Bochs/bochs/iodev/sound/soundosx.cc
Volker Ruppert 192aa1f369 Added global sound control option in bochsrc and config interface
- added 'sound' option to select the driver and the wave output device
  (see bochsrc sample for details)
- the wave device option of sb16 and es1370 is now only used to set up the
  wave input device (if necessary)
- cleaned up the speaker code to initialize the 'sound' case correctly
2013-06-24 19:19:12 +00:00

641 lines
19 KiB
C++

/////////////////////////////////////////////////////////////////////////
// $Id$
/////////////////////////////////////////////////////////////////////////
//
// Copyright (C) 2004-2011 The Bochs Project
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
// This file (SOUNDOSX.CC) written and donated by Brian Huffman
#ifdef PARANOID
#include <MacTypes.h>
#endif
#include "iodev.h"
#include "soundmod.h"
#include "soundosx.h"
#if defined(macintosh) && BX_SUPPORT_SOUNDLOW
#define LOG_THIS
#if BX_WITH_MACOS
#include <QuickTimeMusic.h>
#else
#include <CoreServices/CoreServices.h>
#include <CoreAudio/CoreAudio.h>
#include <AudioUnit/AudioUnit.h>
#include <AudioToolbox/DefaultAudioOutput.h>
#include <AudioToolbox/AudioConverter.h>
#include <AudioToolbox/AUGraph.h>
#include <QuickTime/QuickTimeMusic.h>
#endif
#include <string.h>
#if (MAC_OS_X_VERSION_MAX_ALLOWED >= 1030)
#define BX_SOUND_OSX_CONVERTER_NEW_API 1
#endif
#ifdef BX_SOUND_OSX_use_converter
#ifndef BX_SOUND_OSX_CONVERTER_NEW_API
OSStatus MyRenderer (void *inRefCon, AudioUnitRenderActionFlags inActionFlags, const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, AudioBuffer *ioData);
OSStatus MyACInputProc (AudioConverterRef inAudioConverter, UInt32* outDataSize, void** outData, void* inUserData);
#else
OSStatus MyRenderer (void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags, const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList *ioData);
OSStatus MyACInputProc (AudioConverterRef inAudioConverter, UInt32 *ioNumberDataPackets, AudioBufferList *ioData, AudioStreamPacketDescription **outDataPacketDescription, void *inUserData);
#endif
#endif
// Global variables
#ifdef BX_SOUND_OSX_use_converter
AUGraph MidiGraph;
AudioUnit synthUnit;
#endif
#ifdef BX_SOUND_OSX_use_quicktime
SndChannelPtr WaveChannel;
ExtSoundHeader WaveInfo;
ExtSoundHeader WaveHeader[BX_SOUND_OSX_NBUF];
#endif
#ifdef BX_SOUND_OSX_use_converter
AudioUnit WaveOutputUnit = NULL;
AudioConverterRef WaveConverter = NULL;
#endif
bx_sound_osx_c::bx_sound_osx_c()
:bx_sound_lowlevel_c()
{
MidiOpen = 0;
WaveOpen = 0;
head = 0;
tail = 0;
for (int i=0; i<BX_SOUND_OSX_NBUF; i++)
WaveLength[i] = 0;
BX_INFO(("Sound output module 'osx' initialized"));
}
bx_sound_osx_c::~bx_sound_osx_c()
{
// nothing for now
}
int bx_sound_osx_c::midiready()
{
return BX_SOUNDLOW_OK;
}
int bx_sound_osx_c::openmidioutput(const char *mididev)
{
#ifdef BX_SOUND_OSX_use_converter
ComponentDescription description;
AUNode synthNode, outputNode;
// Create the graph
NewAUGraph (&MidiGraph);
// Open the DLS Synth
description.componentType = kAudioUnitType_MusicDevice;
description.componentSubType = kAudioUnitSubType_DLSSynth;
description.componentManufacturer = kAudioUnitManufacturer_Apple;
description.componentFlags = 0;
description.componentFlagsMask = 0;
AUGraphNewNode (MidiGraph, &description, 0, NULL, &synthNode);
// Open the output device
description.componentType = kAudioUnitType_Output;
description.componentSubType = kAudioUnitSubType_DefaultOutput;
description.componentManufacturer = kAudioUnitManufacturer_Apple;
description.componentFlags = 0;
description.componentFlagsMask = 0;
AUGraphNewNode (MidiGraph, &description, 0, NULL, &outputNode);
// Connect the devices up
AUGraphConnectNodeInput (MidiGraph, synthNode, 1, outputNode, 0);
AUGraphUpdate (MidiGraph, NULL);
// Open and initialize the audio units
AUGraphOpen (MidiGraph);
AUGraphInitialize (MidiGraph);
// Turn off the reverb on the synth
AUGraphGetNodeInfo (MidiGraph, synthNode, NULL, NULL, NULL, &synthUnit);
UInt32 usesReverb = 0;
AudioUnitSetProperty (synthUnit, kMusicDeviceProperty_UsesInternalReverb,
kAudioUnitScope_Global, 0, &usesReverb, sizeof (usesReverb));
// Start playing
AUGraphStart (MidiGraph);
#endif
BX_DEBUG(("openmidioutput(%s)", mididev));
MidiOpen = 1;
return BX_SOUNDLOW_OK;
}
int bx_sound_osx_c::sendmidicommand(int delta, int command, int length, Bit8u data[])
{
BX_DEBUG(("sendmidicommand(%i,%02x,%i)", delta, command, length));
if (!MidiOpen) return BX_SOUNDLOW_ERR;
#ifdef BX_SOUND_OSX_use_converter
if (length <= 2) {
Bit8u arg1 = (length >=1) ? data[0] : 0;
Bit8u arg2 = (length >=2) ? data[1] : 0;
MusicDeviceMIDIEvent (synthUnit, command, arg1, arg2, delta);
}
else {
MusicDeviceSysEx (synthUnit, data, length);
}
#endif
return BX_SOUNDLOW_OK;
}
int bx_sound_osx_c::closemidioutput()
{
BX_DEBUG(("closemidioutput()"));
MidiOpen = 0;
#ifdef BX_SOUND_OSX_use_converter
AUGraphStop (MidiGraph);
AUGraphClose (MidiGraph);
#endif
return BX_SOUNDLOW_OK;
}
#ifdef BX_SOUND_OSX_use_quicktime
#if BX_WITH_MACOS
pascal
#endif
void WaveCallbackProc (SndChannelPtr chan, SndCommand *cmd)
{
// a new buffer is available, so increment tail pointer
int *tail = (int *) (cmd->param2);
(*tail)++;
}
#endif
int bx_sound_osx_c::openwaveoutput(const char *wavedev)
{
OSStatus err;
BX_DEBUG(("openwaveoutput(%s)", wavedev));
// open the default output unit
#ifdef BX_SOUND_OSX_use_quicktime
err = SndNewChannel (&WaveChannel, sampledSynth, 0, NewSndCallBackUPP(WaveCallbackProc));
if (err != noErr) return BX_SOUNDLOW_ERR;
#endif
#ifdef BX_SOUND_OSX_use_converter
#ifndef BX_SOUND_OSX_CONVERTER_NEW_API
err = OpenDefaultAudioOutput (&WaveOutputUnit);
if (err != noErr) return BX_SOUNDLOW_ERR;
AudioUnitInputCallback input;
input.inputProc = MyRenderer;
input.inputProcRefCon = (void *) this;
AudioUnitSetProperty (WaveOutputUnit, kAudioUnitProperty_SetInputCallback,
kAudioUnitScope_Global, 0, &input, sizeof(input));
#else
ComponentDescription desc;
desc.componentType = kAudioUnitType_Output;
desc.componentSubType = kAudioUnitSubType_DefaultOutput;
desc.componentManufacturer = kAudioUnitManufacturer_Apple;
desc.componentFlags = 0;
desc.componentFlagsMask = 0;
Component c = FindNextComponent(NULL, &desc);
if (c == NULL) {
BX_ERROR(("Core Audio: Unable to find default audio output component\n"));
return BX_SOUNDLOW_ERR;
}
err = OpenAComponent(c, &WaveOutputUnit);
if (err) {
BX_ERROR(("Core Audio: Unable to open audio output (err=%X)\n", (unsigned int)err));
return BX_SOUNDLOW_ERR;
}
AURenderCallbackStruct input;
input.inputProc = MyRenderer;
input.inputProcRefCon = (void *) this;
err = AudioUnitSetProperty (WaveOutputUnit,
kAudioUnitProperty_SetRenderCallback,
kAudioUnitScope_Input, 0, &input, sizeof(input));
if (err) {
BX_ERROR(("Core Audio: AudioUnitSetProperty error (err=%X)\n", (unsigned int)err));
return BX_SOUNDLOW_ERR;
}
#endif
err = AudioUnitInitialize (WaveOutputUnit);
if (err) {
BX_ERROR(("Core Audio: AudioUnitInitialize error (err=%X)\n", (unsigned int)err));
return BX_SOUNDLOW_ERR;
}
#endif
WaveOpen = 1;
return BX_SOUNDLOW_OK;
}
#ifdef BX_SOUND_OSX_use_converter
OSStatus bx_sound_osx_c::core_audio_pause() {
OSStatus err = noErr;
if (WaveOutputUnit) {
err = AudioOutputUnitStop (WaveOutputUnit);
if (err) {
BX_ERROR(("Core Audio: nextbuffer(): AudioOutputUnitStop (err=%X)\n", (unsigned int)err));
}
WavePlaying = 0;
}
return err;
}
OSStatus bx_sound_osx_c::core_audio_resume() {
OSStatus err = noErr;
if (WaveConverter) {
err = AudioConverterReset (WaveConverter);
if (err) {
BX_ERROR(("Core Audio: core_audio_resume(): AudioConverterReset (err=%X)\n", (unsigned int)err));
return err;
}
}
if (WaveOutputUnit) {
err = AudioOutputUnitStart (WaveOutputUnit);
if (err) {
BX_ERROR(("Core Audio: core_audio_resume(): AudioOutputUnitStart (err=%X)\n", (unsigned int)err));
return err;
}
WavePlaying = 1;
}
return err;
}
#endif
int bx_sound_osx_c::startwaveplayback(int frequency, int bits, bx_bool stereo, int format)
{
#ifdef BX_SOUND_OSX_use_converter
static int oldfreq, oldbits, oldformat;
static bx_bool oldstereo;
AudioStreamBasicDescription srcFormat, dstFormat;
UInt32 formatSize = sizeof(AudioStreamBasicDescription);
OSStatus err;
#endif
BX_DEBUG(("startwaveplayback(%d, %d, %d, %x)", frequency, bits, stereo, format));
#ifdef BX_SOUND_OSX_use_quicktime
WaveInfo.samplePtr = NULL;
WaveInfo.numChannels = stereo ? 2 : 1;
WaveInfo.sampleRate = frequency << 16; // sampleRate is a 16.16 fixed-point value
WaveInfo.loopStart = 0;
WaveInfo.loopEnd = 0;
WaveInfo.encode = extSH; // WaveInfo has type ExtSoundHeader
WaveInfo.baseFrequency = 1; // not sure what means. It's only a Uint8.
WaveInfo.numFrames = 0;
//WaveInfo.AIFFSampleRate = frequency; // frequency as float80
WaveInfo.markerChunk = NULL;
WaveInfo.instrumentChunks = NULL;
WaveInfo.AESRecording = NULL;
WaveInfo.sampleSize = bits * WaveInfo.numChannels;
#endif
#ifdef BX_SOUND_OSX_use_converter
if ((frequency == oldfreq) &&
(bits == oldbits) &&
(stereo == oldstereo) &&
(format == oldformat))
return BX_SOUNDLOW_OK; // nothing to do
oldfreq = frequency;
oldbits = bits;
oldstereo = stereo;
oldformat = format;
// update the source audio format
UInt32 bytes = bits / 8;
UInt32 channels = stereo ? 2 : 1;
srcFormat.mSampleRate = (Float64) frequency;
srcFormat.mFormatID = kAudioFormatLinearPCM;
srcFormat.mFormatFlags = kLinearPCMFormatFlagIsPacked;
if (format & 1) srcFormat.mFormatFlags |= kLinearPCMFormatFlagIsSignedInteger;
srcFormat.mBytesPerPacket = channels * bytes;
srcFormat.mFramesPerPacket = 1;
srcFormat.mBytesPerFrame = channels * bytes;
srcFormat.mChannelsPerFrame = channels;
srcFormat.mBitsPerChannel = bytes * 8;
if (WavePlaying) {
err = AudioOutputUnitStop (WaveOutputUnit);
if (err)
BX_ERROR(("Core Audio: startwaveplayback(): AudioOutputUnitStop (err=%X)\n", (unsigned int)err));
}
if (WaveConverter) {
err = AudioConverterDispose (WaveConverter);
if (err)
BX_ERROR(("Core Audio: startwaveplayback(): AudioConverterDispose (err=%X)\n", (unsigned int)err));
}
err = AudioUnitGetProperty (WaveOutputUnit, kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Output, 0, &dstFormat, &formatSize);
if (err) {
BX_ERROR(("Core Audio: startwaveplayback(): AudioUnitGetProperty (err=%X)\n", (unsigned int)err));
return BX_SOUNDLOW_ERR;
}
#ifdef BX_SOUND_OSX_CONVERTER_NEW_API
// force interleaved mode
dstFormat.mFormatFlags |= kAudioFormatFlagIsNonInterleaved;
dstFormat.mBytesPerPacket = dstFormat.mBytesPerFrame = (dstFormat.mBitsPerChannel + 7) / 8;
err = AudioUnitSetProperty (WaveOutputUnit, kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Input, 0, &dstFormat, sizeof(dstFormat));
if (err) {
BX_ERROR(("Core Audio: startwaveplayback(): AudioUnitSetProperty (err=%X)\n", (unsigned int)err));
return BX_SOUNDLOW_ERR;
}
#endif
err = AudioConverterNew (&srcFormat, &dstFormat, &WaveConverter);
if (err) {
BX_ERROR(("Core Audio: startwaveplayback(): AudioConverterNew (err=%X)\n", (unsigned int)err));
return BX_SOUNDLOW_ERR;
}
if (srcFormat.mChannelsPerFrame == 1 && dstFormat.mChannelsPerFrame == 2) {
// map single-channel input to both output channels
SInt32 map[2] = {0,0};
err = AudioConverterSetProperty (WaveConverter,
kAudioConverterChannelMap,
sizeof(map), (void*) map);
if (err) {
BX_ERROR(("Core Audio: startwaveplayback(): AudioConverterSetProperty (err=%X)\n", (unsigned int)err));
return BX_SOUNDLOW_ERR;
}
}
if (WavePlaying) {
if (core_audio_resume() != noErr)
return BX_SOUNDLOW_ERR;
}
#endif
return BX_SOUNDLOW_OK;
}
int bx_sound_osx_c::waveready()
{
// HACK: the -4 is to keep from overwriting buffers that
// have been sent, but possibly not yet played. There
// should be a better way of doing this.
if (WaveOpen && (head - tail < BX_SOUND_OSX_NBUF-4)) {
return BX_SOUNDLOW_OK;
}
else {
#ifdef BX_SOUND_OSX_use_converter
// If buffer is full, make sure sound is playing
if (!WavePlaying) {
if (core_audio_resume() != noErr)
return BX_SOUNDLOW_ERR;
}
#endif
return BX_SOUNDLOW_ERR;
}
}
int bx_sound_osx_c::sendwavepacket(int length, Bit8u data[])
{
#ifdef BX_SOUND_OSX_use_quicktime
SndCommand mySndCommand;
#endif
BX_DEBUG(("sendwavepacket(%d, %p), head=%u", length, data, head));
// sanity check
if ((!WaveOpen) || (head - tail >= BX_SOUND_OSX_NBUF))
return BX_SOUNDLOW_ERR;
// find next available buffer
int n = head++ % BX_SOUND_OSX_NBUF;
// put data in buffer
memcpy(WaveData[n], data, length);
WaveLength[n] = length;
#ifdef BX_SOUND_OSX_use_quicktime
memcpy(&WaveHeader[n], &WaveInfo, sizeof(WaveInfo));
WaveHeader[n].samplePtr = (char *) (WaveData[n]);
WaveHeader[n].numFrames = length * 8 / WaveInfo.sampleSize;
#endif
#ifdef BX_SOUND_OSX_use_converter
// make sure that the sound is playing
if (!WavePlaying) {
if (core_audio_resume() != noErr)
return BX_SOUNDLOW_ERR;
}
#endif
#ifdef BX_SOUND_OSX_use_quicktime
// queue buffer to play
mySndCommand.cmd = bufferCmd;
mySndCommand.param1 = 0;
mySndCommand.param2 = (long)(&WaveHeader[n]);
SndDoCommand(WaveChannel, &mySndCommand, TRUE);
// queue callback for when buffer finishes
mySndCommand.cmd = callBackCmd;
mySndCommand.param1 = 0;
mySndCommand.param2 = (long)(&tail);
SndDoCommand(WaveChannel, &mySndCommand, TRUE);
#endif
return BX_SOUNDLOW_OK;
}
int bx_sound_osx_c::stopwaveplayback()
{
return BX_SOUNDLOW_OK;
}
int bx_sound_osx_c::closewaveoutput()
{
#ifdef BX_SOUND_OSX_use_converter
if (WavePlaying) AudioOutputUnitStop (WaveOutputUnit);
if (WaveConverter) AudioConverterDispose (WaveConverter);
if (WaveOutputUnit) CloseComponent (WaveOutputUnit);
WavePlaying = 0;
WaveOpen = 0;
WaveConverter = NULL;
WaveOutputUnit = NULL;
#endif
return BX_SOUNDLOW_OK;
}
#ifdef BX_SOUND_OSX_use_converter
#ifndef BX_SOUND_OSX_CONVERTER_NEW_API
OSStatus MyRenderer (void *inRefCon, AudioUnitRenderActionFlags inActionFlags,
const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, AudioBuffer *ioData)
{
OSStatus err;
UInt32 size = ioData->mDataByteSize;
err = AudioConverterFillBuffer (WaveConverter, MyACInputProc, inRefCon, &size, ioData->mData);
return err;
}
OSStatus MyACInputProc (AudioConverterRef inAudioConverter,
UInt32* outDataSize, void** outData, void* inUserData)
{
bx_sound_osx_c *self = (bx_sound_osx_c*) inUserData;
self->nextbuffer ((int*) outDataSize, outData);
return noErr;
}
#else
OSStatus MyRenderer (void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags, const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList *ioData)
{
UInt32 packets;
AudioStreamBasicDescription dstFormat;
UInt32 formatSize = sizeof(AudioStreamBasicDescription);
OSStatus err = noErr;
err = AudioConverterGetProperty (WaveConverter,
kAudioConverterCurrentOutputStreamDescription,
&formatSize, &dstFormat);
if (err) {
return err;
}
packets = inNumberFrames / dstFormat.mFramesPerPacket;
err = AudioConverterFillComplexBuffer(WaveConverter,
MyACInputProc, inRefCon, &packets, ioData, NULL);
return err;
}
OSStatus MyACInputProc (AudioConverterRef inAudioConverter, UInt32 *ioNumberDataPackets, AudioBufferList *ioData, AudioStreamPacketDescription **outDataPacketDescription, void *inUserData)
{
OSStatus err;
bx_sound_osx_c *self = (bx_sound_osx_c*) inUserData;
AudioStreamBasicDescription srcFormat;
UInt32 formatSize = sizeof(AudioStreamBasicDescription);
err = AudioConverterGetProperty (inAudioConverter,
kAudioConverterCurrentInputStreamDescription,
&formatSize, &srcFormat);
if (err) {
*ioNumberDataPackets = 0;
return err;
}
int outDataSize = *ioNumberDataPackets * srcFormat.mBytesPerPacket;
void *outData = ioData->mBuffers[0].mData;
self->nextbuffer ((int*) &outDataSize, &outData);
*ioNumberDataPackets = outDataSize / srcFormat.mBytesPerPacket;
ioData->mBuffers[0].mDataByteSize = outDataSize;
ioData->mBuffers[0].mData = outData;
return noErr;
}
#endif
void bx_sound_osx_c::nextbuffer (int *outDataSize, void **outData)
{
BX_DEBUG(("nextbuffer(), tail=%u", tail));
if (head - tail <= 0) {
*outData = NULL;
*outDataSize = 0;
// We are getting behind, so stop the output for now
core_audio_pause();
}
else {
int n = tail % BX_SOUND_OSX_NBUF;
*outData = (void *) (WaveData[n]);
*outDataSize = WaveLength[n];
tail++;
}
}
#endif
int bx_sound_osx_c::openwaveinput(const char *wavedev, sound_record_handler_t rh)
{
UNUSED(wavedev);
record_handler = rh;
if (rh != NULL) {
record_timer_index = bx_pc_system.register_timer(this, record_timer_handler, 1, 1, 0, "soundosx");
// record timer: inactive, continuous, frequency variable
}
// TODO
return BX_SOUNDLOW_OK;
}
int bx_sound_osx_c::startwaverecord(int frequency, int bits, bx_bool stereo, int format)
{
Bit64u timer_val;
Bit8u shift = 0;
UNUSED(format);
if (record_timer_index != BX_NULL_TIMER_HANDLE) {
if (bits == 16) shift++;
if (stereo) shift++;
record_packet_size = (frequency / 10) << shift; // 0.1 sec
if (record_packet_size > BX_SOUNDLOW_WAVEPACKETSIZE) {
record_packet_size = BX_SOUNDLOW_WAVEPACKETSIZE;
}
timer_val = (Bit64u)record_packet_size * 1000000 / (frequency << shift);
bx_pc_system.activate_timer(record_timer_index, (Bit32u)timer_val, 1);
}
// TODO
return BX_SOUNDLOW_OK;
}
int bx_sound_osx_c::getwavepacket(int length, Bit8u data[])
{
// TODO
memset(data, 0, length);
return BX_SOUNDLOW_OK;
}
int bx_sound_osx_c::stopwaverecord()
{
if (record_timer_index != BX_NULL_TIMER_HANDLE) {
bx_pc_system.deactivate_timer(record_timer_index);
}
// TODO
return BX_SOUNDLOW_OK;
}
int bx_sound_osx_c::closewaveinput()
{
stopwaverecord();
// TODO
return BX_SOUNDLOW_OK;
}
void bx_sound_osx_c::record_timer_handler(void *this_ptr)
{
bx_sound_osx_c *class_ptr = (bx_sound_osx_c *) this_ptr;
class_ptr->record_timer();
}
void bx_sound_osx_c::record_timer(void)
{
record_handler(this, record_packet_size);
}
#endif // defined(macintosh)