e1b42d183a
- sb16 parameters stored in the device class and runtime parameter handler added - wx: ParamDialog can currently handle only dependency lists of bool parameters correctly (TODO: add support for numeric parameters)
402 lines
12 KiB
C++
402 lines
12 KiB
C++
/////////////////////////////////////////////////////////////////////////
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// $Id: soundosx.cc,v 1.7 2006-03-03 20:29:50 vruppert Exp $
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/////////////////////////////////////////////////////////////////////////
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// This file (SOUNDOSX.CC) written and donated by Brian Huffman
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#ifdef PARANOID
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#include <MacTypes.h>
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#else
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#define Float32 KLUDGE_Float32
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#define Float64 KLUDGE_Float64
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#endif
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#define NO_DEVICE_INCLUDES
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#include "iodev.h"
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#define BX_SOUNDLOW
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#include "sb16.h"
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#undef Float32
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#undef Float64
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#if defined(macintosh) && BX_SUPPORT_SB16
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#define LOG_THIS bx_devices.pluginSB16Device->
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#include "soundosx.h"
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#if BX_WITH_MACOS
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#include <QuickTimeMusic.h>
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#else
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#include <CoreAudio/CoreAudio.h>
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#include <AudioUnit/AudioUnit.h>
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#include <AudioToolbox/DefaultAudioOutput.h>
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#include <AudioToolbox/AudioConverter.h>
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#include <AudioToolbox/AUGraph.h>
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#include <QuickTime/QuickTimeMusic.h>
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#endif
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#include <string.h>
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#ifdef BX_SOUND_OSX_use_converter
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OSStatus MyRenderer (void *inRefCon, AudioUnitRenderActionFlags inActionFlags,
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const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, AudioBuffer *ioData);
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OSStatus MyACInputProc (AudioConverterRef inAudioConverter, UInt32* outDataSize,
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void** outData, void* inUserData);
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#endif
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// Global variables
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#ifdef BX_SOUND_OSX_use_converter
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AUGraph MidiGraph;
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AudioUnit synthUnit;
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#endif
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#ifdef BX_SOUND_OSX_use_quicktime
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SndChannelPtr WaveChannel;
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ExtSoundHeader WaveInfo;
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ExtSoundHeader WaveHeader[BX_SOUND_OSX_NBUF];
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#endif
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#ifdef BX_SOUND_OSX_use_converter
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AudioUnit WaveOutputUnit = NULL;
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AudioConverterRef WaveConverter = NULL;
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#endif
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bx_sound_osx_c::bx_sound_osx_c(bx_sb16_c *sb16)
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:bx_sound_output_c(sb16)
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{
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this->sb16 = sb16;
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MidiOpen = 0;
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WaveOpen = 0;
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head = 0;
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tail = 0;
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for (int i=0; i<BX_SOUND_OSX_NBUF; i++)
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WaveLength[i] = 0;
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}
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bx_sound_osx_c::~bx_sound_osx_c()
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{
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// nothing for now
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}
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int bx_sound_osx_c::midiready()
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{
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return BX_SOUND_OUTPUT_OK;
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}
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int bx_sound_osx_c::openmidioutput(char *device)
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{
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#ifdef BX_SOUND_OSX_use_converter
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ComponentDescription description;
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AUNode synthNode, outputNode;
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// Create the graph
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NewAUGraph (&MidiGraph);
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// Open the DLS Synth
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description.componentType = kAudioUnitComponentType;
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description.componentSubType = kAudioUnitSubType_MusicDevice;
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description.componentManufacturer = kAudioUnitID_DLSSynth;
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description.componentFlags = 0;
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description.componentFlagsMask = 0;
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AUGraphNewNode (MidiGraph, &description, 0, NULL, &synthNode);
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// Open the output device
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description.componentType = kAudioUnitComponentType;
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description.componentSubType = kAudioUnitSubType_Output;
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description.componentManufacturer = kAudioUnitID_DefaultOutput;
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description.componentFlags = 0;
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description.componentFlagsMask = 0;
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AUGraphNewNode (MidiGraph, &description, 0, NULL, &outputNode);
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// Connect the devices up
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AUGraphConnectNodeInput (MidiGraph, synthNode, 1, outputNode, 0);
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AUGraphUpdate (MidiGraph, NULL);
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// Open and initialize the audio units
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AUGraphOpen (MidiGraph);
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AUGraphInitialize (MidiGraph);
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// Turn off the reverb on the synth
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AUGraphGetNodeInfo (MidiGraph, synthNode, NULL, NULL, NULL, &synthUnit);
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UInt32 usesReverb = 0;
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AudioUnitSetProperty (synthUnit, kMusicDeviceProperty_UsesInternalReverb,
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kAudioUnitScope_Global, 0, &usesReverb, sizeof (usesReverb));
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// Start playing
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AUGraphStart (MidiGraph);
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#endif
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WRITELOG( WAVELOG(4), "openmidioutput(%s)", device);
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MidiOpen = 1;
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return BX_SOUND_OUTPUT_OK;
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}
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int bx_sound_osx_c::sendmidicommand(int delta, int command, int length, Bit8u data[])
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{
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WRITELOG( WAVELOG(5), "sendmidicommand(%i,%02x,%i)", delta, command, length);
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if (!MidiOpen) return BX_SOUND_OUTPUT_ERR;
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#ifdef BX_SOUND_OSX_use_converter
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if (length <= 2) {
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Bit8u arg1 = (length >=1) ? data[0] : 0;
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Bit8u arg2 = (length >=2) ? data[1] : 0;
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MusicDeviceMIDIEvent (synthUnit, command, arg1, arg2, delta);
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}
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else {
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MusicDeviceSysEx (synthUnit, data, length);
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}
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#endif
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return BX_SOUND_OUTPUT_OK;
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}
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int bx_sound_osx_c::closemidioutput()
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{
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WRITELOG( WAVELOG(4), "closemidioutput()");
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MidiOpen = 0;
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#ifdef BX_SOUND_OSX_use_converter
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AUGraphStop (MidiGraph);
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AUGraphClose (MidiGraph);
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#endif
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return BX_SOUND_OUTPUT_OK;
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}
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#ifdef BX_SOUND_OSX_use_quicktime
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#if BX_WITH_MACOS
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pascal
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#endif
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void WaveCallbackProc (SndChannelPtr chan, SndCommand *cmd)
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{
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// a new buffer is available, so increment tail pointer
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int *tail = (int *) (cmd->param2);
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(*tail)++;
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}
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#endif
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int bx_sound_osx_c::openwaveoutput(char *device)
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{
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OSStatus err;
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WRITELOG( WAVELOG(4), "openwaveoutput(%s)", device);
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// open the default output unit
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#ifdef BX_SOUND_OSX_use_quicktime
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err = SndNewChannel (&WaveChannel, sampledSynth, 0, NewSndCallBackUPP(WaveCallbackProc));
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if (err != noErr) return BX_SOUND_OUTPUT_ERR;
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#endif
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#ifdef BX_SOUND_OSX_use_converter
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err = OpenDefaultAudioOutput (&WaveOutputUnit);
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if (err != noErr) return BX_SOUND_OUTPUT_ERR;
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AudioUnitInitialize (WaveOutputUnit);
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// Set up a callback function to generate output to the output unit
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AudioUnitInputCallback input;
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input.inputProc = MyRenderer;
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input.inputProcRefCon = (void *) this;
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AudioUnitSetProperty (WaveOutputUnit, kAudioUnitProperty_SetInputCallback,
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kAudioUnitScope_Global, 0, &input, sizeof(input));
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#endif
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WaveOpen = 1;
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return BX_SOUND_OUTPUT_OK;
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}
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int bx_sound_osx_c::startwaveplayback(int frequency, int bits, int stereo, int format)
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{
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#ifdef BX_SOUND_OSX_use_converter
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static int oldfreq, oldbits, oldstereo, oldformat;
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AudioStreamBasicDescription srcFormat, dstFormat;
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UInt32 formatSize = sizeof(AudioStreamBasicDescription);
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#endif
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WRITELOG( WAVELOG(4), "startwaveplayback(%d, %d, %d, %x)", frequency, bits, stereo, format);
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#ifdef BX_SOUND_OSX_use_quicktime
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WaveInfo.samplePtr = NULL;
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WaveInfo.numChannels = stereo ? 2 : 1;
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WaveInfo.sampleRate = frequency << 16; // sampleRate is a 16.16 fixed-point value
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WaveInfo.loopStart = 0;
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WaveInfo.loopEnd = 0;
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WaveInfo.encode = extSH; // WaveInfo has type ExtSoundHeader
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WaveInfo.baseFrequency = 1; // not sure what means. It's only a Uint8.
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WaveInfo.numFrames = 0;
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//WaveInfo.AIFFSampleRate = frequency; // frequency as float80
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WaveInfo.markerChunk = NULL;
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WaveInfo.instrumentChunks = NULL;
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WaveInfo.AESRecording = NULL;
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WaveInfo.sampleSize = bits * WaveInfo.numChannels;
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#endif
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#ifdef BX_SOUND_OSX_use_converter
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if ( (frequency == oldfreq) &&
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(bits == oldbits) &&
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(stereo == oldstereo) &&
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(format == oldformat) )
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return BX_SOUND_OUTPUT_OK; // nothing to do
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oldfreq = frequency;
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oldbits = bits;
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oldstereo = stereo;
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oldformat = format;
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// update the source audio format
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UInt32 bytes = bits / 8;
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UInt32 channels = stereo ? 2 : 1;
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srcFormat.mSampleRate = (Float64) frequency;
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srcFormat.mFormatID = kAudioFormatLinearPCM;
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srcFormat.mFormatFlags = kLinearPCMFormatFlagIsPacked;
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if (format & 1) srcFormat.mFormatFlags |= kLinearPCMFormatFlagIsSignedInteger;
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srcFormat.mBytesPerPacket = channels * bytes;
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srcFormat.mFramesPerPacket = 1;
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srcFormat.mBytesPerFrame = channels * bytes;
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srcFormat.mChannelsPerFrame = channels;
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srcFormat.mBitsPerChannel = bytes * 8;
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if (WavePlaying) AudioOutputUnitStop (WaveOutputUnit);
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if (WaveConverter) AudioConverterDispose (WaveConverter);
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AudioUnitGetProperty (WaveOutputUnit, kAudioUnitProperty_StreamFormat,
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kAudioUnitScope_Output, 0, &dstFormat, &formatSize);
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AudioConverterNew (&srcFormat, &dstFormat, &WaveConverter);
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if (srcFormat.mChannelsPerFrame == 1 && dstFormat.mChannelsPerFrame == 2) {
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// map single-channel input to both output channels
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SInt32 map[2] = {0,0};
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AudioConverterSetProperty (WaveConverter, kAudioConverterChannelMap,
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sizeof(map), (void*) map);
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}
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if (WavePlaying) AudioOutputUnitStart (WaveOutputUnit);
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#endif
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return BX_SOUND_OUTPUT_OK;
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}
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int bx_sound_osx_c::waveready()
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{
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// HACK: the -4 is to keep from overwriting buffers that
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// have been sent, but possibly not yet played. There
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// should be a better way of doing this.
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if (WaveOpen && (head - tail < BX_SOUND_OSX_NBUF-4)) {
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return BX_SOUND_OUTPUT_OK;
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}
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else {
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#ifdef BX_SOUND_OSX_use_converter
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// If buffer is full, make sure sound is playing
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if (WaveOutputUnit && !WavePlaying) {
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AudioOutputUnitStart (WaveOutputUnit);
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WavePlaying = 1;
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}
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#endif
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return BX_SOUND_OUTPUT_ERR;
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}
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}
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int bx_sound_osx_c::sendwavepacket(int length, Bit8u data[])
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{
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#ifdef BX_SOUND_OSX_use_quicktime
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SndCommand mySndCommand;
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#endif
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WRITELOG( WAVELOG(4), "sendwavepacket(%d, %p), head=%u", length, data, head);
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// sanity check
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if ((!WaveOpen) || (head - tail >= BX_SOUND_OSX_NBUF))
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return BX_SOUND_OUTPUT_ERR;
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// find next available buffer
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int n = head++ % BX_SOUND_OSX_NBUF;
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// put data in buffer
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memcpy(WaveData[n], data, length);
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WaveLength[n] = length;
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#ifdef BX_SOUND_OSX_use_quicktime
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memcpy(&WaveHeader[n], &WaveInfo, sizeof(WaveInfo));
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WaveHeader[n].samplePtr = (char *) (WaveData[n]);
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WaveHeader[n].numFrames = length * 8 / WaveInfo.sampleSize;
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#endif
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#ifdef BX_SOUND_OSX_use_converter
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// make sure that the sound is playing
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if (!WavePlaying) {
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AudioOutputUnitStart (WaveOutputUnit);
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WavePlaying = 1;
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}
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#endif
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#ifdef BX_SOUND_OSX_use_quicktime
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// queue buffer to play
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mySndCommand.cmd = bufferCmd;
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mySndCommand.param1 = 0;
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mySndCommand.param2 = (long)(&WaveHeader[n]);
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SndDoCommand(WaveChannel, &mySndCommand, TRUE);
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// queue callback for when buffer finishes
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mySndCommand.cmd = callBackCmd;
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mySndCommand.param1 = 0;
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mySndCommand.param2 = (long)(&tail);
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SndDoCommand(WaveChannel, &mySndCommand, TRUE);
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#endif
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return BX_SOUND_OUTPUT_OK;
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}
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int bx_sound_osx_c::stopwaveplayback()
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{
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return BX_SOUND_OUTPUT_OK;
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}
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int bx_sound_osx_c::closewaveoutput()
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{
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#ifdef BX_SOUND_OSX_use_converter
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if (WavePlaying) AudioOutputUnitStop (WaveOutputUnit);
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if (WaveConverter) AudioConverterDispose (WaveConverter);
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if (WaveOutputUnit) CloseComponent (WaveOutputUnit);
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WavePlaying = 0;
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WaveOpen = 0;
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WaveConverter = NULL;
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WaveOutputUnit = NULL;
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#endif
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return BX_SOUND_OUTPUT_OK;
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}
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#ifdef BX_SOUND_OSX_use_converter
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OSStatus MyRenderer (void *inRefCon, AudioUnitRenderActionFlags inActionFlags,
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const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, AudioBuffer *ioData)
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{
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UInt32 size = ioData->mDataByteSize;
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AudioConverterFillBuffer (WaveConverter, MyACInputProc, inRefCon, &size, ioData->mData);
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return noErr;
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}
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OSStatus MyACInputProc (AudioConverterRef inAudioConverter,
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UInt32* outDataSize, void** outData, void* inUserData)
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{
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bx_sound_osx_c *self = (bx_sound_osx_c*) inUserData;
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self->nextbuffer ((int*) outDataSize, outData);
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return noErr;
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}
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void bx_sound_osx_c::nextbuffer (int *outDataSize, void **outData)
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{
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WRITELOG( WAVELOG(4), "nextbuffer(), tail=%u", tail);
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if (head - tail <= 0) {
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*outData = NULL;
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*outDataSize = 0;
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// We are getting behind, so stop the output for now
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AudioOutputUnitStop (WaveOutputUnit);
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WavePlaying = 0;
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}
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else {
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int n = tail % BX_SOUND_OSX_NBUF;
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*outData = (void *) (WaveData[n]);
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*outDataSize = WaveLength[n];
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tail++;
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}
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}
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#endif
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#endif // defined(macintosh)
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