleRate; + + // 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 + // TODO: retry with input-latency set to 20ms (defaultLowInputLatency might + // not be set correctly in OSS) + with inputParams do + begin + device := deviceIndex; + channelCount := channelCnt; + sampleFormat := paInt16; + suggestedLatency := latency; + hostApiSpecificStreamInfo := nil; + end; + + // check souncard and adjust sample-rate + if (not AudioCore.TestDevice(@inputParams, nil, sampleRate)) then + begin + // ignore device if it does not work + Log.LogError('Device "'+paDevice.Name+'" does not work', + 'TAudioInput_Portaudio.EnumDevices'); + paDevice.Free(); + continue; + end; + + // open device for further info + err := Pa_OpenStream(stream, @inputParams, nil, sampleRate, + paFramesPerBufferUnspecified, paNoFlag, @MicrophoneTestCallback, nil); + if(err <> paNoError) then + begin + // unable to open device -> skip + errMsg := Pa_GetErrorText(err); + Log.LogError('Device error: "'+ deviceName +'" ('+ errMsg +')', + 'TAudioInput_Portaudio.EnumDevices'); + paDevice.Free(); + continue; + end; + + // adjust sample-rate (might be changed by portaudio) + streamInfo := Pa_GetStreamInfo(stream); + if (streamInfo <> nil) then + begin + if (sampleRate <> streamInfo^.sampleRate) then + begin + Log.LogStatus('Portaudio changed Samplerate from ' + FloatToStr(sampleRate) + + ' to ' + FloatToStr(streamInfo^.sampleRate), + 'TAudioInput_Portaudio.InitializeRecord'); + sampleRate := streamInfo^.sampleRate; + end; + end; + + // create audio-format info and resize capture-buffer array + paDevice.AudioFormat := TAudioFormatInfo.Create( + channelCnt, + sampleRate, + asfS16 + ); + SetLength(paDevice.CaptureChannel, paDevice.AudioFormat.Channels); + + Log.LogStatus('InputDevice "'+paDevice.Name+'"@' + + IntToStr(paDevice.AudioFormat.Channels)+'x'+ + FloatToStr(paDevice.AudioFormat.SampleRate)+'Hz ('+ + FloatTostr(inputParams.suggestedLatency)+'sec)' , + 'Portaudio.EnumDevices'); + + // portaudio does not provide a source-type check + paDevice.MicSource := -1; + paDevice.SourceRestore := -1; + + // add a virtual default source (will not change mixer-settings) + SetLength(paDevice.Source, 1); + paDevice.Source[0].Name := DEFAULT_SOURCE_NAME; + + {$IFDEF UsePortmixer} + // use default mixer + mixer := Px_OpenMixer(stream, 0); + + // get input count + sourceCnt := Px_GetNumInputSources(mixer); + SetLength(paDevice.Source, sourceCnt+1); + + // get input names + for sourceIndex := 1 to sourceCnt do + begin + sourceName := Px_GetInputSourceName(mixer, sourceIndex-1); + paDevice.Source[sourceIndex].Name := sourceName; + end; + + Px_CloseMixer(mixer); + {$ENDIF} + + // close test-stream + Pa_CloseStream(stream); + + Inc(SC); + end; + + // adjust size to actual input-device count + SetLength(AudioInputProcessor.DeviceList, SC); + + Log.LogStatus('#Input-Devices: ' + inttostr(SC), 'Portaudio'); + + Result := true; +end; + +function TAudioInput_Portaudio.InitializeRecord(): boolean; +var + err: TPaError; +begin + AudioCore := TAudioCore_Portaudio.GetInstance(); + + // initialize portaudio + err := Pa_Initialize(); + if(err <> paNoError) then + begin + Log.LogError(Pa_GetErrorText(err), 'TAudioInput_Portaudio.InitializeRecord'); + Result := false; + Exit; + end; + + Result := EnumDevices(); +end; + +function TAudioInput_Portaudio.FinalizeRecord: boolean; +begin + CaptureStop; + Pa_Terminate(); + Result := inherited FinalizeRecord(); +end; + +{* + * Portaudio input capture callback. + *} +function MicrophoneCallback(input: Pointer; output: Pointer; frameCount: Longword; + timeInfo: PPaStreamCallbackTimeInfo; statusFlags: TPaStreamCallbackFlags; + inputDevice: Pointer): Integer; cdecl; +begin + AudioInputProcessor.HandleMicrophoneData(input, frameCount*4, inputDevice); + result := paContinue; +end; + +{* + * Portaudio test capture callback. + *} +function MicrophoneTestCallback(input: Pointer; output: Pointer; frameCount: Longword; + timeInfo: PPaStreamCallbackTimeInfo; statusFlags: TPaStreamCallbackFlags; + inputDevice: Pointer): Integer; cdecl; +begin + // this callback is called only once + result := paAbort; +end; + + +initialization + MediaManager.add(TAudioInput_Portaudio.Create); + +end. diff --git a/src/media/UAudioPlaybackBase.pas b/src/media/UAudioPlaybackBase.pas new file mode 100644 index 00000000..2337d43f --- /dev/null +++ b/src/media/UAudioPlaybackBase.pas @@ -0,0 +1,292 @@ +unit UAudioPlaybackBase; + +interface + +{$IFDEF FPC} + {$MODE Delphi} +{$ENDIF} + +{$I switches.inc} + +uses + UMusic; + +type + TAudioPlaybackBase = class(TInterfacedObject, IAudioPlayback) + protected + OutputDeviceList: TAudioOutputDeviceList; + MusicStream: TAudioPlaybackStream; + function CreatePlaybackStream(): TAudioPlaybackStream; virtual; abstract; + procedure ClearOutputDeviceList(); + function GetLatency(): double; virtual; abstract; + + // open sound or music stream (used by Open() and OpenSound()) + function OpenStream(const Filename: string): TAudioPlaybackStream; + function OpenDecodeStream(const Filename: string): TAudioDecodeStream; + public + function GetName: string; virtual; abstract; + + function Open(const Filename: string): boolean; // true if succeed + procedure Close; + + procedure Play; + procedure Pause; + procedure Stop; + procedure FadeIn(Time: real; TargetVolume: single); + + procedure SetSyncSource(SyncSource: TSyncSource); + + procedure SetPosition(Time: real); + function GetPosition: real; + + function InitializePlayback: boolean; virtual; abstract; + function FinalizePlayback: boolean; virtual; + + //function SetOutputDevice(Device: TAudioOutputDevice): boolean; + function GetOutputDeviceList(): TAudioOutputDeviceList; + + procedure SetAppVolume(Volume: single); virtual; abstract; + procedure SetVolume(Volume: single); + procedure SetLoop(Enabled: boolean); + + procedure Rewind; + function Finished: boolean; + function Length: real; + + // Sounds + function OpenSound(const Filename: string): TAudioPlaybackStream; + procedure PlaySound(Stream: TAudioPlaybackStream); + procedure StopSound(Stream: TAudioPlaybackStream); + + // Equalizer + procedure GetFFTData(var Data: TFFTData); + + // Interface for Visualizer + function GetPCMData(var Data: TPCMData): Cardinal; + + function CreateVoiceStream(Channel: integer; FormatInfo: TAudioFormatInfo): TAudioVoiceStream; virtual; abstract; + end; + + +implementation + +uses + ULog, + SysUtils; + +{ TAudioPlaybackBase } + +function TAudioPlaybackBase.FinalizePlayback: boolean; +begin + FreeAndNil(MusicStream); + ClearOutputDeviceList(); + Result := true; +end; + +function TAudioPlaybackBase.Open(const Filename: string): boolean; +begin + // free old MusicStream + MusicStream.Free; + + MusicStream := OpenStream(Filename); + if not assigned(MusicStream) then + begin + Result := false; + Exit; + end; + + //MusicStream.AddSoundEffect(TVoiceRemoval.Create()); + + Result := true; +end; + +procedure TAudioPlaybackBase.Close; +begin + FreeAndNil(MusicStream); +end; + +function TAudioPlaybackBase.OpenDecodeStream(const Filename: String): TAudioDecodeStream; +var + i: integer; +begin + for i := 0 to AudioDecoders.Count-1 do + begin + Result := IAudioDecoder(AudioDecoders[i]).Open(Filename); + if (assigned(Result)) then + begin + Log.LogInfo('Using decoder ' + IAudioDecoder(AudioDecoders[i]).GetName() + + ' for "' + Filename + '"', 'TAudioPlaybackBase.OpenDecodeStream'); + Exit; + end; + end; + Result := nil; +end; + +procedure OnClosePlaybackStream(Stream: TAudioProcessingStream); +var + PlaybackStream: TAudioPlaybackStream; + SourceStream: TAudioSourceStream; +begin + PlaybackStream := TAudioPlaybackStream(Stream); + SourceStream := PlaybackStream.GetSourceStream(); + SourceStream.Free; +end; + +function TAudioPlaybackBase.OpenStream(const Filename: string): TAudioPlaybackStream; +var + PlaybackStream: TAudioPlaybackStream; + DecodeStream: TAudioDecodeStream; +begin + Result := nil; + + //Log.LogStatus('Loading Sound: "' + Filename + '"', 'TAudioPlayback_Bass.OpenStream'); + + DecodeStream := OpenDecodeStream(Filename); + if (not assigned(DecodeStream)) then + begin + Log.LogStatus('Could not open "' + Filename + '"', 'TAudioPlayback_Bass.OpenStream'); + Exit; + end; + + // create a matching playback-stream for the decoder + PlaybackStream := CreatePlaybackStream(); + if (not PlaybackStream.Open(DecodeStream)) then + begin + FreeAndNil(PlaybackStream); + FreeAndNil(DecodeStream); + Exit; + end; + + PlaybackStream.AddOnCloseHandler(OnClosePlaybackStream); + + Result := PlaybackStream; +end; + +procedure TAudioPlaybackBase.Play; +begin + if assigned(MusicStream) then + MusicStream.Play(); +end; + +procedure TAudioPlaybackBase.Pause; +begin + if assigned(MusicStream) then + MusicStream.Pause(); +end; + +procedure TAudioPlaybackBase.Stop; +begin + if assigned(MusicStream) then + MusicStream.Stop(); +end; + +function TAudioPlaybackBase.Length: real; +begin + if assigned(MusicStream) then + Result := MusicStream.Length + else + Result := 0; +end; + +function TAudioPlaybackBase.GetPosition: real; +begin + if assigned(MusicStream) then + Result := MusicStream.Position + else + Result := 0; +end; + +procedure TAudioPlaybackBase.SetPosition(Time: real); +begin + if assigned(MusicStream) then + MusicStream.Position := Time; +end; + +procedure TAudioPlaybackBase.SetSyncSource(SyncSource: TSyncSource); +begin + if assigned(MusicStream) then + MusicStream.SetSyncSource(SyncSource); +end; + +procedure TAudioPlaybackBase.Rewind; +begin + SetPosition(0); +end; + +function TAudioPlaybackBase.Finished: boolean; +begin + if assigned(MusicStream) then + Result := (MusicStream.Status = ssStopped) + else + Result := true; +end; + +procedure TAudioPlaybackBase.SetVolume(Volume: single); +begin + if assigned(MusicStream) then + MusicStream.Volume := Volume; +end; + +procedure TAudioPlaybackBase.FadeIn(Time: real; TargetVolume: single); +begin + if assigned(MusicStream) then + MusicStream.FadeIn(Time, TargetVolume); +end; + +procedure TAudioPlaybackBase.SetLoop(Enabled: boolean); +begin + if assigned(MusicStream) then + MusicStream.Loop := Enabled; +end; + +// Equalizer +procedure TAudioPlaybackBase.GetFFTData(var data: TFFTData); +begin + if assigned(MusicStream) then + MusicStream.GetFFTData(data); +end; + +{* + * Copies interleaved PCM SInt16 stereo samples into data. + * Returns the number of frames + *} +function TAudioPlaybackBase.GetPCMData(var data: TPCMData): Cardinal; +begin + if assigned(MusicStream) then + Result := MusicStream.GetPCMData(data) + else + Result := 0; +end; + +function TAudioPlaybackBase.OpenSound(const Filename: string): TAudioPlaybackStream; +begin + Result := OpenStream(Filename); +end; + +procedure TAudioPlaybackBase.PlaySound(stream: TAudioPlaybackStream); +begin + if assigned(stream) then + stream.Play(); +end; + +procedure TAudioPlaybackBase.StopSound(stream: TAudioPlaybackStream); +begin + if assigned(stream) then + stream.Stop(); +end; + +procedure TAudioPlaybackBase.ClearOutputDeviceList(); +var + DeviceIndex: integer; +begin + for DeviceIndex := 0 to High(OutputDeviceList) do + OutputDeviceList[DeviceIndex].Free(); + SetLength(OutputDeviceList, 0); +end; + +function TAudioPlaybackBase.GetOutputDeviceList(): TAudioOutputDeviceList; +begin + Result := OutputDeviceList; +end; + +end. diff --git a/src/media/UAudioPlayback_Bass.pas b/src/media/UAudioPlayback_Bass.pas new file mode 100644 index 00000000..41a91173 --- /dev/null +++ b/src/media/UAudioPlayback_Bass.pas @@ -0,0 +1,731 @@ +unit UAudioPlayback_Bass; + +interface + +{$IFDEF FPC} + {$MODE Delphi} +{$ENDIF} + +{$I switches.inc} + +implementation + +uses + Classes, + SysUtils, + Math, + UIni, + UMain, + UMusic, + UAudioPlaybackBase, + UAudioCore_Bass, + ULog, + sdl, + bass; + +type + PHDSP = ^HDSP; + +type + TBassPlaybackStream = class(TAudioPlaybackStream) + private + Handle: HSTREAM; + NeedsRewind: boolean; + PausedSeek: boolean; // true if a seek was performed in pause state + + procedure Reset(); + function IsEOF(): boolean; + protected + function GetLatency(): double; override; + function GetLoop(): boolean; override; + procedure SetLoop(Enabled: boolean); override; + function GetLength(): real; override; + function GetStatus(): TStreamStatus; override; + function GetVolume(): single; override; + procedure SetVolume(Volume: single); override; + function GetPosition: real; override; + procedure SetPosition(Time: real); override; + public + constructor Create(); + destructor Destroy(); override; + + function Open(SourceStream: TAudioSourceStream): boolean; override; + procedure Close(); override; + + procedure Play(); override; + procedure Pause(); override; + procedure Stop(); override; + procedure FadeIn(Time: real; TargetVolume: single); override; + + procedure AddSoundEffect(Effect: TSoundEffect); override; + procedure RemoveSoundEffect(Effect: TSoundEffect); override; + + procedure GetFFTData(var Data: TFFTData); override; + function GetPCMData(var Data: TPCMData): Cardinal; override; + + function GetAudioFormatInfo(): TAudioFormatInfo; override; + + function ReadData(Buffer: PChar; BufferSize: integer): integer; + + property EOF: boolean READ IsEOF; + end; + +const + MAX_VOICE_DELAY = 0.020; // 20ms + +type + TBassVoiceStream = class(TAudioVoiceStream) + private + Handle: HSTREAM; + public + function Open(ChannelMap: integer; FormatInfo: TAudioFormatInfo): boolean; override; + procedure Close(); override; + + procedure WriteData(Buffer: PChar; BufferSize: integer); override; + function ReadData(Buffer: PChar; BufferSize: integer): integer; override; + function IsEOF(): boolean; override; + function IsError(): boolean; override; + end; + +type + TAudioPlayback_Bass = class(TAudioPlaybackBase) + private + function EnumDevices(): boolean; + protected + function GetLatency(): double; override; + function CreatePlaybackStream(): TAudioPlaybackStream; override; + public + function GetName: String; override; + function InitializePlayback(): boolean; override; + function FinalizePlayback: boolean; override; + procedure SetAppVolume(Volume: single); override; + function CreateVoiceStream(ChannelMap: integer; FormatInfo: TAudioFormatInfo): TAudioVoiceStream; override; + end; + + TBassOutputDevice = class(TAudioOutputDevice) + private + BassDeviceID: DWORD; // DeviceID used by BASS + end; + +var + BassCore: TAudioCore_Bass; + + +{ TBassPlaybackStream } + +function PlaybackStreamHandler(handle: HSTREAM; buffer: Pointer; length: DWORD; user: Pointer): DWORD; +{$IFDEF MSWINDOWS}stdcall;{$ELSE}cdecl;{$ENDIF} +var + PlaybackStream: TBassPlaybackStream; + BytesRead: integer; +begin + PlaybackStream := TBassPlaybackStream(user); + if (not assigned (PlaybackStream)) then + begin + Result := BASS_STREAMPROC_END; + Exit; + end; + + BytesRead := PlaybackStream.ReadData(buffer, length); + // check for errors + if (BytesRead < 0) then + Result := BASS_STREAMPROC_END + // check for EOF + else if (PlaybackStream.EOF) then + Result := BytesRead or BASS_STREAMPROC_END + // no error/EOF + else + Result := BytesRead; +end; + +function TBassPlaybackStream.ReadData(Buffer: PChar; BufferSize: integer): integer; +var + AdjustedSize: integer; + RequestedSourceSize, SourceSize: integer; + SkipCount: integer; + SourceFormatInfo: TAudioFormatInfo; + FrameSize: integer; + PadFrame: PChar; + //Info: BASS_INFO; + //Latency: double; +begin + Result := -1; + + if (not assigned(SourceStream)) then + Exit; + + // sanity check + if (BufferSize = 0) then + begin + Result := 0; + Exit; + end; + + SourceFormatInfo := SourceStream.GetAudioFormatInfo(); + FrameSize := SourceFormatInfo.FrameSize; + + // check how much data to fetch to be in synch + AdjustedSize := Synchronize(BufferSize, SourceFormatInfo); + + // skip data if we are too far behind + SkipCount := AdjustedSize - BufferSize; + while (SkipCount > 0) do + begin + RequestedSourceSize := Min(SkipCount, BufferSize); + SourceSize := SourceStream.ReadData(Buffer, RequestedSourceSize); + // if an error or EOF occured stop skipping and handle error/EOF with the next ReadData() + if (SourceSize <= 0) then + break; + Dec(SkipCount, SourceSize); + end; + + // get source data (e.g. from a decoder) + RequestedSourceSize := Min(AdjustedSize, BufferSize); + SourceSize := SourceStream.ReadData(Buffer, RequestedSourceSize); + if (SourceSize < 0) then + Exit; + + // set preliminary result + Result := SourceSize; + + // if we are to far ahead, fill output-buffer with last frame of source data + // Note that AdjustedSize is used instead of SourceSize as the SourceSize might + // be less than expected because of errors etc. + if (AdjustedSize < BufferSize) then + begin + // use either the last frame for padding or fill with zero + if (SourceSize >= FrameSize) then + PadFrame := @Buffer[SourceSize-FrameSize] + else + PadFrame := nil; + + FillBufferWithFrame(@Buffer[SourceSize], BufferSize - SourceSize, + PadFrame, FrameSize); + Result := BufferSize; + end; +end; + +constructor TBassPlaybackStream.Create(); +begin + inherited; + Reset(); +end; + +destructor TBassPlaybackStream.Destroy(); +begin + Close(); + inherited; +end; + +function TBassPlaybackStream.Open(SourceStream: TAudioSourceStream): boolean; +var + FormatInfo: TAudioFormatInfo; + FormatFlags: DWORD; +begin + Result := false; + + // close previous stream and reset state + Reset(); + + // sanity check if stream is valid + if not assigned(SourceStream) then + Exit; + + Self.SourceStream := SourceStream; + FormatInfo := SourceStream.GetAudioFormatInfo(); + if (not BassCore.ConvertAudioFormatToBASSFlags(FormatInfo.Format, FormatFlags)) then + begin + Log.LogError('Unhandled sample-format', 'TBassPlaybackStream.Open'); + Exit; + end; + + // create matching playback stream + Handle := BASS_StreamCreate(Round(FormatInfo.SampleRate), FormatInfo.Channels, formatFlags, + @PlaybackStreamHandler, Self); + if (Handle = 0) then + begin + Log.LogError('BASS_StreamCreate failed: ' + BassCore.ErrorGetString(BASS_ErrorGetCode()), + 'TBassPlaybackStream.Open'); + Exit; + end; + + Result := true; +end; + +procedure TBassPlaybackStream.Close(); +begin + // stop and free stream + if (Handle <> 0) then + begin + Bass_StreamFree(Handle); + Handle := 0; + end; + + // Note: PerformOnClose must be called before SourceStream is invalidated + PerformOnClose(); + // unset source-stream + SourceStream := nil; +end; + +procedure TBassPlaybackStream.Reset(); +begin + Close(); + NeedsRewind := false; + PausedSeek := false; +end; + +procedure TBassPlaybackStream.Play(); +var + NeedsFlush: boolean; +begin + if (not assigned(SourceStream)) then + Exit; + + NeedsFlush := true; + + if (BASS_ChannelIsActive(Handle) = BASS_ACTIVE_PAUSED) then + begin + // only paused (and not seeked while paused) streams are not flushed + if (not PausedSeek) then + NeedsFlush := false; + // paused streams do not need a rewind + NeedsRewind := false; + end; + + // rewind if necessary. Cases that require no rewind are: + // - stream was created and never played + // - stream was paused and is resumed now + // - stream was stopped and set to a new position already + if (NeedsRewind) then + SourceStream.Position := 0; + + NeedsRewind := true; + PausedSeek := false; + + // start playing and flush buffers on rewind + BASS_ChannelPlay(Handle, NeedsFlush); +end; + +procedure TBassPlaybackStream.FadeIn(Time: real; TargetVolume: single); +begin + // start stream + Play(); + // start fade-in: slide from fadeStart- to fadeEnd-volume in FadeInTime + BASS_ChannelSlideAttribute(Handle, BASS_ATTRIB_VOL, TargetVolume, Trunc(Time * 1000)); +end; + +procedure TBassPlaybackStream.Pause(); +begin + BASS_ChannelPause(Handle); +end; + +procedure TBassPlaybackStream.Stop(); +begin + BASS_ChannelStop(Handle); +end; + +function TBassPlaybackStream.IsEOF(): boolean; +begin + if (assigned(SourceStream)) then + Result := SourceStream.EOF + else + Result := true; +end; + +function TBassPlaybackStream.GetLatency(): double; +begin + // TODO: should we consider output latency for synching (needs BASS_DEVICE_LATENCY)? + //if (BASS_GetInfo(Info)) then + // Latency := Info.latency / 1000 + //else + // Latency := 0; + Result := 0; +end; + +function TBassPlaybackStream.GetVolume(): single; +var + lVolume: single; +begin + if (not BASS_ChannelGetAttribute(Handle, BASS_ATTRIB_VOL, lVolume)) then + begin + Log.LogError('BASS_ChannelGetAttribute: ' + BassCore.ErrorGetString(), + 'TBassPlaybackStream.GetVolume'); + Result := 0; + Exit; + end; + Result := Round(lVolume); +end; + +procedure TBassPlaybackStream.SetVolume(Volume: single); +begin + // clamp volume + if Volume < 0 then + Volume := 0; + if Volume > 1.0 then + Volume := 1.0; + // set volume + BASS_ChannelSetAttribute(Handle, BASS_ATTRIB_VOL, Volume); +end; + +function TBassPlaybackStream.GetPosition: real; +var + BufferPosByte: QWORD; + BufferPosSec: double; +begin + if assigned(SourceStream) then + begin + BufferPosByte := BASS_ChannelGetData(Handle, nil, BASS_DATA_AVAILABLE); + BufferPosSec := BASS_ChannelBytes2Seconds(Handle, BufferPosByte); + // decrease the decoding position by the amount buffered (and hence not played) + // in the BASS playback stream. + Result := SourceStream.Position - BufferPosSec; + end + else + begin + Result := -1; + end; +end; + +procedure TBassPlaybackStream.SetPosition(Time: real); +var + ChannelState: DWORD; +begin + if assigned(SourceStream) then + begin + ChannelState := BASS_ChannelIsActive(Handle); + if (ChannelState = BASS_ACTIVE_STOPPED) then + begin + // if the stream is stopped, do not rewind when the stream is played next time + NeedsRewind := false + end + else if (ChannelState = BASS_ACTIVE_PAUSED) then + begin + // buffers must be flushed if in paused state but there is no + // BASS_ChannelFlush() function so we have to use BASS_ChannelPlay() called in Play(). + PausedSeek := true; + end; + + // set new position + SourceStream.Position := Time; + end; +end; + +function TBassPlaybackStream.GetLength(): real; +begin + if assigned(SourceStream) then + Result := SourceStream.Length + else + Result := -1; +end; + +function TBassPlaybackStream.GetStatus(): TStreamStatus; +var + State: DWORD; +begin + State := BASS_ChannelIsActive(Handle); + case State of + BASS_ACTIVE_PLAYING, + BASS_ACTIVE_STALLED: + Result := ssPlaying; + BASS_ACTIVE_PAUSED: + Result := ssPaused; + BASS_ACTIVE_STOPPED: + Result := ssStopped; + else + begin + Log.LogError('Unknown status', 'TBassPlaybackStream.GetStatus'); + Result := ssStopped; + end; + end; +end; + +function TBassPlaybackStream.GetLoop(): boolean; +begin + if assigned(SourceStream) then + Result := SourceStream.Loop + else + Result := false; +end; + +procedure TBassPlaybackStream.SetLoop(Enabled: boolean); +begin + if assigned(SourceStream) then + SourceStream.Loop := Enabled; +end; + +procedure DSPProcHandler(handle: HDSP; channel: DWORD; buffer: Pointer; length: DWORD; user: Pointer); +{$IFDEF MSWINDOWS}stdcall;{$ELSE}cdecl;{$ENDIF} +var + Effect: TSoundEffect; +begin + Effect := TSoundEffect(user); + if assigned(Effect) then + Effect.Callback(buffer, length); +end; + +procedure TBassPlaybackStream.AddSoundEffect(Effect: TSoundEffect); +var + DspHandle: HDSP; +begin + if assigned(Effect.engineData) then + begin + Log.LogError('TSoundEffect.engineData already set', 'TBassPlaybackStream.AddSoundEffect'); + Exit; + end; + + DspHandle := BASS_ChannelSetDSP(Handle, @DSPProcHandler, Effect, 0); + if (DspHandle = 0) then + begin + Log.LogError(BassCore.ErrorGetString(), 'TBassPlaybackStream.AddSoundEffect'); + Exit; + end; + + GetMem(Effect.EngineData, SizeOf(HDSP)); + PHDSP(Effect.EngineData)^ := DspHandle; +end; + +procedure TBassPlaybackStream.RemoveSoundEffect(Effect: TSoundEffect); +begin + if not assigned(Effect.EngineData) then + begin + Log.LogError('TSoundEffect.engineData invalid', 'TBassPlaybackStream.RemoveSoundEffect'); + Exit; + end; + + if not BASS_ChannelRemoveDSP(Handle, PHDSP(Effect.EngineData)^) then + begin + Log.LogError(BassCore.ErrorGetString(), 'TBassPlaybackStream.RemoveSoundEffect'); + Exit; + end; + + FreeMem(Effect.EngineData); + Effect.EngineData := nil; +end; + +procedure TBassPlaybackStream.GetFFTData(var Data: TFFTData); +begin + // get FFT channel data (Mono, FFT512 -> 256 values) + BASS_ChannelGetData(Handle, @Data, BASS_DATA_FFT512); +end; + +{* + * Copies interleaved PCM SInt16 stereo samples into data. + * Returns the number of frames + *} +function TBassPlaybackStream.GetPCMData(var Data: TPCMData): Cardinal; +var + Info: BASS_CHANNELINFO; + nBytes: DWORD; +begin + Result := 0; + + FillChar(Data, SizeOf(TPCMData), 0); + + // no support for non-stereo files at the moment + BASS_ChannelGetInfo(Handle, Info); + if (Info.chans <> 2) then + Exit; + + nBytes := BASS_ChannelGetData(Handle, @Data, SizeOf(TPCMData)); + if(nBytes <= 0) then + Result := 0 + else + Result := nBytes div SizeOf(TPCMStereoSample); +end; + +function TBassPlaybackStream.GetAudioFormatInfo(): TAudioFormatInfo; +begin + if assigned(SourceStream) then + Result := SourceStream.GetAudioFormatInfo() + else + Result := nil; +end; + + +{ TBassVoiceStream } + +function TBassVoiceStream.Open(ChannelMap: integer; FormatInfo: TAudioFormatInfo): boolean; +var + Flags: DWORD; +begin + Result := false; + + Close(); + + if (not inherited Open(ChannelMap, FormatInfo)) then + Exit; + + // get channel flags + BassCore.ConvertAudioFormatToBASSFlags(FormatInfo.Format, Flags); + + (* + // distribute the mics equally to both speakers + if ((ChannelMap and CHANNELMAP_LEFT) <> 0) then + Flags := Flags or BASS_SPEAKER_FRONTLEFT; + if ((ChannelMap and CHANNELMAP_RIGHT) <> 0) then + Flags := Flags or BASS_SPEAKER_FRONTRIGHT; + *) + + // create the channel + Handle := BASS_StreamCreate(Round(FormatInfo.SampleRate), 1, Flags, STREAMPROC_PUSH, nil); + + // start the channel + BASS_ChannelPlay(Handle, true); + + Result := true; +end; + +procedure TBassVoiceStream.Close(); +begin + if (Handle <> 0) then + begin + BASS_ChannelStop(Handle); + BASS_StreamFree(Handle); + end; + inherited Close(); +end; + +procedure TBassVoiceStream.WriteData(Buffer: PChar; BufferSize: integer); +var QueueSize: DWORD; +begin + if ((Handle <> 0) and (BufferSize > 0)) then + begin + // query the queue size (normally 0) + QueueSize := BASS_StreamPutData(Handle, nil, 0); + // flush the buffer if the delay would be too high + if (QueueSize > MAX_VOICE_DELAY * FormatInfo.BytesPerSec) then + BASS_ChannelPlay(Handle, true); + // send new data to playback buffer + BASS_StreamPutData(Handle, Buffer, BufferSize); + end; +end; + +// Note: we do not need the read-function for the BASS implementation +function TBassVoiceStream.ReadData(Buffer: PChar; BufferSize: integer): integer; +begin + Result := -1; +end; + +function TBassVoiceStream.IsEOF(): boolean; +begin + Result := false; +end; + +function TBassVoiceStream.IsError(): boolean; +begin + Result := false; +end; + + +{ TAudioPlayback_Bass } + +function TAudioPlayback_Bass.GetName: String; +begin + Result := 'BASS_Playback'; +end; + +function TAudioPlayback_Bass.EnumDevices(): boolean; +var + BassDeviceID: DWORD; + DeviceIndex: integer; + Device: TBassOutputDevice; + DeviceInfo: BASS_DEVICEINFO; +begin + Result := true; + + ClearOutputDeviceList(); + + // skip "no sound"-device (ID = 0) + BassDeviceID := 1; + + while (true) do + begin + // check for device + if (not BASS_GetDeviceInfo(BassDeviceID, DeviceInfo)) then + Break; + + // set device info + Device := TBassOutputDevice.Create(); + Device.Name := DeviceInfo.name; + Device.BassDeviceID := BassDeviceID; + + // add device to list + SetLength(OutputDeviceList, BassDeviceID); + OutputDeviceList[BassDeviceID-1] := Device; + + Inc(BassDeviceID); + end; +end; + +function TAudioPlayback_Bass.InitializePlayback(): boolean; +begin + result := false; + + BassCore := TAudioCore_Bass.GetInstance(); + + EnumDevices(); + + //Log.BenchmarkStart(4); + //Log.LogStatus('Initializing Playback Subsystem', 'Music Initialize'); + + // TODO: use BASS_DEVICE_LATENCY to determine the latency + if not BASS_Init(-1, 44100, 0, 0, nil) then + begin + Log.LogError('Could not initialize BASS', 'TAudioPlayback_Bass.InitializePlayback'); + Exit; + end; + + //Log.BenchmarkEnd(4); Log.LogBenchmark('--> Bass Init', 4); + + // config playing buffer + //BASS_SetConfig(BASS_CONFIG_UPDATEPERIOD, 10); + //BASS_SetConfig(BASS_CONFIG_BUFFER, 100); + + result := true; +end; + +function TAudioPlayback_Bass.FinalizePlayback(): boolean; +begin + Close; + BASS_Free; + inherited FinalizePlayback(); + Result := true; +end; + +function TAudioPlayback_Bass.CreatePlaybackStream(): TAudioPlaybackStream; +begin + Result := TBassPlaybackStream.Create(); +end; + +procedure TAudioPlayback_Bass.SetAppVolume(Volume: single); +begin + // set volume for this application (ranges from 0..10000 since BASS 2.4) + BASS_SetConfig(BASS_CONFIG_GVOL_STREAM, Round(Volume*10000)); +end; + +function TAudioPlayback_Bass.CreateVoiceStream(ChannelMap: integer; FormatInfo: TAudioFormatInfo): TAudioVoiceStream; +var + VoiceStream: TAudioVoiceStream; +begin + Result := nil; + + VoiceStream := TBassVoiceStream.Create(); + if (not VoiceStream.Open(ChannelMap, FormatInfo)) then + begin + VoiceStream.Free; + Exit; + end; + + Result := VoiceStream; +end; + +function TAudioPlayback_Bass.GetLatency(): double; +begin + Result := 0; +end; + + +initialization + MediaManager.Add(TAudioPlayback_Bass.Create); + +end. diff --git a/src/media/UAudioPlayback_Portaudio.pas b/src/media/UAudioPlayback_Portaudio.pas new file mode 100644 index 00000000..c3717ba6 --- /dev/null +++ b/src/media/UAudioPlayback_Portaudio.pas @@ -0,0 +1,361 @@ +unit UAudioPlayback_Portaudio; + +interface + +{$IFDEF FPC} + {$MODE Delphi} +{$ENDIF} + +{$I switches.inc} + + +uses + Classes, + SysUtils, + UMusic; + +implementation + +uses + portaudio, + UAudioCore_Portaudio, + UAudioPlayback_SoftMixer, + ULog, + UIni, + UMain; + +type + TAudioPlayback_Portaudio = class(TAudioPlayback_SoftMixer) + private + paStream: PPaStream; + AudioCore: TAudioCore_Portaudio; + Latency: double; + function OpenDevice(deviceIndex: TPaDeviceIndex): boolean; + function EnumDevices(): boolean; + protected + function InitializeAudioPlaybackEngine(): boolean; override; + function StartAudioPlaybackEngine(): boolean; override; + procedure StopAudioPlaybackEngine(); override; + function FinalizeAudioPlaybackEngine(): boolean; override; + function GetLatency(): double; override; + public + function GetName: String; override; + end; + + TPortaudioOutputDevice = class(TAudioOutputDevice) + private + PaDeviceIndex: TPaDeviceIndex; + end; + + +{ TAudioPlayback_Portaudio } + +function PortaudioAudioCallback(input: Pointer; output: Pointer; frameCount: Longword; + timeInfo: PPaStreamCallbackTimeInfo; statusFlags: TPaStreamCallbackFlags; + userData: Pointer): Integer; cdecl; +var + Engine: TAudioPlayback_Portaudio; +begin + Engine := TAudioPlayback_Portaudio(userData); + // update latency + Engine.Latency := timeInfo.outputBufferDacTime - timeInfo.currentTime; + // call superclass callback + Engine.AudioCallback(output, frameCount * Engine.FormatInfo.FrameSize); + Result := paContinue; +end; + +function TAudioPlayback_Portaudio.GetName: String; +begin + Result := 'Portaudio_Playback'; +end; + +function TAudioPlayback_Portaudio.OpenDevice(deviceIndex: TPaDeviceIndex): boolean; +var + DeviceInfo : PPaDeviceInfo; + SampleRate : double; + OutParams : TPaStreamParameters; + StreamInfo : PPaStreamInfo; + err : TPaError; +begin + Result := false; + + DeviceInfo := Pa_GetDeviceInfo(deviceIndex); + + Log.LogInfo('Audio-Output Device: ' + DeviceInfo^.name, 'TAudioPlayback_Portaudio.OpenDevice'); + + SampleRate := DeviceInfo^.defaultSampleRate; + + with OutParams do + begin + device := deviceIndex; + channelCount := 2; + sampleFormat := paInt16; + suggestedLatency := DeviceInfo^.defaultLowOutputLatency; + hostApiSpecificStreamInfo := nil; + end; + + // check souncard and adjust sample-rate + if not AudioCore.TestDevice(nil, @OutParams, SampleRate) then + begin + Log.LogStatus('TestDevice failed!', 'TAudioPlayback_Portaudio.OpenDevice'); + Exit; + end; + + // open output stream + err := Pa_OpenStream(paStream, nil, @OutParams, SampleRate, + paFramesPerBufferUnspecified, + paNoFlag, @PortaudioAudioCallback, Self); + if(err <> paNoError) then + begin + Log.LogStatus(Pa_GetErrorText(err), 'TAudioPlayback_Portaudio.OpenDevice'); + paStream := nil; + Exit; + end; + + // get estimated latency (will be updated with real latency in the callback) + StreamInfo := Pa_GetStreamInfo(paStream); + if (StreamInfo <> nil) then + Latency := StreamInfo^.outputLatency + else + Latency := 0; + + FormatInfo := TAudioFormatInfo.Create( + OutParams.channelCount, + SampleRate, + asfS16 // FIXME: is paInt16 system-dependant or -independant? + ); + + Result := true; +end; + +function TAudioPlayback_Portaudio.EnumDevices(): boolean; +var + i: integer; + paApiIndex: TPaHostApiIndex; + paApiInfo: PPaHostApiInfo; + deviceName: string; + deviceIndex: TPaDeviceIndex; + deviceInfo: PPaDeviceInfo; + channelCnt: integer; + SC: integer; // soundcard + err: TPaError; + errMsg: string; + paDevice: TPortaudioOutputDevice; + outputParams: TPaStreamParameters; + stream: PPaStream; + streamInfo: PPaStreamInfo; + sampleRate: double; + latency: TPaTime; + cbPolls: integer; + cbWorks: boolean; +begin + Result := false; + +(* + // choose the best available Audio-API + paApiIndex := AudioCore.GetPreferredApiIndex(); + if(paApiIndex = -1) then + begin + Log.LogError('No working Audio-API found', 'TAudioPlayback_Portaudio.EnumDevices'); + Exit; + end; + + paApiInfo := Pa_GetHostApiInfo(paApiIndex); + + SC := 0; + + // init array-size to max. output-devices count + SetLength(OutputDeviceList, paApiInfo^.deviceCount); + for i:= 0 to High(OutputDeviceList) do + begin + // convert API-specific device-index to global index + deviceIndex := Pa_HostApiDeviceIndexToDeviceIndex(paApiIndex, i); + deviceInfo := Pa_GetDeviceInfo(deviceIndex); + + channelCnt := deviceInfo^.maxOutputChannels; + + // current device is no output device -> skip + if (channelCnt <= 0) then + continue; + + // portaudio returns a channel-count of 128 for some devices + // (e.g. the "default"-device), so we have to detect those + // fantasy channel counts. + if (channelCnt > 8) then + channelCnt := 2; + + paDevice := TPortaudioOutputDevice.Create(); + OutputDeviceList[SC] := paDevice; + + // retrieve device-name + deviceName := deviceInfo^.name; + paDevice.Name := deviceName; + paDevice.PaDeviceIndex := deviceIndex; + + if (deviceInfo^.defaultSampleRate > 0) then + sampleRate := deviceInfo^.defaultSampleRate + 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 output parameters + // TODO: retry with input-latency set to 20ms (defaultLowOutputLatency might + // not be set correctly in OSS) + with outputParams do + begin + device := deviceIndex; + channelCount := channelCnt; + sampleFormat := paInt16; + suggestedLatency := latency; + hostApiSpecificStreamInfo := nil; + end; + + // check if mic-callback works (might not be called on some devices) + if (not TAudioCore_Portaudio.TestDevice(nil, @outputParams, sampleRate)) then + begin + // ignore device if callback did not work + Log.LogError('Device "'+paDevice.Name+'" does not respond', + 'TAudioPlayback_Portaudio.InitializeRecord'); + paDevice.Free(); + continue; + end; + + // open device for further info + err := Pa_OpenStream(stream, nil, @outputParams, sampleRate, + paFramesPerBufferUnspecified, paNoFlag, @MicrophoneTestCallback, nil); + if(err <> paNoError) then + begin + // unable to open device -> skip + errMsg := Pa_GetErrorText(err); + Log.LogError('Device error: "'+ deviceName +'" ('+ errMsg +')', + 'TAudioPlayback_Portaudio.InitializeRecord'); + paDevice.Free(); + continue; + end; + + // adjust sample-rate (might be changed by portaudio) + streamInfo := Pa_GetStreamInfo(stream); + if (streamInfo <> nil) then + begin + if (sampleRate <> streamInfo^.sampleRate) then + begin + Log.LogStatus('Portaudio changed Samplerate from ' + FloatToStr(sampleRate) + + ' to ' + FloatToStr(streamInfo^.sampleRate), + 'TAudioInput_Portaudio.InitializeRecord'); + sampleRate := streamInfo^.sampleRate; + end; + end; + + // create audio-format info and resize capture-buffer array + paDevice.AudioFormat := TAudioFormatInfo.Create( + channelCnt, + sampleRate, + asfS16 + ); + SetLength(paDevice.CaptureChannel, paDevice.AudioFormat.Channels); + + Log.LogStatus('OutputDevice "'+paDevice.Name+'"@' + + IntToStr(paDevice.AudioFormat.Channels)+'x'+ + FloatToStr(paDevice.AudioFormat.SampleRate)+'Hz ('+ + FloatTostr(outputParams.suggestedLatency)+'sec)' , + 'TAudioInput_Portaudio.InitializeRecord'); + + // close test-stream + Pa_CloseStream(stream); + + Inc(SC); + end; + + // adjust size to actual input-device count + SetLength(OutputDeviceList, SC); + + Log.LogStatus('#Output-Devices: ' + inttostr(SC), 'Portaudio'); +*) + + Result := true; +end; + +function TAudioPlayback_Portaudio.InitializeAudioPlaybackEngine(): boolean; +var + paApiIndex : TPaHostApiIndex; + paApiInfo : PPaHostApiInfo; + paOutDevice : TPaDeviceIndex; + err: TPaError; +begin + Result := false; + + AudioCore := TAudioCore_Portaudio.GetInstance(); + + // initialize portaudio + err := Pa_Initialize(); + if(err <> paNoError) then + begin + Log.LogError(Pa_GetErrorText(err), 'TAudioInput_Portaudio.InitializeRecord'); + Exit; + end; + + paApiIndex := AudioCore.GetPreferredApiIndex(); + if(paApiIndex = -1) then + begin + Log.LogError('No working Audio-API found', 'TAudioPlayback_Portaudio.InitializeAudioPlaybackEngine'); + Exit; + end; + + EnumDevices(); + + paApiInfo := Pa_GetHostApiInfo(paApiIndex); + Log.LogInfo('Audio-Output API-Type: ' + paApiInfo^.name, 'TAudioPlayback_Portaudio.OpenDevice'); + + paOutDevice := paApiInfo^.defaultOutputDevice; + if (not OpenDevice(paOutDevice)) then + begin + Exit; + end; + + Result := true; +end; + +function TAudioPlayback_Portaudio.StartAudioPlaybackEngine(): boolean; +var + err: TPaError; +begin + Result := false; + + if (paStream = nil) then + Exit; + + err := Pa_StartStream(paStream); + if(err <> paNoError) then + begin + Log.LogStatus('Pa_StartStream: '+Pa_GetErrorText(err), 'UAudioPlayback_Portaudio'); + Exit; + end; + + Result := true; +end; + +procedure TAudioPlayback_Portaudio.StopAudioPlaybackEngine(); +begin + if (paStream <> nil) then + Pa_StopStream(paStream); +end; + +function TAudioPlayback_Portaudio.FinalizeAudioPlaybackEngine(): boolean; +begin + Pa_Terminate(); + Result := true; +end; + +function TAudioPlayback_Portaudio.GetLatency(): double; +begin + Result := Latency; +end; + + +initialization + MediaManager.Add(TAudioPlayback_Portaudio.Create); + +end. diff --git a/src/media/UAudioPlayback_SDL.pas b/src/media/UAudioPlayback_SDL.pas new file mode 100644 index 00000000..deef91e8 --- /dev/null +++ b/src/media/UAudioPlayback_SDL.pas @@ -0,0 +1,160 @@ +unit UAudioPlayback_SDL; + +interface + +{$IFDEF FPC} + {$MODE Delphi} +{$ENDIF} + +{$I switches.inc} + + +uses + Classes, + SysUtils, + UMusic; + +implementation + +uses + sdl, + UAudioPlayback_SoftMixer, + ULog, + UIni, + UMain; + +type + TAudioPlayback_SDL = class(TAudioPlayback_SoftMixer) + private + Latency: double; + function EnumDevices(): boolean; + protected + function InitializeAudioPlaybackEngine(): boolean; override; + function StartAudioPlaybackEngine(): boolean; override; + procedure StopAudioPlaybackEngine(); override; + function FinalizeAudioPlaybackEngine(): boolean; override; + function GetLatency(): double; override; + public + function GetName: String; override; + procedure MixBuffers(dst, src: PChar; size: Cardinal; volume: Single); override; + end; + + +{ TAudioPlayback_SDL } + +procedure SDLAudioCallback(userdata: Pointer; stream: PChar; len: integer); cdecl; +var + Engine: TAudioPlayback_SDL; +begin + Engine := TAudioPlayback_SDL(userdata); + Engine.AudioCallback(stream, len); +end; + +function TAudioPlayback_SDL.GetName: String; +begin + Result := 'SDL_Playback'; +end; + +function TAudioPlayback_SDL.EnumDevices(): boolean; +begin + // Note: SDL does not provide Device-Selection capabilities (will be introduced in 1.3) + ClearOutputDeviceList(); + SetLength(OutputDeviceList, 1); + OutputDeviceList[0] := TAudioOutputDevice.Create(); + OutputDeviceList[0].Name := '[SDL Default-Device]'; + Result := true; +end; + +function TAudioPlayback_SDL.InitializeAudioPlaybackEngine(): boolean; +var + DesiredAudioSpec, ObtainedAudioSpec: TSDL_AudioSpec; + SampleBufferSize: integer; +begin + Result := false; + + EnumDevices(); + + if (SDL_InitSubSystem(SDL_INIT_AUDIO) = -1) then + begin + Log.LogError('SDL_InitSubSystem failed!', 'TAudioPlayback_SDL.InitializeAudioPlaybackEngine'); + Exit; + end; + + SampleBufferSize := IAudioOutputBufferSizeVals[Ini.AudioOutputBufferSizeIndex]; + if (SampleBufferSize <= 0) then + begin + // Automatic setting default + // FIXME: too much glitches with 1024 samples + SampleBufferSize := 2048; //1024; + end; + + FillChar(DesiredAudioSpec, SizeOf(DesiredAudioSpec), 0); + with DesiredAudioSpec do + begin + freq := 44100; + format := AUDIO_S16SYS; + channels := 2; + samples := SampleBufferSize; + callback := @SDLAudioCallback; + userdata := Self; + end; + + // Note: always use the "obtained" parameter, otherwise SDL might try to convert + // the samples itself if the desired format is not available. This might lead + // to problems if for example ALSA does not support 44100Hz and proposes 48000Hz. + // Without the obtained parameter, SDL would try to convert 44.1kHz to 48kHz with + // its crappy (non working) converter resulting in a wrong (too high) pitch. + if(SDL_OpenAudio(@DesiredAudioSpec, @ObtainedAudioSpec) = -1) then + begin + Log.LogStatus('SDL_OpenAudio: ' + SDL_GetError(), 'TAudioPlayback_SDL.InitializeAudioPlaybackEngine'); + Exit; + end; + + FormatInfo := TAudioFormatInfo.Create( + ObtainedAudioSpec.channels, + ObtainedAudioSpec.freq, + asfS16 + ); + + // Note: SDL does not provide info of the internal buffer state. + // So we use the average buffer-size. + Latency := (ObtainedAudioSpec.samples/2) / FormatInfo.SampleRate; + + Log.LogStatus('Opened audio device', 'TAudioPlayback_SDL.InitializeAudioPlaybackEngine'); + + Result := true; +end; + +function TAudioPlayback_SDL.StartAudioPlaybackEngine(): boolean; +begin + SDL_PauseAudio(0); + Result := true; +end; + +procedure TAudioPlayback_SDL.StopAudioPlaybackEngine(); +begin + SDL_PauseAudio(1); +end; + +function TAudioPlayback_SDL.FinalizeAudioPlaybackEngine(): boolean; +begin + SDL_CloseAudio(); + SDL_QuitSubSystem(SDL_INIT_AUDIO); + Result := true; +end; + +function TAudioPlayback_SDL.GetLatency(): double; +begin + Result := Latency; +end; + +procedure TAudioPlayback_SDL.MixBuffers(dst, src: PChar; size: Cardinal; volume: Single); +begin + SDL_MixAudio(PUInt8(dst), PUInt8(src), size, Round(volume * SDL_MIX_MAXVOLUME)); +end; + + +initialization + MediaManager.add(TAudioPlayback_SDL.Create); + +end. diff --git a/src/media/UAudioPlayback_SoftMixer.pas b/src/media/UAudioPlayback_SoftMixer.pas new file mode 100644 index 00000000..6ddae980 --- /dev/null +++ b/src/media/UAudioPlayback_SoftMixer.pas @@ -0,0 +1,1132 @@ +unit UAudioPlayback_SoftMixer; + +interface + +{$IFDEF FPC} + {$MODE Delphi} +{$ENDIF} + +{$I switches.inc} + + +uses + Classes, + SysUtils, + sdl, + URingBuffer, + UMusic, + UAudioPlaybackBase; + +type + TAudioPlayback_SoftMixer = class; + + TGenericPlaybackStream = class(TAudioPlaybackStream) + private + Engine: TAudioPlayback_SoftMixer; + + SampleBuffer: PChar; + SampleBufferSize: integer; + SampleBufferCount: integer; // number of available bytes in SampleBuffer + SampleBufferPos: cardinal; + + SourceBuffer: PChar; + SourceBufferSize: integer; + SourceBufferCount: integer; // number of available bytes in SourceBuffer + + Converter: TAudioConverter; + Status: TStreamStatus; + InternalLock: PSDL_Mutex; + SoundEffects: TList; + fVolume: single; + + FadeInStartTime, FadeInTime: cardinal; + FadeInStartVolume, FadeInTargetVolume: single; + + NeedsRewind: boolean; + + procedure Reset(); + + procedure ApplySoundEffects(Buffer: PChar; BufferSize: integer); + function InitFormatConversion(): boolean; + procedure FlushBuffers(); + + procedure LockSampleBuffer(); {$IFDEF HasInline}inline;{$ENDIF} + procedure UnlockSampleBuffer(); {$IFDEF HasInline}inline;{$ENDIF} + protected + function GetLatency(): double; override; + function GetStatus(): TStreamStatus; override; + function GetVolume(): single; override; + procedure SetVolume(Volume: single); override; + function GetLength(): real; override; + function GetLoop(): boolean; override; + procedure SetLoop(Enabled: boolean); override; + function GetPosition: real; override; + procedure SetPosition(Time: real); override; + public + constructor Create(Engine: TAudioPlayback_SoftMixer); + destructor Destroy(); override; + + function Open(SourceStream: TAudioSourceStream): boolean; override; + procedure Close(); override; + + procedure Play(); override; + procedure Pause(); override; + procedure Stop(); override; + procedure FadeIn(Time: real; TargetVolume: single); override; + + function GetAudioFormatInfo(): TAudioFormatInfo; override; + + function ReadData(Buffer: PChar; BufferSize: integer): integer; + + function GetPCMData(var Data: TPCMData): Cardinal; override; + procedure GetFFTData(var Data: TFFTData); override; + + procedure AddSoundEffect(Effect: TSoundEffect); override; + procedure RemoveSoundEffect(Effect: TSoundEffect); override; + end; + + TAudioMixerStream = class + private + Engine: TAudioPlayback_SoftMixer; + + ActiveStreams: TList; + MixerBuffer: PChar; + InternalLock: PSDL_Mutex; + + AppVolume: single; + + procedure Lock(); {$IFDEF HasInline}inline;{$ENDIF} + procedure Unlock(); {$IFDEF HasInline}inline;{$ENDIF} + + function GetVolume(): single; + procedure SetVolume(Volume: single); + public + constructor Create(Engine: TAudioPlayback_SoftMixer); + destructor Destroy(); override; + procedure AddStream(Stream: TAudioPlaybackStream); + procedure RemoveStream(Stream: TAudioPlaybackStream); + function ReadData(Buffer: PChar; BufferSize: integer): integer; + + property Volume: single read GetVolume write SetVolume; + end; + + TAudioPlayback_SoftMixer = class(TAudioPlaybackBase) + private + MixerStream: TAudioMixerStream; + protected + FormatInfo: TAudioFormatInfo; + + function InitializeAudioPlaybackEngine(): boolean; virtual; abstract; + function StartAudioPlaybackEngine(): boolean; virtual; abstract; + procedure StopAudioPlaybackEngine(); virtual; abstract; + function FinalizeAudioPlaybackEngine(): boolean; virtual; abstract; + procedure AudioCallback(Buffer: PChar; Size: integer); {$IFDEF HasInline}inline;{$ENDIF} + + function CreatePlaybackStream(): TAudioPlaybackStream; override; + public + function GetName: String; override; abstract; + function InitializePlayback(): boolean; override; + function FinalizePlayback: boolean; override; + + procedure SetAppVolume(Volume: single); override; + + function CreateVoiceStream(ChannelMap: integer; FormatInfo: TAudioFormatInfo): TAudioVoiceStream; override; + + function GetMixer(): TAudioMixerStream; {$IFDEF HasInline}inline;{$ENDIF} + function GetAudioFormatInfo(): TAudioFormatInfo; + + procedure MixBuffers(DstBuffer, SrcBuffer: PChar; Size: Cardinal; Volume: Single); virtual; + end; + +type + TGenericVoiceStream = class(TAudioVoiceStream) + private + VoiceBuffer: TRingBuffer; + BufferLock: PSDL_Mutex; + PlaybackStream: TGenericPlaybackStream; + Engine: TAudioPlayback_SoftMixer; + public + constructor Create(Engine: TAudioPlayback_SoftMixer); + + function Open(ChannelMap: integer; FormatInfo: TAudioFormatInfo): boolean; override; + procedure Close(); override; + procedure WriteData(Buffer: PChar; BufferSize: integer); override; + function ReadData(Buffer: PChar; BufferSize: integer): integer; override; + function IsEOF(): boolean; override; + function IsError(): boolean; override; + end; + +const + SOURCE_BUFFER_FRAMES = 4096; + +const + MAX_VOICE_DELAY = 0.500; // 20ms + +implementation + +uses + Math, + ULog, + UIni, + UFFT, + UAudioConverter, + UMain; + +{ TAudioMixerStream } + +constructor TAudioMixerStream.Create(Engine: TAudioPlayback_SoftMixer); +begin + inherited Create(); + + Self.Engine := Engine; + + ActiveStreams := TList.Create; + InternalLock := SDL_CreateMutex(); + AppVolume := 1.0; +end; + +destructor TAudioMixerStream.Destroy(); +begin + if assigned(MixerBuffer) then + Freemem(MixerBuffer); + ActiveStreams.Free; + SDL_DestroyMutex(InternalLock); + inherited; +end; + +procedure TAudioMixerStream.Lock(); +begin + SDL_mutexP(InternalLock); +end; + +procedure TAudioMixerStream.Unlock(); +begin + SDL_mutexV(InternalLock); +end; + +function TAudioMixerStream.GetVolume(): single; +begin + Lock(); + Result := AppVolume; + Unlock(); +end; + +procedure TAudioMixerStream.SetVolume(Volume: single); +begin + Lock(); + AppVolume := Volume; + Unlock(); +end; + +procedure TAudioMixerStream.AddStream(Stream: TAudioPlaybackStream); +begin + if not assigned(Stream) then + Exit; + + Lock(); + // check if stream is already in list to avoid duplicates + if (ActiveStreams.IndexOf(Pointer(Stream)) = -1) then + ActiveStreams.Add(Pointer(Stream)); + Unlock(); +end; + +(* + * Sets the entry of stream in the ActiveStreams-List to nil + * but does not remove it from the list (Count is not changed!). + * Otherwise iterations over the elements might fail due to a + * changed Count-property. + * Call ActiveStreams.Pack() to remove the nil-pointers + * or check for nil-pointers when accessing ActiveStreams. + *) +procedure TAudioMixerStream.RemoveStream(Stream: TAudioPlaybackStream); +var + Index: integer; +begin + Lock(); + Index := activeStreams.IndexOf(Pointer(Stream)); + if (Index <> -1) then + begin + // remove entry but do not decrease count-property + ActiveStreams[Index] := nil; + end; + Unlock(); +end; + +function TAudioMixerStream.ReadData(Buffer: PChar; BufferSize: integer): integer; +var + i: integer; + Size: integer; + Stream: TGenericPlaybackStream; + NeedsPacking: boolean; +begin + Result := BufferSize; + + // zero target-buffer (silence) + FillChar(Buffer^, BufferSize, 0); + + // resize mixer-buffer if necessary + ReallocMem(MixerBuffer, BufferSize); + if not assigned(MixerBuffer) then + Exit; + + Lock(); + + NeedsPacking := false; + + // mix streams to one stream + for i := 0 to ActiveStreams.Count-1 do + begin + if (ActiveStreams[i] = nil) then + begin + NeedsPacking := true; + continue; + end; + + Stream := TGenericPlaybackStream(ActiveStreams[i]); + // fetch data from current stream + Size := Stream.ReadData(MixerBuffer, BufferSize); + if (Size > 0) then + begin + // mix stream-data with mixer-buffer + // Note: use Self.appVolume instead of Self.Volume to prevent recursive locking + Engine.MixBuffers(Buffer, MixerBuffer, Size, AppVolume * Stream.Volume); + end; + end; + + // remove nil-pointers from list + if (NeedsPacking) then + begin + ActiveStreams.Pack(); + end; + + Unlock(); +end; + + +{ TGenericPlaybackStream } + +constructor TGenericPlaybackStream.Create(Engine: TAudioPlayback_SoftMixer); +begin + inherited Create(); + Self.Engine := Engine; + InternalLock := SDL_CreateMutex(); + SoundEffects := TList.Create; + Status := ssStopped; + Reset(); +end; + +destructor TGenericPlaybackStream.Destroy(); +begin + Close(); + SDL_DestroyMutex(InternalLock); + FreeAndNil(SoundEffects); + inherited; +end; + +procedure TGenericPlaybackStream.Reset(); +begin + SourceStream := nil; + + FreeAndNil(Converter); + + FreeMem(SampleBuffer); + SampleBuffer := nil; + SampleBufferPos := 0; + SampleBufferSize := 0; + SampleBufferCount := 0; + + FreeMem(SourceBuffer); + SourceBuffer := nil; + SourceBufferSize := 0; + SourceBufferCount := 0; + + NeedsRewind := false; + + fVolume := 0; + SoundEffects.Clear; + FadeInTime := 0; +end; + +function TGenericPlaybackStream.Open(SourceStream: TAudioSourceStream): boolean; +begin + Result := false; + + Close(); + + if (not assigned(SourceStream)) then + Exit; + Self.SourceStream := SourceStream; + + if (not InitFormatConversion()) then + begin + // reset decode-stream so it will not be freed on destruction + Self.SourceStream := nil; + Exit; + end; + + SourceBufferSize := SOURCE_BUFFER_FRAMES * SourceStream.GetAudioFormatInfo().FrameSize; + GetMem(SourceBuffer, SourceBufferSize); + fVolume := 1.0; + + Result := true; +end; + +procedure TGenericPlaybackStream.Close(); +begin + // stop audio-callback on this stream + Stop(); + + // Note: PerformOnClose must be called before SourceStream is invalidated + PerformOnClose(); + // and free data + Reset(); +end; + +procedure TGenericPlaybackStream.LockSampleBuffer(); +begin + SDL_mutexP(InternalLock); +end; + +procedure TGenericPlaybackStream.UnlockSampleBuffer(); +begin + SDL_mutexV(InternalLock); +end; + +function TGenericPlaybackStream.InitFormatConversion(): boolean; +var + SrcFormatInfo: TAudioFormatInfo; + DstFormatInfo: TAudioFormatInfo; +begin + Result := false; + + SrcFormatInfo := SourceStream.GetAudioFormatInfo(); + DstFormatInfo := GetAudioFormatInfo(); + + // TODO: selection should not be done here, use a factory (TAudioConverterFactory) instead + {$IF Defined(UseFFmpegResample)} + Converter := TAudioConverter_FFmpeg.Create(); + {$ELSEIF Defined(UseSRCResample)} + Converter := TAudioConverter_SRC.Create(); + {$ELSE} + Converter := TAudioConverter_SDL.Create(); + {$IFEND} + + Result := Converter.Init(SrcFormatInfo, DstFormatInfo); +end; + +procedure TGenericPlaybackStream.Play(); +var + Mixer: TAudioMixerStream; +begin + // only paused streams are not flushed + if (Status = ssPaused) then + NeedsRewind := false; + + // rewind if necessary. Cases that require no rewind are: + // - stream was created and never played + // - stream was paused and is resumed now + // - stream was stopped and set to a new position already + if (NeedsRewind) then + SetPosition(0); + + // update status + Status := ssPlaying; + + NeedsRewind := true; + + // add this stream to the mixer + Mixer := Engine.GetMixer(); + if (Mixer <> nil) then + Mixer.AddStream(Self); +end; + +procedure TGenericPlaybackStream.FadeIn(Time: real; TargetVolume: single); +begin + FadeInTime := Trunc(Time * 1000); + FadeInStartTime := SDL_GetTicks(); + FadeInStartVolume := fVolume; + FadeInTargetVolume := TargetVolume; + Play(); +end; + +procedure TGenericPlaybackStream.Pause(); +var + Mixer: TAudioMixerStream; +begin + if (Status <> ssPlaying) then + Exit; + + Status := ssPaused; + + Mixer := Engine.GetMixer(); + if (Mixer <> nil) then + Mixer.RemoveStream(Self); +end; + +procedure TGenericPlaybackStream.Stop(); +var + Mixer: TAudioMixerStream; +begin + if (Status = ssStopped) then + Exit; + + Status := ssStopped; + + Mixer := Engine.GetMixer(); + if (Mixer <> nil) then + Mixer.RemoveStream(Self); +end; + +function TGenericPlaybackStream.GetLoop(): boolean; +begin + if assigned(SourceStream) then + Result := SourceStream.Loop + else + Result := false; +end; + +procedure TGenericPlaybackStream.SetLoop(Enabled: boolean); +begin + if assigned(SourceStream) then + SourceStream.Loop := Enabled; +end; + +function TGenericPlaybackStream.GetLength(): real; +begin + if assigned(SourceStream) then + Result := SourceStream.Length + else + Result := -1; +end; + +function TGenericPlaybackStream.GetLatency(): double; +begin + Result := Engine.GetLatency(); +end; + +function TGenericPlaybackStream.GetStatus(): TStreamStatus; +begin + Result := Status; +end; + +function TGenericPlaybackStream.GetAudioFormatInfo(): TAudioFormatInfo; +begin + Result := Engine.GetAudioFormatInfo(); +end; + +procedure TGenericPlaybackStream.FlushBuffers(); +begin + SampleBufferCount := 0; + SampleBufferPos := 0; + SourceBufferCount := 0; +end; + +procedure TGenericPlaybackStream.ApplySoundEffects(Buffer: PChar; BufferSize: integer); +var + i: integer; +begin + for i := 0 to SoundEffects.Count-1 do + begin + if (SoundEffects[i] <> nil) then + begin + TSoundEffect(SoundEffects[i]).Callback(Buffer, BufferSize); + end; + end; +end; + +function TGenericPlaybackStream.ReadData(Buffer: PChar; BufferSize: integer): integer; +var + ConversionInputCount: integer; + ConversionOutputSize: integer; // max. number of converted data (= buffer size) + ConversionOutputCount: integer; // actual number of converted data + SourceSize: integer; + RequestedSourceSize: integer; + NeededSampleBufferSize: integer; + BytesNeeded, BytesAvail: integer; + SourceFormatInfo, OutputFormatInfo: TAudioFormatInfo; + SourceFrameSize, OutputFrameSize: integer; + SkipOutputCount: integer; // number of output-data bytes to skip + SkipSourceCount: integer; // number of source-data bytes to skip + FillCount: integer; // number of bytes to fill with padding data + CopyCount: integer; + PadFrame: PChar; + i: integer; +begin + Result := -1; + + // sanity check for the source-stream + if (not assigned(SourceStream)) then + Exit; + + SkipOutputCount := 0; + SkipSourceCount := 0; + FillCount := 0; + + SourceFormatInfo := SourceStream.GetAudioFormatInfo(); + SourceFrameSize := SourceFormatInfo.FrameSize; + OutputFormatInfo := GetAudioFormatInfo(); + OutputFrameSize := OutputFormatInfo.FrameSize; + + // synchronize (adjust buffer size) + BytesNeeded := Synchronize(BufferSize, OutputFormatInfo); + if (BytesNeeded > BufferSize) then + begin + SkipOutputCount := BytesNeeded - BufferSize; + BytesNeeded := BufferSize; + end + else if (BytesNeeded < BufferSize) then + begin + FillCount := BufferSize - BytesNeeded; + end; + + // lock access to sample-buffer + LockSampleBuffer(); + try + + // skip sample-buffer data + SampleBufferPos := SampleBufferPos + SkipOutputCount; + // size of available bytes in SampleBuffer after skipping + SampleBufferCount := SampleBufferCount - SampleBufferPos; + // update byte skip-count + SkipOutputCount := -SampleBufferCount; + + // now that we skipped all buffered data from the last pass, we have to skip + // data directly after fetching it from the source-stream. + if (SkipOutputCount > 0) then + begin + SampleBufferCount := 0; + // convert skip-count to source-format units and resize to a multiple of + // the source frame-size. + SkipSourceCount := Round((SkipOutputCount * OutputFormatInfo.GetRatio(SourceFormatInfo)) / + SourceFrameSize) * SourceFrameSize; + SkipOutputCount := 0; + end; + + // copy data to front of buffer + if ((SampleBufferCount > 0) and (SampleBufferPos > 0)) then + Move(SampleBuffer[SampleBufferPos], SampleBuffer[0], SampleBufferCount); + SampleBufferPos := 0; + + // resize buffer to a reasonable size + if (BufferSize > SampleBufferCount) then + begin + // Note: use BufferSize instead of BytesNeeded to minimize the need for resizing + SampleBufferSize := BufferSize; + ReallocMem(SampleBuffer, SampleBufferSize); + if (not assigned(SampleBuffer)) then + Exit; + end; + + // fill sample-buffer (fetch and convert one block of source data per loop) + while (SampleBufferCount < BytesNeeded) do + begin + // move remaining source data from the previous pass to front of buffer + if (SourceBufferCount > 0) then + begin + Move(SourceBuffer[SourceBufferSize-SourceBufferCount], + SourceBuffer[0], + SourceBufferCount); + end; + + SourceSize := SourceStream.ReadData( + @SourceBuffer[SourceBufferCount], SourceBufferSize-SourceBufferCount); + // break on error (-1) or if no data is available (0), e.g. while seeking + if (SourceSize <= 0) then + begin + // if we do not have data -> exit + if (SourceBufferCount = 0) then + begin + FlushBuffers(); + Exit; + end; + // if we have some data, stop retrieving data from the source stream + // and use the data we have so far + Break; + end; + + SourceBufferCount := SourceBufferCount + SourceSize; + + // end-of-file reached -> stop playback + if (SourceStream.EOF) then + begin + if (Loop) then + SourceStream.Position := 0 + else + Stop(); + end; + + if (SkipSourceCount > 0) then + begin + // skip data and update source buffer count + SourceBufferCount := SourceBufferCount - SkipSourceCount; + SkipSourceCount := -SourceBufferCount; + // continue with next pass if we skipped all data + if (SourceBufferCount <= 0) then + begin + SourceBufferCount := 0; + Continue; + end; + end; + + // calc buffer size (might be bigger than actual resampled byte count) + ConversionOutputSize := Converter.GetOutputBufferSize(SourceBufferCount); + NeededSampleBufferSize := SampleBufferCount + ConversionOutputSize; + + // resize buffer if necessary + if (SampleBufferSize < NeededSampleBufferSize) then + begin + SampleBufferSize := NeededSampleBufferSize; + ReallocMem(SampleBuffer, SampleBufferSize); + if (not assigned(SampleBuffer)) then + begin + FlushBuffers(); + Exit; + end; + end; + + // resample source data (Note: ConversionInputCount might be adjusted by Convert()) + ConversionInputCount := SourceBufferCount; + ConversionOutputCount := Converter.Convert( + SourceBuffer, @SampleBuffer[SampleBufferCount], ConversionInputCount); + if (ConversionOutputCount = -1) then + begin + FlushBuffers(); + Exit; + end; + + // adjust sample- and source-buffer count by the number of converted bytes + SampleBufferCount := SampleBufferCount + ConversionOutputCount; + SourceBufferCount := SourceBufferCount - ConversionInputCount; + end; + + // apply effects + ApplySoundEffects(SampleBuffer, SampleBufferCount); + + // copy data to result buffer + CopyCount := Min(BytesNeeded, SampleBufferCount); + Move(SampleBuffer[0], Buffer[BufferSize - BytesNeeded], CopyCount); + Dec(BytesNeeded, CopyCount); + SampleBufferPos := CopyCount; + + // release buffer lock + finally + UnlockSampleBuffer(); + end; + + // pad the buffer with the last frame if we are to fast + if (FillCount > 0) then + begin + if (CopyCount >= OutputFrameSize) then + PadFrame := @Buffer[CopyCount-OutputFrameSize] + else + PadFrame := nil; + FillBufferWithFrame(@Buffer[CopyCount], FillCount, + PadFrame, OutputFrameSize); + end; + + // BytesNeeded now contains the number of remaining bytes we were not able to fetch + Result := BufferSize - BytesNeeded; +end; + +function TGenericPlaybackStream.GetPCMData(var Data: TPCMData): Cardinal; +var + ByteCount: integer; +begin + Result := 0; + + // just SInt16 stereo support for now + if ((Engine.GetAudioFormatInfo().Format <> asfS16) or + (Engine.GetAudioFormatInfo().Channels <> 2)) then + begin + Exit; + end; + + // zero memory + FillChar(Data, SizeOf(Data), 0); + + // TODO: At the moment just the first samples of the SampleBuffer + // are returned, even if there is newer data in the upper samples. + + LockSampleBuffer(); + ByteCount := Min(SizeOf(Data), SampleBufferCount); + if (ByteCount > 0) then + begin + Move(SampleBuffer[0], Data, ByteCount); + end; + UnlockSampleBuffer(); + + Result := ByteCount div SizeOf(TPCMStereoSample); +end; + +procedure TGenericPlaybackStream.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 := GetAudioFormatInfo(); + + DataIn := AllocMem(FFTSize * SizeOf(Single)); + if (DataIn = nil) then + Exit; + + LockSampleBuffer(); + // TODO: We just use the first Frames frames, the others are ignored. + 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; + UnlockSampleBuffer(); + + WindowFunc(fwfHanning, FFTSize, DataIn); + PowerSpectrum(FFTSize, DataIn, @Data); + FreeMem(DataIn); + + // resize data to a 0..1 range + for i := 0 to High(TFFTData) do + begin + Data[i] := Sqrt(Data[i]) / 100; + end; +end; + +procedure TGenericPlaybackStream.AddSoundEffect(Effect: TSoundEffect); +begin + if (not assigned(Effect)) then + Exit; + + LockSampleBuffer(); + // check if effect is already in list to avoid duplicates + if (SoundEffects.IndexOf(Pointer(Effect)) = -1) then + SoundEffects.Add(Pointer(Effect)); + UnlockSampleBuffer(); +end; + +procedure TGenericPlaybackStream.RemoveSoundEffect(Effect: TSoundEffect); +begin + LockSampleBuffer(); + SoundEffects.Remove(Effect); + UnlockSampleBuffer(); +end; + +function TGenericPlaybackStream.GetPosition: real; +var + BufferedTime: double; +begin + if assigned(SourceStream) then + begin + LockSampleBuffer(); + + // calc the time of source data that is buffered (in the SampleBuffer and SourceBuffer) + // but not yet outputed + BufferedTime := (SampleBufferCount - SampleBufferPos) / Engine.FormatInfo.BytesPerSec + + SourceBufferCount / SourceStream.GetAudioFormatInfo().BytesPerSec; + // and subtract it from the source position + Result := SourceStream.Position - BufferedTime; + + UnlockSampleBuffer(); + end + else + begin + Result := -1; + end; +end; + +procedure TGenericPlaybackStream.SetPosition(Time: real); +begin + if assigned(SourceStream) then + begin + LockSampleBuffer(); + + SourceStream.Position := Time; + if (Status = ssStopped) then + NeedsRewind := false; + // do not use outdated data + FlushBuffers(); + + AvgSyncDiff := -1; + + UnlockSampleBuffer(); + end; +end; + +function TGenericPlaybackStream.GetVolume(): single; +var + FadeAmount: Single; +begin + LockSampleBuffer(); + // adjust volume if fading is enabled + if (FadeInTime > 0) then + begin + FadeAmount := (SDL_GetTicks() - FadeInStartTime) / FadeInTime; + // check if fade-target is reached + if (FadeAmount >= 1) then + begin + // target reached -> stop fading + FadeInTime := 0; + fVolume := FadeInTargetVolume; + end + else + begin + // fading in progress + fVolume := FadeAmount*FadeInTargetVolume + (1-FadeAmount)*FadeInStartVolume; + end; + end; + // return current volume + Result := fVolume; + UnlockSampleBuffer(); +end; + +procedure TGenericPlaybackStream.SetVolume(Volume: single); +begin + LockSampleBuffer(); + // stop fading + FadeInTime := 0; + // clamp volume + if (Volume > 1.0) then + fVolume := 1.0 + else if (Volume < 0) then + fVolume := 0 + else + fVolume := Volume; + UnlockSampleBuffer(); +end; + + +{ TGenericVoiceStream } + +constructor TGenericVoiceStream.Create(Engine: TAudioPlayback_SoftMixer); +begin + inherited Create(); + Self.Engine := Engine; +end; + +function TGenericVoiceStream.Open(ChannelMap: integer; FormatInfo: TAudioFormatInfo): boolean; +var + BufferSize: integer; +begin + Result := false; + + Close(); + + if (not inherited Open(ChannelMap, FormatInfo)) then + Exit; + + // Note: + // - use Self.FormatInfo instead of FormatInfo as the latter one might have a + // channel size of 2. + // - the buffer-size must be a multiple of the FrameSize + BufferSize := (Ceil(MAX_VOICE_DELAY * Self.FormatInfo.BytesPerSec) div Self.FormatInfo.FrameSize) * + Self.FormatInfo.FrameSize; + VoiceBuffer := TRingBuffer.Create(BufferSize); + + BufferLock := SDL_CreateMutex(); + + + // create a matching playback stream for the voice-stream + PlaybackStream := TGenericPlaybackStream.Create(Engine); + // link voice- and playback-stream + if (not PlaybackStream.Open(Self)) then + begin + PlaybackStream.Free; + Exit; + end; + + // start voice passthrough + PlaybackStream.Play(); + + Result := true; +end; + +procedure TGenericVoiceStream.Close(); +begin + // stop and free the playback stream + FreeAndNil(PlaybackStream); + + // free data + FreeAndNil(VoiceBuffer); + if (BufferLock <> nil) then + SDL_DestroyMutex(BufferLock); + + inherited Close(); +end; + +procedure TGenericVoiceStream.WriteData(Buffer: PChar; BufferSize: integer); +begin + // lock access to buffer + SDL_mutexP(BufferLock); + try + if (VoiceBuffer = nil) then + Exit; + VoiceBuffer.Write(Buffer, BufferSize); + finally + SDL_mutexV(BufferLock); + end; +end; + +function TGenericVoiceStream.ReadData(Buffer: PChar; BufferSize: integer): integer; +begin + Result := -1; + + // lock access to buffer + SDL_mutexP(BufferLock); + try + if (VoiceBuffer = nil) then + Exit; + Result := VoiceBuffer.Read(Buffer, BufferSize); + finally + SDL_mutexV(BufferLock); + end; +end; + +function TGenericVoiceStream.IsEOF(): boolean; +begin + SDL_mutexP(BufferLock); + Result := (VoiceBuffer = nil); + SDL_mutexV(BufferLock); +end; + +function TGenericVoiceStream.IsError(): boolean; +begin + Result := false; +end; + + +{ TAudioPlayback_SoftMixer } + +function TAudioPlayback_SoftMixer.InitializePlayback: boolean; +begin + Result := false; + + //Log.LogStatus('InitializePlayback', 'UAudioPlayback_SoftMixer'); + + if(not InitializeAudioPlaybackEngine()) then + Exit; + + MixerStream := TAudioMixerStream.Create(Self); + + if(not StartAudioPlaybackEngine()) then + Exit; + + Result := true; +end; + +function TAudioPlayback_SoftMixer.FinalizePlayback: boolean; +begin + Close; + StopAudioPlaybackEngine(); + + FreeAndNil(MixerStream); + FreeAndNil(FormatInfo); + + FinalizeAudioPlaybackEngine(); + inherited FinalizePlayback; + Result := true; +end; + +procedure TAudioPlayback_SoftMixer.AudioCallback(Buffer: PChar; Size: integer); +begin + MixerStream.ReadData(Buffer, Size); +end; + +function TAudioPlayback_SoftMixer.GetMixer(): TAudioMixerStream; +begin + Result := MixerStream; +end; + +function TAudioPlayback_SoftMixer.GetAudioFormatInfo(): TAudioFormatInfo; +begin + Result := FormatInfo; +end; + +function TAudioPlayback_SoftMixer.CreatePlaybackStream(): TAudioPlaybackStream; +begin + Result := TGenericPlaybackStream.Create(Self); +end; + +function TAudioPlayback_SoftMixer.CreateVoiceStream(ChannelMap: integer; FormatInfo: TAudioFormatInfo): TAudioVoiceStream; +var + VoiceStream: TGenericVoiceStream; +begin + Result := nil; + + // create a voice stream + VoiceStream := TGenericVoiceStream.Create(Self); + if (not VoiceStream.Open(ChannelMap, FormatInfo)) then + begin + VoiceStream.Free; + Exit; + end; + + Result := VoiceStream; +end; + +procedure TAudioPlayback_SoftMixer.SetAppVolume(Volume: single); +begin + // sets volume only for this application + MixerStream.Volume := Volume; +end; + +procedure TAudioPlayback_SoftMixer.MixBuffers(DstBuffer, SrcBuffer: PChar; Size: Cardinal; Volume: Single); +var + SampleIndex: Cardinal; + SampleInt: Integer; + SampleFlt: Single; +begin + SampleIndex := 0; + case FormatInfo.Format of + asfS16: + begin + while (SampleIndex < Size) do + begin + // apply volume and sum with previous mixer value + SampleInt := PSmallInt(@DstBuffer[SampleIndex])^ + + Round(PSmallInt(@SrcBuffer[SampleIndex])^ * Volume); + // clip result + if (SampleInt > High(SmallInt)) then + SampleInt := High(SmallInt) + else if (SampleInt < Low(SmallInt)) then + SampleInt := Low(SmallInt); + // assign result + PSmallInt(@DstBuffer[SampleIndex])^ := SampleInt; + // increase index by one sample + Inc(SampleIndex, SizeOf(SmallInt)); + end; + end; + asfFloat: + begin + while (SampleIndex < Size) do + begin + // apply volume and sum with previous mixer value + SampleFlt := PSingle(@DstBuffer[SampleIndex])^ + + PSingle(@SrcBuffer[SampleIndex])^ * Volume; + // clip result + if (SampleFlt > 1.0) then + SampleFlt := 1.0 + else if (SampleFlt < -1.0) then + SampleFlt := -1.0; + // assign result + PSingle(@DstBuffer[SampleIndex])^ := SampleFlt; + // increase index by one sample + Inc(SampleIndex, SizeOf(Single)); + end; + end; + else + begin + Log.LogError('Incompatible format', 'TAudioMixerStream.MixAudio'); + end; + end; +end; + +end. diff --git a/src/media/UMediaCore_FFmpeg.pas b/src/media/UMediaCore_FFmpeg.pas new file mode 100644 index 00000000..cdd320ac --- /dev/null +++ b/src/media/UMediaCore_FFmpeg.pas @@ -0,0 +1,405 @@ +unit UMediaCore_FFmpeg; + +interface + +{$IFDEF FPC} + {$MODE Delphi} +{$ENDIF} + +{$I switches.inc} + +uses + UMusic, + avcodec, + avformat, + avutil, + ULog, + sdl; + +type + PPacketQueue = ^TPacketQueue; + TPacketQueue = class + private + FirstListEntry: PAVPacketList; + LastListEntry: PAVPacketList; + PacketCount: integer; + Mutex: PSDL_Mutex; + Condition: PSDL_Cond; + Size: integer; + AbortRequest: boolean; + public + constructor Create(); + destructor Destroy(); override; + + function Put(Packet : PAVPacket): integer; + function PutStatus(StatusFlag: integer; StatusInfo: Pointer): integer; + procedure FreeStatusInfo(var Packet: TAVPacket); + function GetStatusInfo(var Packet: TAVPacket): Pointer; + function Get(var Packet: TAVPacket; Blocking: boolean): integer; + function GetSize(): integer; + procedure Flush(); + procedure Abort(); + function IsAborted(): boolean; + end; + +const + STATUS_PACKET: PChar = 'STATUS_PACKET'; +const + PKT_STATUS_FLAG_EOF = 1; // signal end-of-file + PKT_STATUS_FLAG_FLUSH = 2; // request the decoder to flush its avcodec decode buffers + PKT_STATUS_FLAG_ERROR = 3; // signal an error state + PKT_STATUS_FLAG_EMPTY = 4; // request the decoder to output empty data (silence or black frames) + +type + TMediaCore_FFmpeg = class + private + AVCodecLock: PSDL_Mutex; + public + constructor Create(); + destructor Destroy(); override; + class function GetInstance(): TMediaCore_FFmpeg; + + function GetErrorString(ErrorNum: integer): string; + function FindStreamIDs(FormatCtx: PAVFormatContext; out FirstVideoStream, FirstAudioStream: integer ): boolean; + function FindAudioStreamIndex(FormatCtx: PAVFormatContext): integer; + function ConvertFFmpegToAudioFormat(FFmpegFormat: TSampleFormat; out Format: TAudioSampleFormat): boolean; + procedure LockAVCodec(); + procedure UnlockAVCodec(); + end; + +implementation + +uses + SysUtils; + +var + Instance: TMediaCore_FFmpeg; + +constructor TMediaCore_FFmpeg.Create(); +begin + inherited; + AVCodecLock := SDL_CreateMutex(); +end; + +destructor TMediaCore_FFmpeg.Destroy(); +begin + SDL_DestroyMutex(AVCodecLock); + inherited; +end; + +class function TMediaCore_FFmpeg.GetInstance(): TMediaCore_FFmpeg; +begin + if (not Assigned(Instance)) then + Instance := TMediaCore_FFmpeg.Create(); + Result := Instance; +end; + +procedure TMediaCore_FFmpeg.LockAVCodec(); +begin + SDL_mutexP(AVCodecLock); +end; + +procedure TMediaCore_FFmpeg.UnlockAVCodec(); +begin + SDL_mutexV(AVCodecLock); +end; + +function TMediaCore_FFmpeg.GetErrorString(ErrorNum: integer): string; +begin + case ErrorNum of + AVERROR_IO: Result := 'AVERROR_IO'; + AVERROR_NUMEXPECTED: Result := 'AVERROR_NUMEXPECTED'; + AVERROR_INVALIDDATA: Result := 'AVERROR_INVALIDDATA'; + AVERROR_NOMEM: Result := 'AVERROR_NOMEM'; + AVERROR_NOFMT: Result := 'AVERROR_NOFMT'; + AVERROR_NOTSUPP: Result := 'AVERROR_NOTSUPP'; + AVERROR_NOENT: Result := 'AVERROR_NOENT'; + AVERROR_PATCHWELCOME: Result := 'AVERROR_PATCHWELCOME'; + else Result := 'AVERROR_#'+inttostr(ErrorNum); + end; +end; + +{ + @param(FormatCtx is a PAVFormatContext returned from av_open_input_file ) + @param(FirstVideoStream is an OUT value of type integer, this is the index of the video stream) + @param(FirstAudioStream is an OUT value of type integer, this is the index of the audio stream) + @returns(@true on success, @false otherwise) +} +function TMediaCore_FFmpeg.FindStreamIDs(FormatCtx: PAVFormatContext; out FirstVideoStream, FirstAudioStream: integer): boolean; +var + i: integer; + Stream: PAVStream; +begin + // find the first video stream + FirstAudioStream := -1; + FirstVideoStream := -1; + + for i := 0 to FormatCtx.nb_streams-1 do + begin + Stream := FormatCtx.streams[i]; + + if (Stream.codec.codec_type = CODEC_TYPE_VIDEO) and + (FirstVideoStream < 0) then + begin + FirstVideoStream := i; + end; + + if (Stream.codec.codec_type = CODEC_TYPE_AUDIO) and + (FirstAudioStream < 0) then + begin + FirstAudioStream := i; + end; + end; + + // return true if either an audio- or video-stream was found + Result := (FirstAudioStream > -1) or + (FirstVideoStream > -1) ; +end; + +function TMediaCore_FFmpeg.FindAudioStreamIndex(FormatCtx: PAVFormatContext): integer; +var + i: integer; + StreamIndex: integer; + Stream: PAVStream; +begin + // find the first audio stream + StreamIndex := -1; + + for i := 0 to FormatCtx^.nb_streams-1 do + begin + Stream := FormatCtx^.streams[i]; + + if (Stream.codec^.codec_type = CODEC_TYPE_AUDIO) then + begin + StreamIndex := i; + Break; + end; + end; + + Result := StreamIndex; +end; + +function TMediaCore_FFmpeg.ConvertFFmpegToAudioFormat(FFmpegFormat: TSampleFormat; out Format: TAudioSampleFormat): boolean; +begin + case FFmpegFormat of + SAMPLE_FMT_U8: Format := asfU8; + SAMPLE_FMT_S16: Format := asfS16; + SAMPLE_FMT_S24: Format := asfS24; + SAMPLE_FMT_S32: Format := asfS32; + SAMPLE_FMT_FLT: Format := asfFloat; + else begin + Result := false; + Exit; + end; + end; + Result := true; +end; + +{ TPacketQueue } + +constructor TPacketQueue.Create(); +begin + inherited; + + FirstListEntry := nil; + LastListEntry := nil; + PacketCount := 0; + Size := 0; + + Mutex := SDL_CreateMutex(); + Condition := SDL_CreateCond(); +end; + +destructor TPacketQueue.Destroy(); +begin + Flush(); + SDL_DestroyMutex(Mutex); + SDL_DestroyCond(Condition); + inherited; +end; + +procedure TPacketQueue.Abort(); +begin + SDL_LockMutex(Mutex); + + AbortRequest := true; + + SDL_CondBroadcast(Condition); + SDL_UnlockMutex(Mutex); +end; + +function TPacketQueue.IsAborted(): boolean; +begin + SDL_LockMutex(Mutex); + Result := AbortRequest; + SDL_UnlockMutex(Mutex); +end; + +function TPacketQueue.Put(Packet : PAVPacket): integer; +var + CurrentListEntry : PAVPacketList; +begin + Result := -1; + + if (Packet = nil) then + Exit; + + if (PChar(Packet^.data) <> STATUS_PACKET) then + begin + if (av_dup_packet(Packet) < 0) then + Exit; + end; + + CurrentListEntry := av_malloc(SizeOf(TAVPacketList)); + if (CurrentListEntry = nil) then + Exit; + + CurrentListEntry^.pkt := Packet^; + CurrentListEntry^.next := nil; + + SDL_LockMutex(Mutex); + try + if (LastListEntry = nil) then + FirstListEntry := CurrentListEntry + else + LastListEntry^.next := CurrentListEntry; + + LastListEntry := CurrentListEntry; + Inc(PacketCount); + + Size := Size + CurrentListEntry^.pkt.size; + SDL_CondSignal(Condition); + finally + SDL_UnlockMutex(Mutex); + end; + + Result := 0; +end; + +(** + * Adds a status packet (EOF, Flush, etc.) to the end of the queue. + * StatusInfo can be used to pass additional information to the decoder. + * Only assign nil or a valid pointer to data allocated with Getmem() to + * StatusInfo because the pointer will be disposed with Freemem() on a call + * to Flush(). If the packet is removed from the queue it is the decoder's + * responsibility to free the StatusInfo data with FreeStatusInfo(). + *) +function TPacketQueue.PutStatus(StatusFlag: integer; StatusInfo: Pointer): integer; +var + TempPacket: PAVPacket; +begin + // create temp. package + TempPacket := av_malloc(SizeOf(TAVPacket)); + if (TempPacket = nil) then + begin + Result := -1; + Exit; + end; + // init package + av_init_packet(TempPacket^); + TempPacket^.data := Pointer(STATUS_PACKET); + TempPacket^.flags := StatusFlag; + TempPacket^.priv := StatusInfo; + // put a copy of the package into the queue + Result := Put(TempPacket); + // data has been copied -> delete temp. package + av_free(TempPacket); +end; + +procedure TPacketQueue.FreeStatusInfo(var Packet: TAVPacket); +begin + if (Packet.priv <> nil) then + FreeMem(Packet.priv); +end; + +function TPacketQueue.GetStatusInfo(var Packet: TAVPacket): Pointer; +begin + Result := Packet.priv; +end; + +function TPacketQueue.Get(var Packet: TAVPacket; Blocking: boolean): integer; +var + CurrentListEntry: PAVPacketList; +const + WAIT_TIMEOUT = 10; // timeout in ms +begin + Result := -1; + + SDL_LockMutex(Mutex); + try + while (true) do + begin + if (AbortRequest) then + Exit; + + CurrentListEntry := FirstListEntry; + if (CurrentListEntry <> nil) then + begin + FirstListEntry := CurrentListEntry^.next; + if (FirstListEntry = nil) then + LastListEntry := nil; + Dec(PacketCount); + + Size := Size - CurrentListEntry^.pkt.size; + Packet := CurrentListEntry^.pkt; + av_free(CurrentListEntry); + + Result := 1; + Break; + end + else if (not Blocking) then + begin + Result := 0; + Break; + end + else + begin + // block until a new package arrives, + // but do not wait till infinity to avoid deadlocks + if (SDL_CondWaitTimeout(Condition, Mutex, WAIT_TIMEOUT) = SDL_MUTEX_TIMEDOUT) then + begin + Result := 0; + Break; + end; + end; + end; + finally + SDL_UnlockMutex(Mutex); + end; +end; + +function TPacketQueue.GetSize(): integer; +begin + SDL_LockMutex(Mutex); + Result := Size; + SDL_UnlockMutex(Mutex); +end; + +procedure TPacketQueue.Flush(); +var + CurrentListEntry, TempListEntry: PAVPacketList; +begin + SDL_LockMutex(Mutex); + + CurrentListEntry := FirstListEntry; + while(CurrentListEntry <> nil) do + begin + TempListEntry := CurrentListEntry^.next; + // free status data + if (PChar(CurrentListEntry^.pkt.data) = STATUS_PACKET) then + FreeStatusInfo(CurrentListEntry^.pkt); + // free packet data + av_free_packet(@CurrentListEntry^.pkt); + // Note: param must be a pointer to a pointer! + av_freep(@CurrentListEntry); + CurrentListEntry := TempListEntry; + end; + LastListEntry := nil; + FirstListEntry := nil; + PacketCount := 0; + Size := 0; + + SDL_UnlockMutex(Mutex); +end; + +end. diff --git a/src/media/UMediaCore_SDL.pas b/src/media/UMediaCore_SDL.pas new file mode 100644 index 00000000..252f72a0 --- /dev/null +++ b/src/media/UMediaCore_SDL.pas @@ -0,0 +1,38 @@ +unit UMediaCore_SDL; + +interface + +{$IFDEF FPC} + {$MODE Delphi} +{$ENDIF} + +{$I switches.inc} + +uses + UMusic, + sdl; + +function ConvertAudioFormatToSDL(Format: TAudioSampleFormat; out SDLFormat: UInt16): boolean; + +implementation + +function ConvertAudioFormatToSDL(Format: TAudioSampleFormat; out SDLFormat: UInt16): boolean; +begin + case Format of + asfU8: SDLFormat := AUDIO_U8; + asfS8: SDLFormat := AUDIO_S8; + asfU16LSB: SDLFormat := AUDIO_U16LSB; + asfS16LSB: SDLFormat := AUDIO_S16LSB; + asfU16MSB: SDLFormat := AUDIO_U16MSB; + asfS16MSB: SDLFormat := AUDIO_S16MSB; + asfU16: SDLFormat := AUDIO_U16; + asfS16: SDLFormat := AUDIO_S16; + else begin + Result := false; + Exit; + end; + end; + Result := true; +end; + +end. diff --git a/src/media/UMedia_dummy.pas b/src/media/UMedia_dummy.pas new file mode 100644 index 00000000..438b89ab --- /dev/null +++ b/src/media/UMedia_dummy.pas @@ -0,0 +1,243 @@ +unit UMedia_dummy; + +interface + +{$IFDEF FPC} + {$MODE Delphi} +{$ENDIF} + +{$I switches.inc} + +implementation + +uses + SysUtils, + math, + UMusic; + +type + TMedia_dummy = class( TInterfacedObject, IVideoPlayback, IVideoVisualization, IAudioPlayback, IAudioInput ) + private + DummyOutputDeviceList: TAudioOutputDeviceList; + public + constructor Create(); + function GetName: string; + + function Init(): boolean; + function Finalize(): boolean; + + function Open(const aFileName : string): boolean; // true if succeed + procedure Close; + + procedure Play; + procedure Pause; + procedure Stop; + + procedure SetPosition(Time: real); + function GetPosition: real; + + procedure SetSyncSource(SyncSource: TSyncSource); + + procedure GetFrame(Time: Extended); + procedure DrawGL(Screen: integer); + + // IAudioInput + function InitializeRecord: boolean; + function FinalizeRecord: boolean; + procedure CaptureStart; + procedure CaptureStop; + procedure GetFFTData(var data: TFFTData); + function GetPCMData(var data: TPCMData): Cardinal; + + // IAudioPlayback + function InitializePlayback: boolean; + function FinalizePlayback: boolean; + + function GetOutputDeviceList(): TAudioOutputDeviceList; + procedure FadeIn(Time: real; TargetVolume: single); + procedure SetAppVolume(Volume: single); + procedure SetVolume(Volume: single); + procedure SetLoop(Enabled: boolean); + procedure Rewind; + + function Finished: boolean; + function Length: real; + + function OpenSound(const Filename: string): TAudioPlaybackStream; + procedure CloseSound(var PlaybackStream: TAudioPlaybackStream); + procedure PlaySound(stream: TAudioPlaybackStream); + procedure StopSound(stream: TAudioPlaybackStream); + + function CreateVoiceStream(Channel: integer; FormatInfo: TAudioFormatInfo): TAudioVoiceStream; + procedure CloseVoiceStream(var VoiceStream: TAudioVoiceStream); + end; + +function TMedia_dummy.GetName: string; +begin + Result := 'dummy'; +end; + +procedure TMedia_dummy.GetFrame(Time: Extended); +begin +end; + +procedure TMedia_dummy.DrawGL(Screen: integer); +begin +end; + +constructor TMedia_dummy.Create(); +begin + inherited; +end; + +function TMedia_dummy.Init(): boolean; +begin + Result := true; +end; + +function TMedia_dummy.Finalize(): boolean; +begin + Result := true; +end; + +function TMedia_dummy.Open(const aFileName : string): boolean; // true if succeed +begin + Result := false; +end; + +procedure TMedia_dummy.Close; +begin +end; + +procedure TMedia_dummy.Play; +begin +end; + +procedure TMedia_dummy.Pause; +begin +end; + +procedure TMedia_dummy.Stop; +begin +end; + +procedure TMedia_dummy.SetPosition(Time: real); +begin +end; + +function TMedia_dummy.GetPosition: real; +begin + Result := 0; +end; + +procedure TMedia_dummy.SetSyncSource(SyncSource: TSyncSource); +begin +end; + +// IAudioInput +function TMedia_dummy.InitializeRecord: boolean; +begin + Result := true; +end; + +function TMedia_dummy.FinalizeRecord: boolean; +begin + Result := true; +end; + +procedure TMedia_dummy.CaptureStart; +begin +end; + +procedure TMedia_dummy.CaptureStop; +begin +end; + +procedure TMedia_dummy.GetFFTData(var data: TFFTData); +begin +end; + +function TMedia_dummy.GetPCMData(var data: TPCMData): Cardinal; +begin + Result := 0; +end; + +// IAudioPlayback +function TMedia_dummy.InitializePlayback: boolean; +begin + SetLength(DummyOutputDeviceList, 1); + DummyOutputDeviceList[0] := TAudioOutputDevice.Create(); + DummyOutputDeviceList[0].Name := '[Dummy Device]'; + Result := true; +end; + +function TMedia_dummy.FinalizePlayback: boolean; +begin + Result := true; +end; + +function TMedia_dummy.GetOutputDeviceList(): TAudioOutputDeviceList; +begin + Result := DummyOutputDeviceList; +end; + +procedure TMedia_dummy.SetAppVolume(Volume: single); +begin +end; + +procedure TMedia_dummy.SetVolume(Volume: single); +begin +end; + +procedure TMedia_dummy.SetLoop(Enabled: boolean); +begin +end; + +procedure TMedia_dummy.FadeIn(Time: real; TargetVolume: single); +begin +end; + +procedure TMedia_dummy.Rewind; +begin +end; + +function TMedia_dummy.Finished: boolean; +begin + Result := false; +end; + +function TMedia_dummy.Length: real; +begin + Result := 60; +end; + +function TMedia_dummy.OpenSound(const Filename: string): TAudioPlaybackStream; +begin + Result := nil; +end; + +procedure TMedia_dummy.CloseSound(var PlaybackStream: TAudioPlaybackStream); +begin +end; + +procedure TMedia_dummy.PlaySound(stream: TAudioPlaybackStream); +begin +end; + +procedure TMedia_dummy.StopSound(stream: TAudioPlaybackStream); +begin +end; + +function TMedia_dummy.CreateVoiceStream(Channel: integer; FormatInfo: TAudioFormatInfo): TAudioVoiceStream; +begin + Result := nil; +end; + +procedure TMedia_dummy.CloseVoiceStream(var VoiceStream: TAudioVoiceStream); +begin +end; + +initialization + MediaManager.Add(TMedia_dummy.Create); + +end. diff --git a/src/media/UVideo.pas b/src/media/UVideo.pas new file mode 100644 index 00000000..0ab1d350 --- /dev/null +++ b/src/media/UVideo.pas @@ -0,0 +1,828 @@ +{############################################################################## + # FFmpeg support for UltraStar deluxe # + # # + # Created by b1indy # + # based on 'An ffmpeg and SDL Tutorial' (http://www.dranger.com/ffmpeg/) # + # with modifications by Jay Binks <jaybinks@gmail.com> # + # # + # http://www.mail-archive.com/fpc-pascal@lists.freepascal.org/msg09949.html # + # http://www.nabble.com/file/p11795857/mpegpas01.zip # + # # + ##############################################################################} + +unit UVideo; + +// uncomment if you want to see the debug stuff +{.$define DebugDisplay} +{.$define DebugFrames} +{.$define VideoBenchmark} +{.$define Info} + +interface + +{$IFDEF FPC} + {$MODE Delphi} +{$ENDIF} + +{$I switches.inc} + +// use BGR-format for accelerated colorspace conversion with swscale +{$IFDEF UseSWScale} + {$DEFINE PIXEL_FMT_BGR} +{$ENDIF} + +implementation + +uses + SDL, + textgl, + avcodec, + avformat, + avutil, + avio, + rational, + {$IFDEF UseSWScale} + swscale, + {$ENDIF} + UMediaCore_FFmpeg, + math, + gl, + glext, + SysUtils, + UCommon, + UConfig, + ULog, + UMusic, + UGraphicClasses, + UGraphic; + +const +{$IFDEF PIXEL_FMT_BGR} + PIXEL_FMT_OPENGL = GL_BGR; + PIXEL_FMT_FFMPEG = PIX_FMT_BGR24; +{$ELSE} + PIXEL_FMT_OPENGL = GL_RGB; + PIXEL_FMT_FFMPEG = PIX_FMT_RGB24; +{$ENDIF} + +type + TVideoPlayback_FFmpeg = class( TInterfacedObject, IVideoPlayback ) + private + fVideoOpened, + fVideoPaused: Boolean; + + VideoStream: PAVStream; + VideoStreamIndex : Integer; + VideoFormatContext: PAVFormatContext; + VideoCodecContext: PAVCodecContext; + VideoCodec: PAVCodec; + + AVFrame: PAVFrame; + AVFrameRGB: PAVFrame; + FrameBuffer: PByte; + + {$IFDEF UseSWScale} + SoftwareScaleContext: PSwsContext; + {$ENDIF} + + fVideoTex: GLuint; + TexWidth, TexHeight: Cardinal; + + VideoAspect: Real; + VideoTimeBase, VideoTime: Extended; + fLoopTime: Extended; + + EOF: boolean; + Loop: boolean; + + Initialized: boolean; + + procedure Reset(); + function DecodeFrame(var AVPacket: TAVPacket; out pts: double): boolean; + procedure SynchronizeVideo(Frame: PAVFrame; var pts: double); + public + function GetName: String; + + function Init(): boolean; + function Finalize: boolean; + + function Open(const aFileName : string): boolean; // true if succeed + procedure Close; + + procedure Play; + procedure Pause; + procedure Stop; + + procedure SetPosition(Time: real); + function GetPosition: real; + + procedure GetFrame(Time: Extended); + procedure DrawGL(Screen: integer); + end; + +var + FFmpegCore: TMediaCore_FFmpeg; + + +// These are called whenever we allocate a frame buffer. +// We use this to store the global_pts in a frame at the time it is allocated. +function PtsGetBuffer(CodecCtx: PAVCodecContext; Frame: PAVFrame): integer; cdecl; +var + pts: Pint64; + VideoPktPts: Pint64; +begin + Result := avcodec_default_get_buffer(CodecCtx, Frame); + VideoPktPts := CodecCtx^.opaque; + if (VideoPktPts <> nil) then + begin + // Note: we must copy the pts instead of passing a pointer, because the packet + // (and with it the pts) might change before a frame is returned by av_decode_video. + pts := av_malloc(sizeof(int64)); + pts^ := VideoPktPts^; + Frame^.opaque := pts; + end; +end; + +procedure PtsReleaseBuffer(CodecCtx: PAVCodecContext; Frame: PAVFrame); cdecl; +begin + if (Frame <> nil) then + av_freep(@Frame^.opaque); + avcodec_default_release_buffer(CodecCtx, Frame); +end; + + +{*------------------------------------------------------------------------------ + * TVideoPlayback_ffmpeg + *------------------------------------------------------------------------------} + +function TVideoPlayback_FFmpeg.GetName: String; +begin + result := 'FFmpeg_Video'; +end; + +function TVideoPlayback_FFmpeg.Init(): boolean; +begin + Result := true; + + if (Initialized) then + Exit; + Initialized := true; + + FFmpegCore := TMediaCore_FFmpeg.GetInstance(); + + Reset(); + av_register_all(); + glGenTextures(1, PGLuint(@fVideoTex)); +end; + +function TVideoPlayback_FFmpeg.Finalize(): boolean; +begin + Close(); + glDeleteTextures(1, PGLuint(@fVideoTex)); + Result := true; +end; + +procedure TVideoPlayback_FFmpeg.Reset(); +begin + // close previously opened video + Close(); + + fVideoOpened := False; + fVideoPaused := False; + VideoTimeBase := 0; + VideoTime := 0; + VideoStream := nil; + VideoStreamIndex := -1; + + EOF := false; + + // TODO: do we really want this by default? + Loop := true; + fLoopTime := 0; +end; + +function TVideoPlayback_FFmpeg.Open(const aFileName : string): boolean; // true if succeed +var + errnum: Integer; + AudioStreamIndex: integer; +begin + Result := false; + + Reset(); + + errnum := av_open_input_file(VideoFormatContext, PChar(aFileName), nil, 0, nil); + if (errnum <> 0) then + begin + Log.LogError('Failed to open file "'+aFileName+'" ('+FFmpegCore.GetErrorString(errnum)+')'); + Exit; + end; + + // update video info + if (av_find_stream_info(VideoFormatContext) < 0) then + begin + Log.LogError('No stream info found', 'TVideoPlayback_ffmpeg.Open'); + Close(); + Exit; + end; + Log.LogInfo('VideoStreamIndex : ' + inttostr(VideoStreamIndex), 'TVideoPlayback_ffmpeg.Open'); + + // find video stream + FFmpegCore.FindStreamIDs(VideoFormatContext, VideoStreamIndex, AudioStreamIndex); + if (VideoStreamIndex < 0) then + begin + Log.LogError('No video stream found', 'TVideoPlayback_ffmpeg.Open'); + Close(); + Exit; + end; + + VideoStream := VideoFormatContext^.streams[VideoStreamIndex]; + VideoCodecContext := VideoStream^.codec; + + VideoCodec := avcodec_find_decoder(VideoCodecContext^.codec_id); + if (VideoCodec = nil) then + begin + Log.LogError('No matching codec found', 'TVideoPlayback_ffmpeg.Open'); + Close(); + Exit; + end; + + // set debug options + VideoCodecContext^.debug_mv := 0; + VideoCodecContext^.debug := 0; + + // detect bug-workarounds automatically + VideoCodecContext^.workaround_bugs := FF_BUG_AUTODETECT; + // error resilience strategy (careful/compliant/agressive/very_aggressive) + //VideoCodecContext^.error_resilience := FF_ER_CAREFUL; //FF_ER_COMPLIANT; + // allow non spec compliant speedup tricks. + //VideoCodecContext^.flags2 := VideoCodecContext^.flags2 or CODEC_FLAG2_FAST; + + // Note: avcodec_open() and avcodec_close() are not thread-safe and will + // fail if called concurrently by different threads. + FFmpegCore.LockAVCodec(); + try + errnum := avcodec_open(VideoCodecContext, VideoCodec); + finally + FFmpegCore.UnlockAVCodec(); + end; + if (errnum < 0) then + begin + Log.LogError('No matching codec found', 'TVideoPlayback_ffmpeg.Open'); + Close(); + Exit; + end; + + // register custom callbacks for pts-determination + VideoCodecContext^.get_buffer := PtsGetBuffer; + VideoCodecContext^.release_buffer := PtsReleaseBuffer; + + {$ifdef DebugDisplay} + DebugWriteln('Found a matching Codec: '+ VideoCodecContext^.Codec.Name + sLineBreak + + sLineBreak + + ' Width = '+inttostr(VideoCodecContext^.width) + + ', Height='+inttostr(VideoCodecContext^.height) + sLineBreak + + ' Aspect : '+inttostr(VideoCodecContext^.sample_aspect_ratio.num) + '/' + + inttostr(VideoCodecContext^.sample_aspect_ratio.den) + sLineBreak + + ' Framerate : '+inttostr(VideoCodecContext^.time_base.num) + '/' + + inttostr(VideoCodecContext^.time_base.den)); + {$endif} + + // allocate space for decoded frame and rgb frame + AVFrame := avcodec_alloc_frame(); + AVFrameRGB := avcodec_alloc_frame(); + FrameBuffer := av_malloc(avpicture_get_size(PIXEL_FMT_FFMPEG, + VideoCodecContext^.width, VideoCodecContext^.height)); + + if ((AVFrame = nil) or (AVFrameRGB = nil) or (FrameBuffer = nil)) then + begin + Log.LogError('Failed to allocate buffers', 'TVideoPlayback_ffmpeg.Open'); + Close(); + Exit; + end; + + // TODO: pad data for OpenGL to GL_UNPACK_ALIGNMENT + // (otherwise video will be distorted if width/height is not a multiple of the alignment) + errnum := avpicture_fill(PAVPicture(AVFrameRGB), FrameBuffer, PIXEL_FMT_FFMPEG, + VideoCodecContext^.width, VideoCodecContext^.height); + if (errnum < 0) then + begin + Log.LogError('avpicture_fill failed: ' + FFmpegCore.GetErrorString(errnum), 'TVideoPlayback_ffmpeg.Open'); + Close(); + Exit; + end; + + // calculate some information for video display + VideoAspect := av_q2d(VideoCodecContext^.sample_aspect_ratio); + if (VideoAspect = 0) then + VideoAspect := VideoCodecContext^.width / + VideoCodecContext^.height + else + VideoAspect := VideoAspect * VideoCodecContext^.width / + VideoCodecContext^.height; + + VideoTimeBase := 1/av_q2d(VideoStream^.r_frame_rate); + + // hack to get reasonable timebase (for divx and others) + if (VideoTimeBase < 0.02) then // 0.02 <-> 50 fps + begin + VideoTimeBase := av_q2d(VideoStream^.r_frame_rate); + while (VideoTimeBase > 50) do + VideoTimeBase := VideoTimeBase/10; + VideoTimeBase := 1/VideoTimeBase; + end; + + Log.LogInfo('VideoTimeBase: ' + floattostr(VideoTimeBase), 'TVideoPlayback_ffmpeg.Open'); + Log.LogInfo('Framerate: '+inttostr(floor(1/VideoTimeBase))+'fps', 'TVideoPlayback_ffmpeg.Open'); + + {$IFDEF UseSWScale} + // if available get a SWScale-context -> faster than the deprecated img_convert(). + // SWScale has accelerated support for PIX_FMT_RGB32/PIX_FMT_BGR24/PIX_FMT_BGR565/PIX_FMT_BGR555. + // Note: PIX_FMT_RGB32 is a BGR- and not an RGB-format (maybe a bug)!!! + // The BGR565-formats (GL_UNSIGNED_SHORT_5_6_5) is way too slow because of its + // bad OpenGL support. The BGR formats have MMX(2) implementations but no speed-up + // could be observed in comparison to the RGB versions. + SoftwareScaleContext := sws_getContext( + VideoCodecContext^.width, VideoCodecContext^.height, + integer(VideoCodecContext^.pix_fmt), + VideoCodecContext^.width, VideoCodecContext^.height, + integer(PIXEL_FMT_FFMPEG), + SWS_FAST_BILINEAR, nil, nil, nil); + if (SoftwareScaleContext = nil) then + begin + Log.LogError('Failed to get swscale context', 'TVideoPlayback_ffmpeg.Open'); + Close(); + Exit; + end; + {$ENDIF} + + TexWidth := Round(Power(2, Ceil(Log2(VideoCodecContext^.width)))); + TexHeight := Round(Power(2, Ceil(Log2(VideoCodecContext^.height)))); + + // we retrieve a texture just once with glTexImage2D and update it with glTexSubImage2D later. + // Benefits: glTexSubImage2D is faster and supports non-power-of-two widths/height. + glBindTexture(GL_TEXTURE_2D, fVideoTex); + glTexEnvi(GL_TEXTURE_2D, GL_TEXTURE_ENV_MODE, GL_REPLACE); + glTexImage2D(GL_TEXTURE_2D, 0, 3, TexWidth, TexHeight, 0, + PIXEL_FMT_OPENGL, GL_UNSIGNED_BYTE, nil); + glTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_MAG_FILTER, GL_LINEAR); + glTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_MIN_FILTER, GL_LINEAR); + + + fVideoOpened := True; + + Result := true; +end; + +procedure TVideoPlayback_FFmpeg.Close; +begin + if (FrameBuffer <> nil) then + av_free(FrameBuffer); + if (AVFrameRGB <> nil) then + av_free(AVFrameRGB); + if (AVFrame <> nil) then + av_free(AVFrame); + + AVFrame := nil; + AVFrameRGB := nil; + FrameBuffer := nil; + + if (VideoCodecContext <> nil) then + begin + // avcodec_close() is not thread-safe + FFmpegCore.LockAVCodec(); + try + avcodec_close(VideoCodecContext); + finally + FFmpegCore.UnlockAVCodec(); + end; + end; + + if (VideoFormatContext <> nil) then + av_close_input_file(VideoFormatContext); + + VideoCodecContext := nil; + VideoFormatContext := nil; + + fVideoOpened := False; +end; + +procedure TVideoPlayback_FFmpeg.SynchronizeVideo(Frame: PAVFrame; var pts: double); +var + FrameDelay: double; +begin + if (pts <> 0) then + begin + // if we have pts, set video clock to it + VideoTime := pts; + end else + begin + // if we aren't given a pts, set it to the clock + pts := VideoTime; + end; + // update the video clock + FrameDelay := av_q2d(VideoCodecContext^.time_base); + // if we are repeating a frame, adjust clock accordingly + FrameDelay := FrameDelay + Frame^.repeat_pict * (FrameDelay * 0.5); + VideoTime := VideoTime + FrameDelay; +end; + +function TVideoPlayback_FFmpeg.DecodeFrame(var AVPacket: TAVPacket; out pts: double): boolean; +var + FrameFinished: Integer; + VideoPktPts: int64; + pbIOCtx: PByteIOContext; + errnum: integer; +begin + Result := false; + FrameFinished := 0; + + if EOF then + Exit; + + // read packets until we have a finished frame (or there are no more packets) + while (FrameFinished = 0) do + begin + errnum := av_read_frame(VideoFormatContext, AVPacket); + if (errnum < 0) then + begin + // failed to read a frame, check reason + + {$IF (LIBAVFORMAT_VERSION_MAJOR >= 52)} + pbIOCtx := VideoFormatContext^.pb; + {$ELSE} + pbIOCtx := @VideoFormatContext^.pb; + {$IFEND} + + // check for end-of-file (eof is not an error) + if (url_feof(pbIOCtx) <> 0) then + begin + EOF := true; + Exit; + end; + + // check for errors + if (url_ferror(pbIOCtx) <> 0) then + Exit; + + // url_feof() does not detect an EOF for some mov-files (e.g. deluxe.mov) + // so we have to do it this way. + if ((VideoFormatContext^.file_size <> 0) and + (pbIOCtx^.pos >= VideoFormatContext^.file_size)) then + begin + EOF := true; + Exit; + end; + + // no error -> wait for user input + SDL_Delay(100); + continue; + end; + + // if we got a packet from the video stream, then decode it + if (AVPacket.stream_index = VideoStreamIndex) then + begin + // save pts to be stored in pFrame in first call of PtsGetBuffer() + VideoPktPts := AVPacket.pts; + VideoCodecContext^.opaque := @VideoPktPts; + + // decode packet + avcodec_decode_video(VideoCodecContext, AVFrame, + frameFinished, AVPacket.data, AVPacket.size); + + // reset opaque data + VideoCodecContext^.opaque := nil; + + // update pts + if (AVPacket.dts <> AV_NOPTS_VALUE) then + begin + pts := AVPacket.dts; + end + else if ((AVFrame^.opaque <> nil) and + (Pint64(AVFrame^.opaque)^ <> AV_NOPTS_VALUE)) then + begin + pts := Pint64(AVFrame^.opaque)^; + end + else + begin + pts := 0; + end; + pts := pts * av_q2d(VideoStream^.time_base); + + // synchronize on each complete frame + if (frameFinished <> 0) then + SynchronizeVideo(AVFrame, pts); + end; + + // free the packet from av_read_frame + av_free_packet( @AVPacket ); + end; + + Result := true; +end; + +procedure TVideoPlayback_FFmpeg.GetFrame(Time: Extended); +var + AVPacket: TAVPacket; + errnum: Integer; + myTime: Extended; + TimeDifference: Extended; + DropFrameCount: Integer; + pts: double; + i: Integer; +const + FRAME_DROPCOUNT = 3; +begin + if not fVideoOpened then + Exit; + + if fVideoPaused then + Exit; + + // current time, relative to last loop (if any) + myTime := Time - fLoopTime; + // time since the last frame was returned + TimeDifference := myTime - VideoTime; + + {$IFDEF DebugDisplay} + DebugWriteln('Time: '+inttostr(floor(Time*1000)) + sLineBreak + + 'VideoTime: '+inttostr(floor(VideoTime*1000)) + sLineBreak + + 'TimeBase: '+inttostr(floor(VideoTimeBase*1000)) + sLineBreak + + 'TimeDiff: '+inttostr(floor(TimeDifference*1000))); + {$endif} + + // check if a new frame is needed + if (VideoTime <> 0) and (TimeDifference < VideoTimeBase) then + begin + {$ifdef DebugFrames} + // frame delay debug display + GoldenRec.Spawn(200,15,1,16,0,-1,ColoredStar,$00ff00); + {$endif} + + {$IFDEF DebugDisplay} + DebugWriteln('not getting new frame' + sLineBreak + + 'Time: '+inttostr(floor(Time*1000)) + sLineBreak + + 'VideoTime: '+inttostr(floor(VideoTime*1000)) + sLineBreak + + 'TimeBase: '+inttostr(floor(VideoTimeBase*1000)) + sLineBreak + + 'TimeDiff: '+inttostr(floor(TimeDifference*1000))); + {$endif} + + // we do not need a new frame now + Exit; + end; + + // update video-time to the next frame + VideoTime := VideoTime + VideoTimeBase; + TimeDifference := myTime - VideoTime; + + // check if we have to skip frames + if (TimeDifference >= FRAME_DROPCOUNT*VideoTimeBase) then + begin + {$IFDEF DebugFrames} + //frame drop debug display + GoldenRec.Spawn(200,55,1,16,0,-1,ColoredStar,$ff0000); + {$ENDIF} + {$IFDEF DebugDisplay} + DebugWriteln('skipping frames' + sLineBreak + + 'TimeBase: '+inttostr(floor(VideoTimeBase*1000)) + sLineBreak + + 'TimeDiff: '+inttostr(floor(TimeDifference*1000))); + {$endif} + + // update video-time + DropFrameCount := Trunc(TimeDifference / VideoTimeBase); + VideoTime := VideoTime + DropFrameCount*VideoTimeBase; + + // skip half of the frames, this is much smoother than to skip all at once + for i := 1 to DropFrameCount (*div 2*) do + DecodeFrame(AVPacket, pts); + end; + + {$IFDEF VideoBenchmark} + Log.BenchmarkStart(15); + {$ENDIF} + + if (not DecodeFrame(AVPacket, pts)) then + begin + if Loop then + begin + // Record the time we looped. This is used to keep the loops in time. otherwise they speed + SetPosition(0); + fLoopTime := Time; + end; + Exit; + end; + + // TODO: support for pan&scan + //if (AVFrame.pan_scan <> nil) then + //begin + // Writeln(Format('PanScan: %d/%d', [AVFrame.pan_scan.width, AVFrame.pan_scan.height])); + //end; + + // otherwise we convert the pixeldata from YUV to RGB + {$IFDEF UseSWScale} + errnum := sws_scale(SoftwareScaleContext, @(AVFrame.data), @(AVFrame.linesize), + 0, VideoCodecContext^.Height, + @(AVFrameRGB.data), @(AVFrameRGB.linesize)); + {$ELSE} + errnum := img_convert(PAVPicture(AVFrameRGB), PIXEL_FMT_FFMPEG, + PAVPicture(AVFrame), VideoCodecContext^.pix_fmt, + VideoCodecContext^.width, VideoCodecContext^.height); + {$ENDIF} + + if (errnum < 0) then + begin + Log.LogError('Image conversion failed', 'TVideoPlayback_ffmpeg.GetFrame'); + Exit; + end; + + {$IFDEF VideoBenchmark} + Log.BenchmarkEnd(15); + Log.BenchmarkStart(16); + {$ENDIF} + + // TODO: data is not padded, so we will need to tell OpenGL. + // Or should we add padding with avpicture_fill? (check which one is faster) + //glPixelStorei(GL_UNPACK_ALIGNMENT, 1); + + glBindTexture(GL_TEXTURE_2D, fVideoTex); + glTexSubImage2D(GL_TEXTURE_2D, 0, 0, 0, + VideoCodecContext^.width, VideoCodecContext^.height, + PIXEL_FMT_OPENGL, GL_UNSIGNED_BYTE, AVFrameRGB^.data[0]); + + {$ifdef DebugFrames} + //frame decode debug display + GoldenRec.Spawn(200, 35, 1, 16, 0, -1, ColoredStar, $ffff00); + {$endif} + + {$IFDEF VideoBenchmark} + Log.BenchmarkEnd(16); + Log.LogBenchmark('FFmpeg', 15); + Log.LogBenchmark('Texture', 16); + {$ENDIF} +end; + +procedure TVideoPlayback_FFmpeg.DrawGL(Screen: integer); +var + TexVideoRightPos, TexVideoLowerPos: Single; + ScreenLeftPos, ScreenRightPos: Single; + ScreenUpperPos, ScreenLowerPos: Single; + ScaledVideoWidth, ScaledVideoHeight: Single; + ScreenMidPosX, ScreenMidPosY: Single; + ScreenAspect: Single; +begin + // have a nice black background to draw on (even if there were errors opening the vid) + if (Screen = 1) then + begin + glClearColor(0, 0, 0, 0); + glClear(GL_COLOR_BUFFER_BIT or GL_DEPTH_BUFFER_BIT); + end; + + // exit if there's nothing to draw + if (not fVideoOpened) then + Exit; + + {$IFDEF VideoBenchmark} + Log.BenchmarkStart(15); + {$ENDIF} + + // TODO: add a SetAspectCorrectionMode() function so we can switch + // aspect correction. The screens video backgrounds look very ugly with aspect + // correction because of the white bars at the top and bottom. + + ScreenAspect := ScreenW / ScreenH; + ScaledVideoWidth := RenderW; + ScaledVideoHeight := RenderH * ScreenAspect/VideoAspect; + + // Note: Scaling the width does not look good because the video might contain + // black borders at the top already + //ScaledVideoHeight := RenderH; + //ScaledVideoWidth := RenderW * VideoAspect/ScreenAspect; + + // center the video + ScreenMidPosX := RenderW/2; + ScreenMidPosY := RenderH/2; + ScreenLeftPos := ScreenMidPosX - ScaledVideoWidth/2; + ScreenRightPos := ScreenMidPosX + ScaledVideoWidth/2; + ScreenUpperPos := ScreenMidPosY - ScaledVideoHeight/2; + ScreenLowerPos := ScreenMidPosY + ScaledVideoHeight/2; + // the video-texture contains empty borders because its width and height must be + // a power of 2. So we have to determine the texture coords of the video. + TexVideoRightPos := VideoCodecContext^.width / TexWidth; + TexVideoLowerPos := VideoCodecContext^.height / TexHeight; + + // we could use blending for brightness control, but do we need this? + glDisable(GL_BLEND); + + // TODO: disable other stuff like lightning, etc. + + glEnable(GL_TEXTURE_2D); + glBindTexture(GL_TEXTURE_2D, fVideoTex); + glColor3f(1, 1, 1); + glBegin(GL_QUADS); + // upper-left coord + glTexCoord2f(0, 0); + glVertex2f(ScreenLeftPos, ScreenUpperPos); + // lower-left coord + glTexCoord2f(0, TexVideoLowerPos); + glVertex2f(ScreenLeftPos, ScreenLowerPos); + // lower-right coord + glTexCoord2f(TexVideoRightPos, TexVideoLowerPos); + glVertex2f(ScreenRightPos, ScreenLowerPos); + // upper-right coord + glTexCoord2f(TexVideoRightPos, 0); + glVertex2f(ScreenRightPos, ScreenUpperPos); + glEnd; + glDisable(GL_TEXTURE_2D); + + {$IFDEF VideoBenchmark} + Log.BenchmarkEnd(15); + Log.LogBenchmark('DrawGL', 15); + {$ENDIF} + + {$IFDEF Info} + if (fVideoSkipTime+VideoTime+VideoTimeBase < 0) then + begin + glColor4f(0.7, 1, 0.3, 1); + SetFontStyle (1); + SetFontItalic(False); + SetFontSize(9); + SetFontPos (300, 0); + glPrint('Delay due to negative VideoGap'); + glColor4f(1, 1, 1, 1); + end; + {$ENDIF} + + {$IFDEF DebugFrames} + glColor4f(0, 0, 0, 0.2); + glbegin(GL_QUADS); + glVertex2f(0, 0); + glVertex2f(0, 70); + glVertex2f(250, 70); + glVertex2f(250, 0); + glEnd; + + glColor4f(1, 1, 1, 1); + SetFontStyle (1); + SetFontItalic(False); + SetFontSize(9); + SetFontPos (5, 0); + glPrint('delaying frame'); + SetFontPos (5, 20); + glPrint('fetching frame'); + SetFontPos (5, 40); + glPrint('dropping frame'); + {$ENDIF} +end; + +procedure TVideoPlayback_FFmpeg.Play; +begin +end; + +procedure TVideoPlayback_FFmpeg.Pause; +begin + fVideoPaused := not fVideoPaused; +end; + +procedure TVideoPlayback_FFmpeg.Stop; +begin +end; + +procedure TVideoPlayback_FFmpeg.SetPosition(Time: real); +var + SeekFlags: integer; +begin + if not fVideoOpened then + Exit; + + if (Time < 0) then + Time := 0; + + // TODO: handle loop-times + //Time := Time mod VideoDuration; + + // backward seeking might fail without AVSEEK_FLAG_BACKWARD + SeekFlags := AVSEEK_FLAG_ANY; + if (Time < VideoTime) then + SeekFlags := SeekFlags or AVSEEK_FLAG_BACKWARD; + + VideoTime := Time; + EOF := false; + + if (av_seek_frame(VideoFormatContext, VideoStreamIndex, Floor(Time/VideoTimeBase), SeekFlags) < 0) then + begin + Log.LogError('av_seek_frame() failed', 'TVideoPlayback_ffmpeg.SetPosition'); + Exit; + end; + + avcodec_flush_buffers(VideoCodecContext); +end; + +function TVideoPlayback_FFmpeg.GetPosition: real; +begin + // TODO: return video-position in seconds + Result := VideoTime; +end; + +initialization + MediaManager.Add(TVideoPlayback_FFmpeg.Create); + +end. diff --git a/src/media/UVisualizer.pas b/src/media/UVisualizer.pas new file mode 100644 index 00000000..e2125201 --- /dev/null +++ b/src/media/UVisualizer.pas @@ -0,0 +1,442 @@ +unit UVisualizer; + +(* TODO: + * - fix video/visualizer switching + * - use GL_EXT_framebuffer_object for rendering to a separate framebuffer, + * this will prevent plugins from messing up our render-context + * (-> no stack corruption anymore, no need for Save/RestoreOpenGLState()). + * - create a generic (C-compatible) interface for visualization plugins + * - create a visualization plugin manager + * - write a plugin for projectM in C/C++ (so we need no wrapper anymore) + *) + +{* Note: + * It would be easier to create a seperate Render-Context (RC) for projectM + * and switch to it when necessary. This can be achieved by pbuffers + * (slow and platform specific) or the OpenGL FramebufferObject (FBO) extension + * (fast and plattform-independent but not supported by older graphic-cards/drivers). + * + * See http://oss.sgi.com/projects/ogl-sample/registry/EXT/framebuffer_object.txt + * + * To support as many cards as possible we will stick to the current dirty + * solution for now even if it is a pain to save/restore projectM's state due + * to bugs etc. + * + * This also restricts us to projectM. As other plug-ins might have different + * needs and bugs concerning the OpenGL state, USDX's state would probably be + * corrupted after the plug-in finshed drawing. + *} + +interface + +{$IFDEF FPC} + {$MODE DELPHI} +{$ENDIF} + +{$I switches.inc} + +uses + SDL, + UGraphicClasses, + textgl, + math, + gl, + SysUtils, + UIni, + projectM, + UMusic; + +implementation + +uses + UGraphic, + UMain, + UConfig, + ULog; + +{$IF PROJECTM_VERSION < 1000000} // < 1.0 +// Initialization data used on projectM 0.9x creation. +// Since projectM 1.0 this data is passed via the config-file. +const + meshX = 32; + meshY = 24; + fps = 30; + textureSize = 512; +{$IFEND} + +type + TVideoPlayback_ProjectM = class( TInterfacedObject, IVideoPlayback, IVideoVisualization ) + private + pm: TProjectM; + ProjectMPath : string; + Initialized: boolean; + + VisualizerStarted: boolean; + VisualizerPaused: boolean; + + VisualTex: GLuint; + PCMData: TPCMData; + RndPCMcount: integer; + + projMatrix: array[0..3, 0..3] of GLdouble; + texMatrix: array[0..3, 0..3] of GLdouble; + + procedure VisualizerStart; + procedure VisualizerStop; + + procedure VisualizerTogglePause; + + function GetRandomPCMData(var data: TPCMData): Cardinal; + + procedure SaveOpenGLState(); + procedure RestoreOpenGLState(); + + public + function GetName: String; + + function Init(): boolean; + function Finalize(): boolean; + + function Open(const aFileName : string): boolean; // true if succeed + procedure Close; + + procedure Play; + procedure Pause; + procedure Stop; + + procedure SetPosition(Time: real); + function GetPosition: real; + + procedure GetFrame(Time: Extended); + procedure DrawGL(Screen: integer); + end; + + +function TVideoPlayback_ProjectM.GetName: String; +begin + Result := 'ProjectM'; +end; + +function TVideoPlayback_ProjectM.Init(): boolean; +begin + Result := true; + + if (Initialized) then + Exit; + Initialized := true; + + RndPCMcount := 0; + + ProjectMPath := ProjectM_DataDir + PathDelim; + + VisualizerStarted := False; + VisualizerPaused := False; + + {$IFDEF UseTexture} + glGenTextures(1, PglUint(@VisualTex)); + glBindTexture(GL_TEXTURE_2D, VisualTex); + + glTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_MIN_FILTER, GL_LINEAR); + glTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_MAG_FILTER, GL_LINEAR); + {$ENDIF} +end; + +function TVideoPlayback_ProjectM.Finalize(): boolean; +begin + VisualizerStop(); + {$IFDEF UseTexture} + glDeleteTextures(1, PglUint(@VisualTex)); + {$ENDIF} + Result := true; +end; + +function TVideoPlayback_ProjectM.Open(const aFileName : string): boolean; // true if succeed +begin + Result := false; +end; + +procedure TVideoPlayback_ProjectM.Close; +begin + VisualizerStop(); +end; + +procedure TVideoPlayback_ProjectM.Play; +begin + VisualizerStart(); +end; + +procedure TVideoPlayback_ProjectM.Pause; +begin + VisualizerTogglePause(); +end; + +procedure TVideoPlayback_ProjectM.Stop; +begin + VisualizerStop(); +end; + +procedure TVideoPlayback_ProjectM.SetPosition(Time: real); +begin + if assigned(pm) then + pm.RandomPreset(); +end; + +function TVideoPlayback_ProjectM.GetPosition: real; +begin + Result := 0; +end; + +{** + * Saves the current OpenGL state. + * This is necessary to prevent projectM from corrupting USDX's current + * OpenGL state. + * + * The following steps are performed: + * - All attributes are pushed to the attribute-stack + * - Projection-/Texture-matrices are saved + * - Modelview-matrix is pushed to the Modelview-stack + * - the OpenGL error-state (glGetError) is cleared + *} +procedure TVideoPlayback_ProjectM.SaveOpenGLState(); +begin + // save all OpenGL state-machine attributes + glPushAttrib(GL_ALL_ATTRIB_BITS); + + // Note: we do not use glPushMatrix() for the GL_PROJECTION and GL_TEXTURE stacks. + // OpenGL specifies the depth of those stacks to be at least 2 but projectM + // already uses 2 stack-entries so overflows might be possible on older hardware. + // In contrast to this the GL_MODELVIEW stack-size is at least 32, so we can + // use glPushMatrix() for this stack. + + // save projection-matrix + glMatrixMode(GL_PROJECTION); + glGetDoublev(GL_PROJECTION_MATRIX, @projMatrix); + {$IF PROJECTM_VERSION = 1000000} // 1.0, 1.01 + // bugfix: projection-matrix is popped without being pushed first + glPushMatrix(); + {$IFEND} + + // save texture-matrix + glMatrixMode(GL_TEXTURE); + glGetDoublev(GL_TEXTURE_MATRIX, @texMatrix); + + // save modelview-matrix + glMatrixMode(GL_MODELVIEW); + glPushMatrix(); + {$IF PROJECTM_VERSION = 1000000} // 1.0, 1.01 + // bugfix: modelview-matrix is popped without being pushed first + glPushMatrix(); + {$IFEND} + + // reset OpenGL error-state + glGetError(); +end; + +{** + * Restores the OpenGL state saved by SaveOpenGLState() + * and resets the error-state. + *} +procedure TVideoPlayback_ProjectM.RestoreOpenGLState(); +begin + // reset OpenGL error-state + glGetError(); + + // restore projection-matrix + glMatrixMode(GL_PROJECTION); + glLoadMatrixd(@projMatrix); + + // restore texture-matrix + glMatrixMode(GL_TEXTURE); + glLoadMatrixd(@texMatrix); + + // restore modelview-matrix + glMatrixMode(GL_MODELVIEW); + glPopMatrix(); + + // restore all OpenGL state-machine attributes + glPopAttrib(); +end; + +procedure TVideoPlayback_ProjectM.VisualizerStart; +begin + if VisualizerStarted then + Exit; + + // the OpenGL state must be saved before + SaveOpenGLState(); + try + + try + {$IF PROJECTM_VERSION >= 1000000} // >= 1.0 + pm := TProjectM.Create(ProjectMPath + 'config.inp'); + {$ELSE} + pm := TProjectM.Create( + meshX, meshY, fps, textureSize, ScreenW, ScreenH, + ProjectMPath + 'presets', ProjectMPath + 'fonts'); + {$IFEND} + except on E: Exception do + begin + // Create() might fail if the config-file is not found + Log.LogError('TProjectM.Create: ' + E.Message, 'TVideoPlayback_ProjectM.VisualizerStart'); + Exit; + end; + end; + + // initialize OpenGL + pm.ResetGL(ScreenW, ScreenH); + // skip projectM default-preset + pm.RandomPreset(); + // projectM >= 1.0 uses the OpenGL FramebufferObject (FBO) extension. + // Unfortunately it does NOT reset the framebuffer-context after + // TProjectM.Create. Either glBindFramebufferEXT(GL_FRAMEBUFFER_EXT, 0) for + // a manual reset or TProjectM.RenderFrame() must be called. + // We use the latter so we do not need to load the FBO extension in USDX. + pm.RenderFrame(); + + VisualizerStarted := True; + finally + RestoreOpenGLState(); + end; +end; + +procedure TVideoPlayback_ProjectM.VisualizerStop; +begin + if VisualizerStarted then + begin + VisualizerStarted := False; + FreeAndNil(pm); + end; +end; + +procedure TVideoPlayback_ProjectM.VisualizerTogglePause; +begin + VisualizerPaused := not VisualizerPaused; +end; + +procedure TVideoPlayback_ProjectM.GetFrame(Time: Extended); +var + nSamples: cardinal; + stackDepth: Integer; +begin + if not VisualizerStarted then + Exit; + + if VisualizerPaused then + Exit; + + // get audio data + nSamples := AudioPlayback.GetPCMData(PcmData); + + // generate some data if non is available + if (nSamples = 0) then + nSamples := GetRandomPCMData(PcmData); + + // send audio-data to projectM + if (nSamples > 0) then + pm.AddPCM16Data(PSmallInt(@PcmData), nSamples); + + // store OpenGL state (might be messed up otherwise) + SaveOpenGLState(); + try + // setup projectM's OpenGL state + pm.ResetGL(ScreenW, ScreenH); + + // let projectM render a frame + pm.RenderFrame(); + + {$IFDEF UseTexture} + glBindTexture(GL_TEXTURE_2D, VisualTex); + glFlush(); + glCopyTexImage2D(GL_TEXTURE_2D, 0, GL_RGB, 0, 0, VisualWidth, VisualHeight, 0); + {$ENDIF} + finally + // restore USDX OpenGL state + RestoreOpenGLState(); + end; + + // discard projectM's depth buffer information (avoid overlay) + glClear(GL_DEPTH_BUFFER_BIT); +end; + +{** + * Draws the current frame to screen. + * TODO: this is not used yet. Data is directly drawn on GetFrame(). + *} +procedure TVideoPlayback_ProjectM.DrawGL(Screen: integer); +begin + {$IFDEF UseTexture} + // have a nice black background to draw on + if (Screen = 1) then + begin + glClearColor(0, 0, 0, 0); + glClear(GL_COLOR_BUFFER_BIT or GL_DEPTH_BUFFER_BIT); + end; + + // exit if there's nothing to draw + if not VisualizerStarted then + Exit; + + // setup display + glMatrixMode(GL_PROJECTION); + glPushMatrix(); + glLoadIdentity(); + gluOrtho2D(0, 1, 0, 1); + glMatrixMode(GL_MODELVIEW); + glPushMatrix(); + glLoadIdentity(); + + glEnable(GL_BLEND); + glEnable(GL_TEXTURE_2D); + glTexEnvf(GL_TEXTURE_ENV, GL_TEXTURE_ENV_MODE, GL_REPLACE); + glBindTexture(GL_TEXTURE_2D, VisualTex); + glColor4f(1, 1, 1, 1); + + // draw projectM frame + glBegin(GL_QUADS); + glTexCoord2f(0, 0); glVertex2f(0, 0); + glTexCoord2f(1, 0); glVertex2f(1, 0); + glTexCoord2f(1, 1); glVertex2f(1, 1); + glTexCoord2f(0, 1); glVertex2f(0, 1); + glEnd(); + + glDisable(GL_TEXTURE_2D); + glDisable(GL_BLEND); + + // restore state + glMatrixMode(GL_PROJECTION); + glPopMatrix(); + glMatrixMode(GL_MODELVIEW); + glPopMatrix(); + {$ENDIF} +end; + +{** + * Produces random "sound"-data in case no audio-data is available. + * Otherwise the visualization will look rather boring. + *} +function TVideoPlayback_ProjectM.GetRandomPCMData(var data: TPCMData): Cardinal; +var + i: integer; +begin + // Produce some fake PCM data + if (RndPCMcount mod 500 = 0) then + begin + FillChar(data, SizeOf(TPCMData), 0); + end + else + begin + for i := 0 to 511 do + begin + data[i][0] := Random(High(Word)+1); + data[i][1] := Random(High(Word)+1); + end; + end; + Inc(RndPCMcount); + Result := 512; +end; + + +initialization + MediaManager.Add(TVideoPlayback_ProjectM.Create); + +end. |