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QB64-PE/internal/c/parts/audio/audio.cpp
Samuel Gomes 8425f117a7 Fix #169 & #170
2022-09-10 04:08:31 +05:30

2135 lines
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//----------------------------------------------------------------------------------------------------
// ___ ___ __ _ _ ___ ___ _ _ _ ___ _
// / _ \| _ ) / /| | || _ \ __| /_\ _ _ __| (_)___ | __|_ _ __ _(_)_ _ ___
// | (_) | _ \/ _ \_ _| _/ _| / _ \ || / _` | / _ \ | _|| ' \/ _` | | ' \/ -_)
// \__\_\___/\___/ |_||_| |___| /_/ \_\_,_\__,_|_\___/ |___|_||_\__, |_|_||_\___|
// |___/
//
// QB64-PE Audio Engine powered by miniaudio (https://miniaud.io/)
//
// Copyright (c) 2022 Samuel Gomes
// https://github.com/a740g
//
//-----------------------------------------------------------------------------------------------------
//-----------------------------------------------------------------------------------------------------
// HEADER FILES
//-----------------------------------------------------------------------------------------------------
// Set this to 1 if we want to print debug messages to stderr
#define AUDIO_DEBUG 0
#include "audio.h"
#include <algorithm>
#include <vector>
// Enable Ogg Vorbis decoding
#define STB_VORBIS_HEADER_ONLY
#include "extras/stb_vorbis.c"
// The main miniaudio header
#include "miniaudio.h"
// Although Matt says we should not be doing this, this has worked out to be ok so far
// We need 'qbs' and also the 'mem' stuff from here
// I am not using 'list' anymore and have migrated the code to use C++ vectors instead
// We'll likely keep the 'include' this way because I do not want to duplicate stuff and cause issues
// For now, we'll wait for Matt until he sorts out things to smaller and logical files
#include "../../libqb.h"
//-----------------------------------------------------------------------------------------------------
//-----------------------------------------------------------------------------------------------------
// CONSTANTS
//-----------------------------------------------------------------------------------------------------
// This should be defined elsewhere (in libqb?). Since it is not, we are doing it here
#define INVALID_MEM_LOCK 1073741821
// This should be defined elsewhere (in libqb?). Since it is not, we are doing it here
#define MEM_TYPE_SOUND 5
// In QuickBASIC false means 0 and true means -1 (sad, but true XD)
#define QB_FALSE MA_FALSE
#define QB_TRUE -MA_TRUE
// This is returned to the caller if handle allocation fails with a -1
// AllocateSoundHandle() does not return 0 because it is a valid internal handle
// Handle 0 is 'handled' as a special case
#define INVALID_SOUND_HANDLE 0
// This is the string that must be passed in the requirements parameter to stream a sound from storage
#define REQUIREMENT_STRING_STREAM "STREAM"
//-----------------------------------------------------------------------------------------------------
//-----------------------------------------------------------------------------------------------------
// MACROS
//-----------------------------------------------------------------------------------------------------
#define SAMPLE_FRAME_SIZE(_type_, _channels_) (sizeof(_type_) * (_channels_))
// This basically checks if the handle is within vector limits and 'isUsed' is set to true
// We are relying on C's boolean short-circuit to not evaluate the last 'isUsed' if previous conditions are false
// Here we are checking > 0 because this is meant to check user handles only
#define IS_SOUND_HANDLE_VALID(_handle_) \
((_handle_) > 0 && (_handle_) < audioEngine.soundHandles.size() && audioEngine.soundHandles[_handle_]->isUsed && \
!audioEngine.soundHandles[_handle_]->autoKill)
#ifdef QB64_WINDOWS
# define ZERO_VARIABLE(_v_) ZeroMemory(&(_v_), sizeof(_v_))
#else
# define ZERO_VARIABLE(_v_) memset(&(_v_), 0, sizeof(_v_))
#endif
//-----------------------------------------------------------------------------------------------------
//-----------------------------------------------------------------------------------------------------
// FORWARD DECLARATIONS
//-----------------------------------------------------------------------------------------------------
// This adds our customer backend (format decoders) VTables to our ma_resource_manager_config
void AudioEngineAttachCustomBackendVTables(ma_resource_manager_config *maResourceManagerConfig);
// These are stuff that was not declared anywhere else
// We will wait for Matt to cleanup the C/C++ source file and include header files that declare this stuff
qbs *qbs_new_txt_len(const char *txt, int32 len); // Not declared in libqb.h
int32 func_instr(int32 start, qbs *str, qbs *substr, int32 passed); // Did not find this declared anywhere
void new_mem_lock(); // This is required for MemSound()
void free_mem_lock(mem_lock *lock); // Same as above
#ifndef QB64_WINDOWS
void Sleep(uint32 milliseconds); // There is a non-Windows implementation. However it is not declared anywhere
#endif
extern ptrszint dblock; // Required for Play(). Did not find this declared anywhere
extern uint64 mem_lock_id; // Another one that we need for the mem stuff
extern mem_lock *mem_lock_base; // Same as above
extern mem_lock *mem_lock_tmp; // Same as above
//-----------------------------------------------------------------------------------------------------
//-----------------------------------------------------------------------------------------------------
// STRUCTURES, CLASSES & ENUMERATIONS
//-----------------------------------------------------------------------------------------------------
/// <summary>
/// Type of sound
/// </summary>
enum struct SoundType {
None, // No sound or internal sound whose buffer is managed by the QBPE audio engine
Static, // Static sounds that are completely managed by miniaudio
Raw // Raw sound stream that is managed by the QBPE audio engine
};
/// <summary>
/// This struct encapsulates a sample frame block and it's management.
/// We choose these 'classes' to be barebones and completely transparent for performance reasons.
/// </summary>
struct SampleFrameBlockNode {
SampleFrameBlockNode *next; // Next block node in the chain
ma_uint32 samples; // Size of the block in 'samples'. See below
ma_uint32 offset; // Where is the write cursor in the buffer (in samples!)?
float *buffer; // The actual sample frame block buffer
bool force; // When this is set, the buffer will be processed even when if it is not full
SampleFrameBlockNode() = delete; // No default constructor
SampleFrameBlockNode(const SampleFrameBlockNode &) = delete; // No default copy constructor
SampleFrameBlockNode &operator=(SampleFrameBlockNode &) = delete; // No assignment operator
/// <summary>
/// The constructor parameter is in sample frames.
/// For a stereo sample frame we'll need (sample frames * 2) samples.
/// Each sample is sizeof(float) bytes.
/// </summary>
/// <param name="sampleFrames">Number of sample frames needed</param>
SampleFrameBlockNode(ma_uint32 sampleFrames) {
next = nullptr; // Set this to null. This will managed by the 'Queue' struct
samples = sampleFrames << 1; // 2 channels (stereo)
offset = 0; // Set the write cursor to 0
force = false; // Set the force flag to false by default
buffer = new float[samples](); // Allocate a zeroed float buffer of size floats. Ah, Creative Silence!
}
/// <summary>
/// Free the sample frame block that was allocated.
/// </summary>
~SampleFrameBlockNode() { delete[] buffer; }
/// <summary>
/// Pushes a sample frame in the block and increments the offset.
/// miniaudio expects it's stereo PCM data interleaved (LRLR format).
/// No clipping is required because miniaudio does that for us (sweet!)
/// </summary>
/// <param name="l">Left floating point sample</param>
/// <param name="r">Right floating point sample</param>
/// <returns>Return true if operation was succcessful. False if block is full</returns>
bool PushSampleFrame(float l, float r) {
if (buffer && offset < samples) {
buffer[offset] = l;
++offset;
buffer[offset] = r;
++offset;
return true;
}
return false;
}
/// <summary>
/// Check if the buffer is completely filled.
/// </summary>
/// <returns>Returns true if buffer is full</returns>
bool IsBufferFull() { return offset >= samples || force; }
};
/// <summary>
/// This is a light weight queue of type SampleFrameBlockNode and also the guts of SndRaw.
/// We could have used some std container but I wanted something really lean, simple and transparent.
/// </summary>
struct SampleFrameBlockQueue {
SampleFrameBlockNode *first; // First sample frame block
SampleFrameBlockNode *last; // Last sample frame block
size_t blockCount; // Sample frame block count
size_t frameCount; // Number of sample frames we have in the queue
ma_uint32 sampleRate; // The sample rate reported by ma_engine
ma_uint32 blockSampleFrames; // How many sample frames do we need per 'block'. See below
float *buffer; // This is the ping-pong buffer where the samples block will be 'streamed' to
ma_uint32 bufferSampleFrames; // Size of the ping-pong buffer in *samples frames*
bool updateFlag; // We will only update the buffer with fresh samples when this flag is not equal to the check condition
ma_uint32 bufferUpdatePosition; // The position (in samples) in the buffer where we should be copying a sample block
ma_uint32 sampleFramesPlaying; // The number of sample frames that was sent for playback
ma_uint64 maEngineTime; // miniaudio engine time use for correct length calculation
ma_sound *maSound; // Pointer to a ma_sound object that was passed in the constructor
ma_engine *maEngine; // Pointer to a ma_engine object
ma_audio_buffer maBuffer; // miniaudio buffer object
ma_audio_buffer_config maBufferConfig; // miniaudio buffer configuration
ma_result maResult; // This is the result of the last miniaudio operation (used for trapping errors)
SampleFrameBlockQueue() = delete; // No default constructor
SampleFrameBlockQueue(const SampleFrameBlockQueue &) = delete; // No default copy constructor
SampleFrameBlockQueue &operator=(SampleFrameBlockQueue &) = delete; // No assignment operator
/// <summary>
/// This initializes the queue and calculates the sample frames per block
/// </summary>
/// <param name="pmaEngine">A pointer to a miniaudio engine object</param>
/// <param name="pmaSound">A pointer to a miniaudio sound object</param>
SampleFrameBlockQueue(ma_engine *pmaEngine, ma_sound *pmaSound) {
first = last = nullptr;
blockCount = frameCount = maEngineTime = sampleFramesPlaying = bufferUpdatePosition = 0;
maSound = pmaSound; // Save the pointer to the ma_sound object (this is basically from a QBPE sound handle)
maEngine = pmaEngine; // Save the pointer to the ma_engine object (this should come from the QBPE sound engine)
sampleRate = ma_engine_get_sample_rate(maEngine); // Save the sample rate
// We can get away with '>> 3' because the sound loop function is called @ ~60Hz
// This should work even on entry level systems. Tested on AMD A6-9200 (230.4 GFLOPS), Crostini Linux
// Also note that the nodes will allocates twice this to account for 2 channels
blockSampleFrames = sampleRate >> 3;
bufferSampleFrames = blockSampleFrames * 2; // We want the playback buffer twice the size of a block to do a proper ping-pong
buffer = new float[bufferSampleFrames * 2](); // Allocate a zeroed float buffer of bufferSizeSampleFrames * 2 floats (2 is for 2 channels - stereo)
updateFlag = false; // Set this to false because we want the initial check to fail
if (buffer) {
// Setup the ma buffer
maBufferConfig = ma_audio_buffer_config_init(ma_format::ma_format_f32, 2, bufferSampleFrames, buffer, NULL);
maResult = ma_audio_buffer_init(&maBufferConfig, &maBuffer);
AUDIO_DEBUG_CHECK(maResult == MA_SUCCESS);
// Create a ma_sound from the ma_buffer
maResult = ma_sound_init_from_data_source(maEngine, &maBuffer, 0, NULL, maSound);
AUDIO_DEBUG_CHECK(maResult == MA_SUCCESS);
// Play the ma_sound
maResult = ma_sound_start(maSound);
AUDIO_DEBUG_CHECK(maResult == MA_SUCCESS);
// Set the buffer to loop forever
ma_sound_set_looping(maSound, MA_TRUE);
}
AUDIO_DEBUG_PRINT("Raw sound stream created with %u sample frame block size", blockSampleFrames);
}
/// <summary>
/// This simply pops all sample blocks
/// </summary>
~SampleFrameBlockQueue() {
if (buffer) {
// Stop playback
maResult = ma_sound_stop(maSound);
AUDIO_DEBUG_CHECK(maResult == MA_SUCCESS);
// Delete the ma_sound object
ma_sound_uninit(maSound);
// Delete the ma_buffer object
ma_audio_buffer_uninit(&maBuffer);
}
while (PopSampleFrameBlock())
;
delete[] buffer;
AUDIO_DEBUG_PRINT("Raw sound stream closed");
}
/// <summary>
/// This pushes a sample frame into the queue.
/// If there are no sample frame blocks then it creates one.
/// If the last sample frame block is full it creates a new one and links it to the chain.
/// Note that in QBPE all samples frames are assumed to be stereo.
/// Mono sample frames are simply simulated by playing the same data from left and right.
/// No clipping is required because miniaudio does that for us (sweet!)
/// </summary>
/// <param name="l">The left sample</param>
/// <param name="r">The right sample</param>
/// <returns>Returns true if operation was successful</returns>
bool PushSampleFrame(float l, float r) {
// Attempt to push the frame into the last node if one exists
// If successfull return true
if (last && last->PushSampleFrame(l, r)) {
++frameCount; // Increment the frame count
return true;
}
// If we reached here, then it means that either there are no nodes or the last one is full
// Simply create a new node and then link it to the chain
SampleFrameBlockNode *node = new SampleFrameBlockNode(blockSampleFrames);
// Return false if memory allocation failed or we're mot able to save the sample frame
if (!node || !node->PushSampleFrame(l, r)) {
delete node;
return false; // Ignore the sample frame and exit silently
}
if (last)
last->next = node; // Add the node to the last node if we have nodes in the queue
else
first = node; // Else this is the first node
last = node; // The last item in the queue is node
++blockCount; // Increase the frame block count
++frameCount; // Increment the frame count
return true;
}
/// <summary>
/// This pops a sample frame block from the front of the queue.
/// The sample frame block can be accessed before popping using the 'first' member.
/// Popping a block frees and invalidates the memory it was using. So, pop a block only when we are sure that we do not need it.
/// </summary>
/// <returns>Returns true if we were able to pop. False means the queue is empty</returns>
bool PopSampleFrameBlock() {
// Only if the queue has some sample frame blocks then...
if (blockCount) {
SampleFrameBlockNode *node = first; // Set node to the first frame in the queue
--blockCount; // Decrement the block count now so that we know what to do with 'last'
frameCount -= node->offset >> 1; // Decrease frame count by number of sample frames written in the block (/ 2 for channels)
first = node->next; // Detach the node. If this is the last node then 'first' will be NULL cause node->next is NULL
if (!blockCount)
last = nullptr; // This means that node was the last node
delete node; // Free the node
return true;
}
return false;
}
/// <summary>
/// Returns the length, in sample frames of sound queued.
/// </summary>
/// <returns>The length left to play in sample frames</returns>
ma_uint64 GetSampleFramesRemaining() {
// Calculate the time difference (ma_engine time is really just of a sum of sample frames sent to the device)
ma_uint64 maEngineDeltaTime = ma_engine_get_time(maEngine) - maEngineTime;
// Decrement the delta from the sample frames that are playing
// Using std::min here is probably risky since these are all unsigned types
sampleFramesPlaying = maEngineDeltaTime > sampleFramesPlaying ? 0 : (ma_uint32)(sampleFramesPlaying - maEngineDeltaTime);
// Add this to the frames in the queue
return sampleFramesPlaying + frameCount;
}
/// <summary>
/// Returns the length, in seconds of sound queued.
/// </summary>
/// <returns>The length left to play in seconds</returns>
double GetTimeRemaining() {
ma_uint64 sampleFramesRemaining = GetSampleFramesRemaining();
// This will help us avoid situations where we can get a non-zero value even if GetSampleFramesRemaining returns 0
if (!sampleFramesRemaining)
return 0;
else
return (double)sampleFramesRemaining / sampleRate;
}
/// <summary>
/// Check if everything is ready to go
/// </summary>
/// <returns>Returns true if everything is a go</returns>
bool IsSetupValid() { return buffer && maEngine && maSound && maResult == MA_SUCCESS; }
/// <summary>
/// This keeps the ping-pong (ring? whatever...) buffer fed and the sound stream going
/// </summary>
void Update() {
// Figure out which pcm frame of the buffer is miniaudio playing
ma_uint64 readCursor;
maResult = ma_sound_get_cursor_in_pcm_frames(maSound, &readCursor);
AUDIO_DEBUG_CHECK(maResult == MA_SUCCESS);
bool checkCondition = readCursor < blockSampleFrames; // Since buffer sample frame size = blockSampleFrames * 2
// Only proceed to update if our flag is not the same as our condition
if (checkCondition != updateFlag) {
// The line below does two sneaky things that deserve explanation
// 1. We are using bufferSampleFrames which is set to exactly halfway through the buffer since we are using stereo (see constructor)
// 2. The boolean condition above will always be 1 if the read cursor is in the lower-half and hence push the position to the top-half
// 3. Obviously, this means that if the condition is 0 then position will be set to the lower-half
bufferUpdatePosition = checkCondition * bufferSampleFrames; // This line basically toggles the buffer copy position
// Check if we have any blocks in the queue and stream only if the block is full
if (blockCount && first->IsBufferFull()) {
// We check this here so that even if the buffer is not allocated, the block object will be popped off
if (first->buffer) {
// Simply copy the first block in the queue
std::copy(first->buffer, first->buffer + first->samples, buffer + bufferUpdatePosition);
}
// Save the number of samples frames sent for playback and the current time for correct time calculation
sampleFramesPlaying = blockSampleFrames;
maEngineTime = ma_engine_get_time(maEngine);
// And then pop it off
PopSampleFrameBlock();
} else { // Else we'll stream silence
// We are using bufferSampleFrames here for the same reason as the explanation above
std::fill(buffer + bufferUpdatePosition, buffer + bufferUpdatePosition + bufferSampleFrames, NULL);
}
updateFlag = checkCondition; // Save our check condition to our flag
}
}
};
/// <summary>
/// Sound handle type
/// This describes every sound the system will ever play (including raw streams).
/// </summary>
struct SoundHandle {
bool isUsed; // Is this handle in active use?
SoundType type; // Type of sound (see SoundType enum class)
bool autoKill; // Do we need to auto-clean this sample / stream after playback is done?
ma_sound maSound; // miniaudio sound
ma_uint32 maFlags; // miniaudio flags that were used when initializing the sound
SampleFrameBlockQueue *rawQueue; // Raw sample frame queue
void *memLockOffset; // This is a pointer from new_mem_lock()
uint64 memLockId; // This is mem_lock_id created by new_mem_lock()
SoundHandle(const SoundHandle &) = delete; // No default copy constructor
SoundHandle &operator=(SoundHandle &) = delete; // No assignment operator
/// <summary>
/// Just initializes some important members.
/// 'inUse' will be set to true by AllocateSoundHandle().
/// This is done here, as well as slightly differently in AllocateSoundHandle() for safety.
/// </summary>
SoundHandle() {
isUsed = false;
type = SoundType::None;
autoKill = false;
rawQueue = nullptr;
ZERO_VARIABLE(maSound);
maFlags = MA_SOUND_FLAG_NO_PITCH | MA_SOUND_FLAG_NO_SPATIALIZATION | MA_SOUND_FLAG_WAIT_INIT;
memLockId = INVALID_MEM_LOCK;
memLockOffset = nullptr;
}
};
/// <summary>
/// Type will help us keep track of the audio engine state
/// </summary>
struct AudioEngine {
bool isInitialized; // This is set to true if we were able to initialize miniaudio and allocated all required resources
bool initializationFailed; // This is set to true if a past initialization attempt failed
ma_resource_manager_config maResourceManagerConfig; // miniaudio resource manager configuration
ma_resource_manager maResourceManager; // miniaudio resource manager
ma_engine_config maEngineConfig; // miniaudio engine configuration (will be used to pass in the resource manager)
ma_engine maEngine; // This is the primary miniaudio engine 'context'. Everything happens using this!
ma_result maResult; // This is the result of the last miniaudio operation (used for trapping errors)
ma_uint32 sampleRate; // Sample rate used by the miniaudio engine
int32_t sndInternal; // Internal sound handle that we will use for Play(), Beep() & Sound()
int32_t sndInternalRaw; // Internal sound handle that we will use for the QB64 'handle-less' raw stream
std::vector<SoundHandle *> soundHandles; // This is the audio handle list used by the engine and by everything else
int32_t lowestFreeHandle; // This is the lowest handle then was recently freed. We'll start checking for free handles from here
bool musicBackground; // Should 'Sound' and 'Play' work in the background or block the caller?
AudioEngine(const AudioEngine &) = delete; // No default copy constructor
AudioEngine &operator=(AudioEngine &) = delete; // No assignment operator
/// <summary>
/// Just initializes some important members.
/// </summary>
AudioEngine() {
isInitialized = initializationFailed = false;
sampleRate = 0;
lowestFreeHandle = 0;
sndInternal = sndInternalRaw = INVALID_SOUND_HANDLE;
musicBackground = false;
}
/// <summary>
/// This allocates a sound handle. It will return -1 on error.
/// Handle 0 is used internally for Beep, Sound and Play and thus cannot be used by the user.
/// Basically, we go through the vector and find an object pointer were 'isUsed' is set as false and return the index.
/// If such an object pointer is not found, then we add a pointer to a new object at the end of the vector and return the index.
/// We are using pointers because miniaudio keeps using stuff from ma_sound and these cannot move in memory when the vector is resized.
/// The handle is put-up for recycling simply by setting the 'isUsed' member to false.
/// Note that this means the vector will keep growing until the largest handle (index) and never shrink.
/// The choice of using a vector was simple - performance. Vector performance when using 'indexes' is next to no other.
/// The vector will be pruned only when snd_un_init gets called.
/// We will however, be good citizens and will also 'delete' the objects when snd_un_init gets called.
/// All this means that a sloppy programmer may be able to grow the vector and eventually the system may run out of memory and crash.
/// But that's ok. Sloppy programmers (like me) must be punished until they learn! XD
/// This also increments 'lowestFreeHandle' to allocated handle + 1.
/// </summary>
/// <returns>Returns a non-negative handle if successful</returns>
int32_t AllocateSoundHandle() {
if (!isInitialized)
return -1; // We cannot return 0 here. Since 0 is a valid internal handle
size_t h, vectorSize = soundHandles.size(); // Save the vector size
// Scan the vector starting from lowestFreeHandle
// This will help us quickly allocate a free handle and should be a decent optimization for SndPlayCopy()
for (h = lowestFreeHandle; h < vectorSize; h++) {
if (!soundHandles[h]->isUsed) {
AUDIO_DEBUG_PRINT("Recent sound handle %i recycled", h);
break;
}
}
if (h >= vectorSize) {
// Scan through the entire vector and return a slot that is not being used
// Ideally this should execute in extremely few (if at all) senarios
// Also, this loop should not execute if size is 0
for (h = 0; h < vectorSize; h++) {
if (!soundHandles[h]->isUsed) {
AUDIO_DEBUG_PRINT("Sound handle %i recycled", h);
break;
}
}
}
if (h >= vectorSize) {
// If we have reached here then either the vector is empty or there are no empty slots
// Simply create a new SoundHandle at the back of the vector
SoundHandle *newHandle = new SoundHandle;
if (!newHandle)
return -1; // We cannot return 0 here. Since 0 is a valid internal handle
soundHandles.push_back(newHandle);
size_t newVectorSize = soundHandles.size();
// If newVectorSize == vectorSize then push_back() failed
if (newVectorSize <= vectorSize) {
delete newHandle;
return -1; // We cannot return 0 here. Since 0 is a valid internal handle
}
h = newVectorSize - 1; // The handle is simply newVectorSize - 1
AUDIO_DEBUG_PRINT("Sound handle %i created", h);
}
AUDIO_DEBUG_CHECK(soundHandles[h]->isUsed == false);
// Initializes a sound handle that was just allocated.
// This will set it to 'in use' after applying some defaults.
soundHandles[h]->type = SoundType::None;
soundHandles[h]->autoKill = false;
soundHandles[h]->rawQueue = nullptr;
ZERO_VARIABLE(soundHandles[h]->maSound);
// We do not use pitch shifting, so this will give a little performance boost
// Spatialization is disabled by default but will be enabled on the fly if required
soundHandles[h]->maFlags = MA_SOUND_FLAG_NO_PITCH | MA_SOUND_FLAG_NO_SPATIALIZATION | MA_SOUND_FLAG_WAIT_INIT;
soundHandles[h]->memLockId = INVALID_MEM_LOCK;
soundHandles[h]->memLockOffset = nullptr;
soundHandles[h]->isUsed = true;
AUDIO_DEBUG_PRINT("Sound handle %i returned", h);
lowestFreeHandle = h + 1; // Set lowestFreeHandle to allocated handle + 1
return (int32_t)(h);
}
/// <summary>
/// The frees and unloads an open sound.
/// If the sound is playing or looping, it will be stopped.
/// If the sound is a stream of raw samples then it is stopped and freed.
/// Finally the handle is invalidated and put-up for recycling.
/// If the handle being freed is lower than 'lowestFreeHandle' then this saves the handle to 'lowestFreeHandle'.
/// </summary>
/// <param name="handle">A sound handle</param>
void FreeSoundHandle(int32_t handle) {
if (isInitialized && handle >= 0 && handle < soundHandles.size() && soundHandles[handle]->isUsed) {
// Sound type specific cleanup
switch (soundHandles[handle]->type) {
case SoundType::Static:
ma_sound_uninit(&soundHandles[handle]->maSound);
break;
case SoundType::Raw:
delete soundHandles[handle]->rawQueue;
soundHandles[handle]->rawQueue = nullptr;
break;
case SoundType::None:
if (handle != 0)
AUDIO_DEBUG_PRINT("Sound type is 'None' when handle value is not 0");
break;
default:
AUDIO_DEBUG_PRINT("Condition not handled"); // It should not come here
}
// Invalidate any memsound stuff
if (soundHandles[handle]->memLockOffset) {
free_mem_lock((mem_lock *)soundHandles[handle]->memLockOffset);
soundHandles[handle]->memLockId = INVALID_MEM_LOCK;
soundHandles[handle]->memLockOffset = nullptr;
}
// Now simply set the 'isUsed' member to false so that the handle can be recycled
soundHandles[handle]->isUsed = false;
soundHandles[handle]->type = SoundType::None;
// Save the free hanndle to lowestFreeHandle if it is lower than lowestFreeHandle
if (handle < lowestFreeHandle)
lowestFreeHandle = handle;
AUDIO_DEBUG_PRINT("Sound handle %i marked as free", handle);
}
}
};
//-----------------------------------------------------------------------------------------------------
//-----------------------------------------------------------------------------------------------------
// GLOBAL VARIABLES
//-----------------------------------------------------------------------------------------------------
// This keeps track of the audio engine state
static AudioEngine audioEngine;
//-----------------------------------------------------------------------------------------------------
//-----------------------------------------------------------------------------------------------------
// FUNCTIONS
//-----------------------------------------------------------------------------------------------------
/// <summary>
/// This creates 16-bit signed stereo data. The sound buffer is allocated and then returned.
/// Do we really need stereo for Play(), Sound() and Beep()?
/// </summary>
/// <param name="frequency">The sound frequency</param>
/// <param name="length">The duration of the sound in seconds</param>
/// <param name="volume">The volume of the sound (0.0 - 1.0)</param>
/// <param name="soundwave_bytes">A pointer to an integer that will receive the buffer size in bytes. This cannot be NULL</param>
/// <returns></returns>
static ma_uint8 *GenerateWaveform(double frequency, double length, double volume, ma_int32 *soundwave_bytes) {
static ma_uint8 *data;
static ma_int32 i;
static ma_int16 x, lastx;
static ma_int16 *sp;
static double samples;
static ma_int32 samplesi;
static ma_int32 direction;
static double value;
static double volume_multiplier;
static ma_int32 waveend;
static double gradient;
// calculate total number of samples required
samples = length * audioEngine.sampleRate;
samplesi = samples;
if (!samplesi)
samplesi = 1;
*soundwave_bytes = samplesi * SAMPLE_FRAME_SIZE(ma_int16, 2);
// Frequency equal to or above 20000 will produce silence
// This is per QuickBASIC 4.5 behavior
if (frequency < 20000) {
data = (ma_uint8 *)malloc(*soundwave_bytes);
} else {
data = (ma_uint8 *)calloc(*soundwave_bytes, sizeof(ma_uint8));
return data;
}
if (!data)
return nullptr;
sp = (ma_int16 *)data;
direction = 1;
value = 0;
volume_multiplier = volume * 32767.0;
waveend = 0;
// frequency*4.0*length is the total distance value will travel (+1,-2,+1[repeated])
// samples is the number of steps to do this in
if (samples)
gradient = (frequency * 4.0 * length) / samples;
else
gradient = 0; // avoid division by 0
lastx = 1; // set to 1 to avoid passing initial comparison
for (i = 0; i < samplesi; i++) {
x = value * volume_multiplier;
*sp++ = x;
*sp++ = x;
if (x > 0) {
if (lastx <= 0) {
waveend = i;
}
}
lastx = x;
if (direction) {
if ((value += gradient) >= 1.0) {
direction = 0;
value = 2.0 - value;
}
} else {
if ((value -= gradient) <= -1.0) {
direction = 1;
value = -2.0 - value;
}
}
} // i
if (waveend)
*soundwave_bytes = waveend * SAMPLE_FRAME_SIZE(ma_int16, 2);
return data;
}
/// <summary>
/// Returns the of a sound buffer in bytes.
/// </summary>
/// <param name="length">Length in seconds</param>
/// <returns>Length in bytes</returns>
static ma_int32 WaveformBufferSize(double length) {
static ma_int32 samples;
samples = (ma_int32)(length * audioEngine.sampleRate);
if (!samples)
samples = 1;
return samples * SAMPLE_FRAME_SIZE(ma_int16, 2);
}
/// <summary>
/// This sends a buffer to a raw queue for playback.
/// Buffer required in 16-bit stereo at native frequency.
/// The buffer is freed.
/// </summary>
/// <param name="data">Sound buffer</param>
/// <param name="bytes">Length of buffer in bytes</param>
/// <param name="block">So we have to wait until playback completes</param>
/// <param name="sndRawQueue">A pointer to a raw queue object</param>
static void SendWaveformToQueue(ma_uint8 *data, ma_int32 bytes, bool block) {
static ma_int32 i;
static ma_int64 time_ms;
if (!data)
return;
// Move data into sndraw handle
for (i = 0; i < bytes; i += SAMPLE_FRAME_SIZE(ma_int16, 2)) {
audioEngine.soundHandles[audioEngine.sndInternal]->rawQueue->PushSampleFrame((float)((ma_int16 *)(data + i))[0] / 32768.0f,
(float)((ma_int16 *)(data + i))[1] / 32768.0f);
}
free(data); // free the sound data
// This will push any unfinished block for playback
if (audioEngine.soundHandles[audioEngine.sndInternal]->rawQueue->last)
audioEngine.soundHandles[audioEngine.sndInternal]->rawQueue->last->force = true;
// This will wait for the block to finish (if specified)
// We'll be good citizens and give-up our time-slices while waiting
if (block) {
time_ms = (ma_int64)(audioEngine.soundHandles[audioEngine.sndInternal]->rawQueue->GetTimeRemaining() * 950.0 - 250.0);
if (time_ms > 0)
Sleep(time_ms);
}
}
/// <summary>
/// This generates a sound at the specified frequency for the specified amount of time.
/// </summary>
/// <param name="frequency">Sound frequency</param>
/// <param name="lengthInClockTicks">Duration in clock ticks. There are 18.2 clock ticks per second</param>
void sub_sound(double frequency, double lengthInClockTicks) {
static ma_uint8 *data;
static ma_int32 soundwave_bytes;
if (new_error || !audioEngine.isInitialized || audioEngine.sndInternal != 0)
return;
if ((frequency < 37.0) && (frequency != 0))
goto error;
if (frequency > 32767.0)
goto error;
if (lengthInClockTicks < 0.0)
goto error;
if (lengthInClockTicks > 65535.0)
goto error;
if (lengthInClockTicks == 0.0)
return;
audioEngine.soundHandles[audioEngine.sndInternal]->type = SoundType::Raw; // This will start processing handle 0 as a raw stream
data = GenerateWaveform(frequency, lengthInClockTicks / 18.2, 1, &soundwave_bytes);
SendWaveformToQueue(data, soundwave_bytes, !audioEngine.musicBackground);
return;
error:
error(5);
}
/// <summary>
/// This generates a default 'beep' sound.
/// </summary>
void sub_beep() { sub_sound(900, 5); }
/// <summary>
/// This was designed to returned the number of notes in the background music queue.
/// However, here we'll just return the number of sample frame remaining to play when Play(), Sound() or Beep() are used.
/// This allows programs like the following to compile and work.
///
/// Music$ = "MBT180o2P2P8L8GGGL2E-P24P8L8FFFL2D"
/// PLAY Music$
/// WHILE PLAY(0) > 5: WEND
/// PRINT "Just about done!"
/// </summary>
/// <param name="ignore">Well, it's ignored</param>
/// <returns>Returns the number of sample frames left to play for Play(), Sound() & Beep()</returns>
int32_t func_play(int32_t ignore) {
if (audioEngine.isInitialized && audioEngine.sndInternal == 0) {
// This will push any unfinished block for playback
if (audioEngine.soundHandles[audioEngine.sndInternal]->rawQueue->last)
audioEngine.soundHandles[audioEngine.sndInternal]->rawQueue->last->force = true;
return (int32_t)audioEngine.soundHandles[audioEngine.sndInternal]->rawQueue->GetSampleFramesRemaining();
}
return 0;
}
/// <summary>
/// Processes and plays the MML specified in the string.
/// Mmmmm. Spaghetti goodness.
/// Formats:
/// A[#|+|-][0-64]
/// 0-64 is like temp. Lnumber, 0 is whatever the current default is
/// </summary>
/// <param name="str">The string to play</param>
void sub_play(qbs *str) {
static ma_int32 soundwave_bytes;
static ma_uint8 *b, *wave, *wave2;
static double d;
static ma_int32 i, bytes_left, a, x, x2, x3, x4, wave_bytes, wave_base;
static ma_int32 o = 4;
static double t = 120; // quarter notes per minute (120/60=2 per second)
static double l = 4;
static double pause = 1.0 / 8.0; // ML 0.0, MN 1.0/8.0, MS 1.0/4.0
static double length, length2; // derived from l and t
static double frequency;
static double v = 50;
static ma_int32 n; // the semitone-intervaled note to be played
static ma_int32 n_changed; //+,#,- applied?
static ma_int64 number;
static ma_int32 number_entered;
static ma_int32 followup; // 1=play note
static ma_int32 playit;
static ma_int32 fullstops = 0;
if (new_error || !audioEngine.isInitialized || audioEngine.sndInternal != 0)
return;
audioEngine.soundHandles[audioEngine.sndInternal]->type = SoundType::Raw; // This will start processing handle 0 as a raw stream
b = str->chr;
bytes_left = str->len;
wave = NULL;
wave_bytes = 0;
n_changed = 0;
n = 0;
number_entered = 0;
number = 0;
followup = 0;
length = 1.0 / (t / 60.0) * (4.0 / l);
playit = 0;
wave_base = 0; // point at which new sounds will be inserted
next_byte:
if ((bytes_left--) || followup) {
if (bytes_left < 0) {
i = 32;
goto follow_up;
}
i = *b++;
if (i == 32)
goto next_byte;
if (i >= 97 && i <= 122)
a = i - 32;
else
a = i;
if (i == 61) { //= (+VARPTR$)
if (fullstops) {
error(5);
return;
}
if (number_entered) {
error(5);
return;
}
number_entered = 2;
// VARPTR$ reference
/*
'BYTE=1
'INTEGER=2
'STRING=3 SUB-STRINGS must use "X"+VARPTR$(string$)
'SINGLE=4
'INT64=5
'FLOAT=6
'DOUBLE=8
'LONG=20
'BIT=64+n
*/
if (bytes_left < 3) {
error(5);
return;
}
i = *b++;
bytes_left--; // read type byte
x = *(ma_uint16 *)b;
b += 2;
bytes_left -= 2; // read offset within DBLOCK
// note: allowable _BIT type variables in VARPTR$ are all at a byte offset and are all
// padded until the next byte
d = 0;
switch (i) {
case 1:
d = *(char *)(dblock + x);
break;
case (1 + 128):
d = *(ma_uint8 *)(dblock + x);
break;
case 2:
d = *(ma_int16 *)(dblock + x);
break;
case (2 + 128):
d = *(ma_uint16 *)(dblock + x);
break;
case 4:
d = *(float *)(dblock + x);
break;
case 5:
d = *(ma_int64 *)(dblock + x);
break;
case (5 + 128):
d = *(ma_int64 *)(dblock + x); // unsigned conversion is unsupported!
break;
case 6:
d = *(long double *)(dblock + x);
break;
case 8:
d = *(double *)(dblock + x);
break;
case 20:
d = *(ma_int32 *)(dblock + x);
break;
case (20 + 128):
d = *(ma_uint32 *)(dblock + x);
break;
default:
// bit type?
if ((i & 64) == 0) {
error(5);
return;
}
x2 = i & 63;
if (x2 > 56) {
error(5);
return;
} // valid number of bits?
// create a mask
static ma_int64 i64num, mask, i64x;
mask = (((ma_int64)1) << x2) - 1;
i64num = (*(ma_int64 *)(dblock + x)) & mask;
// signed?
if (i & 128) {
mask = ((ma_int64)1) << (x2 - 1);
if (i64num & mask) { // top bit on?
mask = -1;
mask <<= x2;
i64num += mask;
}
} // signed
d = i64num;
}
if (d > 2147483647.0 || d < -2147483648.0) {
error(5);
return;
} // out of range value!
number = round(d);
goto next_byte;
}
// read in a number
if ((i >= 48) && (i <= 57)) {
if (fullstops || (number_entered == 2)) {
error(5);
return;
}
if (!number_entered) {
number = 0;
number_entered = 1;
}
number = number * 10 + i - 48;
goto next_byte;
}
// read fullstops
if (i == 46) {
if (followup != 7 && followup != 1 && followup != 4) {
error(5);
return;
}
fullstops++;
goto next_byte;
}
follow_up:
if (followup == 8) { // V...
if (!number_entered) {
error(5);
return;
}
number_entered = 0;
if (number > 100) {
error(5);
return;
}
v = number;
followup = 0;
if (bytes_left < 0)
goto done;
} // 8
if (followup == 7) { // P...
if (number_entered) {
number_entered = 0;
if (number < 1 || number > 64) {
error(5);
return;
}
length2 = 1.0 / (t / 60.0) * (4.0 / ((double)number));
} else {
length2 = length;
}
d = length2;
for (x = 1; x <= fullstops; x++) {
d /= 2.0;
length2 = length2 + d;
}
fullstops = 0;
soundwave_bytes = WaveformBufferSize(length2);
if (!wave) {
// create buffer
wave = (ma_uint8 *)calloc(soundwave_bytes, 1);
wave_bytes = soundwave_bytes;
wave_base = 0;
} else {
// increase buffer?
if ((wave_base + soundwave_bytes) > wave_bytes) {
wave = (ma_uint8 *)realloc(wave, wave_base + soundwave_bytes);
memset(wave + wave_base, 0, wave_base + soundwave_bytes - wave_bytes);
wave_bytes = wave_base + soundwave_bytes;
}
}
if (i != 44) {
wave_base += soundwave_bytes;
}
playit = 1;
followup = 0;
if (i == 44)
goto next_byte;
if (bytes_left < 0)
goto done;
} // 7
if (followup == 6) { // T...
if (!number_entered) {
error(5);
return;
}
number_entered = 0;
if (number < 32 || number > 255) {
number = 120;
}
t = number;
length = 1.0 / (t / 60.0) * (4.0 / l);
followup = 0;
if (bytes_left < 0)
goto done;
} // 6
if (followup == 5) { // M...
if (number_entered) {
error(5);
return;
}
switch (a) {
case 76: // L
pause = 0;
break;
case 78: // N
pause = 1.0 / 8.0;
break;
case 83: // S
pause = 1.0 / 4.0;
break;
case 66: // MB
if (!audioEngine.musicBackground) {
audioEngine.musicBackground = true;
if (playit) {
playit = 0;
SendWaveformToQueue(wave, wave_bytes, true);
}
wave = NULL;
}
break;
case 70: // MF
if (audioEngine.musicBackground) {
audioEngine.musicBackground = false;
// preceding MB content incorporated into MF block
}
break;
default:
error(5);
return;
}
followup = 0;
goto next_byte;
} // 5
if (followup == 4) { // N...
if (!number_entered) {
error(5);
return;
}
number_entered = 0;
if (number > 84) {
error(5);
return;
}
n = -33 + number;
goto followup1;
followup = 0;
if (bytes_left < 0)
goto done;
} // 4
if (followup == 3) { // O...
if (!number_entered) {
error(5);
return;
}
number_entered = 0;
if (number > 6) {
error(5);
return;
}
o = number;
followup = 0;
if (bytes_left < 0)
goto done;
} // 3
if (followup == 2) { // L...
if (!number_entered) {
error(5);
return;
}
number_entered = 0;
if (number < 1 || number > 64) {
error(5);
return;
}
l = number;
length = 1.0 / (t / 60.0) * (4.0 / l);
followup = 0;
if (bytes_left < 0)
goto done;
} // 2
if (followup == 1) { // A-G...
if (i == 45) { //-
if (n_changed || number_entered) {
error(5);
return;
}
n_changed = 1;
n--;
goto next_byte;
}
if (i == 43 || i == 35) { //+,#
if (n_changed || number_entered) {
error(5);
return;
}
n_changed = 1;
n++;
goto next_byte;
}
followup1:
if (number_entered) {
number_entered = 0;
if (number < 0 || number > 64) {
error(5);
return;
}
if (!number)
length2 = length;
else
length2 = 1.0 / (t / 60.0) * (4.0 / ((double)number));
} else {
length2 = length;
} // number_entered
d = length2;
for (x = 1; x <= fullstops; x++) {
d /= 2.0;
length2 = length2 + d;
}
fullstops = 0;
// frequency=(2^(note/12))*440
frequency = pow(2.0, ((double)n) / 12.0) * 440.0;
// create wave
wave2 = GenerateWaveform(frequency, length2 * (1.0 - pause), v / 100.0, &soundwave_bytes);
if (pause > 0) {
wave2 = (ma_uint8 *)realloc(wave2, soundwave_bytes + WaveformBufferSize(length2 * pause));
memset(wave2 + soundwave_bytes, 0, WaveformBufferSize(length2 * pause));
soundwave_bytes += WaveformBufferSize(length2 * pause);
}
if (!wave) {
// adopt buffer
wave = wave2;
wave_bytes = soundwave_bytes;
wave_base = 0;
} else {
// mix required?
if (wave_base == wave_bytes)
x = 0;
else
x = 1;
// increase buffer?
if ((wave_base + soundwave_bytes) > wave_bytes) {
wave = (ma_uint8 *)realloc(wave, wave_base + soundwave_bytes);
memset(wave + wave_base, 0, wave_base + soundwave_bytes - wave_bytes);
wave_bytes = wave_base + soundwave_bytes;
}
// mix or copy
if (x) {
// mix
static ma_int16 *sp, *sp2;
sp = (ma_int16 *)(wave + wave_base);
sp2 = (ma_int16 *)wave2;
x2 = soundwave_bytes / 2;
for (x = 0; x < x2; x++) {
x3 = *sp2++;
x4 = *sp;
x4 += x3;
if (x4 > 32767)
x4 = 32767;
if (x4 < -32767)
x4 = -32767;
*sp++ = x4;
} // x
} else {
// copy
memcpy(wave + wave_base, wave2, soundwave_bytes);
} // x
free(wave2);
}
if (i != 44) {
wave_base += soundwave_bytes;
}
playit = 1;
n_changed = 0;
followup = 0;
if (i == 44)
goto next_byte;
if (bytes_left < 0)
goto done;
} // 1
if (a >= 65 && a <= 71) {
// modify a to represent a semitonal note (n) interval
switch (a) {
//[c][ ][d][ ][e][f][ ][g][ ][a][ ][b]
// 0 1 2 3 4 5 6 7 8 9 0 1
case 65:
n = 9;
break;
case 66:
n = 11;
break;
case 67:
n = 0;
break;
case 68:
n = 2;
break;
case 69:
n = 4;
break;
case 70:
n = 5;
break;
case 71:
n = 7;
break;
}
n = n + (o - 2) * 12 - 9;
followup = 1;
goto next_byte;
} // a
if (a == 76) { // L
followup = 2;
goto next_byte;
}
if (a == 77) { // M
followup = 5;
goto next_byte;
}
if (a == 78) { // N
followup = 4;
goto next_byte;
}
if (a == 79) { // O
followup = 3;
goto next_byte;
}
if (a == 84) { // T
followup = 6;
goto next_byte;
}
if (a == 60) { //<
o--;
if (o < 0)
o = 0;
goto next_byte;
}
if (a == 62) { //>
o++;
if (o > 6)
o = 6;
goto next_byte;
}
if (a == 80) { // P
followup = 7;
goto next_byte;
}
if (a == 86) { // V
followup = 8;
goto next_byte;
}
error(5);
return;
} // bytes_left
done:
if (number_entered || followup) {
error(5);
return;
} // unhandled data
if (playit) {
SendWaveformToQueue(wave, wave_bytes, !audioEngine.musicBackground);
} // playit
}
/// <summary>
/// This returns the sample rate from ma engine if ma is initialized.
/// </summary>
/// <returns>miniaudio sample rtate</returns>
int32_t func__sndrate() { return audioEngine.sampleRate; }
/// <summary>
/// This loads a sound file into memory and returns a LONG handle value above 0.
/// </summary>
/// <param name="fileName">The is the pathname for the sound file. This can be any format that miniaudio or a miniaudio plugin supports</param>
/// <param name="requirements">This is leftover from the old QB64-SDL days. But we use this to pass some parameters like 'stream'</param>
/// <param name="passed">How many parameters were passed?</param>
/// <returns>Returns a valid sound handle (> 0) if successful or 0 if it fails</returns>
int32_t func__sndopen(qbs *fileName, qbs *requirements, int32_t passed) {
// Some QB strings that we'll need
static qbs *fileNameZ = nullptr;
static qbs *reqs = nullptr;
if (!audioEngine.isInitialized)
return INVALID_SOUND_HANDLE;
if (!fileNameZ)
fileNameZ = qbs_new(0, 0);
if (!reqs)
reqs = qbs_new(0, 0);
qbs_set(fileNameZ, qbs_add(fileName, qbs_new_txt_len("\0", 1))); // s1 = filename + CHR$(0)
if (fileNameZ->len == 1)
return INVALID_SOUND_HANDLE; // Return INVALID_SOUND_HANDLE if file name is null length string
// Alocate a sound handle
int32_t handle = audioEngine.AllocateSoundHandle();
if (handle < 1) // We are not expected to open files with handle 0
return INVALID_SOUND_HANDLE;
// Set some handle properties
audioEngine.soundHandles[handle]->type = SoundType::Static;
// Set the flags to specifiy how we want the audio file to be opened
if (passed && requirements->len) {
qbs_set(reqs, qbs_ucase(requirements)); // Convert tmp str to perm str
if (func_instr(1, reqs, qbs_new_txt(REQUIREMENT_STRING_STREAM), 1))
audioEngine.soundHandles[handle]->maFlags |= MA_SOUND_FLAG_STREAM; // Check if the user wants to stream the file
} else {
audioEngine.soundHandles[handle]->maFlags |= MA_SOUND_FLAG_DECODE; // Else decode and load the whole sound in memory
}
// Forward the request to miniaudio to open the sound file
audioEngine.maResult = ma_sound_init_from_file(&audioEngine.maEngine, (const char *)fileNameZ->chr, audioEngine.soundHandles[handle]->maFlags, NULL, NULL,
&audioEngine.soundHandles[handle]->maSound);
// If the sound failed to copy, then free the handle and return INVALID_SOUND_HANDLE
if (audioEngine.maResult != MA_SUCCESS) {
AUDIO_DEBUG_PRINT("'%s' failed to open", fileNameZ->chr);
audioEngine.soundHandles[handle]->isUsed = false;
return INVALID_SOUND_HANDLE;
}
AUDIO_DEBUG_PRINT("'%s' successfully opened", fileNameZ->chr);
return handle;
}
/// <summary>
/// The frees and unloads an open sound.
/// If the sound is playing, it'll let it finish. Looping sounds will loop until the program is closed.
/// If the sound is a stream of raw samples then any remaining samples pending for playback will be sent to miniaudio and then the handle will be freed.
/// </summary>
/// <param name="handle">A sound handle</param>
void sub__sndclose(int32_t handle) {
if (audioEngine.isInitialized && IS_SOUND_HANDLE_VALID(handle)) {
// If we have a raw stream then force it to push all it's data to miniaudio
// Note that this will take care of checking if the handle is a raw steam and other stuff
// So it is completly safe to call it this way
sub__sndrawdone(handle, true);
// Simply set the autokill flag to true and let the sound loop handle disposing the sound
audioEngine.soundHandles[handle]->autoKill = true;
}
}
/// <summary>
/// This copies a sound to a new handle so that two or more of the same sound can be played at once.
/// </summary>
/// <param name="src_handle">A source sound handle</param>
/// <returns>A new sound handle if successful or 0 on failure</returns>
int32_t func__sndcopy(int32_t src_handle) {
// Check for all invalid cases
if (!audioEngine.isInitialized || !IS_SOUND_HANDLE_VALID(src_handle) || audioEngine.soundHandles[src_handle]->type != SoundType::Static)
return INVALID_SOUND_HANDLE;
// Alocate a sound handle
int32_t dst_handle = audioEngine.AllocateSoundHandle();
// Initialize the sound handle data
if (dst_handle < 1) // We are not expected to open files with handle 0
return INVALID_SOUND_HANDLE;
audioEngine.soundHandles[dst_handle]->type = SoundType::Static; // Set some handle properties
audioEngine.soundHandles[dst_handle]->maFlags = audioEngine.soundHandles[src_handle]->maFlags; // Copy the flags
// Initialize a new copy of the sound
audioEngine.maResult = ma_sound_init_copy(&audioEngine.maEngine, &audioEngine.soundHandles[src_handle]->maSound,
audioEngine.soundHandles[dst_handle]->maFlags, NULL, &audioEngine.soundHandles[dst_handle]->maSound);
// If the sound failed to copy, then free the handle and return INVALID_SOUND_HANDLE
if (audioEngine.maResult != MA_SUCCESS) {
audioEngine.soundHandles[dst_handle]->isUsed = false;
return INVALID_SOUND_HANDLE;
}
return dst_handle;
}
/// <summary>
/// This plays a sound designated by a sound handle.
/// </summary>
/// <param name="handle">A sound handle</param>
void sub__sndplay(int32_t handle) {
if (audioEngine.isInitialized && IS_SOUND_HANDLE_VALID(handle) && audioEngine.soundHandles[handle]->type == SoundType::Static) {
// Reset position to zero only if we are playing and (not looping or we've reached the end of the sound)
// This is based on the old OpenAl-soft code behavior
if (ma_sound_is_playing(&audioEngine.soundHandles[handle]->maSound) &&
(!ma_sound_is_looping(&audioEngine.soundHandles[handle]->maSound) || ma_sound_at_end(&audioEngine.soundHandles[handle]->maSound))) {
audioEngine.maResult = ma_sound_seek_to_pcm_frame(&audioEngine.soundHandles[handle]->maSound, 0);
AUDIO_DEBUG_CHECK(audioEngine.maResult == MA_SUCCESS);
}
// Kickstart playback
audioEngine.maResult = ma_sound_start(&audioEngine.soundHandles[handle]->maSound);
AUDIO_DEBUG_CHECK(audioEngine.maResult == MA_SUCCESS);
// Stop looping the sound if it is
if (ma_sound_is_looping(&audioEngine.soundHandles[handle]->maSound)) {
ma_sound_set_looping(&audioEngine.soundHandles[handle]->maSound, MA_FALSE);
}
}
}
/// <summary>
/// This copies a sound, plays it, and automatically closes the copy.
/// </summary>
/// <param name="handle">A sound handle to copy</param>
/// <param name="volume">The volume at which the sound should be played (0.0 - 1.0)</param>
/// <param name="passed">How many parameters were passed?</param>
void sub__sndplaycopy(int32_t src_handle, double volume, int32_t passed) {
// We are simply going to use sndcopy, then setup some stuff like volume and autokill and then use sndplay
// We are not checking if the audio engine was initialized because if not we'll get an invalid handle anyway
int32_t dst_handle = func__sndcopy(src_handle);
// Check if we succeeded and then proceed
if (dst_handle > 0) {
// Set the volume if requested
if (passed)
ma_sound_set_volume(&audioEngine.soundHandles[dst_handle]->maSound, volume);
sub__sndplay(dst_handle); // Play the sound
audioEngine.soundHandles[dst_handle]->autoKill = true; // Set to auto kill
}
}
/// <summary>
/// This is a "fire and forget" style of function.
/// The engine will manage the sound handle internally.
/// When the sound finishes playing, the handle will be put up for recycling.
/// Playback starts asynchronously.
/// </summary>
/// <param name="fileName">The is the name of the file to be played</param>
/// <param name="sync">This paramater is ignored</param>
/// <param name="volume">This the sound playback volume (0 - silent ... 1 - full)</param>
/// <param name="passed">How many parameters were passed?</param>
void sub__sndplayfile(qbs *fileName, int32_t sync, double volume, int32_t passed) {
// We need this to send requirements to SndOpen
static qbs *reqs = nullptr;
if (!reqs) {
// Since this never changes, we can get away by doing this just once
reqs = qbs_new(0, 0);
qbs_set(reqs, qbs_new_txt(REQUIREMENT_STRING_STREAM));
}
// We will not wrap this in a 'if initialized' block because SndOpen will take care of that
int32_t handle = func__sndopen(fileName, reqs, 1);
if (handle > 0) {
if (passed & 2)
ma_sound_set_volume(&audioEngine.soundHandles[handle]->maSound, volume);
sub__sndplay(handle); // Play the sound
audioEngine.soundHandles[handle]->autoKill = true; // Set to auto kill
}
}
/// <summary>
/// This pauses a sound using a sound handle.
/// </summary>
/// <param name="handle">A sound handle</param>
void sub__sndpause(int32_t handle) {
if (audioEngine.isInitialized && IS_SOUND_HANDLE_VALID(handle) && audioEngine.soundHandles[handle]->type == SoundType::Static) {
// Stop the sound and just leave it at that
// miniaudio does not reset the play cursor
audioEngine.maResult = ma_sound_stop(&audioEngine.soundHandles[handle]->maSound);
AUDIO_DEBUG_CHECK(audioEngine.maResult == MA_SUCCESS);
}
}
/// <summary>
/// This returns whether a sound is being played.
/// </summary>
/// <param name="handle">A sound handle</param>
/// <returns>Return true if the sound is playing. False otherwise</returns>
int32_t func__sndplaying(int32_t handle) {
if (audioEngine.isInitialized && IS_SOUND_HANDLE_VALID(handle) && audioEngine.soundHandles[handle]->type == SoundType::Static) {
return ma_sound_is_playing(&audioEngine.soundHandles[handle]->maSound) ? QB_TRUE : QB_FALSE;
}
return QB_FALSE;
}
/// <summary>
/// This checks if a sound is paused.
/// </summary>
/// <param name="handle">A sound handle</param>
/// <returns>Returns true if the sound is paused. False otherwise</returns>
int32_t func__sndpaused(int32_t handle) {
if (audioEngine.isInitialized && IS_SOUND_HANDLE_VALID(handle) && audioEngine.soundHandles[handle]->type == SoundType::Static) {
return !ma_sound_is_playing(&audioEngine.soundHandles[handle]->maSound) &&
(ma_sound_is_looping(&audioEngine.soundHandles[handle]->maSound) || !ma_sound_at_end(&audioEngine.soundHandles[handle]->maSound))
? QB_TRUE
: QB_FALSE;
}
return QB_FALSE;
}
/// <summary>
/// This sets the volume of a sound loaded in memory using a sound handle.
/// New: This works for both static and raw sounds.
/// </summary>
/// <param name="handle">A sound handle</param>
/// <param name="volume">A float point value with 0 resulting in silence and anything above 1 resulting in amplification</param>
void sub__sndvol(int32_t handle, float volume) {
if (audioEngine.isInitialized && IS_SOUND_HANDLE_VALID(handle) &&
(audioEngine.soundHandles[handle]->type == SoundType::Static || audioEngine.soundHandles[handle]->type == SoundType::Raw)) {
ma_sound_set_volume(&audioEngine.soundHandles[handle]->maSound, volume);
}
}
/// <summary>
/// This is like sub__sndplay but the sound is looped.
/// </summary>
/// <param name="handle"></param>
void sub__sndloop(int32_t handle) {
if (audioEngine.isInitialized && IS_SOUND_HANDLE_VALID(handle) && audioEngine.soundHandles[handle]->type == SoundType::Static) {
// Reset position to zero only if we are playing and (not looping or we've reached the end of the sound)
// This is based on the old OpenAl-soft code behavior
if (ma_sound_is_playing(&audioEngine.soundHandles[handle]->maSound) &&
(!ma_sound_is_looping(&audioEngine.soundHandles[handle]->maSound) || ma_sound_at_end(&audioEngine.soundHandles[handle]->maSound))) {
audioEngine.maResult = ma_sound_seek_to_pcm_frame(&audioEngine.soundHandles[handle]->maSound, 0);
AUDIO_DEBUG_CHECK(audioEngine.maResult == MA_SUCCESS);
}
// Kickstart playback
audioEngine.maResult = ma_sound_start(&audioEngine.soundHandles[handle]->maSound);
AUDIO_DEBUG_CHECK(audioEngine.maResult == MA_SUCCESS);
// Start looping the sound if it is not
if (!ma_sound_is_looping(&audioEngine.soundHandles[handle]->maSound)) {
ma_sound_set_looping(&audioEngine.soundHandles[handle]->maSound, MA_TRUE);
}
}
}
/// <summary>
/// This will attempt to set the balance or 3D position of a sound.
/// Note that unlike the OpenAL code, we will do pure stereo panning if y & z are absent.
/// New: This works for both static and raw sounds.
/// </summary>
/// <param name="handle">A sound handle</param>
/// <param name="x">x distance values go from left (negative) to right (positive)</param>
/// <param name="y">y distance values go from below (negative) to above (positive).</param>
/// <param name="z">z distance values go from behind (negative) to in front (positive).</param>
/// <param name="channel">channel value 1 denotes left (mono) and 2 denotes right (stereo) channel. This has no meaning for miniaudio and is ignored</param>
/// <param name="passed">How many parameters were passed?</param>
void sub__sndbal(int32_t handle, double x, double y, double z, int32_t channel, int32_t passed) {
if (audioEngine.isInitialized && IS_SOUND_HANDLE_VALID(handle) &&
(audioEngine.soundHandles[handle]->type == SoundType::Static || audioEngine.soundHandles[handle]->type == SoundType::Raw)) {
if (passed & 2 || passed & 4) { // If y or z or both are passed
ma_sound_set_spatialization_enabled(&audioEngine.soundHandles[handle]->maSound, MA_TRUE); // Enable 3D spatialization
ma_vec3f v = ma_sound_get_position(&audioEngine.soundHandles[handle]->maSound); // Get the current position in 3D space
// Set the previous values of x, y, z if these were not passed
if (!(passed & 1))
x = v.x;
if (!(passed & 2))
y = v.y;
if (!(passed & 4))
z = v.z;
ma_sound_set_position(&audioEngine.soundHandles[handle]->maSound, x, y, z); // Use full 3D positioning
} else {
ma_sound_set_spatialization_enabled(&audioEngine.soundHandles[handle]->maSound, MA_FALSE); // Disable spatialization for better stereo sound
ma_sound_set_pan_mode(&audioEngine.soundHandles[handle]->maSound, ma_pan_mode_pan); // Set true panning
ma_sound_set_pan(&audioEngine.soundHandles[handle]->maSound, x); // Just use stereo panning
}
}
}
/// <summary>
/// This returns the length in seconds of a loaded sound using a sound handle.
/// </summary>
/// <param name="handle">A sound handle</param>
/// <returns>Returns the length of a sound in seconds</returns>
double func__sndlen(int32_t handle) {
if (audioEngine.isInitialized && IS_SOUND_HANDLE_VALID(handle) && audioEngine.soundHandles[handle]->type == SoundType::Static) {
float lengthSeconds = 0;
audioEngine.maResult = ma_sound_get_length_in_seconds(&audioEngine.soundHandles[handle]->maSound, &lengthSeconds);
AUDIO_DEBUG_CHECK(audioEngine.maResult == MA_SUCCESS);
return lengthSeconds;
}
return 0;
}
/// <summary>
/// This returns the current playing position in seconds using a sound handle.
/// </summary>
/// <param name="handle">A sound handle</param>
/// <returns>Returns the current playing position in seconds from an open sound file</returns>
double func__sndgetpos(int32_t handle) {
if (audioEngine.isInitialized && IS_SOUND_HANDLE_VALID(handle) && audioEngine.soundHandles[handle]->type == SoundType::Static) {
float playCursorSeconds = 0;
audioEngine.maResult = ma_sound_get_cursor_in_seconds(&audioEngine.soundHandles[handle]->maSound, &playCursorSeconds);
AUDIO_DEBUG_CHECK(audioEngine.maResult == MA_SUCCESS);
return playCursorSeconds;
}
return 0;
}
/// <summary>
/// This changes the current/starting playing position in seconds of a sound.
/// </summary>
/// <param name="handle">A sound handle</param>
/// <param name="seconds">The position to set in seconds</param>
void sub__sndsetpos(int32_t handle, double seconds) {
if (audioEngine.isInitialized && IS_SOUND_HANDLE_VALID(handle) && audioEngine.soundHandles[handle]->type == SoundType::Static) {
float lengthSeconds;
audioEngine.maResult = ma_sound_get_length_in_seconds(&audioEngine.soundHandles[handle]->maSound, &lengthSeconds); // Get the length in seconds
if (audioEngine.maResult != MA_SUCCESS)
return;
if (seconds > lengthSeconds) // If position is beyond length then simply stop playback and exit
{
audioEngine.maResult = ma_sound_stop(&audioEngine.soundHandles[handle]->maSound);
AUDIO_DEBUG_CHECK(audioEngine.maResult == MA_SUCCESS);
return;
}
ma_uint64 lengthSampleFrames;
audioEngine.maResult =
ma_sound_get_length_in_pcm_frames(&audioEngine.soundHandles[handle]->maSound, &lengthSampleFrames); // Get the total sample frames
if (audioEngine.maResult != MA_SUCCESS)
return;
audioEngine.maResult = ma_sound_seek_to_pcm_frame(&audioEngine.soundHandles[handle]->maSound,
lengthSampleFrames * (seconds / lengthSeconds)); // Set the postion in PCM frames
AUDIO_DEBUG_CHECK(audioEngine.maResult == MA_SUCCESS);
}
}
/// <summary>
/// This stops playing a sound after it has been playing for a set number of seconds.
/// </summary>
/// <param name="handle">A sound handle</param>
/// <param name="limit">The number of seconds that the sound will play</param>
void sub__sndlimit(int32_t handle, double limit) {
if (audioEngine.isInitialized && IS_SOUND_HANDLE_VALID(handle) && audioEngine.soundHandles[handle]->type == SoundType::Static) {
ma_sound_set_stop_time_in_milliseconds(&audioEngine.soundHandles[handle]->maSound, limit * 1000);
}
}
/// <summary>
/// This stops a playing or paused sound using a sound handle.
/// </summary>
/// <param name="handle">A sound handle</param>
void sub__sndstop(int32_t handle) {
if (audioEngine.isInitialized && IS_SOUND_HANDLE_VALID(handle) && audioEngine.soundHandles[handle]->type == SoundType::Static) {
// Stop the sound first
audioEngine.maResult = ma_sound_stop(&audioEngine.soundHandles[handle]->maSound);
AUDIO_DEBUG_CHECK(audioEngine.maResult == MA_SUCCESS);
// Also reset the playback cursor to zero
audioEngine.maResult = ma_sound_seek_to_pcm_frame(&audioEngine.soundHandles[handle]->maSound, 0);
AUDIO_DEBUG_CHECK(audioEngine.maResult == MA_SUCCESS);
}
}
/// <summary>
/// This function opens a new channel to fill with _SNDRAW content to manage multiple dynamically generated sounds.
/// </summary>
/// <returns>A new sound handle if successful or 0 on failure</returns>
int32_t func__sndopenraw() {
// Return invalid handle if audio engine is not initialized
if (!audioEngine.isInitialized)
return INVALID_SOUND_HANDLE;
// Alocate a sound handle
int32_t handle = audioEngine.AllocateSoundHandle();
if (handle < 1)
return INVALID_SOUND_HANDLE;
// Set some handle properties
audioEngine.soundHandles[handle]->type = SoundType::Raw;
// Create the raw sound object
audioEngine.soundHandles[handle]->rawQueue = new SampleFrameBlockQueue(&audioEngine.maEngine, &audioEngine.soundHandles[handle]->maSound);
if (!audioEngine.soundHandles[handle]->rawQueue)
return INVALID_SOUND_HANDLE;
// Check if everything was setup correctly
if (!audioEngine.soundHandles[handle]->rawQueue->IsSetupValid()) {
delete audioEngine.soundHandles[handle]->rawQueue;
audioEngine.soundHandles[handle]->rawQueue = nullptr;
return INVALID_SOUND_HANDLE;
}
return handle;
}
/// <summary>
/// This plays sound wave sample frequencies created by a program.
/// </summary>
/// <param name="left">Left channel sample</param>
/// <param name="right">Right channel sample</param>
/// <param name="handle">A sound handle</param>
/// <param name="passed">How many parameters were passed?</param>
void sub__sndraw(float left, float right, int32_t handle, int32_t passed) {
// Use the default raw handle if handle was not passed
if (!(passed & 2)) {
// Check if the default handle was created
if (audioEngine.sndInternalRaw < 1) {
audioEngine.sndInternalRaw = func__sndopenraw();
}
handle = audioEngine.sndInternalRaw;
}
if (audioEngine.isInitialized && IS_SOUND_HANDLE_VALID(handle) && audioEngine.soundHandles[handle]->type == SoundType::Raw) {
if (!(passed & 1))
right = left;
audioEngine.soundHandles[handle]->rawQueue->PushSampleFrame(left, right);
}
}
/// <summary>
/// This ensures that the final buffer portion is played in short sound effects even if it is incomplete.
/// </summary>
/// <param name="handle">A sound handle</param>
/// <param name="passed">How many parameters were passed?</param>
void sub__sndrawdone(int32_t handle, int32_t passed) {
// Use the default raw handle if handle was not passed
if (!passed)
handle = audioEngine.sndInternalRaw;
if (audioEngine.isInitialized && IS_SOUND_HANDLE_VALID(handle) && audioEngine.soundHandles[handle]->type == SoundType::Raw) {
// Set the last block's force flag to true
if (audioEngine.soundHandles[handle]->rawQueue->last) {
audioEngine.soundHandles[handle]->rawQueue->last->force = true;
}
}
}
/// <summary>
/// This function returns the length, in seconds, of a _SNDRAW sound currently queued.
/// </summary>
/// <param name="handle">A sound handle</param>
/// <param name="passed">How many parameters were passed?</param>
/// <returns></returns>
double func__sndrawlen(int32_t handle, int32_t passed) {
// Use the default raw handle if handle was not passed
if (!passed)
handle = audioEngine.sndInternalRaw;
if (audioEngine.isInitialized && IS_SOUND_HANDLE_VALID(handle) && audioEngine.soundHandles[handle]->type == SoundType::Raw) {
// This is for mainitianing compatibility with the SndRaw examples in the wiki
// Ideally, we should use _SNDRAWDONE at least once before checking for _SNDRAWLEN in a loop
// However, none of the examples in the wiki seem to do that
// So, we'll set the last blocks force flag to true only when there are > 1 block
// This should help avoid those examples from locking up in an infinite loop
if (audioEngine.soundHandles[handle]->rawQueue->blockCount > 1)
audioEngine.soundHandles[handle]->rawQueue->last->force = true;
return audioEngine.soundHandles[handle]->rawQueue->GetTimeRemaining();
}
return 0;
}
/// <summary>
/// This function returns a _MEM value referring to a sound's raw data in memory using a designated sound handle created by the _SNDOPEN function.
/// miniaudio supports a variety of sample and channel formats. Translating all of that to basic 2 channel 16-bit format that
/// MemSound was originally supporting would require significant overhead both in terms of system resources and code.
/// For now we are just exposing the underlying PCM data directly from miniaudio. This fits rather well using the existing mem structure.
/// Mono sounds should continue to work just as it was before. Stereo and multi-channel sounds however will be required to be handled correctly
/// by the user by checking the 'elementsize' (for frame size in bytes) and 'type' (for data type) members.
/// </summary>
/// <param name="handle">A sound handle</param>
/// <param name="targetChannel">This should be 0 (for interleaved) or 1 (for mono). Anything else will result in failure</param>
/// <returns>A _MEM value that can be used to access the sound data</returns>
mem_block func__memsound(int32_t handle, int32_t targetChannel) {
static mem_block mb;
static ma_format maFormat;
static ma_uint32 channels;
static ma_uint64 sampleFrames;
static ma_resource_manager_data_buffer *ds;
// The sound cannot be steaming and must be completely decoded in memory
if (!audioEngine.isInitialized || !IS_SOUND_HANDLE_VALID(handle) || audioEngine.soundHandles[handle]->type != SoundType::Static ||
audioEngine.soundHandles[handle]->maFlags & MA_SOUND_FLAG_STREAM || !(audioEngine.soundHandles[handle]->maFlags & MA_SOUND_FLAG_DECODE))
goto error;
// Get the pointer to the data source
ds = (ma_resource_manager_data_buffer *)ma_sound_get_data_source(&audioEngine.soundHandles[handle]->maSound);
if (!ds || !ds->pNode) {
AUDIO_DEBUG_PRINT("Data source pointer OR data source node pointer is NULL");
goto error;
}
// Check if the data is one contigious buffer or a link list of decoded pages
// We cannot have a mem object for a link list of decoded pages for obvious reasons
if (ds->pNode->data.type != ma_resource_manager_data_supply_type::ma_resource_manager_data_supply_type_decoded) {
AUDIO_DEBUG_PRINT("Data is not a contigious buffer. Type = %u", ds->pNode->data.type);
goto error;
}
// Check the data pointer
if (!ds->pNode->data.backend.decoded.pData) {
AUDIO_DEBUG_PRINT("Data source data pointer is NULL");
goto error;
}
AUDIO_DEBUG_PRINT("Data source data pointer = %p", ds->pNode->data.backend.decoded.pData);
// Query the data format
if (ma_sound_get_data_format(&audioEngine.soundHandles[handle]->maSound, &maFormat, &channels, NULL, NULL, 0) != MA_SUCCESS) {
AUDIO_DEBUG_PRINT("Data format query failed");
goto error;
}
// Do not proceed if invalid (unsupported) channel values were passed
if (targetChannel != 0 && targetChannel != 1) {
AUDIO_DEBUG_PRINT("Sound channels = %u, Target channel %i not supported", channels, targetChannel);
goto error;
}
// Get the length in sample frames
if (ma_sound_get_length_in_pcm_frames(&audioEngine.soundHandles[handle]->maSound, &sampleFrames) != MA_SUCCESS) {
AUDIO_DEBUG_PRINT("PCM frames query failed");
goto error;
}
AUDIO_DEBUG_PRINT("Format = %u, Channels = %u, Frames = %llu", maFormat, channels, sampleFrames);
if (audioEngine.soundHandles[handle]->memLockOffset) {
mb.lock_offset = (ptrszint)audioEngine.soundHandles[handle]->memLockOffset;
mb.lock_id = audioEngine.soundHandles[handle]->memLockId;
} else {
new_mem_lock();
mem_lock_tmp->type = MEM_TYPE_SOUND;
mb.lock_offset = (ptrszint)mem_lock_tmp;
mb.lock_id = mem_lock_id;
audioEngine.soundHandles[handle]->memLockOffset = (void *)mem_lock_tmp;
audioEngine.soundHandles[handle]->memLockId = mem_lock_id;
}
// Setup type: This was not done in the old code
// But we are doing it here. By examing the type the user can now figure out if they have to use FP32 or integers
if (maFormat == ma_format::ma_format_f32)
mb.type = 4 + 256; // FP32
else if (maFormat == ma_format::ma_format_s32)
mb.type = 4 + 128; // Int32
else if (maFormat == ma_format::ma_format_s16)
mb.type = 2 + 128; // Int16
else if (maFormat == ma_format::ma_format_u8)
mb.type = 1 + 128 + 1024; // Int8
mb.elementsize = ma_get_bytes_per_frame(maFormat, channels); // Set the element size. This is the size of each PCM frame in bytes
mb.offset = (ptrszint)ds->pNode->data.backend.decoded.pData; // Setup offset
mb.size = sampleFrames * mb.elementsize; // Setup size (in bytes)
mb.sound = handle; // Copy the handle
mb.image = 0; // Not needed. Set to 0
AUDIO_DEBUG_PRINT("ElementSize = %lli, Size = %lli, Type = %lli", mb.elementsize, mb.size, mb.type);
return mb;
error:
mb.offset = 0;
mb.size = 0;
mb.lock_offset = (ptrszint)mem_lock_base;
mb.lock_id = INVALID_MEM_LOCK;
mb.type = 0;
mb.elementsize = 0;
mb.sound = 0;
mb.image = 0;
return mb;
}
/// <summary>
/// This initializes the QBPE audio subsystem.
/// We simply attempt to initialize and then set some globals with the results.
/// </summary>
void snd_init() {
// Exit if engine is initialize or already initialization was attempted but failed
if (audioEngine.isInitialized || audioEngine.initializationFailed)
return;
// Initialize the miniaudio resource manager
audioEngine.maResourceManagerConfig = ma_resource_manager_config_init();
AudioEngineAttachCustomBackendVTables(&audioEngine.maResourceManagerConfig);
audioEngine.maResourceManagerConfig.pCustomDecodingBackendUserData = NULL; // <- pUserData parameter of each function in the decoding backend vtables
audioEngine.maResult = ma_resource_manager_init(&audioEngine.maResourceManagerConfig, &audioEngine.maResourceManager);
if (audioEngine.maResult != MA_SUCCESS) {
audioEngine.initializationFailed = true;
AUDIO_DEBUG_PRINT("Failed to initialize miniaudio resource manager");
return;
}
// Once we have a resource manager we can create the engine
audioEngine.maEngineConfig = ma_engine_config_init();
audioEngine.maEngineConfig.pResourceManager = &audioEngine.maResourceManager;
// Attempt to initialize with miniaudio defaults
audioEngine.maResult = ma_engine_init(&audioEngine.maEngineConfig, &audioEngine.maEngine);
// If failed, then set the global flag so that we don't attempt to initialize again
if (audioEngine.maResult != MA_SUCCESS) {
ma_resource_manager_uninit(&audioEngine.maResourceManager);
audioEngine.initializationFailed = true;
AUDIO_DEBUG_PRINT("miniaudio initialization failed");
return;
}
// Get and save the engine sample rate. We will let miniaudio choose the device sample rate for us
// This ensures we get the lowest latency
// Set the resource manager decorder sample rate to the device sample rate (miniaudio engine bug?)
audioEngine.maResourceManager.config.decodedSampleRate = audioEngine.sampleRate = ma_engine_get_sample_rate(&audioEngine.maEngine);
// Set the initialized flag as true
audioEngine.isInitialized = true;
AUDIO_DEBUG_PRINT("Audio engine initialized at %uHz sample rate", audioEngine.sampleRate);
// Reserve sound handle 0 so that nothing else can use it
// We will use this handle internally for Play(), Beep(), Sound() etc.
audioEngine.sndInternal = audioEngine.AllocateSoundHandle();
AUDIO_DEBUG_CHECK(audioEngine.sndInternal == 0); // The first handle must return 0 and this is what is used by Beep and Sound
// Just do a basic setup and mark the type as 'none'
// If Play(), Sound(), Beep() are called, those will mark it as 'raw'
audioEngine.soundHandles[audioEngine.sndInternal]->rawQueue =
new SampleFrameBlockQueue(&audioEngine.maEngine, &audioEngine.soundHandles[audioEngine.sndInternal]->maSound);
audioEngine.soundHandles[audioEngine.sndInternal]->type = SoundType::None;
}
/// <summary>
/// This shuts down the audio engine and frees any resources used.
/// </summary>
void snd_un_init() {
if (audioEngine.isInitialized) {
// Special handling for handle 0
audioEngine.soundHandles[audioEngine.sndInternal]->type = SoundType::None;
delete audioEngine.soundHandles[audioEngine.sndInternal]->rawQueue;
audioEngine.soundHandles[audioEngine.sndInternal]->rawQueue = nullptr;
// Free all sound handles here
for (size_t handle = 0; handle < audioEngine.soundHandles.size(); handle++) {
audioEngine.FreeSoundHandle(handle); // Let FreeSoundHandle do it's thing
delete audioEngine.soundHandles[handle]; // Now free the object created by AllocateSoundHandle()
}
// Now that all sounds are closed and SoundHandle objects are freed, clear the vector
audioEngine.soundHandles.clear();
// Invalidate internal handles
audioEngine.sndInternal = audioEngine.sndInternalRaw = INVALID_SOUND_HANDLE;
// Shutdown miniaudio
ma_engine_uninit(&audioEngine.maEngine);
// Shutdown the miniaudio resource manager
ma_resource_manager_uninit(&audioEngine.maResourceManager);
// Set engine initialized flag as false
audioEngine.isInitialized = false;
AUDIO_DEBUG_PRINT("Audio engine shutdown");
}
}
/// <summary>
/// This is called by the QBPE library code.
/// We use this for housekeeping and other stuff.
/// </summary>
void snd_mainloop() {
if (audioEngine.isInitialized) {
// Scan through the whole handle vector to find anything we need to update or close
for (size_t handle = 0; handle < audioEngine.soundHandles.size(); handle++) {
// Only process handles that are in use
if (audioEngine.soundHandles[handle]->isUsed) {
// Keep raw audio streams going
if (audioEngine.soundHandles[handle]->type == SoundType::Raw)
audioEngine.soundHandles[handle]->rawQueue->Update();
// Look for stuff that is set to auto-destruct
if (audioEngine.soundHandles[handle]->autoKill) {
switch (audioEngine.soundHandles[handle]->type) {
case SoundType::Static:
// Dispose the sound if it has finished playing
// Note that this means that temporary looping sounds will never close
// Well thats on the programmer. Probably they want it that way
if (!ma_sound_is_playing(&audioEngine.soundHandles[handle]->maSound))
audioEngine.FreeSoundHandle(handle);
break;
case SoundType::Raw:
// Close the raw stream if we have no more frames in the queue or playing
if (!audioEngine.soundHandles[handle]->rawQueue->GetSampleFramesRemaining())
audioEngine.FreeSoundHandle(handle);
break;
case SoundType::None:
if (handle != 0)
AUDIO_DEBUG_PRINT("Sound type is 'None' when handle value is not 0");
break;
default:
AUDIO_DEBUG_PRINT("Condition not handled"); // It should not come here
}
}
}
}
}
}
//-----------------------------------------------------------------------------------------------------
//-----------------------------------------------------------------------------------------------------
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