255 lines
8.7 KiB
C
255 lines
8.7 KiB
C
/* ========================== *
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* WASAPI backend for audio playback
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*
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* Based on mmozeiko's WASAPI example
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* https://gist.github.com/mmozeiko/5a5b168e61aff4c1eaec0381da62808f#file-win32_wasapi-h
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* ========================== */
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#include "playback.h"
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#include "scratch.h"
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#include "sys.h"
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#include "mixer.h"
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#include "atomic.h"
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#include "app.h"
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#define COBJMACROS
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#define WIN32_LEAN_AND_MEAN
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#define UNICODE
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#include <Windows.h>
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#include <initguid.h>
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#include <objbase.h>
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#include <uuids.h>
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#include <avrt.h>
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#include <Audioclient.h>
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#include <mmdeviceapi.h>
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#pragma comment(lib, "avrt")
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DEFINE_GUID(CLSID_MMDeviceEnumerator, 0xbcde0395, 0xe52f, 0x467c, 0x8e, 0x3d, 0xc4, 0x57, 0x92, 0x91, 0x69, 0x2e);
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DEFINE_GUID(IID_IMMDeviceEnumerator, 0xa95664d2, 0x9614, 0x4f35, 0xa7, 0x46, 0xde, 0x8d, 0xb6, 0x36, 0x17, 0xe6);
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DEFINE_GUID(IID_IAudioClient, 0x1cb9ad4c, 0xdbfa, 0x4c32, 0xb1, 0x78, 0xc2, 0xf5, 0x68, 0xa7, 0x03, 0xb2);
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DEFINE_GUID(IID_IAudioClient3, 0x7ed4ee07, 0x8e67, 0x4cd4, 0x8c, 0x1a, 0x2b, 0x7a, 0x59, 0x87, 0xad, 0x42);
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DEFINE_GUID(IID_IAudioRenderClient, 0xf294acfc, 0x3146, 0x4483, 0xa7, 0xbf, 0xad, 0xdc, 0xa7, 0xc2, 0x60, 0xe2);
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struct wasapi_buffer {
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u32 frames_count;
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u8 *frames;
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};
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GLOBAL struct {
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struct atomic_i32 playback_thread_shutdown;
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struct sys_thread playback_thread;
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IAudioClient *client;
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HANDLE event;
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IAudioRenderClient *playback;
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WAVEFORMATEX *buffer_format;
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u32 buffer_frames;
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HANDLE mmtc_handle;
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} G = { 0 }, DEBUG_ALIAS(G, G_playback_wasapi);
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/* ========================== *
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* Startup
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* ========================== */
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INTERNAL void wasapi_initialize(void);
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INTERNAL APP_EXIT_CALLBACK_FUNC_DEF(playback_shutdown);
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INTERNAL SYS_THREAD_ENTRY_POINT_FUNC_DEF(playback_thread_entry_point, arg);
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struct playback_startup_receipt playback_startup(struct mixer_startup_receipt *mixer_sr)
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{
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(UNUSED)mixer_sr;
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wasapi_initialize();
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G.playback_thread = sys_thread_alloc(&playback_thread_entry_point, NULL, STR("[P3] Audio thread"));
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app_register_exit_callback(&playback_shutdown);
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return (struct playback_startup_receipt) { 0 };
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}
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INTERNAL APP_EXIT_CALLBACK_FUNC_DEF(playback_shutdown)
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{
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__prof;
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atomic_i32_eval_exchange(&G.playback_thread_shutdown, true);
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sys_thread_wait_release(&G.playback_thread);
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}
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/* ========================== *
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* Wasapi initialization
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* ========================== */
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INTERNAL void wasapi_initialize(void)
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{
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/* https://learn.microsoft.com/en-us/windows/win32/procthread/multimedia-class-scheduler-service#registry-settings */
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DWORD task = 0;
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G.mmtc_handle = AvSetMmThreadCharacteristicsW(L"Pro Audio", &task);
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u64 sample_rate = PLAYBACK_SAMPLE_RATE;
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u64 channel_count = 2;
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u32 channel_mask = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT;
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/* Create enumerator to get audio device */
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IMMDeviceEnumerator *enumerator;
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CoCreateInstance(&CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL, &IID_IMMDeviceEnumerator, (LPVOID *)&enumerator);
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/* Get default playback device */
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IMMDevice *device;
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IMMDeviceEnumerator_GetDefaultAudioEndpoint(enumerator, eRender, eConsole, &device);
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IMMDeviceEnumerator_Release(enumerator);
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/* Create audio client for device */
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IMMDevice_Activate(device, &IID_IAudioClient, CLSCTX_ALL, NULL, (LPVOID *)&G.client);
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IMMDevice_Release(device);
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WAVEFORMATEXTENSIBLE format_ex = {
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.Format = {
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.wFormatTag = WAVE_FORMAT_EXTENSIBLE,
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.nChannels = (WORD)channel_count,
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.nSamplesPerSec = (WORD)sample_rate,
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.nAvgBytesPerSec = (DWORD)(sample_rate * channel_count * sizeof(f32)),
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.nBlockAlign = (WORD)(channel_count * sizeof(f32)),
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.wBitsPerSample = (WORD)(8 * sizeof(f32)),
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.cbSize = sizeof(format_ex) - sizeof(format_ex.Format),
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},
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.Samples.wValidBitsPerSample = 8 * sizeof(f32),
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.dwChannelMask = channel_mask,
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.SubFormat = MEDIASUBTYPE_IEEE_FLOAT,
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};
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WAVEFORMATEX *wfx = &format_ex.Format;
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#if 0
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b32 client_initialized = FALSE;
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IAudioClient3 *client3;
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if (SUCCEEDED(IAudioClient_QueryInterface(G.client, &IID_IAudioClient3, (LPVOID *)&client3))) {
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/* From Martins: Minimum buffer size will typically be 480 samples (10msec @ 48khz)
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* but it can be 128 samples (2.66 msec @ 48khz) if driver is properly installed
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* see bullet-point instructions here: https://learn.microsoft.com/en-us/windows-hardware/drivers/audio/low-latency-audio#measurement-tools
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*/
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UINT32 default_period_samples, fundamental_period_samples, min_period_samples, max_period_samples;
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IAudioClient3_GetSharedModeEnginePeriod(client3, wfx, &default_period_samples, &fundamental_period_samples, &min_period_samples, &max_period_samples);
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const DWORD flags = AUDCLNT_STREAMFLAGS_EVENTCALLBACK;
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if (SUCCEEDED(IAudioClient3_InitializeSharedAudioStream(client3, flags, min_period_samples, wfx, NULL))) {
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client_initialized = TRUE;
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}
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IAudioClient3_Release(client3);
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}
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#else
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b32 client_initialized = false;
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#endif
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if (!client_initialized) {
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/* Get duration for shared-mode streams, this will typically be 480 samples (10msec @ 48khz) */
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REFERENCE_TIME duration;
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IAudioClient_GetDevicePeriod(G.client, &duration, NULL);
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/* Initialize audio playback
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*
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* NOTE:
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* Passing AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM will tell WASAPI to
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* always convert to native mixing format. This may introduce latency
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* but allows for any input format.
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*/
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const DWORD flags = AUDCLNT_STREAMFLAGS_EVENTCALLBACK | AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM | AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY;
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IAudioClient_Initialize(G.client, AUDCLNT_SHAREMODE_SHARED, flags, duration, 0, wfx, NULL);
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}
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IAudioClient_GetMixFormat(G.client, &G.buffer_format);
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/* Set up event handler to wait on */
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G.event = CreateEventW(NULL, FALSE, FALSE, NULL);
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IAudioClient_SetEventHandle(G.client, G.event);
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/* Get playback client */
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IAudioClient_GetService(G.client, &IID_IAudioRenderClient, (LPVOID *)&G.playback);
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/* Start the playback */
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IAudioClient_Start(G.client);
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/* Get audio buffer size in samples */
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IAudioClient_GetBufferSize(G.client, &G.buffer_frames);
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}
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/* ========================== *
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* Playback thread update
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* ========================== */
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INTERNAL struct wasapi_buffer wasapi_update_begin(void)
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{
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__prof;
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struct wasapi_buffer wspbuf = { 0 };
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/* Wait */
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{
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__profscope(wasapi_wait_on_event);
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WaitForSingleObject(G.event, INFINITE);
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}
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/* Get padding frames */
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u32 padding_frames;
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IAudioClient_GetCurrentPadding(G.client, &padding_frames);
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/* Get output buffer from WASAPI */
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wspbuf.frames_count = 0;
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if (padding_frames <= G.buffer_frames) {
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wspbuf.frames_count = G.buffer_frames - padding_frames;
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}
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IAudioRenderClient_GetBuffer(G.playback, wspbuf.frames_count, &wspbuf.frames);
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return wspbuf;
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}
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INTERNAL void wasapi_update_end(struct wasapi_buffer *wspbuf, struct mixed_pcm_f32 src)
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{
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__prof;
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u32 frames_in_source = src.count / 2;
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u32 frames_in_output = wspbuf->frames_count;
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u32 flags = 0;
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if (frames_in_source == frames_in_output) {
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/* Copy bytes to output */
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u32 bytes_per_sample = G.buffer_format->nBlockAlign / G.buffer_format->nChannels;
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u32 write_size = frames_in_source * 2 * bytes_per_sample;
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MEMCPY(wspbuf->frames, src.samples, write_size);
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} else {
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/* Submit silence if not enough samples */
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flags = AUDCLNT_BUFFERFLAGS_SILENT;
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/* This shouldn't occur, mixer should be generating samples equivilent
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* to value returned from `playback_update_begin`. */
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ASSERT(false);
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}
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#if !AUDIO_ENABLED
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flags = AUDCLNT_BUFFERFLAGS_SILENT;
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#endif
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/* Submit output buffer to WASAPI */
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IAudioRenderClient_ReleaseBuffer(G.playback, frames_in_source, flags);
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__profframe("Audio");
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}
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/* ========================== *
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* Playback thread entry
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* ========================== */
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INTERNAL SYS_THREAD_ENTRY_POINT_FUNC_DEF(playback_thread_entry_point, arg)
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{
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struct temp_arena scratch = scratch_begin_no_conflict();
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(UNUSED)arg;
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/* FIXME: If playback fails at any point and mixer stops advancing, we
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* need to halt mixer to prevent memory leak when sounds are played. */
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while (!atomic_i32_eval(&G.playback_thread_shutdown)) {
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struct temp_arena temp = arena_temp_begin(scratch.arena);
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struct wasapi_buffer wspbuf = wasapi_update_begin();
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struct mixed_pcm_f32 pcm = mixer_update(temp.arena, wspbuf.frames_count);
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wasapi_update_end(&wspbuf, pcm);
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arena_temp_end(temp);
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}
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scratch_end(scratch);
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}
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