Reformat all the things. Have fun with merge conflicts.

This commit is contained in:
Pierre Bourdon
2016-06-24 10:43:46 +02:00
parent 2115e8a4a6
commit 3570c7f03a
1116 changed files with 187350 additions and 180289 deletions
File diff suppressed because it is too large Load Diff
+238 -238
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@@ -9,249 +9,249 @@
namespace ButtonManager
{
enum ButtonType
{
// GC
BUTTON_A = 0,
BUTTON_B = 1,
BUTTON_START = 2,
BUTTON_X = 3,
BUTTON_Y = 4,
BUTTON_Z = 5,
BUTTON_UP = 6,
BUTTON_DOWN = 7,
BUTTON_LEFT = 8,
BUTTON_RIGHT = 9,
STICK_MAIN = 10, // Used on Java Side
STICK_MAIN_UP = 11,
STICK_MAIN_DOWN = 12,
STICK_MAIN_LEFT = 13,
STICK_MAIN_RIGHT = 14,
STICK_C = 15, // Used on Java Side
STICK_C_UP = 16,
STICK_C_DOWN = 17,
STICK_C_LEFT = 18,
STICK_C_RIGHT = 19,
TRIGGER_L = 20,
TRIGGER_R = 21,
// Wiimote
WIIMOTE_BUTTON_A = 100,
WIIMOTE_BUTTON_B = 101,
WIIMOTE_BUTTON_MINUS = 102,
WIIMOTE_BUTTON_PLUS = 103,
WIIMOTE_BUTTON_HOME = 104,
WIIMOTE_BUTTON_1 = 105,
WIIMOTE_BUTTON_2 = 106,
WIIMOTE_UP = 107,
WIIMOTE_DOWN = 108,
WIIMOTE_LEFT = 109,
WIIMOTE_RIGHT = 110,
WIIMOTE_IR = 111, // To Be Used on Java Side
WIIMOTE_IR_UP = 112,
WIIMOTE_IR_DOWN = 113,
WIIMOTE_IR_LEFT = 114,
WIIMOTE_IR_RIGHT = 115,
WIIMOTE_IR_FORWARD = 116,
WIIMOTE_IR_BACKWARD = 117,
WIIMOTE_IR_HIDE = 118,
WIIMOTE_SWING = 119, // To Be Used on Java Side
WIIMOTE_SWING_UP = 120,
WIIMOTE_SWING_DOWN = 121,
WIIMOTE_SWING_LEFT = 122,
WIIMOTE_SWING_RIGHT = 123,
WIIMOTE_SWING_FORWARD = 124,
WIIMOTE_SWING_BACKWARD = 125,
WIIMOTE_TILT = 126, // To Be Used on Java Side
WIIMOTE_TILT_FORWARD = 127,
WIIMOTE_TILT_BACKWARD = 128,
WIIMOTE_TILT_LEFT = 129,
WIIMOTE_TILT_RIGHT = 130,
WIIMOTE_TILT_MODIFIER = 131,
WIIMOTE_SHAKE_X = 132,
WIIMOTE_SHAKE_Y = 133,
WIIMOTE_SHAKE_Z = 134,
//Nunchuk
NUNCHUK_BUTTON_C = 200,
NUNCHUK_BUTTON_Z = 201,
NUNCHUK_STICK = 202, // To Be Used on Java Side
NUNCHUK_STICK_UP = 203,
NUNCHUK_STICK_DOWN = 204,
NUNCHUK_STICK_LEFT = 205,
NUNCHUK_STICK_RIGHT = 206,
NUNCHUK_SWING = 207, // To Be Used on Java Side
NUNCHUK_SWING_UP = 208,
NUNCHUK_SWING_DOWN = 209,
NUNCHUK_SWING_LEFT = 210,
NUNCHUK_SWING_RIGHT = 211,
NUNCHUK_SWING_FORWARD = 212,
NUNCHUK_SWING_BACKWARD = 213,
NUNCHUK_TILT = 214, // To Be Used on Java Side
NUNCHUK_TILT_FORWARD = 215,
NUNCHUK_TILT_BACKWARD = 216,
NUNCHUK_TILT_LEFT = 217,
NUNCHUK_TILT_RIGHT = 218,
NUNCHUK_TILT_MODIFIER = 219,
NUNCHUK_SHAKE_X = 220,
NUNCHUK_SHAKE_Y = 221,
NUNCHUK_SHAKE_Z = 222,
//Classic
CLASSIC_BUTTON_A = 300,
CLASSIC_BUTTON_B = 301,
CLASSIC_BUTTON_X = 302,
CLASSIC_BUTTON_Y = 303,
CLASSIC_BUTTON_MINUS = 304,
CLASSIC_BUTTON_PLUS = 305,
CLASSIC_BUTTON_HOME = 306,
CLASSIC_BUTTON_ZL = 307,
CLASSIC_BUTTON_ZR = 308,
CLASSIC_DPAD_UP = 309,
CLASSIC_DPAD_DOWN = 310,
CLASSIC_DPAD_LEFT = 311,
CLASSIC_DPAD_RIGHT = 312,
CLASSIC_STICK_LEFT = 313, // To Be Used on Java Side
CLASSIC_STICK_LEFT_UP = 314,
CLASSIC_STICK_LEFT_DOWN = 315,
CLASSIC_STICK_LEFT_LEFT = 316,
CLASSIC_STICK_LEFT_RIGHT = 317,
CLASSIC_STICK_RIGHT = 318, // To Be Used on Java Side
CLASSIC_STICK_RIGHT_UP = 319,
CLASSIC_STICK_RIGHT_DOWN = 320,
CLASSIC_STICK_RIGHT_LEFT = 321,
CLASSIC_STICK_RIGHT_RIGHT = 322,
CLASSIC_TRIGGER_L = 323,
CLASSIC_TRIGGER_R = 324,
//Guitar
GUITAR_BUTTON_MINUS = 400,
GUITAR_BUTTON_PLUS = 401,
GUITAR_FRET_GREEN = 402,
GUITAR_FRET_RED = 403,
GUITAR_FRET_YELLOW = 404,
GUITAR_FRET_BLUE = 405,
GUITAR_FRET_ORANGE = 406,
GUITAR_STRUM_UP = 407,
GUITAR_STRUM_DOWN = 408,
GUITAR_STICK = 409, // To Be Used on Java Side
GUITAR_STICK_UP = 410,
GUITAR_STICK_DOWN = 411,
GUITAR_STICK_LEFT = 412,
GUITAR_STICK_RIGHT = 413,
GUITAR_WHAMMY_BAR = 414,
//Drums
DRUMS_BUTTON_MINUS = 500,
DRUMS_BUTTON_PLUS = 501,
DRUMS_PAD_RED = 502,
DRUMS_PAD_YELLOW = 503,
DRUMS_PAD_BLUE = 504,
DRUMS_PAD_GREEN = 505,
DRUMS_PAD_ORANGE = 506,
DRUMS_PAD_BASS = 507,
DRUMS_STICK = 508, // To Be Used on Java Side
DRUMS_STICK_UP = 509,
DRUMS_STICK_DOWN = 510,
DRUMS_STICK_LEFT = 511,
DRUMS_STICK_RIGHT = 512,
//Turntable
TURNTABLE_BUTTON_GREEN_LEFT = 600,
TURNTABLE_BUTTON_RED_LEFT = 601,
TURNTABLE_BUTTON_BLUE_LEFT = 602,
TURNTABLE_BUTTON_GREEN_RIGHT = 603,
TURNTABLE_BUTTON_RED_RIGHT = 604,
TURNTABLE_BUTTON_BLUE_RIGHT = 605,
TURNTABLE_BUTTON_MINUS = 606,
TURNTABLE_BUTTON_PLUS = 607,
TURNTABLE_BUTTON_HOME = 608,
TURNTABLE_BUTTON_EUPHORIA = 609,
TURNTABLE_TABLE_LEFT = 610, // To Be Used on Java Side
TURNTABLE_TABLE_LEFT_LEFT = 611,
TURNTABLE_TABLE_LEFT_RIGHT = 612,
TURNTABLE_TABLE_RIGHT = 613, // To Be Used on Java Side
TURNTABLE_TABLE_RIGHT_LEFT = 614,
TURNTABLE_TABLE_RIGHT_RIGHT = 615,
TURNTABLE_STICK = 616, // To Be Used on Java Side
TURNTABLE_STICK_UP = 617,
TURNTABLE_STICK_DOWN = 618,
TURNTABLE_STICK_LEFT = 619,
TURNTABLE_STICK_RIGHT = 620,
TURNTABLE_EFFECT_DIAL = 621,
TURNTABLE_CROSSFADE = 622, // To Be Used on Java Side
TURNTABLE_CROSSFADE_LEFT = 623,
TURNTABLE_CROSSFADE_RIGHT = 624,
};
enum ButtonState
{
BUTTON_RELEASED = 0,
BUTTON_PRESSED = 1
};
enum BindType
{
BIND_BUTTON = 0,
BIND_AXIS
};
class Button
{
private:
ButtonState m_state;
public:
Button() : m_state(BUTTON_RELEASED) {}
void SetState(ButtonState state) { m_state = state; }
bool Pressed() { return m_state == BUTTON_PRESSED; }
enum ButtonType
{
// GC
BUTTON_A = 0,
BUTTON_B = 1,
BUTTON_START = 2,
BUTTON_X = 3,
BUTTON_Y = 4,
BUTTON_Z = 5,
BUTTON_UP = 6,
BUTTON_DOWN = 7,
BUTTON_LEFT = 8,
BUTTON_RIGHT = 9,
STICK_MAIN = 10, // Used on Java Side
STICK_MAIN_UP = 11,
STICK_MAIN_DOWN = 12,
STICK_MAIN_LEFT = 13,
STICK_MAIN_RIGHT = 14,
STICK_C = 15, // Used on Java Side
STICK_C_UP = 16,
STICK_C_DOWN = 17,
STICK_C_LEFT = 18,
STICK_C_RIGHT = 19,
TRIGGER_L = 20,
TRIGGER_R = 21,
// Wiimote
WIIMOTE_BUTTON_A = 100,
WIIMOTE_BUTTON_B = 101,
WIIMOTE_BUTTON_MINUS = 102,
WIIMOTE_BUTTON_PLUS = 103,
WIIMOTE_BUTTON_HOME = 104,
WIIMOTE_BUTTON_1 = 105,
WIIMOTE_BUTTON_2 = 106,
WIIMOTE_UP = 107,
WIIMOTE_DOWN = 108,
WIIMOTE_LEFT = 109,
WIIMOTE_RIGHT = 110,
WIIMOTE_IR = 111, // To Be Used on Java Side
WIIMOTE_IR_UP = 112,
WIIMOTE_IR_DOWN = 113,
WIIMOTE_IR_LEFT = 114,
WIIMOTE_IR_RIGHT = 115,
WIIMOTE_IR_FORWARD = 116,
WIIMOTE_IR_BACKWARD = 117,
WIIMOTE_IR_HIDE = 118,
WIIMOTE_SWING = 119, // To Be Used on Java Side
WIIMOTE_SWING_UP = 120,
WIIMOTE_SWING_DOWN = 121,
WIIMOTE_SWING_LEFT = 122,
WIIMOTE_SWING_RIGHT = 123,
WIIMOTE_SWING_FORWARD = 124,
WIIMOTE_SWING_BACKWARD = 125,
WIIMOTE_TILT = 126, // To Be Used on Java Side
WIIMOTE_TILT_FORWARD = 127,
WIIMOTE_TILT_BACKWARD = 128,
WIIMOTE_TILT_LEFT = 129,
WIIMOTE_TILT_RIGHT = 130,
WIIMOTE_TILT_MODIFIER = 131,
WIIMOTE_SHAKE_X = 132,
WIIMOTE_SHAKE_Y = 133,
WIIMOTE_SHAKE_Z = 134,
// Nunchuk
NUNCHUK_BUTTON_C = 200,
NUNCHUK_BUTTON_Z = 201,
NUNCHUK_STICK = 202, // To Be Used on Java Side
NUNCHUK_STICK_UP = 203,
NUNCHUK_STICK_DOWN = 204,
NUNCHUK_STICK_LEFT = 205,
NUNCHUK_STICK_RIGHT = 206,
NUNCHUK_SWING = 207, // To Be Used on Java Side
NUNCHUK_SWING_UP = 208,
NUNCHUK_SWING_DOWN = 209,
NUNCHUK_SWING_LEFT = 210,
NUNCHUK_SWING_RIGHT = 211,
NUNCHUK_SWING_FORWARD = 212,
NUNCHUK_SWING_BACKWARD = 213,
NUNCHUK_TILT = 214, // To Be Used on Java Side
NUNCHUK_TILT_FORWARD = 215,
NUNCHUK_TILT_BACKWARD = 216,
NUNCHUK_TILT_LEFT = 217,
NUNCHUK_TILT_RIGHT = 218,
NUNCHUK_TILT_MODIFIER = 219,
NUNCHUK_SHAKE_X = 220,
NUNCHUK_SHAKE_Y = 221,
NUNCHUK_SHAKE_Z = 222,
// Classic
CLASSIC_BUTTON_A = 300,
CLASSIC_BUTTON_B = 301,
CLASSIC_BUTTON_X = 302,
CLASSIC_BUTTON_Y = 303,
CLASSIC_BUTTON_MINUS = 304,
CLASSIC_BUTTON_PLUS = 305,
CLASSIC_BUTTON_HOME = 306,
CLASSIC_BUTTON_ZL = 307,
CLASSIC_BUTTON_ZR = 308,
CLASSIC_DPAD_UP = 309,
CLASSIC_DPAD_DOWN = 310,
CLASSIC_DPAD_LEFT = 311,
CLASSIC_DPAD_RIGHT = 312,
CLASSIC_STICK_LEFT = 313, // To Be Used on Java Side
CLASSIC_STICK_LEFT_UP = 314,
CLASSIC_STICK_LEFT_DOWN = 315,
CLASSIC_STICK_LEFT_LEFT = 316,
CLASSIC_STICK_LEFT_RIGHT = 317,
CLASSIC_STICK_RIGHT = 318, // To Be Used on Java Side
CLASSIC_STICK_RIGHT_UP = 319,
CLASSIC_STICK_RIGHT_DOWN = 320,
CLASSIC_STICK_RIGHT_LEFT = 321,
CLASSIC_STICK_RIGHT_RIGHT = 322,
CLASSIC_TRIGGER_L = 323,
CLASSIC_TRIGGER_R = 324,
// Guitar
GUITAR_BUTTON_MINUS = 400,
GUITAR_BUTTON_PLUS = 401,
GUITAR_FRET_GREEN = 402,
GUITAR_FRET_RED = 403,
GUITAR_FRET_YELLOW = 404,
GUITAR_FRET_BLUE = 405,
GUITAR_FRET_ORANGE = 406,
GUITAR_STRUM_UP = 407,
GUITAR_STRUM_DOWN = 408,
GUITAR_STICK = 409, // To Be Used on Java Side
GUITAR_STICK_UP = 410,
GUITAR_STICK_DOWN = 411,
GUITAR_STICK_LEFT = 412,
GUITAR_STICK_RIGHT = 413,
GUITAR_WHAMMY_BAR = 414,
// Drums
DRUMS_BUTTON_MINUS = 500,
DRUMS_BUTTON_PLUS = 501,
DRUMS_PAD_RED = 502,
DRUMS_PAD_YELLOW = 503,
DRUMS_PAD_BLUE = 504,
DRUMS_PAD_GREEN = 505,
DRUMS_PAD_ORANGE = 506,
DRUMS_PAD_BASS = 507,
DRUMS_STICK = 508, // To Be Used on Java Side
DRUMS_STICK_UP = 509,
DRUMS_STICK_DOWN = 510,
DRUMS_STICK_LEFT = 511,
DRUMS_STICK_RIGHT = 512,
// Turntable
TURNTABLE_BUTTON_GREEN_LEFT = 600,
TURNTABLE_BUTTON_RED_LEFT = 601,
TURNTABLE_BUTTON_BLUE_LEFT = 602,
TURNTABLE_BUTTON_GREEN_RIGHT = 603,
TURNTABLE_BUTTON_RED_RIGHT = 604,
TURNTABLE_BUTTON_BLUE_RIGHT = 605,
TURNTABLE_BUTTON_MINUS = 606,
TURNTABLE_BUTTON_PLUS = 607,
TURNTABLE_BUTTON_HOME = 608,
TURNTABLE_BUTTON_EUPHORIA = 609,
TURNTABLE_TABLE_LEFT = 610, // To Be Used on Java Side
TURNTABLE_TABLE_LEFT_LEFT = 611,
TURNTABLE_TABLE_LEFT_RIGHT = 612,
TURNTABLE_TABLE_RIGHT = 613, // To Be Used on Java Side
TURNTABLE_TABLE_RIGHT_LEFT = 614,
TURNTABLE_TABLE_RIGHT_RIGHT = 615,
TURNTABLE_STICK = 616, // To Be Used on Java Side
TURNTABLE_STICK_UP = 617,
TURNTABLE_STICK_DOWN = 618,
TURNTABLE_STICK_LEFT = 619,
TURNTABLE_STICK_RIGHT = 620,
TURNTABLE_EFFECT_DIAL = 621,
TURNTABLE_CROSSFADE = 622, // To Be Used on Java Side
TURNTABLE_CROSSFADE_LEFT = 623,
TURNTABLE_CROSSFADE_RIGHT = 624,
};
enum ButtonState
{
BUTTON_RELEASED = 0,
BUTTON_PRESSED = 1
};
enum BindType
{
BIND_BUTTON = 0,
BIND_AXIS
};
class Button
{
private:
ButtonState m_state;
~Button() {}
};
class Axis
{
private:
float m_value;
public:
Axis() : m_value(0.0f) {}
void SetValue(float value) { m_value = value; }
float AxisValue() { return m_value; }
public:
Button() : m_state(BUTTON_RELEASED) {}
void SetState(ButtonState state) { m_state = state; }
bool Pressed() { return m_state == BUTTON_PRESSED; }
~Button() {}
};
class Axis
{
private:
float m_value;
~Axis() {}
};
public:
Axis() : m_value(0.0f) {}
void SetValue(float value) { m_value = value; }
float AxisValue() { return m_value; }
~Axis() {}
};
struct sBind
{
const int _padID;
const ButtonType _buttontype;
const BindType _bindtype;
const int _bind;
const float _neg;
sBind(int padID, ButtonType buttontype, BindType bindtype, int bind, float neg)
: _padID(padID), _buttontype(buttontype), _bindtype(bindtype), _bind(bind), _neg(neg)
{}
};
struct sBind
{
const int _padID;
const ButtonType _buttontype;
const BindType _bindtype;
const int _bind;
const float _neg;
sBind(int padID, ButtonType buttontype, BindType bindtype, int bind, float neg)
: _padID(padID), _buttontype(buttontype), _bindtype(bindtype), _bind(bind), _neg(neg)
{
}
};
class InputDevice
{
private:
const std::string _dev;
std::map<ButtonType, bool> _buttons;
std::map<ButtonType, float> _axises;
class InputDevice
{
private:
const std::string _dev;
std::map<ButtonType, bool> _buttons;
std::map<ButtonType, float> _axises;
// Key is padID and ButtonType
std::map<std::pair<int, ButtonType>, sBind*> _inputbinds;
// Key is padID and ButtonType
std::map<std::pair<int, ButtonType>, sBind*> _inputbinds;
public:
InputDevice(std::string dev)
: _dev(dev) {}
~InputDevice()
{
for (const auto& bind : _inputbinds)
delete bind.second;
_inputbinds.clear();
}
void AddBind(sBind* bind) { _inputbinds[std::make_pair(bind->_padID, bind->_buttontype)] = bind; }
bool PressEvent(int button, int action);
void AxisEvent(int axis, float value);
bool ButtonValue(int padID, ButtonType button);
float AxisValue(int padID, ButtonType axis);
};
public:
InputDevice(std::string dev) : _dev(dev) {}
~InputDevice()
{
for (const auto& bind : _inputbinds)
delete bind.second;
_inputbinds.clear();
}
void AddBind(sBind* bind) { _inputbinds[std::make_pair(bind->_padID, bind->_buttontype)] = bind; }
bool PressEvent(int button, int action);
void AxisEvent(int axis, float value);
bool ButtonValue(int padID, ButtonType button);
float AxisValue(int padID, ButtonType axis);
};
void Init();
bool GetButtonPressed(int padID, ButtonType button);
float GetAxisValue(int padID, ButtonType axis);
bool GamepadEvent(const std::string& dev, int button, int action);
void GamepadAxisEvent(const std::string& dev, int axis, float value);
void Shutdown();
void Init();
bool GetButtonPressed(int padID, ButtonType button);
float GetAxisValue(int padID, ButtonType axis);
bool GamepadEvent(const std::string& dev, int button, int action);
void GamepadAxisEvent(const std::string& dev, int axis, float value);
void Shutdown();
}
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+42 -42
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@@ -6,77 +6,77 @@
#include "AudioCommon/AOSoundStream.h"
#include "AudioCommon/Mixer.h"
#include "Common/MsgHandler.h"
#include "Common/Logging/Log.h"
#include "Common/MsgHandler.h"
#if defined(HAVE_AO) && HAVE_AO
void AOSound::SoundLoop()
{
Common::SetCurrentThreadName("Audio thread - ao");
Common::SetCurrentThreadName("Audio thread - ao");
uint_32 numBytesToRender = 256;
ao_initialize();
default_driver = ao_default_driver_id();
format.bits = 16;
format.channels = 2;
format.rate = m_mixer->GetSampleRate();
format.byte_format = AO_FMT_LITTLE;
uint_32 numBytesToRender = 256;
ao_initialize();
default_driver = ao_default_driver_id();
format.bits = 16;
format.channels = 2;
format.rate = m_mixer->GetSampleRate();
format.byte_format = AO_FMT_LITTLE;
device = ao_open_live(default_driver, &format, nullptr /* no options */);
if (!device)
{
PanicAlertT("AudioCommon: Error opening AO device.\n");
ao_shutdown();
Stop();
return;
}
device = ao_open_live(default_driver, &format, nullptr /* no options */);
if (!device)
{
PanicAlertT("AudioCommon: Error opening AO device.\n");
ao_shutdown();
Stop();
return;
}
buf_size = format.bits/8 * format.channels * format.rate;
buf_size = format.bits / 8 * format.channels * format.rate;
while (m_run_thread.load())
{
m_mixer->Mix(realtimeBuffer, numBytesToRender >> 2);
while (m_run_thread.load())
{
m_mixer->Mix(realtimeBuffer, numBytesToRender >> 2);
{
std::lock_guard<std::mutex> lk(soundCriticalSection);
ao_play(device, (char*)realtimeBuffer, numBytesToRender);
}
{
std::lock_guard<std::mutex> lk(soundCriticalSection);
ao_play(device, (char*)realtimeBuffer, numBytesToRender);
}
soundSyncEvent.Wait();
}
soundSyncEvent.Wait();
}
}
bool AOSound::Start()
{
m_run_thread.store(true);
memset(realtimeBuffer, 0, sizeof(realtimeBuffer));
m_run_thread.store(true);
memset(realtimeBuffer, 0, sizeof(realtimeBuffer));
thread = std::thread(&AOSound::SoundLoop, this);
return true;
thread = std::thread(&AOSound::SoundLoop, this);
return true;
}
void AOSound::Update()
{
soundSyncEvent.Set();
soundSyncEvent.Set();
}
void AOSound::Stop()
{
m_run_thread.store(false);
soundSyncEvent.Set();
m_run_thread.store(false);
soundSyncEvent.Set();
{
std::lock_guard<std::mutex> lk(soundCriticalSection);
thread.join();
{
std::lock_guard<std::mutex> lk(soundCriticalSection);
thread.join();
if (device)
ao_close(device);
if (device)
ao_close(device);
ao_shutdown();
ao_shutdown();
device = nullptr;
}
device = nullptr;
}
}
#endif
+14 -18
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@@ -19,29 +19,25 @@
class AOSound final : public SoundStream
{
#if defined(HAVE_AO) && HAVE_AO
std::thread thread;
std::atomic<bool> m_run_thread;
std::mutex soundCriticalSection;
Common::Event soundSyncEvent;
std::thread thread;
std::atomic<bool> m_run_thread;
std::mutex soundCriticalSection;
Common::Event soundSyncEvent;
int buf_size;
int buf_size;
ao_device *device;
ao_sample_format format;
int default_driver;
ao_device* device;
ao_sample_format format;
int default_driver;
short realtimeBuffer[1024 * 1024];
short realtimeBuffer[1024 * 1024];
public:
bool Start() override;
void SoundLoop() override;
void Stop() override;
void Update() override;
static bool isValid()
{
return true;
}
bool Start() override;
void SoundLoop() override;
void Stop() override;
void Update() override;
static bool isValid() { return true; }
#endif
};
+170 -171
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@@ -6,230 +6,229 @@
#include "AudioCommon/AlsaSoundStream.h"
#include "Common/CommonTypes.h"
#include "Common/Thread.h"
#include "Common/Logging/Log.h"
#include "Common/Thread.h"
AlsaSound::AlsaSound()
: m_thread_status(ALSAThreadStatus::STOPPED)
, handle(nullptr)
, frames_to_deliver(FRAME_COUNT_MIN)
: m_thread_status(ALSAThreadStatus::STOPPED), handle(nullptr),
frames_to_deliver(FRAME_COUNT_MIN)
{
}
bool AlsaSound::Start()
{
m_thread_status.store(ALSAThreadStatus::RUNNING);
if (!AlsaInit())
{
m_thread_status.store(ALSAThreadStatus::STOPPED);
return false;
}
m_thread_status.store(ALSAThreadStatus::RUNNING);
if (!AlsaInit())
{
m_thread_status.store(ALSAThreadStatus::STOPPED);
return false;
}
thread = std::thread(&AlsaSound::SoundLoop, this);
return true;
thread = std::thread(&AlsaSound::SoundLoop, this);
return true;
}
void AlsaSound::Stop()
{
m_thread_status.store(ALSAThreadStatus::STOPPING);
m_thread_status.store(ALSAThreadStatus::STOPPING);
//Give the opportunity to the audio thread
//to realize we are stopping the emulation
cv.notify_one();
thread.join();
// Give the opportunity to the audio thread
// to realize we are stopping the emulation
cv.notify_one();
thread.join();
}
void AlsaSound::Update()
{
// don't need to do anything here.
// don't need to do anything here.
}
// Called on audio thread.
void AlsaSound::SoundLoop()
{
Common::SetCurrentThreadName("Audio thread - alsa");
while (m_thread_status.load() != ALSAThreadStatus::STOPPING)
{
while (m_thread_status.load() == ALSAThreadStatus::RUNNING)
{
m_mixer->Mix(mix_buffer, frames_to_deliver);
int rc = snd_pcm_writei(handle, mix_buffer, frames_to_deliver);
if (rc == -EPIPE)
{
// Underrun
snd_pcm_prepare(handle);
}
else if (rc < 0)
{
ERROR_LOG(AUDIO, "writei fail: %s", snd_strerror(rc));
}
}
if (m_thread_status.load() == ALSAThreadStatus::PAUSED)
{
snd_pcm_drop(handle); // Stop sound output
Common::SetCurrentThreadName("Audio thread - alsa");
while (m_thread_status.load() != ALSAThreadStatus::STOPPING)
{
while (m_thread_status.load() == ALSAThreadStatus::RUNNING)
{
m_mixer->Mix(mix_buffer, frames_to_deliver);
int rc = snd_pcm_writei(handle, mix_buffer, frames_to_deliver);
if (rc == -EPIPE)
{
// Underrun
snd_pcm_prepare(handle);
}
else if (rc < 0)
{
ERROR_LOG(AUDIO, "writei fail: %s", snd_strerror(rc));
}
}
if (m_thread_status.load() == ALSAThreadStatus::PAUSED)
{
snd_pcm_drop(handle); // Stop sound output
// Block until thread status changes.
std::unique_lock<std::mutex> lock(cv_m);
cv.wait(lock, [this]{ return m_thread_status.load() != ALSAThreadStatus::PAUSED; });
// Block until thread status changes.
std::unique_lock<std::mutex> lock(cv_m);
cv.wait(lock, [this] { return m_thread_status.load() != ALSAThreadStatus::PAUSED; });
snd_pcm_prepare(handle); // resume sound output
}
}
AlsaShutdown();
m_thread_status.store(ALSAThreadStatus::STOPPED);
snd_pcm_prepare(handle); // resume sound output
}
}
AlsaShutdown();
m_thread_status.store(ALSAThreadStatus::STOPPED);
}
void AlsaSound::Clear(bool muted)
{
m_muted = muted;
m_thread_status.store(muted ? ALSAThreadStatus::PAUSED : ALSAThreadStatus::RUNNING);
cv.notify_one(); // Notify thread that status has changed
m_muted = muted;
m_thread_status.store(muted ? ALSAThreadStatus::PAUSED : ALSAThreadStatus::RUNNING);
cv.notify_one(); // Notify thread that status has changed
}
bool AlsaSound::AlsaInit()
{
unsigned int sample_rate = m_mixer->GetSampleRate();
int err;
int dir;
snd_pcm_sw_params_t *swparams;
snd_pcm_hw_params_t *hwparams;
snd_pcm_uframes_t buffer_size,buffer_size_max;
unsigned int periods;
unsigned int sample_rate = m_mixer->GetSampleRate();
int err;
int dir;
snd_pcm_sw_params_t* swparams;
snd_pcm_hw_params_t* hwparams;
snd_pcm_uframes_t buffer_size, buffer_size_max;
unsigned int periods;
err = snd_pcm_open(&handle, "default", SND_PCM_STREAM_PLAYBACK, 0);
if (err < 0)
{
ERROR_LOG(AUDIO, "Audio open error: %s\n", snd_strerror(err));
return false;
}
err = snd_pcm_open(&handle, "default", SND_PCM_STREAM_PLAYBACK, 0);
if (err < 0)
{
ERROR_LOG(AUDIO, "Audio open error: %s\n", snd_strerror(err));
return false;
}
snd_pcm_hw_params_alloca(&hwparams);
snd_pcm_hw_params_alloca(&hwparams);
err = snd_pcm_hw_params_any(handle, hwparams);
if (err < 0)
{
ERROR_LOG(AUDIO, "Broken configuration for this PCM: %s\n", snd_strerror(err));
return false;
}
err = snd_pcm_hw_params_any(handle, hwparams);
if (err < 0)
{
ERROR_LOG(AUDIO, "Broken configuration for this PCM: %s\n", snd_strerror(err));
return false;
}
err = snd_pcm_hw_params_set_access(handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);
if (err < 0)
{
ERROR_LOG(AUDIO, "Access type not available: %s\n", snd_strerror(err));
return false;
}
err = snd_pcm_hw_params_set_access(handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);
if (err < 0)
{
ERROR_LOG(AUDIO, "Access type not available: %s\n", snd_strerror(err));
return false;
}
err = snd_pcm_hw_params_set_format(handle, hwparams, SND_PCM_FORMAT_S16_LE);
if (err < 0)
{
ERROR_LOG(AUDIO, "Sample format not available: %s\n", snd_strerror(err));
return false;
}
err = snd_pcm_hw_params_set_format(handle, hwparams, SND_PCM_FORMAT_S16_LE);
if (err < 0)
{
ERROR_LOG(AUDIO, "Sample format not available: %s\n", snd_strerror(err));
return false;
}
dir = 0;
err = snd_pcm_hw_params_set_rate_near(handle, hwparams, &sample_rate, &dir);
if (err < 0)
{
ERROR_LOG(AUDIO, "Rate not available: %s\n", snd_strerror(err));
return false;
}
dir = 0;
err = snd_pcm_hw_params_set_rate_near(handle, hwparams, &sample_rate, &dir);
if (err < 0)
{
ERROR_LOG(AUDIO, "Rate not available: %s\n", snd_strerror(err));
return false;
}
err = snd_pcm_hw_params_set_channels(handle, hwparams, CHANNEL_COUNT);
if (err < 0)
{
ERROR_LOG(AUDIO, "Channels count not available: %s\n", snd_strerror(err));
return false;
}
err = snd_pcm_hw_params_set_channels(handle, hwparams, CHANNEL_COUNT);
if (err < 0)
{
ERROR_LOG(AUDIO, "Channels count not available: %s\n", snd_strerror(err));
return false;
}
periods = BUFFER_SIZE_MAX / FRAME_COUNT_MIN;
err = snd_pcm_hw_params_set_periods_max(handle, hwparams, &periods, &dir);
if (err < 0)
{
ERROR_LOG(AUDIO, "Cannot set maximum periods per buffer: %s\n", snd_strerror(err));
return false;
}
periods = BUFFER_SIZE_MAX / FRAME_COUNT_MIN;
err = snd_pcm_hw_params_set_periods_max(handle, hwparams, &periods, &dir);
if (err < 0)
{
ERROR_LOG(AUDIO, "Cannot set maximum periods per buffer: %s\n", snd_strerror(err));
return false;
}
buffer_size_max = BUFFER_SIZE_MAX;
err = snd_pcm_hw_params_set_buffer_size_max(handle, hwparams, &buffer_size_max);
if (err < 0)
{
ERROR_LOG(AUDIO, "Cannot set maximum buffer size: %s\n", snd_strerror(err));
return false;
}
buffer_size_max = BUFFER_SIZE_MAX;
err = snd_pcm_hw_params_set_buffer_size_max(handle, hwparams, &buffer_size_max);
if (err < 0)
{
ERROR_LOG(AUDIO, "Cannot set maximum buffer size: %s\n", snd_strerror(err));
return false;
}
err = snd_pcm_hw_params(handle, hwparams);
if (err < 0)
{
ERROR_LOG(AUDIO, "Unable to install hw params: %s\n", snd_strerror(err));
return false;
}
err = snd_pcm_hw_params(handle, hwparams);
if (err < 0)
{
ERROR_LOG(AUDIO, "Unable to install hw params: %s\n", snd_strerror(err));
return false;
}
err = snd_pcm_hw_params_get_buffer_size(hwparams, &buffer_size);
if (err < 0)
{
ERROR_LOG(AUDIO, "Cannot get buffer size: %s\n", snd_strerror(err));
return false;
}
err = snd_pcm_hw_params_get_buffer_size(hwparams, &buffer_size);
if (err < 0)
{
ERROR_LOG(AUDIO, "Cannot get buffer size: %s\n", snd_strerror(err));
return false;
}
err = snd_pcm_hw_params_get_periods_max(hwparams, &periods, &dir);
if (err < 0)
{
ERROR_LOG(AUDIO, "Cannot get periods: %s\n", snd_strerror(err));
return false;
}
err = snd_pcm_hw_params_get_periods_max(hwparams, &periods, &dir);
if (err < 0)
{
ERROR_LOG(AUDIO, "Cannot get periods: %s\n", snd_strerror(err));
return false;
}
//periods is the number of fragments alsa can wait for during one
//buffer_size
frames_to_deliver = buffer_size / periods;
//limit the minimum size. pulseaudio advertises a minimum of 32 samples.
if (frames_to_deliver < FRAME_COUNT_MIN)
frames_to_deliver = FRAME_COUNT_MIN;
//it is probably a bad idea to try to send more than one buffer of data
if ((unsigned int)frames_to_deliver > buffer_size)
frames_to_deliver = buffer_size;
NOTICE_LOG(AUDIO, "ALSA gave us a %ld sample \"hardware\" buffer with %d periods. Will send %d samples per fragments.\n", buffer_size, periods, frames_to_deliver);
// periods is the number of fragments alsa can wait for during one
// buffer_size
frames_to_deliver = buffer_size / periods;
// limit the minimum size. pulseaudio advertises a minimum of 32 samples.
if (frames_to_deliver < FRAME_COUNT_MIN)
frames_to_deliver = FRAME_COUNT_MIN;
// it is probably a bad idea to try to send more than one buffer of data
if ((unsigned int)frames_to_deliver > buffer_size)
frames_to_deliver = buffer_size;
NOTICE_LOG(AUDIO, "ALSA gave us a %ld sample \"hardware\" buffer with %d periods. Will send %d "
"samples per fragments.\n",
buffer_size, periods, frames_to_deliver);
snd_pcm_sw_params_alloca(&swparams);
snd_pcm_sw_params_alloca(&swparams);
err = snd_pcm_sw_params_current(handle, swparams);
if (err < 0)
{
ERROR_LOG(AUDIO, "cannot init sw params: %s\n", snd_strerror(err));
return false;
}
err = snd_pcm_sw_params_current(handle, swparams);
if (err < 0)
{
ERROR_LOG(AUDIO, "cannot init sw params: %s\n", snd_strerror(err));
return false;
}
err = snd_pcm_sw_params_set_start_threshold(handle, swparams, 0U);
if (err < 0)
{
ERROR_LOG(AUDIO, "cannot set start thresh: %s\n", snd_strerror(err));
return false;
}
err = snd_pcm_sw_params_set_start_threshold(handle, swparams, 0U);
if (err < 0)
{
ERROR_LOG(AUDIO, "cannot set start thresh: %s\n", snd_strerror(err));
return false;
}
err = snd_pcm_sw_params(handle, swparams);
if (err < 0)
{
ERROR_LOG(AUDIO, "cannot set sw params: %s\n", snd_strerror(err));
return false;
}
err = snd_pcm_sw_params(handle, swparams);
if (err < 0)
{
ERROR_LOG(AUDIO, "cannot set sw params: %s\n", snd_strerror(err));
return false;
}
err = snd_pcm_prepare(handle);
if (err < 0)
{
ERROR_LOG(AUDIO, "Unable to prepare: %s\n", snd_strerror(err));
return false;
}
NOTICE_LOG(AUDIO, "ALSA successfully initialized.\n");
return true;
err = snd_pcm_prepare(handle);
if (err < 0)
{
ERROR_LOG(AUDIO, "Unable to prepare: %s\n", snd_strerror(err));
return false;
}
NOTICE_LOG(AUDIO, "ALSA successfully initialized.\n");
return true;
}
void AlsaSound::AlsaShutdown()
{
if (handle != nullptr)
{
snd_pcm_drop(handle);
snd_pcm_close(handle);
handle = nullptr;
}
if (handle != nullptr)
{
snd_pcm_drop(handle);
snd_pcm_close(handle);
handle = nullptr;
}
}
+29 -33
View File
@@ -20,47 +20,43 @@ class AlsaSound final : public SoundStream
{
#if defined(HAVE_ALSA) && HAVE_ALSA
public:
AlsaSound();
AlsaSound();
bool Start() override;
void SoundLoop() override;
void Stop() override;
void Update() override;
void Clear(bool) override;
static bool isValid()
{
return true;
}
bool Start() override;
void SoundLoop() override;
void Stop() override;
void Update() override;
void Clear(bool) override;
static bool isValid() { return true; }
private:
// maximum number of frames the buffer can hold
static constexpr size_t BUFFER_SIZE_MAX = 8192;
// maximum number of frames the buffer can hold
static constexpr size_t BUFFER_SIZE_MAX = 8192;
// minimum number of frames to deliver in one transfer
static constexpr u32 FRAME_COUNT_MIN = 256;
// minimum number of frames to deliver in one transfer
static constexpr u32 FRAME_COUNT_MIN = 256;
// number of channels per frame
static constexpr u32 CHANNEL_COUNT = 2;
// number of channels per frame
static constexpr u32 CHANNEL_COUNT = 2;
enum class ALSAThreadStatus
{
RUNNING,
PAUSED,
STOPPING,
STOPPED,
};
enum class ALSAThreadStatus
{
RUNNING,
PAUSED,
STOPPING,
STOPPED,
};
bool AlsaInit();
void AlsaShutdown();
bool AlsaInit();
void AlsaShutdown();
s16 mix_buffer[BUFFER_SIZE_MAX * CHANNEL_COUNT];
std::thread thread;
std::atomic<ALSAThreadStatus> m_thread_status;
std::condition_variable cv;
std::mutex cv_m;
s16 mix_buffer[BUFFER_SIZE_MAX * CHANNEL_COUNT];
std::thread thread;
std::atomic<ALSAThreadStatus> m_thread_status;
std::condition_variable cv;
std::mutex cv_m;
snd_pcm_t *handle;
unsigned int frames_to_deliver;
snd_pcm_t* handle;
unsigned int frames_to_deliver;
#endif
};
+175 -176
View File
@@ -2,22 +2,21 @@
// Licensed under GPLv2+
// Refer to the license.txt file included.
#include "AudioCommon/AlsaSoundStream.h"
#include "AudioCommon/AOSoundStream.h"
#include "AudioCommon/AudioCommon.h"
#include "AudioCommon/AOSoundStream.h"
#include "AudioCommon/AlsaSoundStream.h"
#include "AudioCommon/CoreAudioSoundStream.h"
#include "AudioCommon/Mixer.h"
#include "AudioCommon/NullSoundStream.h"
#include "AudioCommon/OpenALStream.h"
#include "AudioCommon/OpenSLESStream.h"
#include "AudioCommon/PulseAudioStream.h"
#include "AudioCommon/XAudio2_7Stream.h"
#include "AudioCommon/XAudio2Stream.h"
#include "AudioCommon/XAudio2_7Stream.h"
#include "Common/Common.h"
#include "Common/FileUtil.h"
#include "Common/MsgHandler.h"
#include "Common/Logging/Log.h"
#include "Common/MsgHandler.h"
#include "Core/ConfigManager.h"
#include "Core/Movie.h"
@@ -28,182 +27,182 @@ static bool s_audio_dump_start = false;
namespace AudioCommon
{
static const int AUDIO_VOLUME_MIN = 0;
static const int AUDIO_VOLUME_MAX = 100;
static const int AUDIO_VOLUME_MIN = 0;
static const int AUDIO_VOLUME_MAX = 100;
SoundStream* InitSoundStream()
{
std::string backend = SConfig::GetInstance().sBackend;
if (backend == BACKEND_OPENAL && OpenALStream::isValid())
g_sound_stream = new OpenALStream();
else if (backend == BACKEND_NULLSOUND && NullSound::isValid())
g_sound_stream = new NullSound();
else if (backend == BACKEND_XAUDIO2)
{
if (XAudio2::isValid())
g_sound_stream = new XAudio2();
else if (XAudio2_7::isValid())
g_sound_stream = new XAudio2_7();
}
else if (backend == BACKEND_AOSOUND && AOSound::isValid())
g_sound_stream = new AOSound();
else if (backend == BACKEND_ALSA && AlsaSound::isValid())
g_sound_stream = new AlsaSound();
else if (backend == BACKEND_COREAUDIO && CoreAudioSound::isValid())
g_sound_stream = new CoreAudioSound();
else if (backend == BACKEND_PULSEAUDIO && PulseAudio::isValid())
g_sound_stream = new PulseAudio();
else if (backend == BACKEND_OPENSLES && OpenSLESStream::isValid())
g_sound_stream = new OpenSLESStream();
SoundStream* InitSoundStream()
{
std::string backend = SConfig::GetInstance().sBackend;
if (backend == BACKEND_OPENAL && OpenALStream::isValid())
g_sound_stream = new OpenALStream();
else if (backend == BACKEND_NULLSOUND && NullSound::isValid())
g_sound_stream = new NullSound();
else if (backend == BACKEND_XAUDIO2)
{
if (XAudio2::isValid())
g_sound_stream = new XAudio2();
else if (XAudio2_7::isValid())
g_sound_stream = new XAudio2_7();
}
else if (backend == BACKEND_AOSOUND && AOSound::isValid())
g_sound_stream = new AOSound();
else if (backend == BACKEND_ALSA && AlsaSound::isValid())
g_sound_stream = new AlsaSound();
else if (backend == BACKEND_COREAUDIO && CoreAudioSound::isValid())
g_sound_stream = new CoreAudioSound();
else if (backend == BACKEND_PULSEAUDIO && PulseAudio::isValid())
g_sound_stream = new PulseAudio();
else if (backend == BACKEND_OPENSLES && OpenSLESStream::isValid())
g_sound_stream = new OpenSLESStream();
if (!g_sound_stream && NullSound::isValid())
{
WARN_LOG(AUDIO, "Could not initialize backend %s, using %s instead.",
backend.c_str(), BACKEND_NULLSOUND);
g_sound_stream = new NullSound();
}
if (!g_sound_stream && NullSound::isValid())
{
WARN_LOG(AUDIO, "Could not initialize backend %s, using %s instead.", backend.c_str(),
BACKEND_NULLSOUND);
g_sound_stream = new NullSound();
}
if (g_sound_stream)
{
UpdateSoundStream();
if (!g_sound_stream->Start())
{
ERROR_LOG(AUDIO, "Could not start backend %s, using %s instead",
backend.c_str(), BACKEND_NULLSOUND);
delete g_sound_stream;
g_sound_stream = new NullSound();
g_sound_stream->Start();
}
if (g_sound_stream)
{
UpdateSoundStream();
if (!g_sound_stream->Start())
{
ERROR_LOG(AUDIO, "Could not start backend %s, using %s instead", backend.c_str(),
BACKEND_NULLSOUND);
delete g_sound_stream;
g_sound_stream = new NullSound();
g_sound_stream->Start();
}
if (SConfig::GetInstance().m_DumpAudio && !s_audio_dump_start)
StartAudioDump();
if (SConfig::GetInstance().m_DumpAudio && !s_audio_dump_start)
StartAudioDump();
return g_sound_stream;
}
return g_sound_stream;
}
PanicAlertT("Sound backend %s is not valid.", backend.c_str());
PanicAlertT("Sound backend %s is not valid.", backend.c_str());
delete g_sound_stream;
g_sound_stream = nullptr;
return nullptr;
}
void ShutdownSoundStream()
{
INFO_LOG(AUDIO, "Shutting down sound stream");
if (g_sound_stream)
{
g_sound_stream->Stop();
if (SConfig::GetInstance().m_DumpAudio && s_audio_dump_start)
StopAudioDump();
delete g_sound_stream;
g_sound_stream = nullptr;
}
INFO_LOG(AUDIO, "Done shutting down sound stream");
}
std::vector<std::string> GetSoundBackends()
{
std::vector<std::string> backends;
if (NullSound::isValid())
backends.push_back(BACKEND_NULLSOUND);
if (XAudio2_7::isValid() || XAudio2::isValid())
backends.push_back(BACKEND_XAUDIO2);
if (AOSound::isValid())
backends.push_back(BACKEND_AOSOUND);
if (AlsaSound::isValid())
backends.push_back(BACKEND_ALSA);
if (CoreAudioSound::isValid())
backends.push_back(BACKEND_COREAUDIO);
if (PulseAudio::isValid())
backends.push_back(BACKEND_PULSEAUDIO);
if (OpenALStream::isValid())
backends.push_back(BACKEND_OPENAL);
if (OpenSLESStream::isValid())
backends.push_back(BACKEND_OPENSLES);
return backends;
}
void UpdateSoundStream()
{
if (g_sound_stream)
{
int volume = SConfig::GetInstance().m_IsMuted ? 0 : SConfig::GetInstance().m_Volume;
g_sound_stream->SetVolume(volume);
}
}
void ClearAudioBuffer(bool mute)
{
if (g_sound_stream)
g_sound_stream->Clear(mute);
}
void SendAIBuffer(short *samples, unsigned int num_samples)
{
if (!g_sound_stream)
return;
if (SConfig::GetInstance().m_DumpAudio && !s_audio_dump_start)
StartAudioDump();
else if (!SConfig::GetInstance().m_DumpAudio && s_audio_dump_start)
StopAudioDump();
CMixer* pMixer = g_sound_stream->GetMixer();
if (pMixer && samples)
{
pMixer->PushSamples(samples, num_samples);
}
g_sound_stream->Update();
}
void StartAudioDump()
{
std::string audio_file_name_dtk = File::GetUserPath(D_DUMPAUDIO_IDX) + "dtkdump.wav";
std::string audio_file_name_dsp = File::GetUserPath(D_DUMPAUDIO_IDX) + "dspdump.wav";
File::CreateFullPath(audio_file_name_dtk);
File::CreateFullPath(audio_file_name_dsp);
g_sound_stream->GetMixer()->StartLogDTKAudio(audio_file_name_dtk);
g_sound_stream->GetMixer()->StartLogDSPAudio(audio_file_name_dsp);
s_audio_dump_start = true;
}
void StopAudioDump()
{
g_sound_stream->GetMixer()->StopLogDTKAudio();
g_sound_stream->GetMixer()->StopLogDSPAudio();
s_audio_dump_start = false;
}
void IncreaseVolume(unsigned short offset)
{
SConfig::GetInstance().m_IsMuted = false;
int& currentVolume = SConfig::GetInstance().m_Volume;
currentVolume += offset;
if (currentVolume > AUDIO_VOLUME_MAX)
currentVolume = AUDIO_VOLUME_MAX;
UpdateSoundStream();
}
void DecreaseVolume(unsigned short offset)
{
SConfig::GetInstance().m_IsMuted = false;
int& currentVolume = SConfig::GetInstance().m_Volume;
currentVolume -= offset;
if (currentVolume < AUDIO_VOLUME_MIN)
currentVolume = AUDIO_VOLUME_MIN;
UpdateSoundStream();
}
void ToggleMuteVolume()
{
bool& isMuted = SConfig::GetInstance().m_IsMuted;
isMuted = !isMuted;
UpdateSoundStream();
}
delete g_sound_stream;
g_sound_stream = nullptr;
return nullptr;
}
void ShutdownSoundStream()
{
INFO_LOG(AUDIO, "Shutting down sound stream");
if (g_sound_stream)
{
g_sound_stream->Stop();
if (SConfig::GetInstance().m_DumpAudio && s_audio_dump_start)
StopAudioDump();
delete g_sound_stream;
g_sound_stream = nullptr;
}
INFO_LOG(AUDIO, "Done shutting down sound stream");
}
std::vector<std::string> GetSoundBackends()
{
std::vector<std::string> backends;
if (NullSound::isValid())
backends.push_back(BACKEND_NULLSOUND);
if (XAudio2_7::isValid() || XAudio2::isValid())
backends.push_back(BACKEND_XAUDIO2);
if (AOSound::isValid())
backends.push_back(BACKEND_AOSOUND);
if (AlsaSound::isValid())
backends.push_back(BACKEND_ALSA);
if (CoreAudioSound::isValid())
backends.push_back(BACKEND_COREAUDIO);
if (PulseAudio::isValid())
backends.push_back(BACKEND_PULSEAUDIO);
if (OpenALStream::isValid())
backends.push_back(BACKEND_OPENAL);
if (OpenSLESStream::isValid())
backends.push_back(BACKEND_OPENSLES);
return backends;
}
void UpdateSoundStream()
{
if (g_sound_stream)
{
int volume = SConfig::GetInstance().m_IsMuted ? 0 : SConfig::GetInstance().m_Volume;
g_sound_stream->SetVolume(volume);
}
}
void ClearAudioBuffer(bool mute)
{
if (g_sound_stream)
g_sound_stream->Clear(mute);
}
void SendAIBuffer(short* samples, unsigned int num_samples)
{
if (!g_sound_stream)
return;
if (SConfig::GetInstance().m_DumpAudio && !s_audio_dump_start)
StartAudioDump();
else if (!SConfig::GetInstance().m_DumpAudio && s_audio_dump_start)
StopAudioDump();
CMixer* pMixer = g_sound_stream->GetMixer();
if (pMixer && samples)
{
pMixer->PushSamples(samples, num_samples);
}
g_sound_stream->Update();
}
void StartAudioDump()
{
std::string audio_file_name_dtk = File::GetUserPath(D_DUMPAUDIO_IDX) + "dtkdump.wav";
std::string audio_file_name_dsp = File::GetUserPath(D_DUMPAUDIO_IDX) + "dspdump.wav";
File::CreateFullPath(audio_file_name_dtk);
File::CreateFullPath(audio_file_name_dsp);
g_sound_stream->GetMixer()->StartLogDTKAudio(audio_file_name_dtk);
g_sound_stream->GetMixer()->StartLogDSPAudio(audio_file_name_dsp);
s_audio_dump_start = true;
}
void StopAudioDump()
{
g_sound_stream->GetMixer()->StopLogDTKAudio();
g_sound_stream->GetMixer()->StopLogDSPAudio();
s_audio_dump_start = false;
}
void IncreaseVolume(unsigned short offset)
{
SConfig::GetInstance().m_IsMuted = false;
int& currentVolume = SConfig::GetInstance().m_Volume;
currentVolume += offset;
if (currentVolume > AUDIO_VOLUME_MAX)
currentVolume = AUDIO_VOLUME_MAX;
UpdateSoundStream();
}
void DecreaseVolume(unsigned short offset)
{
SConfig::GetInstance().m_IsMuted = false;
int& currentVolume = SConfig::GetInstance().m_Volume;
currentVolume -= offset;
if (currentVolume < AUDIO_VOLUME_MIN)
currentVolume = AUDIO_VOLUME_MIN;
UpdateSoundStream();
}
void ToggleMuteVolume()
{
bool& isMuted = SConfig::GetInstance().m_IsMuted;
isMuted = !isMuted;
UpdateSoundStream();
}
}
+12 -13
View File
@@ -7,22 +7,21 @@
#include "AudioCommon/SoundStream.h"
#include "Common/CommonTypes.h"
class CMixer;
extern SoundStream *g_sound_stream;
extern SoundStream* g_sound_stream;
namespace AudioCommon
{
SoundStream* InitSoundStream();
void ShutdownSoundStream();
std::vector<std::string> GetSoundBackends();
void UpdateSoundStream();
void ClearAudioBuffer(bool mute);
void SendAIBuffer(short* samples, unsigned int num_samples);
void StartAudioDump();
void StopAudioDump();
void IncreaseVolume(unsigned short offset);
void DecreaseVolume(unsigned short offset);
void ToggleMuteVolume();
SoundStream* InitSoundStream();
void ShutdownSoundStream();
std::vector<std::string> GetSoundBackends();
void UpdateSoundStream();
void ClearAudioBuffer(bool mute);
void SendAIBuffer(short* samples, unsigned int num_samples);
void StartAudioDump();
void StopAudioDump();
void IncreaseVolume(unsigned short offset);
void DecreaseVolume(unsigned short offset);
void ToggleMuteVolume();
}
@@ -7,105 +7,94 @@
#include "AudioCommon/CoreAudioSoundStream.h"
#include "Common/Logging/Log.h"
OSStatus CoreAudioSound::callback(void *inRefCon,
AudioUnitRenderActionFlags *ioActionFlags,
const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber,
UInt32 inNumberFrames, AudioBufferList *ioData)
OSStatus CoreAudioSound::callback(void* inRefCon, AudioUnitRenderActionFlags* ioActionFlags,
const AudioTimeStamp* inTimeStamp, UInt32 inBusNumber,
UInt32 inNumberFrames, AudioBufferList* ioData)
{
for (UInt32 i = 0; i < ioData->mNumberBuffers; i++)
((CoreAudioSound *)inRefCon)->m_mixer->
Mix((short *)ioData->mBuffers[i].mData,
ioData->mBuffers[i].mDataByteSize / 4);
for (UInt32 i = 0; i < ioData->mNumberBuffers; i++)
((CoreAudioSound*)inRefCon)
->m_mixer->Mix((short*)ioData->mBuffers[i].mData, ioData->mBuffers[i].mDataByteSize / 4);
return noErr;
return noErr;
}
bool CoreAudioSound::Start()
{
OSStatus err;
AURenderCallbackStruct callback_struct;
AudioStreamBasicDescription format;
AudioComponentDescription desc;
AudioComponent component;
OSStatus err;
AURenderCallbackStruct callback_struct;
AudioStreamBasicDescription format;
AudioComponentDescription desc;
AudioComponent component;
desc.componentType = kAudioUnitType_Output;
desc.componentSubType = kAudioUnitSubType_DefaultOutput;
desc.componentFlags = 0;
desc.componentFlagsMask = 0;
desc.componentManufacturer = kAudioUnitManufacturer_Apple;
component = AudioComponentFindNext(nullptr, &desc);
if (component == nullptr)
{
ERROR_LOG(AUDIO, "error finding audio component");
return false;
}
desc.componentType = kAudioUnitType_Output;
desc.componentSubType = kAudioUnitSubType_DefaultOutput;
desc.componentFlags = 0;
desc.componentFlagsMask = 0;
desc.componentManufacturer = kAudioUnitManufacturer_Apple;
component = AudioComponentFindNext(nullptr, &desc);
if (component == nullptr)
{
ERROR_LOG(AUDIO, "error finding audio component");
return false;
}
err = AudioComponentInstanceNew(component, &audioUnit);
if (err != noErr)
{
ERROR_LOG(AUDIO, "error opening audio component");
return false;
}
err = AudioComponentInstanceNew(component, &audioUnit);
if (err != noErr)
{
ERROR_LOG(AUDIO, "error opening audio component");
return false;
}
FillOutASBDForLPCM(format, m_mixer->GetSampleRate(),
2, 16, 16, false, false, false);
err = AudioUnitSetProperty(audioUnit,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Input, 0, &format,
sizeof(AudioStreamBasicDescription));
if (err != noErr)
{
ERROR_LOG(AUDIO, "error setting audio format");
return false;
}
FillOutASBDForLPCM(format, m_mixer->GetSampleRate(), 2, 16, 16, false, false, false);
err = AudioUnitSetProperty(audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0,
&format, sizeof(AudioStreamBasicDescription));
if (err != noErr)
{
ERROR_LOG(AUDIO, "error setting audio format");
return false;
}
callback_struct.inputProc = callback;
callback_struct.inputProcRefCon = this;
err = AudioUnitSetProperty(audioUnit,
kAudioUnitProperty_SetRenderCallback,
kAudioUnitScope_Input, 0, &callback_struct,
sizeof callback_struct);
if (err != noErr)
{
ERROR_LOG(AUDIO, "error setting audio callback");
return false;
}
callback_struct.inputProc = callback;
callback_struct.inputProcRefCon = this;
err = AudioUnitSetProperty(audioUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input,
0, &callback_struct, sizeof callback_struct);
if (err != noErr)
{
ERROR_LOG(AUDIO, "error setting audio callback");
return false;
}
err = AudioUnitSetParameter(audioUnit,
kHALOutputParam_Volume,
kAudioUnitScope_Output, 0,
m_volume / 100., 0);
if (err != noErr)
ERROR_LOG(AUDIO, "error setting volume");
err = AudioUnitSetParameter(audioUnit, kHALOutputParam_Volume, kAudioUnitScope_Output, 0,
m_volume / 100., 0);
if (err != noErr)
ERROR_LOG(AUDIO, "error setting volume");
err = AudioUnitInitialize(audioUnit);
if (err != noErr)
{
ERROR_LOG(AUDIO, "error initializing audiounit");
return false;
}
err = AudioUnitInitialize(audioUnit);
if (err != noErr)
{
ERROR_LOG(AUDIO, "error initializing audiounit");
return false;
}
err = AudioOutputUnitStart(audioUnit);
if (err != noErr)
{
ERROR_LOG(AUDIO, "error starting audiounit");
return false;
}
err = AudioOutputUnitStart(audioUnit);
if (err != noErr)
{
ERROR_LOG(AUDIO, "error starting audiounit");
return false;
}
return true;
return true;
}
void CoreAudioSound::SetVolume(int volume)
{
OSStatus err;
m_volume = volume;
OSStatus err;
m_volume = volume;
err = AudioUnitSetParameter(audioUnit,
kHALOutputParam_Volume,
kAudioUnitScope_Output, 0,
volume / 100., 0);
if (err != noErr)
ERROR_LOG(AUDIO, "error setting volume");
err = AudioUnitSetParameter(audioUnit, kHALOutputParam_Volume, kAudioUnitScope_Output, 0,
volume / 100., 0);
if (err != noErr)
ERROR_LOG(AUDIO, "error setting volume");
}
void CoreAudioSound::SoundLoop()
@@ -114,19 +103,19 @@ void CoreAudioSound::SoundLoop()
void CoreAudioSound::Stop()
{
OSStatus err;
OSStatus err;
err = AudioOutputUnitStop(audioUnit);
if (err != noErr)
ERROR_LOG(AUDIO, "error stopping audiounit");
err = AudioOutputUnitStop(audioUnit);
if (err != noErr)
ERROR_LOG(AUDIO, "error stopping audiounit");
err = AudioUnitUninitialize(audioUnit);
if (err != noErr)
ERROR_LOG(AUDIO, "error uninitializing audiounit");
err = AudioUnitUninitialize(audioUnit);
if (err != noErr)
ERROR_LOG(AUDIO, "error uninitializing audiounit");
err = AudioComponentInstanceDispose(audioUnit);
if (err != noErr)
ERROR_LOG(AUDIO, "error closing audio component");
err = AudioComponentInstanceDispose(audioUnit);
if (err != noErr)
ERROR_LOG(AUDIO, "error closing audio component");
}
void CoreAudioSound::Update()
+11 -17
View File
@@ -14,25 +14,19 @@ class CoreAudioSound final : public SoundStream
{
#ifdef __APPLE__
public:
bool Start() override;
void SetVolume(int volume) override;
void SoundLoop() override;
void Stop() override;
void Update() override;
static bool isValid()
{
return true;
}
bool Start() override;
void SetVolume(int volume) override;
void SoundLoop() override;
void Stop() override;
void Update() override;
static bool isValid() { return true; }
private:
AudioUnit audioUnit;
int m_volume;
AudioUnit audioUnit;
int m_volume;
static OSStatus callback(void *inRefCon,
AudioUnitRenderActionFlags *ioActionFlags,
const AudioTimeStamp *inTimeStamp,
UInt32 inBusNumber, UInt32 inNumberFrames,
AudioBufferList *ioData);
static OSStatus callback(void* inRefCon, AudioUnitRenderActionFlags* ioActionFlags,
const AudioTimeStamp* inTimeStamp, UInt32 inBusNumber,
UInt32 inNumberFrames, AudioBufferList* ioData);
#endif
};
File diff suppressed because it is too large Load Diff
+1 -1
View File
@@ -4,5 +4,5 @@
#pragma once
void DPL2Decode(float *samples, int numsamples, float *out);
void DPL2Decode(float* samples, int numsamples, float* out);
void DPL2Reset();
+172 -169
View File
@@ -8,18 +8,17 @@
#include "AudioCommon/Mixer.h"
#include "Common/CommonFuncs.h"
#include "Common/CommonTypes.h"
#include "Common/MathUtil.h"
#include "Common/Logging/Log.h"
#include "Common/MathUtil.h"
#include "Core/ConfigManager.h"
#if _M_SSE >= 0x301 && !(defined __GNUC__ && !defined __SSSE3__)
#include <tmmintrin.h>
#endif
CMixer::CMixer(unsigned int BackendSampleRate)
: m_sampleRate(BackendSampleRate)
CMixer::CMixer(unsigned int BackendSampleRate) : m_sampleRate(BackendSampleRate)
{
INFO_LOG(AUDIO_INTERFACE, "Mixer is initialized");
INFO_LOG(AUDIO_INTERFACE, "Mixer is initialized");
}
CMixer::~CMixer()
@@ -27,249 +26,253 @@ CMixer::~CMixer()
}
// Executed from sound stream thread
unsigned int CMixer::MixerFifo::Mix(short* samples, unsigned int numSamples, bool consider_framelimit)
unsigned int CMixer::MixerFifo::Mix(short* samples, unsigned int numSamples,
bool consider_framelimit)
{
unsigned int currentSample = 0;
unsigned int currentSample = 0;
// Cache access in non-volatile variable
// This is the only function changing the read value, so it's safe to
// cache it locally although it's written here.
// The writing pointer will be modified outside, but it will only increase,
// so we will just ignore new written data while interpolating.
// Without this cache, the compiler wouldn't be allowed to optimize the
// interpolation loop.
u32 indexR = m_indexR.load();
u32 indexW = m_indexW.load();
// Cache access in non-volatile variable
// This is the only function changing the read value, so it's safe to
// cache it locally although it's written here.
// The writing pointer will be modified outside, but it will only increase,
// so we will just ignore new written data while interpolating.
// Without this cache, the compiler wouldn't be allowed to optimize the
// interpolation loop.
u32 indexR = m_indexR.load();
u32 indexW = m_indexW.load();
u32 low_waterwark = m_input_sample_rate * SConfig::GetInstance().iTimingVariance / 1000;
low_waterwark = std::min(low_waterwark, MAX_SAMPLES / 2);
u32 low_waterwark = m_input_sample_rate * SConfig::GetInstance().iTimingVariance / 1000;
low_waterwark = std::min(low_waterwark, MAX_SAMPLES / 2);
float numLeft = (float)(((indexW - indexR) & INDEX_MASK) / 2);
m_numLeftI = (numLeft + m_numLeftI*(CONTROL_AVG-1)) / CONTROL_AVG;
float offset = (m_numLeftI - low_waterwark) * CONTROL_FACTOR;
if (offset > MAX_FREQ_SHIFT) offset = MAX_FREQ_SHIFT;
if (offset < -MAX_FREQ_SHIFT) offset = -MAX_FREQ_SHIFT;
float numLeft = (float)(((indexW - indexR) & INDEX_MASK) / 2);
m_numLeftI = (numLeft + m_numLeftI * (CONTROL_AVG - 1)) / CONTROL_AVG;
float offset = (m_numLeftI - low_waterwark) * CONTROL_FACTOR;
if (offset > MAX_FREQ_SHIFT)
offset = MAX_FREQ_SHIFT;
if (offset < -MAX_FREQ_SHIFT)
offset = -MAX_FREQ_SHIFT;
//render numleft sample pairs to samples[]
//advance indexR with sample position
//remember fractional offset
// render numleft sample pairs to samples[]
// advance indexR with sample position
// remember fractional offset
float emulationspeed = SConfig::GetInstance().m_EmulationSpeed;
float aid_sample_rate = m_input_sample_rate + offset;
if (consider_framelimit && emulationspeed > 0.0f)
{
aid_sample_rate = aid_sample_rate * emulationspeed;
}
float emulationspeed = SConfig::GetInstance().m_EmulationSpeed;
float aid_sample_rate = m_input_sample_rate + offset;
if (consider_framelimit && emulationspeed > 0.0f)
{
aid_sample_rate = aid_sample_rate * emulationspeed;
}
const u32 ratio = (u32)(65536.0f * aid_sample_rate / (float)m_mixer->m_sampleRate);
const u32 ratio = (u32)(65536.0f * aid_sample_rate / (float)m_mixer->m_sampleRate);
s32 lvolume = m_LVolume.load();
s32 rvolume = m_RVolume.load();
s32 lvolume = m_LVolume.load();
s32 rvolume = m_RVolume.load();
// TODO: consider a higher-quality resampling algorithm.
for (; currentSample < numSamples * 2 && ((indexW-indexR) & INDEX_MASK) > 2; currentSample += 2)
{
u32 indexR2 = indexR + 2; //next sample
// TODO: consider a higher-quality resampling algorithm.
for (; currentSample < numSamples * 2 && ((indexW - indexR) & INDEX_MASK) > 2; currentSample += 2)
{
u32 indexR2 = indexR + 2; // next sample
s16 l1 = Common::swap16(m_buffer[indexR & INDEX_MASK]); //current
s16 l2 = Common::swap16(m_buffer[indexR2 & INDEX_MASK]); //next
int sampleL = ((l1 << 16) + (l2 - l1) * (u16)m_frac) >> 16;
sampleL = (sampleL * lvolume) >> 8;
sampleL += samples[currentSample + 1];
samples[currentSample + 1] = MathUtil::Clamp(sampleL, -32767, 32767);
s16 l1 = Common::swap16(m_buffer[indexR & INDEX_MASK]); // current
s16 l2 = Common::swap16(m_buffer[indexR2 & INDEX_MASK]); // next
int sampleL = ((l1 << 16) + (l2 - l1) * (u16)m_frac) >> 16;
sampleL = (sampleL * lvolume) >> 8;
sampleL += samples[currentSample + 1];
samples[currentSample + 1] = MathUtil::Clamp(sampleL, -32767, 32767);
s16 r1 = Common::swap16(m_buffer[(indexR + 1) & INDEX_MASK]); //current
s16 r2 = Common::swap16(m_buffer[(indexR2 + 1) & INDEX_MASK]); //next
int sampleR = ((r1 << 16) + (r2 - r1) * (u16)m_frac) >> 16;
sampleR = (sampleR * rvolume) >> 8;
sampleR += samples[currentSample];
samples[currentSample] = MathUtil::Clamp(sampleR, -32767, 32767);
s16 r1 = Common::swap16(m_buffer[(indexR + 1) & INDEX_MASK]); // current
s16 r2 = Common::swap16(m_buffer[(indexR2 + 1) & INDEX_MASK]); // next
int sampleR = ((r1 << 16) + (r2 - r1) * (u16)m_frac) >> 16;
sampleR = (sampleR * rvolume) >> 8;
sampleR += samples[currentSample];
samples[currentSample] = MathUtil::Clamp(sampleR, -32767, 32767);
m_frac += ratio;
indexR += 2 * (u16)(m_frac >> 16);
m_frac &= 0xffff;
}
m_frac += ratio;
indexR += 2 * (u16)(m_frac >> 16);
m_frac &= 0xffff;
}
// Padding
short s[2];
s[0] = Common::swap16(m_buffer[(indexR - 1) & INDEX_MASK]);
s[1] = Common::swap16(m_buffer[(indexR - 2) & INDEX_MASK]);
s[0] = (s[0] * rvolume) >> 8;
s[1] = (s[1] * lvolume) >> 8;
for (; currentSample < numSamples * 2; currentSample += 2)
{
int sampleR = MathUtil::Clamp(s[0] + samples[currentSample + 0], -32767, 32767);
int sampleL = MathUtil::Clamp(s[1] + samples[currentSample + 1], -32767, 32767);
// Padding
short s[2];
s[0] = Common::swap16(m_buffer[(indexR - 1) & INDEX_MASK]);
s[1] = Common::swap16(m_buffer[(indexR - 2) & INDEX_MASK]);
s[0] = (s[0] * rvolume) >> 8;
s[1] = (s[1] * lvolume) >> 8;
for (; currentSample < numSamples * 2; currentSample += 2)
{
int sampleR = MathUtil::Clamp(s[0] + samples[currentSample + 0], -32767, 32767);
int sampleL = MathUtil::Clamp(s[1] + samples[currentSample + 1], -32767, 32767);
samples[currentSample + 0] = sampleR;
samples[currentSample + 1] = sampleL;
}
samples[currentSample + 0] = sampleR;
samples[currentSample + 1] = sampleL;
}
// Flush cached variable
m_indexR.store(indexR);
// Flush cached variable
m_indexR.store(indexR);
return numSamples;
return numSamples;
}
unsigned int CMixer::Mix(short* samples, unsigned int num_samples, bool consider_framelimit)
{
if (!samples)
return 0;
if (!samples)
return 0;
memset(samples, 0, num_samples * 2 * sizeof(short));
memset(samples, 0, num_samples * 2 * sizeof(short));
m_dma_mixer.Mix(samples, num_samples, consider_framelimit);
m_streaming_mixer.Mix(samples, num_samples, consider_framelimit);
m_wiimote_speaker_mixer.Mix(samples, num_samples, consider_framelimit);
return num_samples;
m_dma_mixer.Mix(samples, num_samples, consider_framelimit);
m_streaming_mixer.Mix(samples, num_samples, consider_framelimit);
m_wiimote_speaker_mixer.Mix(samples, num_samples, consider_framelimit);
return num_samples;
}
void CMixer::MixerFifo::PushSamples(const short *samples, unsigned int num_samples)
void CMixer::MixerFifo::PushSamples(const short* samples, unsigned int num_samples)
{
// Cache access in non-volatile variable
// indexR isn't allowed to cache in the audio throttling loop as it
// needs to get updates to not deadlock.
u32 indexW = m_indexW.load();
// Cache access in non-volatile variable
// indexR isn't allowed to cache in the audio throttling loop as it
// needs to get updates to not deadlock.
u32 indexW = m_indexW.load();
// Check if we have enough free space
// indexW == m_indexR results in empty buffer, so indexR must always be smaller than indexW
if (num_samples * 2 + ((indexW - m_indexR.load()) & INDEX_MASK) >= MAX_SAMPLES * 2)
return;
// Check if we have enough free space
// indexW == m_indexR results in empty buffer, so indexR must always be smaller than indexW
if (num_samples * 2 + ((indexW - m_indexR.load()) & INDEX_MASK) >= MAX_SAMPLES * 2)
return;
// AyuanX: Actual re-sampling work has been moved to sound thread
// to alleviate the workload on main thread
// and we simply store raw data here to make fast mem copy
int over_bytes = num_samples * 4 - (MAX_SAMPLES * 2 - (indexW & INDEX_MASK)) * sizeof(short);
if (over_bytes > 0)
{
memcpy(&m_buffer[indexW & INDEX_MASK], samples, num_samples * 4 - over_bytes);
memcpy(&m_buffer[0], samples + (num_samples * 4 - over_bytes) / sizeof(short), over_bytes);
}
else
{
memcpy(&m_buffer[indexW & INDEX_MASK], samples, num_samples * 4);
}
// AyuanX: Actual re-sampling work has been moved to sound thread
// to alleviate the workload on main thread
// and we simply store raw data here to make fast mem copy
int over_bytes = num_samples * 4 - (MAX_SAMPLES * 2 - (indexW & INDEX_MASK)) * sizeof(short);
if (over_bytes > 0)
{
memcpy(&m_buffer[indexW & INDEX_MASK], samples, num_samples * 4 - over_bytes);
memcpy(&m_buffer[0], samples + (num_samples * 4 - over_bytes) / sizeof(short), over_bytes);
}
else
{
memcpy(&m_buffer[indexW & INDEX_MASK], samples, num_samples * 4);
}
m_indexW.fetch_add(num_samples * 2);
m_indexW.fetch_add(num_samples * 2);
}
void CMixer::PushSamples(const short *samples, unsigned int num_samples)
void CMixer::PushSamples(const short* samples, unsigned int num_samples)
{
m_dma_mixer.PushSamples(samples, num_samples);
if (m_log_dsp_audio)
m_wave_writer_dsp.AddStereoSamplesBE(samples, num_samples);
m_dma_mixer.PushSamples(samples, num_samples);
if (m_log_dsp_audio)
m_wave_writer_dsp.AddStereoSamplesBE(samples, num_samples);
}
void CMixer::PushStreamingSamples(const short *samples, unsigned int num_samples)
void CMixer::PushStreamingSamples(const short* samples, unsigned int num_samples)
{
m_streaming_mixer.PushSamples(samples, num_samples);
if (m_log_dtk_audio)
m_wave_writer_dtk.AddStereoSamplesBE(samples, num_samples);
m_streaming_mixer.PushSamples(samples, num_samples);
if (m_log_dtk_audio)
m_wave_writer_dtk.AddStereoSamplesBE(samples, num_samples);
}
void CMixer::PushWiimoteSpeakerSamples(const short *samples, unsigned int num_samples, unsigned int sample_rate)
void CMixer::PushWiimoteSpeakerSamples(const short* samples, unsigned int num_samples,
unsigned int sample_rate)
{
short samples_stereo[MAX_SAMPLES * 2];
short samples_stereo[MAX_SAMPLES * 2];
if (num_samples < MAX_SAMPLES)
{
m_wiimote_speaker_mixer.SetInputSampleRate(sample_rate);
if (num_samples < MAX_SAMPLES)
{
m_wiimote_speaker_mixer.SetInputSampleRate(sample_rate);
for (unsigned int i = 0; i < num_samples; ++i)
{
samples_stereo[i * 2] = Common::swap16(samples[i]);
samples_stereo[i * 2 + 1] = Common::swap16(samples[i]);
}
for (unsigned int i = 0; i < num_samples; ++i)
{
samples_stereo[i * 2] = Common::swap16(samples[i]);
samples_stereo[i * 2 + 1] = Common::swap16(samples[i]);
}
m_wiimote_speaker_mixer.PushSamples(samples_stereo, num_samples);
}
m_wiimote_speaker_mixer.PushSamples(samples_stereo, num_samples);
}
}
void CMixer::SetDMAInputSampleRate(unsigned int rate)
{
m_dma_mixer.SetInputSampleRate(rate);
m_dma_mixer.SetInputSampleRate(rate);
}
void CMixer::SetStreamInputSampleRate(unsigned int rate)
{
m_streaming_mixer.SetInputSampleRate(rate);
m_streaming_mixer.SetInputSampleRate(rate);
}
void CMixer::SetStreamingVolume(unsigned int lvolume, unsigned int rvolume)
{
m_streaming_mixer.SetVolume(lvolume, rvolume);
m_streaming_mixer.SetVolume(lvolume, rvolume);
}
void CMixer::SetWiimoteSpeakerVolume(unsigned int lvolume, unsigned int rvolume)
{
m_wiimote_speaker_mixer.SetVolume(lvolume, rvolume);
m_wiimote_speaker_mixer.SetVolume(lvolume, rvolume);
}
void CMixer::StartLogDTKAudio(const std::string& filename)
{
if (!m_log_dtk_audio)
{
m_log_dtk_audio = true;
m_wave_writer_dtk.Start(filename, 48000);
m_wave_writer_dtk.SetSkipSilence(false);
NOTICE_LOG(AUDIO, "Starting DTK Audio logging");
}
else
{
WARN_LOG(AUDIO, "DTK Audio logging has already been started");
}
if (!m_log_dtk_audio)
{
m_log_dtk_audio = true;
m_wave_writer_dtk.Start(filename, 48000);
m_wave_writer_dtk.SetSkipSilence(false);
NOTICE_LOG(AUDIO, "Starting DTK Audio logging");
}
else
{
WARN_LOG(AUDIO, "DTK Audio logging has already been started");
}
}
void CMixer::StopLogDTKAudio()
{
if (m_log_dtk_audio)
{
m_log_dtk_audio = false;
m_wave_writer_dtk.Stop();
NOTICE_LOG(AUDIO, "Stopping DTK Audio logging");
}
else
{
WARN_LOG(AUDIO, "DTK Audio logging has already been stopped");
}
if (m_log_dtk_audio)
{
m_log_dtk_audio = false;
m_wave_writer_dtk.Stop();
NOTICE_LOG(AUDIO, "Stopping DTK Audio logging");
}
else
{
WARN_LOG(AUDIO, "DTK Audio logging has already been stopped");
}
}
void CMixer::StartLogDSPAudio(const std::string& filename)
{
if (!m_log_dsp_audio)
{
m_log_dsp_audio = true;
m_wave_writer_dsp.Start(filename, 32000);
m_wave_writer_dsp.SetSkipSilence(false);
NOTICE_LOG(AUDIO, "Starting DSP Audio logging");
}
else
{
WARN_LOG(AUDIO, "DSP Audio logging has already been started");
}
if (!m_log_dsp_audio)
{
m_log_dsp_audio = true;
m_wave_writer_dsp.Start(filename, 32000);
m_wave_writer_dsp.SetSkipSilence(false);
NOTICE_LOG(AUDIO, "Starting DSP Audio logging");
}
else
{
WARN_LOG(AUDIO, "DSP Audio logging has already been started");
}
}
void CMixer::StopLogDSPAudio()
{
if (m_log_dsp_audio)
{
m_log_dsp_audio = false;
m_wave_writer_dsp.Stop();
NOTICE_LOG(AUDIO, "Stopping DSP Audio logging");
}
else
{
WARN_LOG(AUDIO, "DSP Audio logging has already been stopped");
}
if (m_log_dsp_audio)
{
m_log_dsp_audio = false;
m_wave_writer_dsp.Stop();
NOTICE_LOG(AUDIO, "Stopping DSP Audio logging");
}
else
{
WARN_LOG(AUDIO, "DSP Audio logging has already been stopped");
}
}
void CMixer::MixerFifo::SetInputSampleRate(unsigned int rate)
{
m_input_sample_rate = rate;
m_input_sample_rate = rate;
}
void CMixer::MixerFifo::SetVolume(unsigned int lvolume, unsigned int rvolume)
{
m_LVolume.store(lvolume + (lvolume >> 7));
m_RVolume.store(rvolume + (rvolume >> 7));
m_LVolume.store(lvolume + (lvolume >> 7));
m_RVolume.store(rvolume + (rvolume >> 7));
}
+59 -60
View File
@@ -13,74 +13,73 @@
class CMixer final
{
public:
explicit CMixer(unsigned int BackendSampleRate);
~CMixer();
explicit CMixer(unsigned int BackendSampleRate);
~CMixer();
// Called from audio threads
unsigned int Mix(short* samples, unsigned int numSamples, bool consider_framelimit = true);
// Called from audio threads
unsigned int Mix(short* samples, unsigned int numSamples, bool consider_framelimit = true);
// Called from main thread
void PushSamples(const short* samples, unsigned int num_samples);
void PushStreamingSamples(const short* samples, unsigned int num_samples);
void PushWiimoteSpeakerSamples(const short* samples, unsigned int num_samples, unsigned int sample_rate);
unsigned int GetSampleRate() const { return m_sampleRate; }
// Called from main thread
void PushSamples(const short* samples, unsigned int num_samples);
void PushStreamingSamples(const short* samples, unsigned int num_samples);
void PushWiimoteSpeakerSamples(const short* samples, unsigned int num_samples,
unsigned int sample_rate);
unsigned int GetSampleRate() const { return m_sampleRate; }
void SetDMAInputSampleRate(unsigned int rate);
void SetStreamInputSampleRate(unsigned int rate);
void SetStreamingVolume(unsigned int lvolume, unsigned int rvolume);
void SetWiimoteSpeakerVolume(unsigned int lvolume, unsigned int rvolume);
void SetDMAInputSampleRate(unsigned int rate);
void SetStreamInputSampleRate(unsigned int rate);
void SetStreamingVolume(unsigned int lvolume, unsigned int rvolume);
void SetWiimoteSpeakerVolume(unsigned int lvolume, unsigned int rvolume);
void StartLogDTKAudio(const std::string& filename);
void StopLogDTKAudio();
void StartLogDTKAudio(const std::string& filename);
void StopLogDTKAudio();
void StartLogDSPAudio(const std::string& filename);
void StopLogDSPAudio();
float GetCurrentSpeed() const { return m_speed.load(); }
void UpdateSpeed(float val) { m_speed.store(val); }
void StartLogDSPAudio(const std::string& filename);
void StopLogDSPAudio();
float GetCurrentSpeed() const { return m_speed.load(); }
void UpdateSpeed(float val) { m_speed.store(val); }
private:
static constexpr u32 MAX_SAMPLES = 1024 * 4; // 128 ms
static constexpr u32 INDEX_MASK = MAX_SAMPLES * 2 - 1;
static constexpr int MAX_FREQ_SHIFT = 200; // Per 32000 Hz
static constexpr float CONTROL_FACTOR = 0.2f;
static constexpr u32 CONTROL_AVG = 32; // In freq_shift per FIFO size offset
static constexpr u32 MAX_SAMPLES = 1024 * 4; // 128 ms
static constexpr u32 INDEX_MASK = MAX_SAMPLES * 2 - 1;
static constexpr int MAX_FREQ_SHIFT = 200; // Per 32000 Hz
static constexpr float CONTROL_FACTOR = 0.2f;
static constexpr u32 CONTROL_AVG = 32; // In freq_shift per FIFO size offset
class MixerFifo final
{
public:
MixerFifo(CMixer* mixer, unsigned sample_rate)
: m_mixer(mixer)
, m_input_sample_rate(sample_rate)
{
}
void PushSamples(const short* samples, unsigned int num_samples);
unsigned int Mix(short* samples, unsigned int numSamples, bool consider_framelimit = true);
void SetInputSampleRate(unsigned int rate);
void SetVolume(unsigned int lvolume, unsigned int rvolume);
private:
CMixer* m_mixer;
unsigned m_input_sample_rate;
std::array<short, MAX_SAMPLES * 2> m_buffer{};
std::atomic<u32> m_indexW{0};
std::atomic<u32> m_indexR{0};
// Volume ranges from 0-256
std::atomic<s32> m_LVolume{256};
std::atomic<s32> m_RVolume{256};
float m_numLeftI = 0.0f;
u32 m_frac = 0;
};
MixerFifo m_dma_mixer{this, 32000};
MixerFifo m_streaming_mixer{this, 48000};
MixerFifo m_wiimote_speaker_mixer{this, 3000};
unsigned int m_sampleRate;
class MixerFifo final
{
public:
MixerFifo(CMixer* mixer, unsigned sample_rate)
: m_mixer(mixer), m_input_sample_rate(sample_rate)
{
}
void PushSamples(const short* samples, unsigned int num_samples);
unsigned int Mix(short* samples, unsigned int numSamples, bool consider_framelimit = true);
void SetInputSampleRate(unsigned int rate);
void SetVolume(unsigned int lvolume, unsigned int rvolume);
WaveFileWriter m_wave_writer_dtk;
WaveFileWriter m_wave_writer_dsp;
private:
CMixer* m_mixer;
unsigned m_input_sample_rate;
std::array<short, MAX_SAMPLES * 2> m_buffer{};
std::atomic<u32> m_indexW{0};
std::atomic<u32> m_indexR{0};
// Volume ranges from 0-256
std::atomic<s32> m_LVolume{256};
std::atomic<s32> m_RVolume{256};
float m_numLeftI = 0.0f;
u32 m_frac = 0;
};
MixerFifo m_dma_mixer{this, 32000};
MixerFifo m_streaming_mixer{this, 48000};
MixerFifo m_wiimote_speaker_mixer{this, 3000};
unsigned int m_sampleRate;
bool m_log_dtk_audio = false;
bool m_log_dsp_audio = false;
WaveFileWriter m_wave_writer_dtk;
WaveFileWriter m_wave_writer_dsp;
// Current rate of emulation (1.0 = 100% speed)
std::atomic<float> m_speed{0.0f};
bool m_log_dtk_audio = false;
bool m_log_dsp_audio = false;
// Current rate of emulation (1.0 = 100% speed)
std::atomic<float> m_speed{0.0f};
};
+12 -9
View File
@@ -13,7 +13,7 @@ void NullSound::SoundLoop()
bool NullSound::Start()
{
return true;
return true;
}
void NullSound::SetVolume(int volume)
@@ -22,19 +22,22 @@ void NullSound::SetVolume(int volume)
void NullSound::Update()
{
// num_samples_to_render in this update - depends on SystemTimers::AUDIO_DMA_PERIOD.
constexpr u32 stereo_16_bit_size = 4;
constexpr u32 dma_length = 32;
const u64 audio_dma_period = SystemTimers::GetTicksPerSecond() / (AudioInterface::GetAIDSampleRate() * stereo_16_bit_size / dma_length);
const u64 ais_samples_per_second = 48000 * stereo_16_bit_size;
const u64 num_samples_to_render = (audio_dma_period * ais_samples_per_second) / SystemTimers::GetTicksPerSecond();
// num_samples_to_render in this update - depends on SystemTimers::AUDIO_DMA_PERIOD.
constexpr u32 stereo_16_bit_size = 4;
constexpr u32 dma_length = 32;
const u64 audio_dma_period =
SystemTimers::GetTicksPerSecond() /
(AudioInterface::GetAIDSampleRate() * stereo_16_bit_size / dma_length);
const u64 ais_samples_per_second = 48000 * stereo_16_bit_size;
const u64 num_samples_to_render =
(audio_dma_period * ais_samples_per_second) / SystemTimers::GetTicksPerSecond();
m_mixer->Mix(m_realtime_buffer.data(), (unsigned int)num_samples_to_render);
m_mixer->Mix(m_realtime_buffer.data(), (unsigned int)num_samples_to_render);
}
void NullSound::Clear(bool mute)
{
m_muted = mute;
m_muted = mute;
}
void NullSound::Stop()
+10 -11
View File
@@ -10,18 +10,17 @@
class NullSound final : public SoundStream
{
public:
bool Start() override;
void SoundLoop() override;
void SetVolume(int volume) override;
void Stop() override;
void Clear(bool mute) override;
void Update() override;
static bool isValid() { return true; }
bool Start() override;
void SoundLoop() override;
void SetVolume(int volume) override;
void Stop() override;
void Clear(bool mute) override;
void Update() override;
static bool isValid() { return true; }
private:
static constexpr size_t BUFFER_SIZE = 48000 * 4 / 32;
static constexpr size_t BUFFER_SIZE = 48000 * 4 / 32;
// Playback position
std::array<short, BUFFER_SIZE / sizeof(short)> m_realtime_buffer;
// Playback position
std::array<short, BUFFER_SIZE / sizeof(short)> m_realtime_buffer;
};
File diff suppressed because it is too large Load Diff
+30 -34
View File
@@ -32,56 +32,52 @@
#define BOOL SoundTouch_BOOL
#endif
#include <soundtouch/SoundTouch.h>
#include <soundtouch/STTypes.h>
#include <soundtouch/SoundTouch.h>
#ifdef __APPLE__
#undef BOOL
#endif
// 16 bit Stereo
#define SFX_MAX_SOURCE 1
#define OAL_MAX_BUFFERS 32
#define OAL_MAX_SAMPLES 256
#define STEREO_CHANNELS 2
#define SURROUND_CHANNELS 6 // number of channels in surround mode
#define SIZE_SHORT 2
#define SIZE_FLOAT 4 // size of a float in bytes
#define FRAME_STEREO_SHORT STEREO_CHANNELS * SIZE_SHORT
#define FRAME_STEREO_FLOAT STEREO_CHANNELS * SIZE_FLOAT
#define FRAME_SURROUND_FLOAT SURROUND_CHANNELS * SIZE_FLOAT
#define FRAME_SURROUND_SHORT SURROUND_CHANNELS * SIZE_SHORT
#define SFX_MAX_SOURCE 1
#define OAL_MAX_BUFFERS 32
#define OAL_MAX_SAMPLES 256
#define STEREO_CHANNELS 2
#define SURROUND_CHANNELS 6 // number of channels in surround mode
#define SIZE_SHORT 2
#define SIZE_FLOAT 4 // size of a float in bytes
#define FRAME_STEREO_SHORT STEREO_CHANNELS* SIZE_SHORT
#define FRAME_STEREO_FLOAT STEREO_CHANNELS* SIZE_FLOAT
#define FRAME_SURROUND_FLOAT SURROUND_CHANNELS* SIZE_FLOAT
#define FRAME_SURROUND_SHORT SURROUND_CHANNELS* SIZE_SHORT
#endif
class OpenALStream final : public SoundStream
{
#if defined HAVE_OPENAL && HAVE_OPENAL
public:
OpenALStream() : uiSource(0)
{
}
bool Start() override;
void SoundLoop() override;
void SetVolume(int volume) override;
void Stop() override;
void Clear(bool mute) override;
void Update() override;
static bool isValid() { return true; }
OpenALStream() : uiSource(0) {}
bool Start() override;
void SoundLoop() override;
void SetVolume(int volume) override;
void Stop() override;
void Clear(bool mute) override;
void Update() override;
static bool isValid() { return true; }
private:
std::thread thread;
std::atomic<bool> m_run_thread;
std::thread thread;
std::atomic<bool> m_run_thread;
Common::Event soundSyncEvent;
Common::Event soundSyncEvent;
short realtimeBuffer[OAL_MAX_SAMPLES * STEREO_CHANNELS];
soundtouch::SAMPLETYPE sampleBuffer[OAL_MAX_SAMPLES * SURROUND_CHANNELS * OAL_MAX_BUFFERS];
ALuint uiBuffers[OAL_MAX_BUFFERS];
ALuint uiSource;
ALfloat fVolume;
short realtimeBuffer[OAL_MAX_SAMPLES * STEREO_CHANNELS];
soundtouch::SAMPLETYPE sampleBuffer[OAL_MAX_SAMPLES * SURROUND_CHANNELS * OAL_MAX_BUFFERS];
ALuint uiBuffers[OAL_MAX_BUFFERS];
ALuint uiSource;
ALfloat fVolume;
u8 numBuffers;
#endif // HAVE_OPENAL
u8 numBuffers;
#endif // HAVE_OPENAL
};
+82 -81
View File
@@ -24,7 +24,7 @@ static SLPlayItf bqPlayerPlay;
static SLAndroidSimpleBufferQueueItf bqPlayerBufferQueue;
static SLMuteSoloItf bqPlayerMuteSolo;
static SLVolumeItf bqPlayerVolume;
static CMixer *g_mixer;
static CMixer* g_mixer;
#define BUFFER_SIZE 512
#define BUFFER_SIZE_IN_SAMPLES (BUFFER_SIZE / 2)
@@ -32,106 +32,107 @@ static CMixer *g_mixer;
static short buffer[2][BUFFER_SIZE];
static int curBuffer = 0;
static void bqPlayerCallback(SLAndroidSimpleBufferQueueItf bq, void *context)
static void bqPlayerCallback(SLAndroidSimpleBufferQueueItf bq, void* context)
{
assert(bq == bqPlayerBufferQueue);
assert(nullptr == context);
assert(bq == bqPlayerBufferQueue);
assert(nullptr == context);
// Render to the fresh buffer
g_mixer->Mix(reinterpret_cast<short *>(buffer[curBuffer]), BUFFER_SIZE_IN_SAMPLES);
SLresult result = (*bqPlayerBufferQueue)->Enqueue(bqPlayerBufferQueue, buffer[curBuffer], sizeof(buffer[0]));
curBuffer ^= 1; // Switch buffer
// Render to the fresh buffer
g_mixer->Mix(reinterpret_cast<short*>(buffer[curBuffer]), BUFFER_SIZE_IN_SAMPLES);
SLresult result =
(*bqPlayerBufferQueue)->Enqueue(bqPlayerBufferQueue, buffer[curBuffer], sizeof(buffer[0]));
curBuffer ^= 1; // Switch buffer
// Comment from sample code:
// the most likely other result is SL_RESULT_BUFFER_INSUFFICIENT,
// which for this code example would indicate a programming error
_assert_msg_(AUDIO, SL_RESULT_SUCCESS == result, "Couldn't enqueue audio stream.");
// Comment from sample code:
// the most likely other result is SL_RESULT_BUFFER_INSUFFICIENT,
// which for this code example would indicate a programming error
_assert_msg_(AUDIO, SL_RESULT_SUCCESS == result, "Couldn't enqueue audio stream.");
}
bool OpenSLESStream::Start()
{
SLresult result;
// create engine
result = slCreateEngine(&engineObject, 0, nullptr, 0, nullptr, nullptr);
assert(SL_RESULT_SUCCESS == result);
result = (*engineObject)->Realize(engineObject, SL_BOOLEAN_FALSE);
assert(SL_RESULT_SUCCESS == result);
result = (*engineObject)->GetInterface(engineObject, SL_IID_ENGINE, &engineEngine);
assert(SL_RESULT_SUCCESS == result);
result = (*engineEngine)->CreateOutputMix(engineEngine, &outputMixObject, 0, 0, 0);
assert(SL_RESULT_SUCCESS == result);
result = (*outputMixObject)->Realize(outputMixObject, SL_BOOLEAN_FALSE);
assert(SL_RESULT_SUCCESS == result);
SLresult result;
// create engine
result = slCreateEngine(&engineObject, 0, nullptr, 0, nullptr, nullptr);
assert(SL_RESULT_SUCCESS == result);
result = (*engineObject)->Realize(engineObject, SL_BOOLEAN_FALSE);
assert(SL_RESULT_SUCCESS == result);
result = (*engineObject)->GetInterface(engineObject, SL_IID_ENGINE, &engineEngine);
assert(SL_RESULT_SUCCESS == result);
result = (*engineEngine)->CreateOutputMix(engineEngine, &outputMixObject, 0, 0, 0);
assert(SL_RESULT_SUCCESS == result);
result = (*outputMixObject)->Realize(outputMixObject, SL_BOOLEAN_FALSE);
assert(SL_RESULT_SUCCESS == result);
SLDataLocator_AndroidSimpleBufferQueue loc_bufq = {SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE, 2};
SLDataFormat_PCM format_pcm = {
SL_DATAFORMAT_PCM,
2,
m_mixer->GetSampleRate() * 1000,
SL_PCMSAMPLEFORMAT_FIXED_16,
SL_PCMSAMPLEFORMAT_FIXED_16,
SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT,
SL_BYTEORDER_LITTLEENDIAN
};
SLDataLocator_AndroidSimpleBufferQueue loc_bufq = {SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE, 2};
SLDataFormat_PCM format_pcm = {SL_DATAFORMAT_PCM,
2,
m_mixer->GetSampleRate() * 1000,
SL_PCMSAMPLEFORMAT_FIXED_16,
SL_PCMSAMPLEFORMAT_FIXED_16,
SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT,
SL_BYTEORDER_LITTLEENDIAN};
SLDataSource audioSrc = {&loc_bufq, &format_pcm};
SLDataSource audioSrc = {&loc_bufq, &format_pcm};
// configure audio sink
SLDataLocator_OutputMix loc_outmix = {SL_DATALOCATOR_OUTPUTMIX, outputMixObject};
SLDataSink audioSnk = {&loc_outmix, nullptr};
// configure audio sink
SLDataLocator_OutputMix loc_outmix = {SL_DATALOCATOR_OUTPUTMIX, outputMixObject};
SLDataSink audioSnk = {&loc_outmix, nullptr};
// create audio player
const SLInterfaceID ids[2] = {SL_IID_BUFFERQUEUE, SL_IID_VOLUME};
const SLboolean req[2] = {SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE};
result = (*engineEngine)->CreateAudioPlayer(engineEngine, &bqPlayerObject, &audioSrc, &audioSnk, 2, ids, req);
assert(SL_RESULT_SUCCESS == result);
// create audio player
const SLInterfaceID ids[2] = {SL_IID_BUFFERQUEUE, SL_IID_VOLUME};
const SLboolean req[2] = {SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE};
result =
(*engineEngine)
->CreateAudioPlayer(engineEngine, &bqPlayerObject, &audioSrc, &audioSnk, 2, ids, req);
assert(SL_RESULT_SUCCESS == result);
result = (*bqPlayerObject)->Realize(bqPlayerObject, SL_BOOLEAN_FALSE);
assert(SL_RESULT_SUCCESS == result);
result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_PLAY, &bqPlayerPlay);
assert(SL_RESULT_SUCCESS == result);
result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_BUFFERQUEUE,
&bqPlayerBufferQueue);
assert(SL_RESULT_SUCCESS == result);
result = (*bqPlayerBufferQueue)->RegisterCallback(bqPlayerBufferQueue, bqPlayerCallback, nullptr);
assert(SL_RESULT_SUCCESS == result);
result = (*bqPlayerPlay)->SetPlayState(bqPlayerPlay, SL_PLAYSTATE_PLAYING);
assert(SL_RESULT_SUCCESS == result);
result = (*bqPlayerObject)->Realize(bqPlayerObject, SL_BOOLEAN_FALSE);
assert(SL_RESULT_SUCCESS == result);
result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_PLAY, &bqPlayerPlay);
assert(SL_RESULT_SUCCESS == result);
result =
(*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_BUFFERQUEUE, &bqPlayerBufferQueue);
assert(SL_RESULT_SUCCESS == result);
result = (*bqPlayerBufferQueue)->RegisterCallback(bqPlayerBufferQueue, bqPlayerCallback, nullptr);
assert(SL_RESULT_SUCCESS == result);
result = (*bqPlayerPlay)->SetPlayState(bqPlayerPlay, SL_PLAYSTATE_PLAYING);
assert(SL_RESULT_SUCCESS == result);
// Render and enqueue a first buffer.
curBuffer ^= 1;
g_mixer = m_mixer.get();
// Render and enqueue a first buffer.
curBuffer ^= 1;
g_mixer = m_mixer.get();
result = (*bqPlayerBufferQueue)->Enqueue(bqPlayerBufferQueue, buffer[0], sizeof(buffer[0]));
if (SL_RESULT_SUCCESS != result)
return false;
result = (*bqPlayerBufferQueue)->Enqueue(bqPlayerBufferQueue, buffer[0], sizeof(buffer[0]));
if (SL_RESULT_SUCCESS != result)
return false;
return true;
return true;
}
void OpenSLESStream::Stop()
{
if (bqPlayerObject != nullptr)
{
(*bqPlayerObject)->Destroy(bqPlayerObject);
bqPlayerObject = nullptr;
bqPlayerPlay = nullptr;
bqPlayerBufferQueue = nullptr;
bqPlayerMuteSolo = nullptr;
bqPlayerVolume = nullptr;
}
if (bqPlayerObject != nullptr)
{
(*bqPlayerObject)->Destroy(bqPlayerObject);
bqPlayerObject = nullptr;
bqPlayerPlay = nullptr;
bqPlayerBufferQueue = nullptr;
bqPlayerMuteSolo = nullptr;
bqPlayerVolume = nullptr;
}
if (outputMixObject != nullptr)
{
(*outputMixObject)->Destroy(outputMixObject);
outputMixObject = nullptr;
}
if (outputMixObject != nullptr)
{
(*outputMixObject)->Destroy(outputMixObject);
outputMixObject = nullptr;
}
if (engineObject != nullptr)
{
(*engineObject)->Destroy(engineObject);
engineObject = nullptr;
engineEngine = nullptr;
}
if (engineObject != nullptr)
{
(*engineObject)->Destroy(engineObject);
engineObject = nullptr;
engineEngine = nullptr;
}
}
#endif

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