update DivaSound source
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@@ -12,43 +12,127 @@
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#include <mmdeviceapi.h>
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#include <audioclient.h>
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void (__cdecl* divaAudioInit)(void* cls, uint64_t unk, uint64_t unk2) = (void(__cdecl*)(void* cls, uint64_t unk, uint64_t unk2))0x1406269F0;
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void (*divaAudioInit)(void* cls, uint64_t unk, uint64_t unk2) = (void(*)(void* cls, uint64_t unk, uint64_t unk2))0x1406269F0;
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void (__cdecl* divaAudioFillbuffer)(void* mixer, int16_t* buf, uint64_t nFrames, bool disableHpVol, bool invertPhase) = (void(__cdecl*)(void* mixer, int16_t* buf, uint64_t nFrames, bool disableHpVol, bool invertPhase))0x140627370;
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void (*divaAudioFillbuffer)(void* mixer, int16_t* buf, uint64_t nFrames, bool disableHpVol, bool invertPhase) = (void(*)(void* mixer, int16_t* buf, uint64_t nFrames, bool disableHpVol, bool invertPhase))0x140627370;
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// mixer is a pointer to an audioMixer (from audio init's cls + 0x70)
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// disableHpVol uses speaker volume for headphones (only works in 4ch mode)
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// invertPhase inverts the output signal (only works if disableHpVol == false)
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int (__cdecl* divaAudioAllocMixer)(void* cls, uint64_t unk, uint64_t unk2, int64_t nFrames) = (int(__cdecl*)(void* cls, uint64_t unk, uint64_t unk2, int64_t nFrames))0x140626710;
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int (*divaAudioAllocMixer)(void* cls, uint64_t unk, uint64_t unk2, int64_t nFrames) = (int(*)(void* cls, uint64_t unk, uint64_t unk2, int64_t nFrames))0x140626710;
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// unks are the same as from the init call. they seem to set the internal mixing channel counts(?)
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// cls is the same as mixer in divaAudioFillbuffer
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// nFrames is number of audio frames to hold in the mixing buffers (only used when divaAudioFillbuffer is called). Internally this is multiplied by 16 (buffers are built using 32bit floats)
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int(__cdecl* mtx_init)(void* mutex, int type) = (int(__cdecl*)(void* mutex, int type))0x14081DEF4;
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int(*mtx_init)(void* mutex, int type) = (int(*)(void* mutex, int type))0x14081DEF4;
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int(*mtx_lock_0)(void* mutex) = (int(*)(void* mutex))0x14081DF50;
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int(*mtx_unlock)(void* mutex) = (int(*)(void* mutex))0x14081DFD4;
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typedef void streamingCallback(float* buffer, int64_t nFrames, void* userdata);
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#pragma pack(push, 1)
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struct _50 {
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byte padding00[0x50];
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};
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struct _70 {
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byte padding00[0x70];
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};
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struct initClass; // define these earlier so they can be used in other structs
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struct audioMixer;
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// known streaming channel variables
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// + 0x00 = pointer to mixer class
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// + 0x08 = buffer pointer (float)
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// + 0x10 = buffer size (frame count * 16)
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// + 0x18 = playing (32 bits, set to 0 for pause)
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// + 0x1c = reset (32 bits, set to 0 to kill channel)
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// + 0x20 = mutex pointer
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// + 0x28 = master volume float (multiply other volumes by this)
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// + 0x2c = ch1 volume
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// + 0x30 = ch2 volume
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// + 0x34 = ch3 volume
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// + 0x38 = ch4 volume
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//
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// + 0x40 = callback func
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// + 0x48 = callback userdata pointer
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struct streamingChannel {
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audioMixer* mixer;
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struct streamingState {
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std::vector<_50>* state;
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float* buffer;
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uint64_t buffer_size;
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uint64_t buffer_size; // in bytes
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uint32_t playing; // set to 0 for pause (usually ==1)
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uint32_t reset; // set to 1 to kill (usually ==0)
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void* mutex;
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float volume_master;
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float volume_channels[4];
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byte padding3c[0x4];
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streamingCallback callback;
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void* userdata;
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};
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struct initClass;
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// known SE channel variables
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// + 0x00 = pointer to mixer class
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// + 0x08 = some kind of toggle (is playing)?
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// + 0x10 = mutex pointer
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// + 0x18 = master volume float (multiply other volumes by this)
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// + 0x1c = ch1 volume
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// + 0x20 = ch2 volume
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// + 0x24 = ch3 volume
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// + 0x28 = ch4 volume
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// + 0x30 = buffer pointer (float)
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// + 0x38 = buffer size????? (frame count * 16)??? (buffers seem to hold whole sounds)
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// + 0x40 = num channels
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// + 0x48 = num frames
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// + 0x50 = sample rate???
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// + 0x58 = loop start frame???
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// + 0x60 = loop end frame???
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// + 0x68 = current playback time
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struct seChannel {
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audioMixer* mixer;
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uint32_t unk1; // toggle on/off?
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byte padding0c[0x4];
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void* mutex;
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float volume_master;
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float volume_channels[4];
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byte padding2c[0x4];
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float* buffer;
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byte padding38[0x8]; // probably buffer size
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int64_t nChannels; // seems to affect mapping
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uint64_t length; // in frames
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byte padding50[0x8]; // sample rate?
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uint64_t loop_start; // ?
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uint64_t loop_end; // ?
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uint64_t curTime;
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};
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// known internal audio mixer variables (used by divaAudioFillbuffer and divaAudioAllocMixer)
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// cls + 0x00 = pointer to main audio/init class
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// cls + 0x08 = SE channels pointer
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// cls + 0x10 = SE channels count
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// cls + 0x18 = streaming channels pointer
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// cls + 0x20 = streaming channels count
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// cls + 0x28 = mixing buffer pointer (when a buffer is generated, this is filled with 32bit floats (four per frame)
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// cls + 0x30 = mixing buffer size (frame count * 16)
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//
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// cls + 0x38 = mutex lock for volume
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// cls + 0x40 = master volume float (multiply other volumes by this)
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// cls + 0x44 = ch1 volume
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// cls + 0x48 = ch2 volume
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// cls + 0x4c = ch3 volume
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// cls + 0x50 = ch4 volume
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struct audioMixer {
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initClass* audioClass;
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std::vector<_70>** state1;
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uint64_t state1_len;
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streamingState* state2;
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uint64_t state2_len;
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seChannel* seChannels;
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uint64_t seChannels_len;
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streamingChannel* streamingChannels;
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uint64_t streamingChannels_len;
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float* mixbuffer;
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uint64_t mixbuffer_size;
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@@ -60,6 +144,27 @@ struct audioMixer {
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byte padding54[4];
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};
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// known main/init audio class variables (used by divaAudioInit)
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// cls + 0x0 = WAVEFORMATEXTENSIBLE wave_format (null with old method) (length 0x28)
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// cls + 0x28 = IMMDeviceEnumerator *pEnumerator
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// cls + 0x30 = IMMDevice *pDevice (default device)
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// cls + 0x38 = IAudioClient *pAudioClient
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//
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// cls + 0x40 = buffer size in frames
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//
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// cls + 0x48 = IAudioRenderClient *pRenderClient
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//
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// cls + 0x50 = HANDLE hEvent (for original WASAPI buffer timing)
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//
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// cls + 0x58 = number of output channels
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// cls + 0x60 = audio sample rate
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// cls + 0x68 = audio bit depth (only 16bit works)
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//
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// cls + 0x70 = pointer to mixer class
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//
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// cls + 0x78 = HANDLE* hCallback (handle to original callback thread)
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//
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// cls + 0x80 = break flag dword (becomes 1 on exit)
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struct initClass {
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WAVEFORMATEXTENSIBLE wave_format;
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IMMDeviceEnumerator *pEnumerator;
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@@ -83,45 +188,6 @@ struct initClass {
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};
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#pragma pack(pop)
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// known internal audio mixer variables (used by divaAudioFillbuffer and divaAudioAllocMixer)
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// cls + 0x00 = pointer to main audio/init class
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// cls + 0x08 = internal state 1 (SE?) vector pointer (or full class?)
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// cls + 0x10 = internal state 1 (SE?) length (channels?)
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// cls + 0x18 = internal state 2 (streaming?) vector pointer (or full class?)
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// cls + 0x20 = internal state 2 (streaming?) length (channels?)
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// cls + 0x28 = mixing buffer pointer (when a buffer is generated, this is filled with 32bit floats (four per frame)
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// cls + 0x30 = mixing buffer size (frame count * 16)
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//
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// cls + 0x38 = mutex lock for volume
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// cls + 0x40 = master volume float (multiply other volumes by this)
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// cls + 0x44 = ch1 volume
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// cls + 0x48 = ch2 volume
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// cls + 0x4c = ch3 volume
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// cls + 0x50 = ch4 volume
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//
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//
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// known main/init audio class variables (used by divaAudioInit)
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// cls + 0x0 = WAVEFORMATEXTENSIBLE wave_format (null with old method) (length 0x28)
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// cls + 0x28 = IMMDeviceEnumerator *pEnumerator
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// cls + 0x30 = IMMDevice *pDevice (default device)
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// cls + 0x38 = IAudioClient *pAudioClient
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//
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// cls + 0x40 = buffer size in frames
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//
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// cls + 0x48 = IAudioRenderClient *pRenderClient
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//
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// cls + 0x50 = HANDLE hEvent (for original WASAPI buffer timing)
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//
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// cls + 0x58 = number of output channels
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// cls + 0x60 = audio sample rate
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// cls + 0x68 = audio bit depth (only 16bit works)
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//
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// cls + 0x70 = pointer to mixer class
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//
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// cls + 0x78 = HANDLE* hCallback (handle to original callback thread)
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//
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// cls + 0x80 = break flag dword (becomes 1 on exit)
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initClass *divaAudCls;
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audioMixer *divaAudioMixCls;
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