661 lines
28 KiB
Python
661 lines
28 KiB
Python
import ctypes
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import platform
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import os
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from pyglet.libs.win32.constants import *
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from pyglet.libs.win32.types import *
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from pyglet.libs.win32 import com
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from pyglet.util import debug_print
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_debug = debug_print('debug_media')
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def load_xaudio2(dll_name):
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"""This will attempt to load a version of XAudio2. Versions supported: 2.9, 2.8.
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While Windows 8 ships with 2.8 and Windows 10 ships with version 2.9, it is possible to install 2.9 on 8/8.1.
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"""
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xaudio2 = dll_name
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# System32 and SysWOW64 folders are opposite perception in Windows x64.
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# System32 = x64 dll's | SysWOW64 = x86 dlls
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# By default ctypes only seems to look in system32 regardless of Python architecture, which has x64 dlls.
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if platform.architecture()[0] == '32bit':
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if platform.machine().endswith('64'): # Machine is 64 bit, Python is 32 bit.
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xaudio2 = os.path.join(os.environ['WINDIR'], 'SysWOW64', '{}.dll'.format(xaudio2))
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xaudio2_lib = ctypes.windll.LoadLibrary(xaudio2)
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# Somehow x3d uses different calling structure than the rest of the DLL; Only affects 32 bit? Microsoft...
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x3d_lib = ctypes.cdll.LoadLibrary(xaudio2)
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return xaudio2_lib, x3d_lib
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try:
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xaudio2_lib, x3d_lib = load_xaudio2("xaudio2_9")
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except OSError:
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_debug("Could not load XAudio2.9 library")
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try:
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xaudio2_lib, x3d_lib = load_xaudio2("xaudio2_8")
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except OSError:
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_debug("Could not load XAudio2.8 library")
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raise ImportError('Could not locate a supported XAudio2 library.')
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UINT32 = c_uint32
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FLOAT32 = c_float
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class XAUDIO2_DEBUG_CONFIGURATION(ctypes.Structure):
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_fields_ = [
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('TraceMask', UINT32),
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('BreakMask', UINT32),
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('LogThreadID', BOOL),
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('LogFileline', BOOL),
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('LogFunctionName', BOOL),
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('LogTiming', BOOL),
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]
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class XAUDIO2_PERFORMANCE_DATA(ctypes.Structure):
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_fields_ = [
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('AudioCyclesSinceLastQuery', c_uint64),
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('TotalCyclesSinceLastQuery', c_uint64),
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('MinimumCyclesPerQuantum', UINT32),
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('MaximumCyclesPerQuantum', UINT32),
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('MemoryUsageInBytes', UINT32),
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('CurrentLatencyInSamples', UINT32),
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('GlitchesSinceEngineStarted', UINT32),
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('ActiveSourceVoiceCount', UINT32),
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('TotalSourceVoiceCount', UINT32),
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('ActiveSubmixVoiceCount', UINT32),
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('ActiveResamplerCount', UINT32),
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('ActiveMatrixMixCount', UINT32),
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('ActiveXmaSourceVoices', UINT32),
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('ActiveXmaStreams', UINT32),
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]
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def __repr__(self):
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return "XAUDIO2PerformanceData(active_voices={}, total_voices={}, glitches={}, latency={} samples, memory_usage={} bytes)".format(self.ActiveSourceVoiceCount, self.TotalSourceVoiceCount, self.GlitchesSinceEngineStarted, self.CurrentLatencyInSamples, self.MemoryUsageInBytes)
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class XAUDIO2_VOICE_SENDS(ctypes.Structure):
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_fields_ = [
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('SendCount', UINT32),
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('pSends', c_void_p),
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]
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class XAUDIO2_BUFFER(ctypes.Structure):
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_fields_ = [
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('Flags', UINT32),
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('AudioBytes', UINT32),
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('pAudioData', POINTER(c_char)),
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('PlayBegin', UINT32),
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('PlayLength', UINT32),
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('LoopBegin', UINT32),
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('LoopLength', UINT32),
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('LoopCount', UINT32),
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('pContext', c_void_p),
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]
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class XAUDIO2_VOICE_STATE(ctypes.Structure):
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_fields_ = [
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('pCurrentBufferContext', c_void_p),
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('BuffersQueued', UINT32),
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('SamplesPlayed', UINT32)
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]
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def __repr__(self):
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return "XAUDIO2_VOICE_STATE(BuffersQueued={0}, SamplesPlayed={1})".format(self.BuffersQueued, self.SamplesPlayed)
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class WAVEFORMATEX(ctypes.Structure):
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_fields_ = [
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('wFormatTag', WORD),
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('nChannels', WORD),
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('nSamplesPerSec', DWORD),
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('nAvgBytesPerSec', DWORD),
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('nBlockAlign', WORD),
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('wBitsPerSample', WORD),
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('cbSize', WORD),
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]
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def __repr__(self):
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return 'WAVEFORMATEX(wFormatTag={}, nChannels={}, nSamplesPerSec={}, nAvgBytesPersec={}' \
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', nBlockAlign={}, wBitsPerSample={}, cbSize={})'.format(
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self.wFormatTag, self.nChannels, self.nSamplesPerSec,
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self.nAvgBytesPerSec, self.nBlockAlign, self.wBitsPerSample,
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self.cbSize)
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XAUDIO2_USE_DEFAULT_PROCESSOR = 0x00000000 # Win 10+
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if WINDOWS_10_ANNIVERSARY_UPDATE_OR_GREATER:
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XAUDIO2_DEFAULT_PROCESSOR = XAUDIO2_USE_DEFAULT_PROCESSOR
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else:
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XAUDIO2_DEFAULT_PROCESSOR = 0x00000001 # Windows 8/8.1
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XAUDIO2_LOG_ERRORS = 0x0001 # For handled errors with serious effects.
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XAUDIO2_LOG_WARNINGS = 0x0002 # For handled errors that may be recoverable.
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XAUDIO2_LOG_INFO = 0x0004 # Informational chit-chat (e.g. state changes).
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XAUDIO2_LOG_DETAIL = 0x0008 # More detailed chit-chat.
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XAUDIO2_LOG_API_CALLS = 0x0010 # Public API function entries and exits.
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XAUDIO2_LOG_FUNC_CALLS = 0x0020 # Internal function entries and exits.
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XAUDIO2_LOG_TIMING = 0x0040 # Delays detected and other timing data.
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XAUDIO2_LOG_LOCKS = 0x0080 # Usage of critical sections and mutexes.
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XAUDIO2_LOG_MEMORY = 0x0100 # Memory heap usage information.
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XAUDIO2_LOG_STREAMING = 0x1000 # Audio streaming information.
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# Some XAUDIO2 global settings, most not used, but useful information
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XAUDIO2_MAX_BUFFER_BYTES = 0x80000000 # Maximum bytes allowed in a source buffer
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XAUDIO2_MAX_QUEUED_BUFFERS = 64 # Maximum buffers allowed in a voice queue
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XAUDIO2_MAX_BUFFERS_SYSTEM = 2 # Maximum buffers allowed for system threads (Xbox 360 only)
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XAUDIO2_MAX_AUDIO_CHANNELS = 64 # Maximum channels in an audio stream
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XAUDIO2_MIN_SAMPLE_RATE = 1000 # Minimum audio sample rate supported
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XAUDIO2_MAX_SAMPLE_RATE = 200000 # Maximum audio sample rate supported
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XAUDIO2_MAX_VOLUME_LEVEL = 16777216.0 # Maximum acceptable volume level (2^24)
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XAUDIO2_MIN_FREQ_RATIO = (1/1024.0) # Minimum SetFrequencyRatio argument
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XAUDIO2_MAX_FREQ_RATIO = 1024.0 # Maximum MaxFrequencyRatio argument
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XAUDIO2_DEFAULT_FREQ_RATIO = 2.0 # Default MaxFrequencyRatio argument
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XAUDIO2_MAX_FILTER_ONEOVERQ = 1.5 # Maximum XAUDIO2_FILTER_PARAMETERS.OneOverQ
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XAUDIO2_MAX_FILTER_FREQUENCY = 1.0 # Maximum XAUDIO2_FILTER_PARAMETERS.Frequency
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XAUDIO2_MAX_LOOP_COUNT = 254 # Maximum non-infinite XAUDIO2_BUFFER.LoopCount
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XAUDIO2_MAX_INSTANCES = 8 # Maximum simultaneous XAudio2 objects on Xbox 360
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XAUDIO2_FILTER_TYPE = UINT
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LowPassFilter = 0 # Attenuates frequencies above the cutoff frequency (state-variable filter).
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BandPassFilter = 1 # Attenuates frequencies outside a given range (state-variable filter).
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HighPassFilter = 2 # Attenuates frequencies below the cutoff frequency (state-variable filter).
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NotchFilter = 3 # Attenuates frequencies inside a given range (state-variable filter).
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LowPassOnePoleFilter = 4 # Attenuates frequencies above the cutoff frequency (one-pole filter, XAUDIO2_FILTER_PARAMETERS.OneOverQ has no effect)
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HighPassOnePoleFilter = 5 # Attenuates frequencies below the cutoff frequency (one-pole filter, XAUDIO2_FILTER_PARAMETERS.OneOverQ has no effect)
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XAUDIO2_NO_LOOP_REGION = 0 # Used in XAUDIO2_BUFFER.LoopCount
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XAUDIO2_LOOP_INFINITE = 255 # Used in XAUDIO2_BUFFER.LoopCount
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XAUDIO2_DEFAULT_CHANNELS = 0 # Used in CreateMasteringVoice
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XAUDIO2_DEFAULT_SAMPLERATE = 0 # Used in CreateMasteringVoice
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WAVE_FORMAT_PCM = 1
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XAUDIO2_DEBUG_ENGINE = 0x0001 # Used in XAudio2Create
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XAUDIO2_VOICE_NOPITCH = 0x0002 # Used in IXAudio2::CreateSourceVoice
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XAUDIO2_VOICE_NOSRC = 0x0004 # Used in IXAudio2::CreateSourceVoice
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XAUDIO2_VOICE_USEFILTER = 0x0008 # Used in IXAudio2::CreateSource/SubmixVoice
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XAUDIO2_PLAY_TAILS = 0x0020 # Used in IXAudio2SourceVoice::Stop
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XAUDIO2_END_OF_STREAM = 0x0040 # Used in XAUDIO2_BUFFER.Flags
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XAUDIO2_SEND_USEFILTER = 0x0080 # Used in XAUDIO2_SEND_DESCRIPTOR.Flags
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XAUDIO2_VOICE_NOSAMPLESPLAYED = 0x0100 # Used in IXAudio2SourceVoice::GetState
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XAUDIO2_STOP_ENGINE_WHEN_IDLE = 0x2000 # Used in XAudio2Create to force the engine to Stop when no source voices are Started, and Start when a voice is Started
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XAUDIO2_1024_QUANTUM = 0x8000 # Used in XAudio2Create to specify nondefault processing quantum of 21.33 ms (1024 samples at 48KHz)
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XAUDIO2_NO_VIRTUAL_AUDIO_CLIENT = 0x10000 # Used in CreateMasteringVoice to create a virtual audio client
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class IXAudio2VoiceCallback(com.Interface):
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_methods_ = [
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('OnVoiceProcessingPassStart',
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com.VOIDMETHOD(UINT32)),
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('OnVoiceProcessingPassEnd',
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com.VOIDMETHOD()),
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('OnStreamEnd',
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com.VOIDMETHOD()),
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('OnBufferStart',
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com.VOIDMETHOD(ctypes.c_void_p)),
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('OnBufferEnd',
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com.VOIDMETHOD(ctypes.c_void_p)),
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('OnLoopEnd',
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com.VOIDMETHOD(ctypes.c_void_p)),
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('OnVoiceError',
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com.VOIDMETHOD(ctypes.c_void_p, HRESULT))
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]
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class XA2SourceCallback(com.COMObject):
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"""Callback class used to trigger when buffers or streams end..
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WARNING: Whenever a callback is running, XAudio2 cannot generate audio.
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Make sure these functions run as fast as possible and do not block/delay more than a few milliseconds.
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MS Recommendation:
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At a minimum, callback functions must not do the following:
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- Access the hard disk or other permanent storage
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- Make expensive or blocking API calls
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- Synchronize with other parts of client code
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- Require significant CPU usage
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"""
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_interfaces_ = [IXAudio2VoiceCallback]
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def __init__(self, xa2_player):
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super().__init__()
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self.xa2_player = xa2_player
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def OnBufferEnd(self, pBufferContext):
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"""At the end of playing one buffer, attempt to refill again.
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Even if the player is out of sources, it needs to be called to purge all buffers.
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"""
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if self.xa2_player:
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self.xa2_player.refill_source_player()
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def OnVoiceError(self, pBufferContext, hresult):
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raise Exception("Error occurred during audio playback.", hresult)
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class XAUDIO2_EFFECT_DESCRIPTOR(Structure):
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_fields_ = [
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('pEffect', com.pIUnknown),
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('InitialState', c_bool),
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('OutputChannels', UINT32)
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]
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class XAUDIO2_EFFECT_CHAIN(ctypes.Structure):
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_fields_ = [
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('EffectCount', UINT32),
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('pEffectDescriptors', POINTER(XAUDIO2_EFFECT_DESCRIPTOR)),
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]
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class XAUDIO2_FILTER_PARAMETERS(Structure):
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_fields_ = [
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('Type', XAUDIO2_FILTER_TYPE),
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('Frequency', FLOAT),
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('OneOverQ', FLOAT)
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]
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class XAUDIO2_VOICE_DETAILS(Structure):
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_fields_ = [
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('CreationFlags', UINT32),
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('ActiveFlags', UINT32),
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('InputChannels', UINT32),
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('InputSampleRate', UINT32)
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]
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class IXAudio2Voice(com.pInterface):
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_methods_ = [
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('GetVoiceDetails',
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com.STDMETHOD(POINTER(XAUDIO2_VOICE_DETAILS))),
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('SetOutputVoices',
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com.STDMETHOD()),
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('SetEffectChain',
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com.STDMETHOD(POINTER(XAUDIO2_EFFECT_CHAIN))),
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('EnableEffect',
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com.STDMETHOD()),
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('DisableEffect',
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com.STDMETHOD()),
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('GetEffectState',
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com.STDMETHOD()),
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('SetEffectParameters',
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com.STDMETHOD()),
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('GetEffectParameters',
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com.STDMETHOD()),
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('SetFilterParameters',
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com.STDMETHOD(POINTER(XAUDIO2_FILTER_PARAMETERS), UINT32)),
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('GetFilterParameters',
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com.STDMETHOD()),
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('SetOutputFilterParameters',
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com.STDMETHOD()),
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('GetOutputFilterParameters',
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com.STDMETHOD()),
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('SetVolume',
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com.STDMETHOD(ctypes.c_float, UINT32)),
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('GetVolume',
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com.STDMETHOD(POINTER(c_float))),
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('SetChannelVolumes',
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com.STDMETHOD()),
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('GetChannelVolumes',
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com.STDMETHOD()),
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('SetOutputMatrix',
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com.STDMETHOD(c_void_p, UINT32, UINT32, POINTER(FLOAT), UINT32)),
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('GetOutputMatrix',
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com.STDMETHOD()),
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('DestroyVoice',
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com.STDMETHOD())
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]
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class IXAudio2SubmixVoice(IXAudio2Voice):
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pass
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class IXAudio2SourceVoice(IXAudio2Voice):
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_methods_ = [
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('Start',
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com.STDMETHOD(UINT32, UINT32)),
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('Stop',
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com.STDMETHOD(UINT32, UINT32)),
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('SubmitSourceBuffer',
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com.STDMETHOD(POINTER(XAUDIO2_BUFFER), c_void_p)),
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('FlushSourceBuffers',
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com.STDMETHOD()),
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('Discontinuity',
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com.STDMETHOD()),
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('ExitLoop',
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com.STDMETHOD()),
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('GetState',
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com.STDMETHOD(POINTER(XAUDIO2_VOICE_STATE), UINT32)),
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('SetFrequencyRatio',
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com.STDMETHOD(FLOAT, UINT32)),
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('GetFrequencyRatio',
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com.STDMETHOD(POINTER(c_float))),
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('SetSourceSampleRate',
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com.STDMETHOD()),
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]
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class IXAudio2MasteringVoice(IXAudio2Voice):
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_methods_ = [
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('GetChannelMask',
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com.STDMETHOD(POINTER(DWORD)))
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]
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class IXAudio2EngineCallback(com.Interface):
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_methods_ = [
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('OnProcessingPassStart',
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com.VOIDMETHOD()),
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('OnProcessingPassEnd',
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com.VOIDMETHOD()),
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('OnCriticalError',
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com.VOIDMETHOD(HRESULT)),
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]
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class XA2EngineCallback(com.COMObject):
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_interfaces_ = [IXAudio2EngineCallback]
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def OnCriticalError(self, hresult):
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raise Exception("Critical Error:", hresult)
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# -------------- 3D Audio Positioning----------
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class X3DAUDIO_DISTANCE_CURVE_POINT(ctypes.Structure):
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_fields_ = [
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('Distance', FLOAT32),
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('DSPSetting', FLOAT32)
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]
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class X3DAUDIO_DISTANCE_CURVE(ctypes.Structure):
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_fields_ = [
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('pPoints', POINTER(X3DAUDIO_DISTANCE_CURVE_POINT)),
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('PointCount', UINT32)
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]
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class X3DAUDIO_VECTOR(ctypes.Structure):
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_fields_ = [
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('x', c_float),
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('y', c_float),
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('z', c_float),
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]
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"""Cone:
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Specifies directionality for a listener or single-channel emitter by
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modifying DSP behaviour with respect to its front orientation.
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This is modeled using two sound cones: an inner cone and an outer cone.
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On/within the inner cone, DSP settings are scaled by the inner values.
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On/beyond the outer cone, DSP settings are scaled by the outer values.
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If on both the cones, DSP settings are scaled by the inner values only.
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Between the two cones, the scaler is linearly interpolated between the
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inner and outer values. Set both cone angles to 0 or X3DAUDIO_2PI for
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omnidirectionality using only the outer or inner values respectively."""
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class X3DAUDIO_CONE(Structure):
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_fields_ = [
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('InnerAngle', FLOAT32), # inner cone angle in radians, must be within [0.0f, X3DAUDIO_2PI]
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('OuterAngle', FLOAT32), # outer cone angle in radians, must be within [InnerAngle, X3DAUDIO_2PI]
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('InnerVolume', FLOAT32), # volume level scaler on/within inner cone, used only for matrix calculations, must be within [0.0f, 2.0f] when used
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('OuterVolume', FLOAT32), # volume level scaler on/beyond outer cone, used only for matrix calculations, must be within [0.0f, 2.0f] when used
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('InnerLPF', FLOAT32), # LPF (both direct and reverb paths) coefficient subtrahend on/within inner cone, used only for LPF (both direct and reverb paths) calculations, must be within [0.0f, 1.0f] when used
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('OuterLPF', FLOAT32), # LPF (both direct and reverb paths) coefficient subtrahend on/beyond outer cone, used only for LPF (both direct and reverb paths) calculations, must be within [0.0f, 1.0f] when used
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('InnerReverb', FLOAT32), # reverb send level scaler on/within inner cone, used only for reverb calculations, must be within [0.0f, 2.0f] when used
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('OuterReverb', FLOAT32) # reverb send level scaler on/beyond outer cone, used only for reverb calculations, must be within [0.0f, 2.0f] when used
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]
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class X3DAUDIO_LISTENER(Structure):
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_fields_ = [
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('OrientFront', X3DAUDIO_VECTOR), # orientation of front direction, used only for matrix and delay calculations or listeners with cones for matrix, LPF (both direct and reverb paths), and reverb calculations, must be normalized when used
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('OrientTop', X3DAUDIO_VECTOR), # orientation of top direction, used only for matrix and delay calculations, must be orthonormal with OrientFront when used
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('Position', X3DAUDIO_VECTOR), # position in user-defined world units, does not affect Velocity
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('Velocity', X3DAUDIO_VECTOR), # velocity vector in user-defined world units/second, used only for doppler calculations, does not affect Position
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('pCone', POINTER(X3DAUDIO_CONE)) # sound cone, used only for matrix, LPF (both direct and reverb paths), and reverb calculations, NULL specifies omnidirectionality
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]
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class X3DAUDIO_EMITTER(Structure):
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_fields_ = [
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('pCone', POINTER(X3DAUDIO_CONE)),
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('OrientFront', X3DAUDIO_VECTOR),
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('OrientTop', X3DAUDIO_VECTOR),
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('Position', X3DAUDIO_VECTOR),
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('Velocity', X3DAUDIO_VECTOR),
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('InnerRadius', FLOAT32),
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('InnerRadiusAngle', FLOAT32),
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('ChannelCount', UINT32),
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('ChannelRadius', FLOAT32),
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('pChannelAzimuths', POINTER(FLOAT32)),
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('pVolumeCurve', POINTER(X3DAUDIO_DISTANCE_CURVE)),
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('pLFECurve', POINTER(X3DAUDIO_DISTANCE_CURVE)),
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('pLPFDirectCurve', POINTER(X3DAUDIO_DISTANCE_CURVE)),
|
|
('pLPFReverbCurve', POINTER(X3DAUDIO_DISTANCE_CURVE)),
|
|
('pReverbCurve', POINTER(X3DAUDIO_DISTANCE_CURVE)),
|
|
('CurveDistanceScaler', FLOAT32),
|
|
('DopplerScaler', FLOAT32)
|
|
]
|
|
|
|
|
|
class X3DAUDIO_DSP_SETTINGS(Structure):
|
|
_fields_ = [
|
|
('pMatrixCoefficients', POINTER(FLOAT)), # float array
|
|
('pDelayTimes', POINTER(FLOAT32)),
|
|
('SrcChannelCount', UINT32),
|
|
('DstChannelCount', UINT32),
|
|
('LPFDirectCoefficient', FLOAT32),
|
|
('LPFReverbCoefficient', FLOAT32),
|
|
('ReverbLevel', FLOAT32),
|
|
('DopplerFactor', FLOAT32),
|
|
('EmitterToListenerAngle', FLOAT32),
|
|
('EmitterToListenerDistance', FLOAT32),
|
|
('EmitterVelocityComponent', FLOAT32),
|
|
('ListenerVelocityComponent', FLOAT32)
|
|
]
|
|
|
|
# Other constants that may or may not be used in X3D.
|
|
|
|
SPEAKER_FRONT_LEFT = 0x00000001
|
|
SPEAKER_FRONT_RIGHT = 0x00000002
|
|
SPEAKER_FRONT_CENTER = 0x00000004
|
|
SPEAKER_LOW_FREQUENCY = 0x00000008
|
|
SPEAKER_BACK_LEFT = 0x00000010
|
|
SPEAKER_BACK_RIGHT = 0x00000020
|
|
SPEAKER_FRONT_LEFT_OF_CENTER = 0x00000040
|
|
SPEAKER_FRONT_RIGHT_OF_CENTER = 0x00000080
|
|
SPEAKER_BACK_CENTER = 0x00000100
|
|
SPEAKER_SIDE_LEFT = 0x00000200
|
|
SPEAKER_SIDE_RIGHT = 0x00000400
|
|
SPEAKER_TOP_CENTER = 0x00000800
|
|
SPEAKER_TOP_FRONT_LEFT = 0x00001000
|
|
SPEAKER_TOP_FRONT_CENTER = 0x00002000
|
|
SPEAKER_TOP_FRONT_RIGHT = 0x00004000
|
|
SPEAKER_TOP_BACK_LEFT = 0x00008000
|
|
SPEAKER_TOP_BACK_CENTER = 0x00010000
|
|
SPEAKER_TOP_BACK_RIGHT = 0x00020000
|
|
SPEAKER_RESERVED = 0x7FFC0000 # bit mask locations reserved for future use
|
|
SPEAKER_ALL = 0x80000000 # used to specify that any possible permutation of speaker configurations
|
|
|
|
SPEAKER_MONO = SPEAKER_FRONT_CENTER
|
|
SPEAKER_STEREO = (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT)
|
|
SPEAKER_2POINT1 = (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_LOW_FREQUENCY)
|
|
SPEAKER_SURROUND = (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_BACK_CENTER)
|
|
SPEAKER_QUAD = (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT)
|
|
SPEAKER_4POINT1 = (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_LOW_FREQUENCY | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT)
|
|
SPEAKER_5POINT1 = (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT)
|
|
SPEAKER_7POINT1 = (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | SPEAKER_FRONT_LEFT_OF_CENTER | SPEAKER_FRONT_RIGHT_OF_CENTER)
|
|
SPEAKER_5POINT1_SURROUND = (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_SIDE_LEFT | SPEAKER_SIDE_RIGHT)
|
|
SPEAKER_7POINT1_SURROUND = (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | SPEAKER_SIDE_LEFT | SPEAKER_SIDE_RIGHT)
|
|
|
|
|
|
DBL_DECIMAL_DIG = 17 # # of decimal digits of rounding precision
|
|
DBL_DIG = 15 # # of decimal digits of precision
|
|
DBL_EPSILON = 2.2204460492503131e-016 # smallest such that 1.0+DBL_EPSILON != 1.0
|
|
DBL_HAS_SUBNORM = 1 # type does support subnormal numbers
|
|
DBL_MANT_DIG = 53 # # of bits in mantissa
|
|
DBL_MAX = 1.7976931348623158e+308 # max value
|
|
DBL_MAX_10_EXP = 308 # max decimal exponent
|
|
DBL_MAX_EXP = 1024 # max binary exponent
|
|
DBL_MIN = 2.2250738585072014e-308 # min positive value
|
|
DBL_MIN_10_EXP = (-307) # min decimal exponent
|
|
DBL_MIN_EXP = (-1021) # min binary exponent
|
|
_DBL_RADIX = 2 # exponent radix
|
|
DBL_TRUE_MIN = 4.9406564584124654e-324 # min positive value
|
|
|
|
FLT_DECIMAL_DIG = 9 # # of decimal digits of rounding precision
|
|
FLT_DIG = 6 # # of decimal digits of precision
|
|
FLT_EPSILON = 1.192092896e-07 # smallest such that 1.0+FLT_EPSILON != 1.0
|
|
FLT_HAS_SUBNORM = 1 # type does support subnormal numbers
|
|
FLT_GUARD = 0
|
|
FLT_MANT_DIG = 24 # # of bits in mantissa
|
|
FLT_MAX = 3.402823466e+38 # max value
|
|
FLT_MAX_10_EXP = 38 # max decimal exponent
|
|
FLT_MAX_EXP = 128 # max binary exponent
|
|
FLT_MIN = 1.175494351e-38 # min normalized positive value
|
|
FLT_MIN_10_EXP = (-37) # min decimal exponent
|
|
FLT_MIN_EXP = (-125) # min binary exponent
|
|
FLT_NORMALIZE = 0
|
|
FLT_RADIX = 2 # exponent radix
|
|
FLT_TRUE_MIN = 1.401298464e-45 # min positive value
|
|
|
|
LDBL_DIG = DBL_DIG # # of decimal digits of precision
|
|
LDBL_EPSILON = DBL_EPSILON # smallest such that 1.0+LDBL_EPSILON != 1.0
|
|
LDBL_HAS_SUBNORM = DBL_HAS_SUBNORM # type does support subnormal numbers
|
|
LDBL_MANT_DIG = DBL_MANT_DIG # # of bits in mantissa
|
|
LDBL_MAX = DBL_MAX # max value
|
|
LDBL_MAX_10_EXP = DBL_MAX_10_EXP # max decimal exponent
|
|
LDBL_MAX_EXP = DBL_MAX_EXP # max binary exponent
|
|
LDBL_MIN = DBL_MIN # min normalized positive value
|
|
LDBL_MIN_10_EXP = DBL_MIN_10_EXP # min decimal exponent
|
|
LDBL_MIN_EXP = DBL_MIN_EXP # min binary exponent
|
|
_LDBL_RADIX = _DBL_RADIX # exponent radix
|
|
LDBL_TRUE_MIN = DBL_TRUE_MIN # min positive value
|
|
|
|
DECIMAL_DIG = DBL_DECIMAL_DIG
|
|
|
|
|
|
X3DAUDIO_HANDLE_BYTESIZE = 20
|
|
X3DAUDIO_HANDLE = (BYTE * X3DAUDIO_HANDLE_BYTESIZE)
|
|
|
|
|
|
# speed of sound in meters per second for dry air at approximately 20C, used with X3DAudioInitialize
|
|
X3DAUDIO_SPEED_OF_SOUND = 343.5
|
|
|
|
|
|
X3DAUDIO_CALCULATE_MATRIX = 0x00000001 # enable matrix coefficient table calculation
|
|
X3DAUDIO_CALCULATE_DELAY = 0x00000002 # enable delay time array calculation (stereo final mix only)
|
|
X3DAUDIO_CALCULATE_LPF_DIRECT = 0x00000004 # enable LPF direct-path coefficient calculation
|
|
X3DAUDIO_CALCULATE_LPF_REVERB = 0x00000008 # enable LPF reverb-path coefficient calculation
|
|
X3DAUDIO_CALCULATE_REVERB = 0x00000010 # enable reverb send level calculation
|
|
X3DAUDIO_CALCULATE_DOPPLER = 0x00000020 # enable doppler shift factor calculation
|
|
X3DAUDIO_CALCULATE_EMITTER_ANGLE = 0x00000040 # enable emitter-to-listener interior angle calculation
|
|
X3DAUDIO_CALCULATE_ZEROCENTER = 0x00010000 # do not position to front center speaker, signal positioned to remaining speakers instead, front center destination channel will be zero in returned matrix coefficient table, valid only for matrix calculations with final mix formats that have a front center channel
|
|
X3DAUDIO_CALCULATE_REDIRECT_TO_LFE = 0x00020000 # apply equal mix of all source channels to LFE destination channel, valid only for matrix calculations with sources that have no LFE channel and final mix formats that have an LFE channel
|
|
|
|
default_dsp_calculation = X3DAUDIO_CALCULATE_MATRIX | X3DAUDIO_CALCULATE_DOPPLER
|
|
|
|
X3DAudioInitialize = x3d_lib.X3DAudioInitialize
|
|
X3DAudioInitialize.restype = HRESULT
|
|
X3DAudioInitialize.argtypes = [c_int, c_float, c_void_p]
|
|
|
|
|
|
X3DAudioCalculate = x3d_lib.X3DAudioCalculate
|
|
X3DAudioCalculate.restype = c_void
|
|
X3DAudioCalculate.argtypes = [POINTER(X3DAUDIO_HANDLE), POINTER(X3DAUDIO_LISTENER), POINTER(X3DAUDIO_EMITTER), UINT32, POINTER(X3DAUDIO_DSP_SETTINGS)]
|
|
|
|
|
|
AudioCategory_Other = 0
|
|
AudioCategory_ForegroundOnlyMedia = 1
|
|
AudioCategory_Communications = 3
|
|
AudioCategory_Alerts = 4
|
|
AudioCategory_SoundEffects = 5
|
|
AudioCategory_GameEffects = 6
|
|
AudioCategory_GameMedia = 7
|
|
AudioCategory_GameChat = 8
|
|
AudioCategory_Speech = 9
|
|
AudioCategory_Movie = 10
|
|
AudioCategory_Media = 11
|
|
|
|
# Reverb not implemented but if someone wants to take a stab at it.
|
|
class XAUDIO2FX_REVERB_PARAMETERS(Structure):
|
|
_fields_ = [
|
|
('WetDryMix', c_float), # ratio of wet (processed) signal to dry (original) signal
|
|
|
|
# Delay times
|
|
('ReflectionsDelay', UINT32), # [0, 300] in ms
|
|
('ReverbDelay', BYTE), # [0, 85] in ms
|
|
('RearDelay', UINT32), # 7.1: [0, 20] in ms, all other: [0, 5] in ms
|
|
('SideDelay', UINT32), # .1: [0, 5] in ms, all other: not used, but still validated # WIN 10 only.
|
|
|
|
# Indexed Paremeters
|
|
('PositionLeft', BYTE), # [0, 30] no units
|
|
('PositionRight', BYTE), # 0, 30] no units, ignored when configured to mono
|
|
('PositionMatrixLeft', BYTE), # [0, 30] no units
|
|
('PositionMatrixRight', BYTE), # [0, 30] no units, ignored when configured to mono
|
|
('EarlyDiffusion', BYTE), # [0, 15] no units
|
|
('LateDiffusion', BYTE), # [0, 15] no units
|
|
('LowEQGain', BYTE), # [0, 12] no units
|
|
('LowEQCutoff', BYTE), # [0, 9] no units
|
|
('LowEQCutoff', BYTE), # [0, 8] no units
|
|
('HighEQCutoff', BYTE), # [0, 14] no units
|
|
|
|
# Direct parameters
|
|
('RoomFilterFreq', c_float), # [20, 20000] in Hz
|
|
('RoomFilterMain', c_float), # [-100, 0] in dB
|
|
('RoomFilterHF', c_float), # [-100, 0] in dB
|
|
('ReflectionsGain', c_float), # [-100, 20] in dB
|
|
('ReverbGain', c_float), # [-100, 20] in dB
|
|
('DecayTime', c_float), # [0.1, inf] in seconds
|
|
('Density', c_float), # [0, 100] (percentage)
|
|
('RoomSize', c_float), # [1, 100] in feet
|
|
|
|
# component control
|
|
('DisableLateField', c_bool), # TRUE to disable late field reflections
|
|
]
|
|
|
|
|
|
class IXAudio2(com.pIUnknown):
|
|
_methods_ = [
|
|
('RegisterForCallbacks',
|
|
com.STDMETHOD(POINTER(IXAudio2EngineCallback))),
|
|
('UnregisterForCallbacks',
|
|
com.METHOD(ctypes.c_void_p, POINTER(IXAudio2EngineCallback))),
|
|
('CreateSourceVoice',
|
|
com.STDMETHOD(POINTER(IXAudio2SourceVoice), POINTER(WAVEFORMATEX), UINT32, c_float,
|
|
POINTER(IXAudio2VoiceCallback), POINTER(XAUDIO2_VOICE_SENDS), POINTER(XAUDIO2_EFFECT_CHAIN))),
|
|
('CreateSubmixVoice',
|
|
com.STDMETHOD(POINTER(IXAudio2SubmixVoice), UINT32, UINT32, UINT32, UINT32,
|
|
POINTER(XAUDIO2_VOICE_SENDS), POINTER(XAUDIO2_EFFECT_CHAIN))),
|
|
('CreateMasteringVoice',
|
|
com.STDMETHOD(POINTER(IXAudio2MasteringVoice), UINT32, UINT32, UINT32, LPCWSTR, POINTER(XAUDIO2_EFFECT_CHAIN),
|
|
UINT32)),
|
|
('StartEngine',
|
|
com.STDMETHOD()),
|
|
('StopEngine',
|
|
com.STDMETHOD()),
|
|
('CommitChanges',
|
|
com.STDMETHOD(UINT32)),
|
|
('GetPerformanceData',
|
|
com.METHOD(c_void, POINTER(XAUDIO2_PERFORMANCE_DATA))),
|
|
('SetDebugConfiguration',
|
|
com.STDMETHOD(POINTER(XAUDIO2_DEBUG_CONFIGURATION), c_void_p)),
|
|
]
|
|
|
|
|
|
XAudio2Create = xaudio2_lib.XAudio2Create
|
|
XAudio2Create.restype = HRESULT
|
|
XAudio2Create.argtypes = [POINTER(IXAudio2), UINT32, UINT32]
|
|
|
|
CreateAudioReverb = xaudio2_lib.CreateAudioReverb
|
|
CreateAudioReverb.restype = HRESULT
|
|
CreateAudioReverb.argtypes = [POINTER(com.pIUnknown)]
|
|
|