libretro/scummvm
1
0
Fork 0
mirror of https://github.com/libretro/scummvm.git synced 2024-12-04 08:17:40 +00:00
Code Issues Actions 2 Packages Projects Releases Wiki Activity

Patch ##1956946 (Audio::Mixer internal API revision) with some tweaks

Browse Source 
svn-id: r32828
This commit is contained in:
Max Horn 2008-06-28 15:28:29 +00:00
parent e68efca5a1
commit c45d632f3b
13 changed files with 336 additions and 199 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

4
backends/platform/gp2x/gp2x-common.h
View File

@ -37,7 +37,7 @@
#include <SDL_gp2x.h>
namespace Audio {
class Mixer;
class MixerImpl;
}
namespace Common {
@ -367,7 +367,7 @@ protected:
Common::SaveFileManager *_savefile;
FilesystemFactory *getFilesystemFactory();
Audio::Mixer *_mixer;
Audio::MixerImpl *_mixer;
SDL_TimerID _timerID;
Common::TimerManager *_timer;

7
backends/platform/gp2x/gp2x.cpp
View File

@ -40,7 +40,7 @@
#include "backends/timer/default/default-timer.h"
#include "backends/plugins/posix/posix-provider.h"
#include "backends/fs/posix/posix-fs-factory.h" // for getFilesystemFactory()
#include "sound/mixer.h"
#include "sound/mixer_intern.h"
#include <stdio.h>
#include <stdlib.h>
@ -225,8 +225,7 @@ void OSystem_GP2X::initBackend() {
// Create and hook up the mixer, if none exists yet (we check for this to
// allow subclasses to provide their own).
if (_mixer == 0) {
_mixer = new Audio::Mixer();
setSoundCallback(Audio::Mixer::mixCallback, _mixer);
setupMixer();
}
// Create and hook up the timer manager, if none exists yet (we check for
@ -445,7 +444,7 @@ void OSystem_GP2X::deleteMutex(MutexRef mutex) {
#pragma mark --- Audio ---
#pragma mark -
bool OSystem_GP2X::setSoundCallback(SoundProc proc, void *param) {
void OSystem_GP2X::setupMixer() {
SDL_AudioSpec desired;
SDL_AudioSpec obtained;

4
backends/platform/psp/osys_psp.cpp
View File

@ -34,7 +34,7 @@
#include "backends/timer/default/default-timer.h"
#include "graphics/surface.h"
#include "graphics/scaler.h"
#include "sound/mixer.h"
#include "sound/mixer_intern.h"
#include <pspgu.h>
@ -99,7 +99,7 @@ OSystem_PSP::~OSystem_PSP() {
void OSystem_PSP::initBackend() {
_savefile = new DefaultSaveFileManager();
_mixer = new Audio::Mixer();
_mixer = new Audio::MixerImpl(this);
_timer = new DefaultTimerManager();
setSoundCallback(Audio::Mixer::mixCallback, _mixer);
setTimerCallback(&timer_handler, 10);

51
backends/platform/sdl/sdl.cpp
View File

@ -30,7 +30,7 @@
#include "backends/saves/default/default-saves.h"
#include "backends/timer/default/default-timer.h"
#include "sound/mixer.h"
#include "sound/mixer_intern.h"
#include "icons/scummvm.xpm"
@ -131,9 +131,7 @@ void OSystem_SDL::initBackend() {
// Create and hook up the mixer, if none exists yet (we check for this to
// allow subclasses to provide their own).
if (_mixer == 0) {
_mixer = new Audio::Mixer();
bool result = setSoundCallback(Audio::Mixer::mixCallback, _mixer);
_mixer->setReady(result);
setupMixer();
}
// Create and hook up the timer manager, if none exists yet (we check for
@ -391,7 +389,15 @@ void OSystem_SDL::deleteMutex(MutexRef mutex) {
#pragma mark --- Audio ---
#pragma mark -
bool OSystem_SDL::setSoundCallback(SoundProc proc, void *param) {
void OSystem_SDL::mixCallback(void *sys, byte *samples, int len) {
OSystem_SDL *this_ = (OSystem_SDL *)sys;
assert(this_);
if (this_->_mixer)
this_->_mixer->mixCallback(samples, len);
}
void OSystem_SDL::setupMixer() {
SDL_AudioSpec desired;
SDL_AudioSpec obtained;
@ -415,23 +421,30 @@ bool OSystem_SDL::setSoundCallback(SoundProc proc, void *param) {
desired.format = AUDIO_S16SYS;
desired.channels = 2;
desired.samples = (uint16)samples;
desired.callback = proc;
desired.userdata = param;
desired.callback = mixCallback;
desired.userdata = this;
// Create the mixer instance
assert(!_mixer);
_mixer = new Audio::MixerImpl(this);
assert(_mixer);
if (SDL_OpenAudio(&desired, &obtained) != 0) {
warning("Could not open audio device: %s", SDL_GetError());
return false;
_samplesPerSec = 0;
_mixer->setReady(false);
} else {
// Note: This should be the obtained output rate, but it seems that at
// least on some platforms SDL will lie and claim it did get the rate
// even if it didn't. Probably only happens for "weird" rates, though.
_samplesPerSec = obtained.freq;
debug(1, "Output sample rate: %d Hz", _samplesPerSec);
// Tell the mixer that we are ready and start the sound processing
_mixer->setOutputRate(_samplesPerSec);
_mixer->setReady(true);
SDL_PauseAudio(0);
}
// Note: This should be the obtained output rate, but it seems that at
// least on some platforms SDL will lie and claim it did get the rate
// even if it didn't. Probably only happens for "weird" rates, though.
_samplesPerSec = obtained.freq;
debug(1, "Output sample rate: %d Hz", _samplesPerSec);
SDL_PauseAudio(0);
return true;
}
int OSystem_SDL::getOutputSampleRate() const {
return _samplesPerSec;
}
Audio::Mixer *OSystem_SDL::getMixer() {

10
backends/platform/sdl/sdl.h
View File

@ -38,7 +38,7 @@
namespace Audio {
class Mixer;
class MixerImpl;
}
namespace Common {
@ -134,8 +134,9 @@ public:
virtual bool pollEvent(Common::Event &event); // overloaded by CE backend
// Set function that generates samples
typedef void (*SoundProc)(void *param, byte *buf, int len);
virtual bool setSoundCallback(SoundProc proc, void *param); // overloaded by CE backend
virtual void setupMixer();
static void mixCallback(void *s, byte *samples, int len);
virtual Audio::Mixer *getMixer();
// Poll CD status
@ -186,7 +187,6 @@ public:
virtual void setWindowCaption(const char *caption);
virtual bool openCD(int drive);
virtual int getOutputSampleRate() const;
virtual bool hasFeature(Feature f);
virtual void setFeatureState(Feature f, bool enable);
@ -371,7 +371,7 @@ protected:
Common::SaveFileManager *_savefile;
Audio::Mixer *_mixer;
Audio::MixerImpl *_mixer;
SDL_TimerID _timerID;
Common::TimerManager *_timer;

69
backends/platform/symbian/src/SymbianOS.cpp
View File

@ -173,11 +173,8 @@ void OSystem_SDL_Symbian::quit() {
OSystem_SDL::quit();
}
bool OSystem_SDL_Symbian::setSoundCallback(SoundProc proc, void *param) {
void OSystem_SDL_Symbian::setupMixer() {
// First save the proc and param
_sound_proc_param = param;
_sound_proc = proc;
SDL_AudioSpec desired;
SDL_AudioSpec obtained;
@ -207,48 +204,53 @@ bool OSystem_SDL_Symbian::setSoundCallback(SoundProc proc, void *param) {
desired.format = AUDIO_S16SYS;
desired.channels = 2;
desired.samples = (uint16)samples;
#ifdef S60
desired.callback = symbianMixCallback;
desired.userdata = this;
#else
desired.callback = proc;
desired.userdata = param;
#endif
// Create the mixer instance
assert(!_mixer);
_mixer = new Audio::MixerImpl(this);
assert(_mixer);
if (SDL_OpenAudio(&desired, &obtained) != 0) {
warning("Could not open audio device: %s", SDL_GetError());
return false;
_samplesPerSec = 0;
_mixer->setReady(false);
} else {
// Note: This should be the obtained output rate, but it seems that at
// least on some platforms SDL will lie and claim it did get the rate
// even if it didn't. Probably only happens for "weird" rates, though.
_samplesPerSec = obtained.freq;
_channels = obtained.channels;
// Need to create mixbuffer for stereo mix to downmix
if (_channels != 2) {
_stereo_mix_buffer = new byte [obtained.size*2];//*2 for stereo values
}
// Tell the mixer that we are ready and start the sound processing
_mixer->setOutputRate(_samplesPerSec);
_mixer->setReady(true);
SDL_PauseAudio(0);
}
// Note: This should be the obtained output rate, but it seems that at
// least on some platforms SDL will lie and claim it did get the rate
// even if it didn't. Probably only happens for "weird" rates, though.
_samplesPerSec = obtained.freq;
_channels = obtained.channels;
// Need to create mixbuffer for stereo mix to downmix
if (_channels != 2) {
_stereo_mix_buffer = new byte [obtained.size*2];//*2 for stereo values
}
SDL_PauseAudio(0);
return true;
}
/**
* The mixer callback function, passed on to OSystem::setSoundCallback().
* This simply calls the mix() method.
*/
void OSystem_SDL_Symbian::symbianMixCallback(void *s, byte *samples, int len) {
static_cast <OSystem_SDL_Symbian*>(s)->symbianMix(samples,len);
}
void OSystem_SDL_Symbian::symbianMixCallback(void *sys, byte *samples, int len) {
OSystem_SDL_Symbian *this_ = (OSystem_SDL_Symbian *)sys;
assert(this_);
if (!this_->_mixer)
return;
/**
* Actual mixing implementation
*/
void OSystem_SDL_Symbian::symbianMix(byte *samples, int len) {
#ifdef S60
// If not stereo then we need to downmix
if (_channels != 2) {
_sound_proc(_sound_proc_param, _stereo_mix_buffer, len * 2);
this_->_mixer->mixCallback(_stereo_mix_buffer, len * 2);
int16 *bitmixDst = (int16 *)samples;
int16 *bitmixSrc = (int16 *)_stereo_mix_buffer;
@ -258,9 +260,12 @@ void OSystem_SDL_Symbian::symbianMix(byte *samples, int len) {
bitmixSrc += 2;
}
} else
_sound_proc(_sound_proc_param, samples, len);
#else
this_->_mixer->mixCallback(samples, len);
#endif
}
/**
* This is an implementation by the remapKey function
* @param SDL_Event to remap

9
backends/platform/symbian/src/SymbianOS.h
View File

@ -58,7 +58,7 @@ public:
// This function is overridden by the symbian port in order to provide MONO audio
// downmix is done by supplying our own audiocallback
//
virtual bool setSoundCallback(SoundProc proc, void *param); // overloaded by CE backend
virtual void setupMixer(); // overloaded by CE backend
// Overloaded from SDL_Commmon
void quit();
@ -70,11 +70,6 @@ protected:
//
static void symbianMixCallback(void *s, byte *samples, int len);
//
// Actual mixing implementation
//
void symbianMix(byte *samples, int len);
virtual FilesystemFactory *getFilesystemFactory();
public:
// vibration support
@ -121,8 +116,6 @@ protected:
// Audio
int _channels;
SoundProc _sound_proc;
void *_sound_proc_param;
byte *_stereo_mix_buffer;
// Used to handle joystick navi zones

45
backends/platform/wince/wince-sdl.cpp
View File

@ -35,7 +35,7 @@
#include "base/main.h"
#include "base/plugins.h"
#include "sound/mixer.h"
#include "sound/mixer_intern.h"
#include "sound/fmopl.h"
#include "backends/timer/default/default-timer.h"
@ -404,7 +404,7 @@ void OSystem_WINCE3::initBackend()
// Instantiate our own sound mixer
// mixer init is postponed until a game engine is selected.
if (_mixer == 0) {
_mixer = new Audio::Mixer();
_mixer = new Audio::Mixer(this);
}
// Create the timer. CE SDL does not support multiple timers (SDL_AddTimer).
@ -770,7 +770,7 @@ void OSystem_WINCE3::create_toolbar() {
_toolbarHandler.setVisible(false);
}
bool OSystem_WINCE3::setSoundCallback(SoundProc proc, void *param) {
void OSystem_WINCE3::setupMixer(SoundProc proc, void *param) {
SDL_AudioSpec desired;
int thread_priority;
@ -785,12 +785,16 @@ bool OSystem_WINCE3::setSoundCallback(SoundProc proc, void *param) {
desired.channels = 2;
desired.samples = 128;
desired.callback = private_sound_proc;
desired.userdata = param;
desired.userdata = this;
// Create the mixer instance
assert(!_mixer);
_mixer = new Audio::MixerImpl(this);
assert(_mixer);
// Add sound thread priority
if (!ConfMan.hasKey("sound_thread_priority")) {
if (!ConfMan.hasKey("sound_thread_priority"))
thread_priority = THREAD_PRIORITY_NORMAL;
}
else
thread_priority = ConfMan.getInt("sound_thread_priority");
@ -799,16 +803,24 @@ bool OSystem_WINCE3::setSoundCallback(SoundProc proc, void *param) {
SDL_CloseAudio();
if (SDL_OpenAudio(&desired, NULL) != 0) {
warning("Could not open audio device: %s", SDL_GetError());
return false;
}
else
_mixer->setReady(false);
} else {
debug(1, "Sound opened OK, mixing at %d Hz", _sampleRate);
SDL_PauseAudio(0);
return true;
// Tell the mixer that we are ready and start the sound processing
_mixer->setOutputRate(_sampleRate);
_mixer->setReady(true);
SDL_PauseAudio(0);
}
}
void OSystem_WINCE3::private_sound_proc(void *param, byte *buf, int len) {
(*_originalSoundProc)(param, buf, len);
OSystem_WINCE3 *this_ = (OSystem_WINCE3 *)param;
assert(this_);
if (this_->_mixer)
this_->_mixer->mixCallback(buf, len);
if (!_soundMaster)
memset(buf, 0, len);
}
@ -838,7 +850,7 @@ bool OSystem_WINCE3::checkOggHighSampleRate() {
}
#endif
void OSystem_WINCE3::get_sample_rate() {
void OSystem_WINCE3::compute_sample_rate() {
// Force at least medium quality FM synthesis for FOTAQ
Common::String gameid(ConfMan.get("gameid"));
if (gameid == "queen") {
@ -875,9 +887,8 @@ void OSystem_WINCE3::setWindowCaption(const char *caption) {
//update_game_settings();
// finalize mixer init
get_sample_rate();
bool result = setSoundCallback(Audio::Mixer::mixCallback, _mixer);
_mixer->setReady(result);
compute_sample_rate();
setupMixer();
// handle the actual event
OSystem_SDL::setWindowCaption(caption);
@ -1050,7 +1061,7 @@ void OSystem_WINCE3::update_game_settings() {
}
}
get_sample_rate();
compute_sample_rate();
}
void OSystem_WINCE3::initSize(uint w, uint h) {

5
backends/platform/wince/wince-sdl.h
View File

@ -82,7 +82,7 @@ public:
// Overloaded from SDL_Commmon
void quit();
// Overloaded from SDL_Commmon (master volume and sample rate subtleties)
bool setSoundCallback(SoundProc proc, void *param);
void setupMixer();
// Overloaded from OSystem
//void engineInit();
void getTimeAndDate(struct tm &t) const;
@ -160,13 +160,12 @@ private:
#endif
static void private_sound_proc(void *param, byte *buf, int len);
static SoundProc _originalSoundProc;
bool update_scalers();
void create_toolbar();
void update_game_settings();
void check_mappings();
void get_sample_rate();
void compute_sample_rate();
void retrieve_mouse_location(int &x, int &y);

9
common/system.h
View File

@ -814,15 +814,6 @@ public:
*/
virtual Audio::Mixer *getMixer() = 0;
/**
* Determine the output sample rate. Audio data provided by the sound
* callback will be played using this rate.
* @note Client code other than the sound mixer should _not_ use this
* method. Instead, call Mixer::getOutputRate()!
* @return the output sample rate
*/
virtual int getOutputSampleRate() const = 0;
//@}

83
sound/mixer.cpp
View File

@ -27,7 +27,7 @@
#include "common/util.h"
#include "common/system.h"
#include "sound/mixer.h"
#include "sound/mixer_intern.h"
#include "sound/rate.h"
#include "sound/audiostream.h"
@ -103,32 +103,38 @@ public:
#pragma mark -
Mixer::Mixer() {
_syst = g_system;
MixerImpl::MixerImpl(OSystem *system)
: _syst(system), _sampleRate(0), _mixerReady(false), _handleSeed(0) {
_handleSeed = 0;
int i = 0;
int i;
for (i = 0; i < ARRAYSIZE(_volumeForSoundType); i++)
_volumeForSoundType[i] = kMaxMixerVolume;
for (i = 0; i != NUM_CHANNELS; i++)
_channels[i] = 0;
_mixerReady = false;
}
Mixer::~Mixer() {
MixerImpl::~MixerImpl() {
for (int i = 0; i != NUM_CHANNELS; i++)
delete _channels[i];
}
uint Mixer::getOutputRate() const {
return (uint)_syst->getOutputSampleRate();
void MixerImpl::setReady(bool ready) {
_mixerReady = ready;
}
void Mixer::insertChannel(SoundHandle *handle, Channel *chan) {
uint MixerImpl::getOutputRate() const {
return _sampleRate;
}
void MixerImpl::setOutputRate(uint sampleRate) {
if (_sampleRate != 0 && _sampleRate != sampleRate)
error("Changing the Audio::Mixer output sample rate is not supported");
_sampleRate = sampleRate;
}
void MixerImpl::insertChannel(SoundHandle *handle, Channel *chan) {
int index = -1;
for (int i = 0; i != NUM_CHANNELS; i++) {
@ -138,7 +144,7 @@ void Mixer::insertChannel(SoundHandle *handle, Channel *chan) {
}
}
if (index == -1) {
warning("Mixer::out of mixer slots");
warning("MixerImpl::out of mixer slots");
delete chan;
return;
}
@ -151,7 +157,7 @@ void Mixer::insertChannel(SoundHandle *handle, Channel *chan) {
}
}
void Mixer::playRaw(
void MixerImpl::playRaw(
SoundType type,
SoundHandle *handle,
void *sound,
@ -166,7 +172,7 @@ void Mixer::playRaw(
playInputStream(type, handle, input, id, volume, balance, true, false, ((flags & Mixer::FLAG_REVERSE_STEREO) != 0));
}
void Mixer::playInputStream(
void MixerImpl::playInputStream(
SoundType type,
SoundHandle *handle,
AudioStream *input,
@ -198,8 +204,13 @@ void Mixer::playInputStream(
insertChannel(handle, chan);
}
void Mixer::mix(int16 *buf, uint len) {
void MixerImpl::mixCallback(byte *samples, uint len) {
assert(samples);
Common::StackLock lock(_mutex);
int16 *buf = (int16 *)samples;
len >>= 2;
// Since the mixer callback has been called, the mixer must be ready...
_mixerReady = true;
@ -218,15 +229,7 @@ void Mixer::mix(int16 *buf, uint len) {
}
}
void Mixer::mixCallback(void *s, byte *samples, int len) {
assert(s);
assert(samples);
// Len is the number of bytes in the buffer; we divide it by
// four to get the number of samples (stereo 16 bit).
((Mixer *)s)->mix((int16 *)samples, len >> 2);
}
void Mixer::stopAll() {
void MixerImpl::stopAll() {
Common::StackLock lock(_mutex);
for (int i = 0; i != NUM_CHANNELS; i++) {
if (_channels[i] != 0 && !_channels[i]->isPermanent()) {
@ -236,7 +239,7 @@ void Mixer::stopAll() {
}
}
void Mixer::stopID(int id) {
void MixerImpl::stopID(int id) {
Common::StackLock lock(_mutex);
for (int i = 0; i != NUM_CHANNELS; i++) {
if (_channels[i] != 0 && _channels[i]->getId() == id) {
@ -246,7 +249,7 @@ void Mixer::stopID(int id) {
}
}
void Mixer::stopHandle(SoundHandle handle) {
void MixerImpl::stopHandle(SoundHandle handle) {
Common::StackLock lock(_mutex);
// Simply ignore stop requests for handles of sounds that already terminated
@ -258,7 +261,7 @@ void Mixer::stopHandle(SoundHandle handle) {
_channels[index] = 0;
}
void Mixer::setChannelVolume(SoundHandle handle, byte volume) {
void MixerImpl::setChannelVolume(SoundHandle handle, byte volume) {
Common::StackLock lock(_mutex);
const int index = handle._val % NUM_CHANNELS;
@ -268,7 +271,7 @@ void Mixer::setChannelVolume(SoundHandle handle, byte volume) {
_channels[index]->setVolume(volume);
}
void Mixer::setChannelBalance(SoundHandle handle, int8 balance) {
void MixerImpl::setChannelBalance(SoundHandle handle, int8 balance) {
Common::StackLock lock(_mutex);
const int index = handle._val % NUM_CHANNELS;
@ -278,7 +281,7 @@ void Mixer::setChannelBalance(SoundHandle handle, int8 balance) {
_channels[index]->setBalance(balance);
}
uint32 Mixer::getSoundElapsedTime(SoundHandle handle) {
uint32 MixerImpl::getSoundElapsedTime(SoundHandle handle) {
Common::StackLock lock(_mutex);
const int index = handle._val % NUM_CHANNELS;
@ -288,7 +291,7 @@ uint32 Mixer::getSoundElapsedTime(SoundHandle handle) {
return _channels[index]->getElapsedTime();
}
void Mixer::pauseAll(bool paused) {
void MixerImpl::pauseAll(bool paused) {
Common::StackLock lock(_mutex);
for (int i = 0; i != NUM_CHANNELS; i++) {
if (_channels[i] != 0) {
@ -297,7 +300,7 @@ void Mixer::pauseAll(bool paused) {
}
}
void Mixer::pauseID(int id, bool paused) {
void MixerImpl::pauseID(int id, bool paused) {
Common::StackLock lock(_mutex);
for (int i = 0; i != NUM_CHANNELS; i++) {
if (_channels[i] != 0 && _channels[i]->getId() == id) {
@ -307,7 +310,7 @@ void Mixer::pauseID(int id, bool paused) {
}
}
void Mixer::pauseHandle(SoundHandle handle, bool paused) {
void MixerImpl::pauseHandle(SoundHandle handle, bool paused) {
Common::StackLock lock(_mutex);
// Simply ignore (un)pause requests for sounds that already terminated
@ -318,7 +321,7 @@ void Mixer::pauseHandle(SoundHandle handle, bool paused) {
_channels[index]->pause(paused);
}
bool Mixer::isSoundIDActive(int id) {
bool MixerImpl::isSoundIDActive(int id) {
Common::StackLock lock(_mutex);
for (int i = 0; i != NUM_CHANNELS; i++)
if (_channels[i] && _channels[i]->getId() == id)
@ -326,7 +329,7 @@ bool Mixer::isSoundIDActive(int id) {
return false;
}
int Mixer::getSoundID(SoundHandle handle) {
int MixerImpl::getSoundID(SoundHandle handle) {
Common::StackLock lock(_mutex);
const int index = handle._val % NUM_CHANNELS;
if (_channels[index] && _channels[index]->_handle._val == handle._val)
@ -334,13 +337,13 @@ int Mixer::getSoundID(SoundHandle handle) {
return 0;
}
bool Mixer::isSoundHandleActive(SoundHandle handle) {
bool MixerImpl::isSoundHandleActive(SoundHandle handle) {
Common::StackLock lock(_mutex);
const int index = handle._val % NUM_CHANNELS;
return _channels[index] && _channels[index]->_handle._val == handle._val;
}
bool Mixer::hasActiveChannelOfType(SoundType type) {
bool MixerImpl::hasActiveChannelOfType(SoundType type) {
Common::StackLock lock(_mutex);
for (int i = 0; i != NUM_CHANNELS; i++)
if (_channels[i] && _channels[i]->_type == type)
@ -348,7 +351,7 @@ bool Mixer::hasActiveChannelOfType(SoundType type) {
return false;
}
void Mixer::setVolumeForSoundType(SoundType type, int volume) {
void MixerImpl::setVolumeForSoundType(SoundType type, int volume) {
assert(0 <= type && type < ARRAYSIZE(_volumeForSoundType));
// Check range
@ -363,7 +366,7 @@ void Mixer::setVolumeForSoundType(SoundType type, int volume) {
_volumeForSoundType[type] = volume;
}
int Mixer::getVolumeForSoundType(SoundType type) const {
int MixerImpl::getVolumeForSoundType(SoundType type) const {
assert(0 <= type && type < ARRAYSIZE(_volumeForSoundType));
return _volumeForSoundType[type];
@ -443,7 +446,7 @@ uint32 Channel::getElapsedTime() {
// Convert the number of samples into a time duration. To avoid
// overflow, this has to be done in a somewhat non-obvious way.
uint rate = _mixer->getOutputRate();
uint32 rate = _mixer->getOutputRate();
uint32 seconds = _samplesConsumed / rate;
uint32 milliseconds = (1000 * (_samplesConsumed % rate)) / rate;

85
sound/mixer.h
View File

@ -38,6 +38,7 @@ namespace Audio {
class AudioStream;
class Channel;
class Mixer;
class MixerImpl;
/**
* A SoundHandle instances corresponds to a specific sound
@ -47,7 +48,7 @@ class Mixer;
*/
class SoundHandle {
friend class Channel;
friend class Mixer;
friend class MixerImpl;
uint32 _val;
public:
inline SoundHandle() : _val(0xFFFFFFFF) {}
@ -104,24 +105,9 @@ public:
kMaxMixerVolume = 256
};
private:
enum {
NUM_CHANNELS = 16
};
OSystem *_syst;
Common::Mutex _mutex;
int _volumeForSoundType[4];
uint32 _handleSeed;
Channel *_channels[NUM_CHANNELS];
bool _mixerReady;
public:
Mixer();
~Mixer();
Mixer() {}
virtual ~Mixer() {}
@ -132,8 +118,10 @@ public:
* sync with an audio stream. In particular, the Adlib MIDI emulation...
*
* @return whether the mixer is ready and setup
*
* @todo get rid of this?
*/
bool isReady() const { return _mixerReady; }
virtual bool isReady() const = 0;
@ -143,12 +131,12 @@ public:
* (using the makeLinearInputStream factory function), which is then
* passed on to playInputStream.
*/
void playRaw(
virtual void playRaw(
SoundType type,
SoundHandle *handle,
void *sound, uint32 size, uint rate, byte flags,
int id = -1, byte volume = kMaxChannelVolume, int8 balance = 0,
uint32 loopStart = 0, uint32 loopEnd = 0);
uint32 loopStart = 0, uint32 loopEnd = 0) = 0;
/**
* Start playing the given audio input stream.
@ -170,35 +158,35 @@ public:
* not stop this particular stream
* @param reverseStereo a flag indicating whether left and right channels shall be swapped
*/
void playInputStream(
virtual void playInputStream(
SoundType type,
SoundHandle *handle,
AudioStream *input,
int id = -1, byte volume = kMaxChannelVolume, int8 balance = 0,
bool autofreeStream = true,
bool permanent = false,
bool reverseStereo = false);
bool reverseStereo = false) = 0;
/**
* Stop all currently playing sounds.
*/
void stopAll();
virtual void stopAll() = 0;
/**
* Stop playing the sound with given ID.
*
* @param id the ID of the sound to affect
*/
void stopID(int id);
virtual void stopID(int id) = 0;
/**
* Stop playing the sound corresponding to the given handle.
*
* @param handle the sound to affect
*/
void stopHandle(SoundHandle handle);
virtual void stopHandle(SoundHandle handle) = 0;
@ -208,7 +196,7 @@ public:
*
* @param paused true to pause everything, false to unpause
*/
void pauseAll(bool paused);
virtual void pauseAll(bool paused) = 0;
/**
* Pause/unpause the sound with the given ID.
@ -216,7 +204,7 @@ public:
* @param id the ID of the sound to affect
* @param paused true to pause the sound, false to unpause it
*/
void pauseID(int id, bool paused);
virtual void pauseID(int id, bool paused) = 0;
/**
* Pause/unpause the sound corresponding to the given handle.
@ -224,7 +212,7 @@ public:
* @param handle the sound to affect
* @param paused true to pause the sound, false to unpause it
*/
void pauseHandle(SoundHandle handle, bool paused);
virtual void pauseHandle(SoundHandle handle, bool paused) = 0;
@ -234,7 +222,7 @@ public:
* @param id the ID of the sound to query
* @return true if the sound is active
*/
bool isSoundIDActive(int id);
virtual bool isSoundIDActive(int id) = 0;
/**
* Get the sound ID of handle sound
@ -242,7 +230,7 @@ public:
* @param handle sound to query
* @return sound ID if active
*/
int getSoundID(SoundHandle handle);
virtual int getSoundID(SoundHandle handle) = 0;
/**
* Check if a sound with the given handle is active.
@ -250,7 +238,7 @@ public:
* @param handle sound to query
* @return true if the sound is active
*/
bool isSoundHandleActive(SoundHandle handle);
virtual bool isSoundHandleActive(SoundHandle handle) = 0;
@ -260,7 +248,7 @@ public:
* @param handle the sound to affect
* @param volume the new channel volume (0 - kMaxChannelVolume)
*/
void setChannelVolume(SoundHandle handle, byte volume);
virtual void setChannelVolume(SoundHandle handle, byte volume) = 0;
/**
* Set the channel balance for the given handle.
@ -269,12 +257,12 @@ public:
* @param balance the new channel balance:
* (-127 ... 0 ... 127) corresponds to (left ... center ... right)
*/
void setChannelBalance(SoundHandle handle, int8 balance);
virtual void setChannelBalance(SoundHandle handle, int8 balance) = 0;
/**
* Get approximation of for how long the channel has been playing.
*/
uint32 getSoundElapsedTime(SoundHandle handle);
virtual uint32 getSoundElapsedTime(SoundHandle handle) = 0;
/**
* Check whether any channel of the given sound type is active.
@ -284,7 +272,7 @@ public:
* @param type the sound type to look for
* @return true if any channels of the specified type are active.
*/
bool hasActiveChannelOfType(SoundType type);
virtual bool hasActiveChannelOfType(SoundType type) = 0;
/**
* Set the volume for the given sound type.
@ -292,7 +280,7 @@ public:
* @param type the sound type
* @param volume the new global volume, 0 - kMaxMixerVolume
*/
void setVolumeForSoundType(SoundType type, int volume);
virtual void setVolumeForSoundType(SoundType type, int volume) = 0;
/**
* Query the global volume.
@ -300,7 +288,7 @@ public:
* @param type the sound type
* @return the global music volume, 0 - kMaxMixerVolume
*/
int getVolumeForSoundType(SoundType type) const;
virtual int getVolumeForSoundType(SoundType type) const = 0;
/**
* Query the system's audio output sample rate. This returns
@ -308,26 +296,7 @@ public:
*
* @return the output sample rate in Hz
*/
uint getOutputRate() const;
protected:
void insertChannel(SoundHandle *handle, Channel *chan);
/**
* Internal main method -- all the actual mixing work is done from here.
*/
void mix(int16 * buf, uint len);
// FIXME: temporary "public" to allow access to mixCallback
// from within OSystem::makeMixer()
public:
/**
* The mixer callback function, passed on to OSystem::setSoundCallback().
* This simply calls the mix() method.
*/
static void mixCallback(void *s, byte *samples, int len);
void setReady(bool ready) { _mixerReady = ready; }
virtual uint getOutputRate() const = 0;
};

154
sound/mixer_intern.h Normal file
View File

@ -0,0 +1,154 @@
/* ScummVM - Graphic Adventure Engine
*
* ScummVM is the legal property of its developers, whose names
* are too numerous to list here. Please refer to the COPYRIGHT
* file distributed with this source distribution.
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* as published by the Free Software Foundation; either version 2
* of the License, or (at your option) any later version.
* This program 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 General Public License for more details.
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
*
* $URL$
* $Id$
*
*/
#ifndef SOUND_MIXER_INTERN_H
#define SOUND_MIXER_INTERN_H
#include "common/scummsys.h"
#include "common/mutex.h"
#include "sound/mixer.h"
namespace Audio {
/**
* The (default) implementation of the ScummVM audio mixing subsystem.
*
* Backends are responsible for allocating (and later releasing) an instance
* of this class, which engines can access via OSystem::getMixer().
*
* Initialisation of instances of this class usually happens as follows:
* 1) Creat a new Audio::MixerImpl instance.
* 2) Set the hardware output sample rate via the setSampleRate() method.
* 3) Hook up the mixCallback() in a suitable audio processing thread/callback.
* 4) Change the mixer into ready mode via setReady(true).
* 5) Start audio processing (e.g. by resuming the audio thread, if applicable).
*
* In the future, we might make it possible for backends to provide
* (partial) alternative implementations of the mixer, e.g. to make
* better use of native sound mixing support on low-end devices.
*
* @see OSystem::getMixer()
*/
class MixerImpl : public Mixer {
private:
enum {
NUM_CHANNELS = 16
};
OSystem *_syst;
Common::Mutex _mutex;
uint _sampleRate;
bool _mixerReady;
uint32 _handleSeed;
int _volumeForSoundType[4];
Channel *_channels[NUM_CHANNELS];
public:
MixerImpl(OSystem *system);
~MixerImpl();
virtual bool isReady() const { return _mixerReady; }
virtual void playRaw(
SoundType type,
SoundHandle *handle,
void *sound, uint32 size, uint rate, byte flags,
int id = -1, byte volume = 255, int8 balance = 0,
uint32 loopStart = 0, uint32 loopEnd = 0);
virtual void playInputStream(
SoundType type,
SoundHandle *handle,
AudioStream *input,
int id = -1, byte volume = 255, int8 balance = 0,
bool autofreeStream = true,
bool permanent = false,
bool reverseStereo = false);
virtual void stopAll();
virtual void stopID(int id);
virtual void stopHandle(SoundHandle handle);
virtual void pauseAll(bool paused);
virtual void pauseID(int id, bool paused);
virtual void pauseHandle(SoundHandle handle, bool paused);
virtual bool isSoundIDActive(int id);
virtual int getSoundID(SoundHandle handle);
virtual bool isSoundHandleActive(SoundHandle handle);
virtual void setChannelVolume(SoundHandle handle, byte volume);
virtual void setChannelBalance(SoundHandle handle, int8 balance);
virtual uint32 getSoundElapsedTime(SoundHandle handle);
virtual bool hasActiveChannelOfType(SoundType type);
virtual void setVolumeForSoundType(SoundType type, int volume);
virtual int getVolumeForSoundType(SoundType type) const;
virtual uint getOutputRate() const;
protected:
void insertChannel(SoundHandle *handle, Channel *chan);
public:
/**
* The mixer callback function, to be called at regular intervals by
* the backend (e.g. from an audio mixing thread). All the actual mixing
* work is done from here.
*/
void mixCallback(byte *samples, uint len);
/**
* Set the internal 'is ready' flag of the mixer.
* Backends should invoke Mixer::setReady(true) once initialisation of
* their audio system has been completed (and in particular, *after*
* setOutputRate() has been called).
*/
void setReady(bool ready);
/**
* Set the output sample rate.
*
* @param sampleRate the new output sample rate
*
* @note Right now, this can be done exactly ONCE. That is, the mixer
* currently does not support changing the output sample rate after it
* has been set for the first time. This may change in the future.
*/
void setOutputRate(uint sampleRate);
};
} // End of namespace Audio
#endif