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Merge branch 'dev' into dev-0.12

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This commit is contained in:
David Reid
2024-02-25 09:37:21 +10:00
parent d27c3e03f1 f4e5cf99dc
commit f01ce432be
17 changed files with 120 additions and 76 deletions
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miniaudio.h
+67 -25
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@@ -11306,12 +11306,12 @@ MA_API ma_engine_config ma_engine_config_init(void);
struct ma_engine
{
ma_node_graph nodeGraph; /* An engine is a node graph. It should be able to be plugged into any ma_node_graph API (with a cast) which means this must be the first member of this struct. */
ma_node_graph nodeGraph; /* An engine is a node graph. It should be able to be plugged into any ma_node_graph API (with a cast) which means this must be the first member of this struct. */
#if !defined(MA_NO_RESOURCE_MANAGER)
ma_resource_manager* pResourceManager;
#endif
#if !defined(MA_NO_DEVICE_IO)
ma_device* pDevice; /* Optionally set via the config, otherwise allocated by the engine in ma_engine_init(). */
ma_device* pDevice; /* Optionally set via the config, otherwise allocated by the engine in ma_engine_init(). */
#endif
ma_log* pLog;
ma_uint32 sampleRate;
@@ -11320,10 +11320,10 @@ struct ma_engine
ma_allocation_callbacks allocationCallbacks;
ma_bool8 ownsResourceManager;
ma_bool8 ownsDevice;
ma_spinlock inlinedSoundLock; /* For synchronizing access so the inlined sound list. */
ma_sound_inlined* pInlinedSoundHead; /* The first inlined sound. Inlined sounds are tracked in a linked list. */
MA_ATOMIC(4, ma_uint32) inlinedSoundCount; /* The total number of allocated inlined sound objects. Used for debugging. */
ma_uint32 gainSmoothTimeInFrames; /* The number of frames to interpolate the gain of spatialized sounds across. */
ma_spinlock inlinedSoundLock; /* For synchronizing access so the inlined sound list. */
ma_sound_inlined* pInlinedSoundHead; /* The first inlined sound. Inlined sounds are tracked in a linked list. */
MA_ATOMIC(4, ma_uint32) inlinedSoundCount; /* The total number of allocated inlined sound objects. Used for debugging. */
ma_uint32 gainSmoothTimeInFrames; /* The number of frames to interpolate the gain of spatialized sounds across. */
ma_uint32 defaultVolumeSmoothTimeInPCMFrames;
ma_mono_expansion_mode monoExpansionMode;
ma_engine_process_proc onProcess;
@@ -16201,19 +16201,34 @@ static ma_result ma_thread_create__posix(ma_thread* pThread, ma_thread_priority
if (priority == ma_thread_priority_idle) {
sched.sched_priority = priorityMin;
} else if (priority == ma_thread_priority_realtime) {
sched.sched_priority = priorityMax;
} else {
sched.sched_priority += ((int)priority + 5) * priorityStep; /* +5 because the lowest priority is -5. */
if (sched.sched_priority < priorityMin) {
sched.sched_priority = priorityMin;
#if defined(MA_PTHREAD_REALTIME_THREAD_PRIORITY)
{
sched.sched_priority = MA_PTHREAD_REALTIME_THREAD_PRIORITY;
}
if (sched.sched_priority > priorityMax) {
#else
{
sched.sched_priority = priorityMax;
}
#endif
} else {
sched.sched_priority += ((int)priority + 5) * priorityStep; /* +5 because the lowest priority is -5. */
}
/* I'm not treating a failure of setting the priority as a critical error so not checking the return value here. */
pthread_attr_setschedparam(&attr, &sched);
if (sched.sched_priority < priorityMin) {
sched.sched_priority = priorityMin;
}
if (sched.sched_priority > priorityMax) {
sched.sched_priority = priorityMax;
}
/* I'm not treating a failure of setting the priority as a critical error so not aborting on failure here. */
if (pthread_attr_setschedparam(&attr, &sched) == 0) {
#if !defined(MA_ANDROID) || (defined(__ANDROID_API__) && __ANDROID_API__ >= 28)
{
pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED);
}
#endif
}
}
}
}
@@ -17998,6 +18013,10 @@ DEVICE I/O
#endif
#endif
/* This must be set to at least 26. */
#ifndef MA_AAUDIO_MIN_ANDROID_SDK_VERSION
#define MA_AAUDIO_MIN_ANDROID_SDK_VERSION 27
#endif
@@ -18354,7 +18373,7 @@ MA_API ma_bool32 ma_is_backend_enabled(ma_backend backend)
#if defined(MA_HAS_AAUDIO)
#if defined(MA_ANDROID)
{
return ma_android_sdk_version() >= 26;
return ma_android_sdk_version() >= MA_AAUDIO_MIN_ANDROID_SDK_VERSION;
}
#else
return MA_FALSE;
@@ -19071,7 +19090,7 @@ static void ma_device__handle_data_callback(ma_device* pDevice, void* pFramesOut
unsigned int prevDenormalState = ma_device_disable_denormals(pDevice);
{
/* Volume control of input makes things a bit awkward because the input buffer is read-only. We'll need to use a temp buffer and loop in this case. */
if (pFramesIn != NULL && masterVolumeFactor < 1) {
if (pFramesIn != NULL && masterVolumeFactor != 1) {
ma_uint8 tempFramesIn[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
ma_uint32 bpfCapture = ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels);
ma_uint32 bpfPlayback = ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels);
@@ -19094,7 +19113,7 @@ static void ma_device__handle_data_callback(ma_device* pDevice, void* pFramesOut
/* Volume control and clipping for playback devices. */
if (pFramesOut != NULL) {
if (masterVolumeFactor < 1) {
if (masterVolumeFactor != 1) {
if (pFramesIn == NULL) { /* <-- In full-duplex situations, the volume will have been applied to the input samples before the data callback. Applying it again post-callback will incorrectly compound it. */
ma_apply_volume_factor_pcm_frames(pFramesOut, frameCount, pDevice->playback.format, pDevice->playback.channels, masterVolumeFactor);
}
@@ -23922,6 +23941,13 @@ DirectSound Backend
#define MA_DSBPLAY_TERMINATEBY_DISTANCE 0x00000010
#define MA_DSBPLAY_TERMINATEBY_PRIORITY 0x00000020
#define MA_DSBSTATUS_PLAYING 0x00000001
#define MA_DSBSTATUS_BUFFERLOST 0x00000002
#define MA_DSBSTATUS_LOOPING 0x00000004
#define MA_DSBSTATUS_LOCHARDWARE 0x00000008
#define MA_DSBSTATUS_LOCSOFTWARE 0x00000010
#define MA_DSBSTATUS_TERMINATED 0x00000020
#define MA_DSCBSTART_LOOPING 0x00000001
typedef struct
@@ -25084,6 +25110,7 @@ static ma_result ma_device_data_loop__dsound(ma_device* pDevice)
ma_bool32 isPlaybackDeviceStarted = MA_FALSE;
ma_uint32 framesWrittenToPlaybackDevice = 0; /* For knowing whether or not the playback device needs to be started. */
ma_uint32 waitTimeInMilliseconds = 1;
DWORD playbackBufferStatus = 0;
MA_ASSERT(pDevice != NULL);
@@ -25412,6 +25439,20 @@ static ma_result ma_device_data_loop__dsound(ma_device* pDevice)
break;
}
hr = ma_IDirectSoundBuffer_GetStatus((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, &playbackBufferStatus);
if (SUCCEEDED(hr) && (playbackBufferStatus & MA_DSBSTATUS_PLAYING) == 0 && isPlaybackDeviceStarted) {
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[DirectSound] Attempting to resume audio due to state: %d.", (int)playbackBufferStatus);
hr = ma_IDirectSoundBuffer_Play((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, 0, 0, MA_DSBPLAY_LOOPING);
if (FAILED(hr)) {
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundBuffer_Play() failed after attempting to resume from state %d.", (int)playbackBufferStatus);
return ma_result_from_HRESULT(hr);
}
isPlaybackDeviceStarted = MA_TRUE;
ma_sleep(waitTimeInMilliseconds);
continue;
}
if (physicalPlayCursorInBytes < prevPlayCursorInBytesPlayback) {
physicalPlayCursorLoopFlagPlayback = !physicalPlayCursorLoopFlagPlayback;
}
@@ -39894,7 +39935,7 @@ TODO: Version 0.12: Swap this logic around so that AudioWorklets are used by def
/* The thread stack size must be a multiple of 16. */
#ifndef MA_AUDIO_WORKLETS_THREAD_STACK_SIZE
#define MA_AUDIO_WORKLETS_THREAD_STACK_SIZE 16384
#define MA_AUDIO_WORKLETS_THREAD_STACK_SIZE 131072
#endif
#if defined(MA_USE_AUDIO_WORKLETS)
@@ -40637,6 +40678,10 @@ static ma_result ma_context_uninit__webaudio(ma_context* pContext)
/* Remove the global miniaudio object from window if there are no more references to it. */
EM_ASM({
if (typeof(window.miniaudio) !== 'undefined') {
miniaudio.unlock_event_types.map(function(event_type) {
document.removeEventListener(event_type, miniaudio.unlock, true);
});
window.miniaudio.referenceCount -= 1;
if (window.miniaudio.referenceCount === 0) {
delete window.miniaudio;
@@ -59336,7 +59381,7 @@ static ma_result ma_default_vfs_seek__win32(ma_vfs* pVFS, ma_vfs_file file, ma_i
result = ma_SetFilePointerEx((HANDLE)file, liDistanceToMove, NULL, dwMoveMethod);
} else if (ma_SetFilePointer != NULL) {
/* No SetFilePointerEx() so restrict to 31 bits. */
if (origin > 0x7FFFFFFF) {
if (offset > 0x7FFFFFFF) {
return MA_OUT_OF_RANGE;
}
@@ -59939,7 +59984,7 @@ extern "C" {
#define MA_DR_WAV_XSTRINGIFY(x) MA_DR_WAV_STRINGIFY(x)
#define MA_DR_WAV_VERSION_MAJOR 0
#define MA_DR_WAV_VERSION_MINOR 13
#define MA_DR_WAV_VERSION_REVISION 13
#define MA_DR_WAV_VERSION_REVISION 14
#define MA_DR_WAV_VERSION_STRING MA_DR_WAV_XSTRINGIFY(MA_DR_WAV_VERSION_MAJOR) "." MA_DR_WAV_XSTRINGIFY(MA_DR_WAV_VERSION_MINOR) "." MA_DR_WAV_XSTRINGIFY(MA_DR_WAV_VERSION_REVISION)
#include <stddef.h>
#define MA_DR_WAVE_FORMAT_PCM 0x1
@@ -72457,7 +72502,7 @@ static ma_result ma_node_detach_full(ma_node* pNode)
linked list logic. We don't need to worry about the audio thread referencing these because the step
above severed the connection to the graph.
*/
for (pOutputBus = (ma_node_output_bus*)ma_atomic_load_ptr(&pInputBus->head.pNext); pOutputBus != NULL; pOutputBus = (ma_node_output_bus*)ma_atomic_load_ptr(&pOutputBus->pNext)) {
for (pOutputBus = (ma_node_output_bus*)ma_atomic_load_ptr(&pInputBus->head.pNext); pOutputBus != NULL; pOutputBus = (ma_node_output_bus*)ma_atomic_load_ptr(&pInputBus->head.pNext)) {
ma_node_detach_output_bus(pOutputBus->pNode, pOutputBus->outputBusIndex); /* This won't do any waiting in practice and should be efficient. */
}
}
@@ -78720,7 +78765,6 @@ MA_PRIVATE ma_bool32 ma_dr_wav_init__internal(ma_dr_wav* pWav, ma_dr_wav_chunk_p
}
if (pWav->container == ma_dr_wav_container_riff || pWav->container == ma_dr_wav_container_rifx) {
if (ma_dr_wav_bytes_to_u32_ex(chunkSizeBytes, pWav->container) < 36) {
return MA_FALSE;
}
} else if (pWav->container == ma_dr_wav_container_rf64) {
if (ma_dr_wav_bytes_to_u32_le(chunkSizeBytes) != 0xFFFFFFFF) {
@@ -79047,9 +79091,7 @@ MA_PRIVATE ma_bool32 ma_dr_wav_init__internal(ma_dr_wav* pWav, ma_dr_wav_chunk_p
}
}
if (isProcessingMetadata) {
ma_uint64 metadataBytesRead;
metadataBytesRead = ma_dr_wav__metadata_process_chunk(&metadataParser, &header, ma_dr_wav_metadata_type_all_including_unknown);
MA_DR_WAV_ASSERT(metadataBytesRead <= header.sizeInBytes);
ma_dr_wav__metadata_process_chunk(&metadataParser, &header, ma_dr_wav_metadata_type_all_including_unknown);
if (ma_dr_wav__seek_from_start(pWav->onSeek, cursor, pWav->pUserData) == MA_FALSE) {
break;
}