mirror of
https://github.com/hrydgard/ppsspp.git
synced 2026-07-11 09:35:09 +02:00
680 lines
21 KiB
C++
680 lines
21 KiB
C++
#include <windows.h>
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#include <mmdeviceapi.h>
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#include <functiondiscoverykeys_devpkey.h>
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#include <audioclient.h>
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#include <avrt.h>
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#include <comdef.h>
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#include <atomic>
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#include <thread>
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#include <vector>
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#include <string_view>
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#include <wrl/client.h>
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#include "Common/Data/Encoding/Utf8.h"
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#include "Common/Log.h"
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#include "Common/StringUtils.h"
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#include "Common/Thread/ThreadUtil.h"
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#include "WASAPIContext.h"
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using Microsoft::WRL::ComPtr;
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// We must have one of these already...
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static inline s16 ClampFloatToS16(float f) {
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f *= 32768.0f;
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if (f >= 32767) {
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return 32767;
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} else if (f < -32767) {
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return -32767;
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} else {
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return (s16)(s32)f;
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}
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}
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void BuildStereoFloatFormat(const WAVEFORMATEXTENSIBLE *original, WAVEFORMATEXTENSIBLE *output) {
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// Zero‑init all fields first.
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ZeroMemory(output, sizeof(WAVEFORMATEXTENSIBLE));
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// Fill the WAVEFORMATEX base part.
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output->Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
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output->Format.nChannels = 2;
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output->Format.nSamplesPerSec = original->Format.nSamplesPerSec;
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output->Format.wBitsPerSample = 32; // 32‑bit float
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output->Format.nBlockAlign = output->Format.nChannels *
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output->Format.wBitsPerSample / 8;
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output->Format.nAvgBytesPerSec = output->Format.nSamplesPerSec *
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output->Format.nBlockAlign;
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output->Format.cbSize = sizeof(WAVEFORMATEXTENSIBLE) - sizeof(WAVEFORMATEX);
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// Fill the extensible fields.
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output->Samples.wValidBitsPerSample = 32;
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output->dwChannelMask = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT;
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output->SubFormat = KSDATAFORMAT_SUBTYPE_IEEE_FLOAT;
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}
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WASAPIContext::WASAPIContext() : notificationClient_(this) {
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HRESULT hr = CoCreateInstance(__uuidof(MMDeviceEnumerator), nullptr, CLSCTX_ALL, IID_PPV_ARGS(&enumerator_));
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if (FAILED(hr)) {
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// Bad!
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enumerator_ = nullptr;
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return;
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}
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enumerator_->RegisterEndpointNotificationCallback(¬ificationClient_);
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}
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WASAPIContext::~WASAPIContext() {
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if (!enumerator_) {
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// Nothing can have been happening.
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return;
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}
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Stop();
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enumerator_->UnregisterEndpointNotificationCallback(¬ificationClient_);
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}
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WASAPIContext::AudioFormat WASAPIContext::Classify(const WAVEFORMATEX *format) {
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if (format->wFormatTag == WAVE_FORMAT_EXTENSIBLE) {
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const WAVEFORMATEXTENSIBLE *ex = (const WAVEFORMATEXTENSIBLE *)format;
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if (ex->SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT) {
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if (format->nChannels >= 1)
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return AudioFormat::Float;
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} else {
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wchar_t guid[256]{};
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StringFromGUID2(ex->SubFormat, guid, 256);
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ERROR_LOG(Log::Audio, "Got unexpected WASAPI 0xFFFE stream format (%S), expected float!", guid);
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if (ex->Format.wBitsPerSample == 16 && format->nChannels >= 1) {
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INFO_LOG(Log::Audio, "Got a PCM16 audio output (%d channels)", format->nChannels);
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return AudioFormat::PCM16;
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}
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}
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} else if (format->wFormatTag == WAVE_FORMAT_IEEE_FLOAT && format->nChannels >= 1) {
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return AudioFormat::Float;
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} else if (format->wFormatTag == WAVE_FORMAT_PCM && format->wBitsPerSample == 16 && format->nChannels >= 1) {
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INFO_LOG(Log::Audio, "Got a PCM16 audio output", format->nChannels);
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return AudioFormat::PCM16;
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} else {
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WARN_LOG(Log::Audio, "Unhandled output format!");
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}
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return AudioFormat::Unhandled;
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}
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bool GetDeviceDesc(IMMDevice *device, AudioDeviceDesc *desc) {
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ComPtr<IPropertyStore> props;
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device->OpenPropertyStore(STGM_READ, &props);
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PROPVARIANT nameProp;
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PropVariantInit(&nameProp);
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props->GetValue(PKEY_Device_FriendlyName, &nameProp);
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LPWSTR id_str = 0;
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bool success = false;
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if (SUCCEEDED(device->GetId(&id_str))) {
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desc->name = ConvertWStringToUTF8(nameProp.pwszVal);
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desc->uniqueId = ConvertWStringToUTF8(id_str);
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CoTaskMemFree(id_str);
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success = true;
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}
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PropVariantClear(&nameProp);
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return success;
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}
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void WASAPIContext::EnumerateDevices(std::vector<AudioDeviceDesc> *output, bool captureDevices) {
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ComPtr<IMMDeviceCollection> collection;
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enumerator_->EnumAudioEndpoints(captureDevices ? eCapture : eRender, DEVICE_STATE_ACTIVE, &collection);
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if (!collection) {
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ERROR_LOG(Log::Audio, "Failed to enumerate devices");
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return;
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}
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UINT count = 0;
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collection->GetCount(&count);
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for (UINT i = 0; i < count; ++i) {
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ComPtr<IMMDevice> device;
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collection->Item(i, &device);
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AudioDeviceDesc desc{};
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if (GetDeviceDesc(device.Get(), &desc)) {
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output->push_back(desc);
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}
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}
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}
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// Also logs.
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void WASAPIContext::SetErrorString(std::string_view str, HRESULT hr) {
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std::string temp = StringFromFormat("%s (HRESULT: %08lx)", str.data(), hr);
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ERROR_LOG(Log::Audio, "%s", temp.c_str());
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std::lock_guard<std::mutex> guard(errorLock_);
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errorString_ = temp;
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}
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void WASAPIContext::ClearErrorString() {
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std::lock_guard<std::mutex> guard(errorLock_);
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errorString_.clear();
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}
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bool WASAPIContext::TryInitAudioClient3(IMMDevice *device, LatencyMode latencyMode) {
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HRESULT hr = E_FAIL;
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// Try IAudioClient3 first if not in "safe" mode. It's probably safe anyway, but still, let's use the legacy client as a safe fallback option.
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if (latencyMode != LatencyMode::Safe) {
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hr = device->Activate(__uuidof(IAudioClient3), CLSCTX_ALL, nullptr, (void**)&audioClient3_);
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} else {
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// Proceed to AudioClient.
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INFO_LOG(Log::Audio, "LatencyMode::Safe is set, skipping AudioClient3 and going directly to AudioClient");
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return false;
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}
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if (!SUCCEEDED(hr)) {
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audioClient3_.Reset();
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return false;
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}
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hr = audioClient3_->GetMixFormat(&format_);
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if (FAILED(hr)) {
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audioClient3_.Reset();
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SetErrorString("AudioClient3 GetMixFormat failed", hr);
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return false;
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}
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curSamplesPerSec_ = format_->nSamplesPerSec;
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// We only use AudioClient3 if we got the format we wanted (stereo float).
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if (format_->nChannels != 2 || Classify(format_) != AudioFormat::Float) {
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// Let's fall back to the old path. The docs seem to be wrong, if you try to create an
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// AudioClient3 with low latency audio with AUTOCONVERTPCM, you get the error 0x88890021.
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INFO_LOG(Log::Audio, "AudioClient3: Got %d channels or non-float format, falling back to AudioClient", format_->nChannels);
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audioClient3_.Reset();
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// Free the format before falling through - AudioClient will allocate a new one
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CoTaskMemFree(format_);
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format_ = nullptr;
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return false;
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} else {
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hr = audioClient3_->GetSharedModeEnginePeriod(format_, &defaultPeriodFrames_, &fundamentalPeriodFrames_, &minPeriodFrames_, &maxPeriodFrames_);
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if (FAILED(hr)) {
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audioClient3_.Reset();
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CoTaskMemFree(format_);
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format_ = nullptr;
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SetErrorString("AudioClient3 GetSharedModeEnginePeriod failed", hr);
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return false;
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}
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INFO_LOG(Log::Audio, "AudioClient3: default: %d fundamental: %d min: %d max: %d\n", (int)defaultPeriodFrames_, (int)fundamentalPeriodFrames_, (int)minPeriodFrames_, (int)maxPeriodFrames_);
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INFO_LOG(Log::Audio, "initializing with %d frame period at %d Hz, meaning %0.1fms\n", (int)minPeriodFrames_, (int)format_->nSamplesPerSec, FramesToMs(minPeriodFrames_, format_->nSamplesPerSec));
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hr = audioClient3_->InitializeSharedAudioStream(
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AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
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minPeriodFrames_,
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format_,
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nullptr
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);
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if (FAILED(hr)) {
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WARN_LOG(Log::Audio, "Error initializing AudioClient3 shared audio stream: %08lx", hr);
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audioClient3_.Reset();
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CoTaskMemFree(format_);
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format_ = nullptr;
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SetErrorString("AudioClient3 init failed", hr);
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return false;
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}
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actualPeriodFrames_ = minPeriodFrames_;
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hr = audioClient3_->GetBufferSize(&reportedBufferSize_);
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if (FAILED(hr)) {
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audioClient3_.Reset();
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CoTaskMemFree(format_);
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format_ = nullptr;
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SetErrorString("AudioClient3 GetBufferSize failed", hr);
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return false;
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}
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hr = audioClient3_->SetEventHandle(audioEvent_);
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if (FAILED(hr)) {
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audioClient3_.Reset();
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CoTaskMemFree(format_);
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format_ = nullptr;
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SetErrorString("AudioClient3 SetEventHandle failed", hr);
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return false;
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}
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hr = audioClient3_->GetService(IID_PPV_ARGS(&renderClient_));
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if (FAILED(hr)) {
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audioClient3_.Reset();
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CoTaskMemFree(format_);
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format_ = nullptr;
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SetErrorString("AudioClient3 GetService failed", hr);
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return false;
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}
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}
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return true;
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}
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bool WASAPIContext::TryInitAudioClient(IMMDevice *device, LatencyMode latencyMode) {
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// Fallback to IAudioClient (older OS)
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HRESULT hr = device->Activate(__uuidof(IAudioClient), CLSCTX_ALL, nullptr, (void**)&audioClient_);
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if (FAILED(hr)) {
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SetErrorString("Failed to activate audio device", hr);
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return false;
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}
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hr = audioClient_->GetMixFormat(&format_);
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if (FAILED(hr)) {
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audioClient_.Reset();
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SetErrorString("AudioClient GetMixFormat failed", hr);
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return false;
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}
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// If there are too many channels, try asking for a 2-channel output format.
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DWORD extraStreamFlags = 0;
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const AudioFormat fmt = Classify(format_);
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curSamplesPerSec_ = format_->nSamplesPerSec;
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bool createBuffer = false;
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if (fmt == AudioFormat::Float) {
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if (format_->nChannels != 2) {
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INFO_LOG(Log::Audio, "Got %d channels, asking for stereo instead", format_->nChannels);
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WAVEFORMATEXTENSIBLE stereo;
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BuildStereoFloatFormat((const WAVEFORMATEXTENSIBLE *)format_, &stereo);
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WAVEFORMATEX *closestMatch = nullptr;
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const HRESULT result = audioClient_->IsFormatSupported(AUDCLNT_SHAREMODE_SHARED, (const WAVEFORMATEX *)&stereo, &closestMatch);
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if (result == S_OK) {
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// We got the format! Use it and set as current.
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_dbg_assert_(!closestMatch);
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WAVEFORMATEX *newFormat = (WAVEFORMATEX *)CoTaskMemAlloc(sizeof(WAVEFORMATEXTENSIBLE));
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_dbg_assert_(newFormat);
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memcpy(newFormat, &stereo, sizeof(WAVEFORMATEX) + stereo.Format.cbSize);
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CoTaskMemFree(format_);
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format_ = newFormat;
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extraStreamFlags = AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM | AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY;
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INFO_LOG(Log::Audio, "Successfully asked for two channels");
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} else if (result == S_FALSE) {
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// We got another format. Meh, let's just use what we got.
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if (closestMatch) {
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WARN_LOG(Log::Audio, "Didn't get the format we wanted, but got: %lu ch=%d", closestMatch->nSamplesPerSec, closestMatch->nChannels);
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CoTaskMemFree(closestMatch);
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} else {
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WARN_LOG(Log::Audio, "Failed to fall back to two channels. Using workarounds.");
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}
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createBuffer = true;
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} else {
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WARN_LOG(Log::Audio, "Got other error %08lx", result);
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_dbg_assert_(!closestMatch);
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}
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} else {
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// All good, nothing to convert.
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_dbg_assert_(format_);
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}
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} else {
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// Some other format.
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WARN_LOG(Log::Audio, "Format not float, applying conversion.");
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createBuffer = true;
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}
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// Get engine period info
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REFERENCE_TIME defaultPeriod = 0, minPeriod = 0;
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audioClient_->GetDevicePeriod(&defaultPeriod, &minPeriod);
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const REFERENCE_TIME duration = minPeriod;
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hr = audioClient_->Initialize(
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AUDCLNT_SHAREMODE_SHARED,
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AUDCLNT_STREAMFLAGS_EVENTCALLBACK | extraStreamFlags,
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duration, // This is a minimum, the result might be larger. We use GetBufferSize to check.
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0, // ref duration, always 0 in shared mode.
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format_,
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nullptr
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);
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if (FAILED(hr)) {
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audioClient_.Reset();
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CoTaskMemFree(format_);
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format_ = nullptr;
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SetErrorString("AudioClient init failed", hr);
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return false;
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}
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hr = audioClient_->GetBufferSize(&reportedBufferSize_);
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if (FAILED(hr)) {
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audioClient_.Reset();
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CoTaskMemFree(format_);
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format_ = nullptr;
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SetErrorString("AudioClient GetBufferSize failed", hr);
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return false;
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}
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actualPeriodFrames_ = reportedBufferSize_; // we don't have a better estimate.
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hr = audioClient_->SetEventHandle(audioEvent_);
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if (FAILED(hr)) {
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audioClient_.Reset();
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CoTaskMemFree(format_);
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format_ = nullptr;
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SetErrorString("AudioClient SetEventHandle failed", hr);
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return false;
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}
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hr = audioClient_->GetService(IID_PPV_ARGS(&renderClient_));
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if (FAILED(hr)) {
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audioClient_.Reset();
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CoTaskMemFree(format_);
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format_ = nullptr;
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SetErrorString("AudioClient GetService failed", hr);
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return false;
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}
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if (createBuffer) {
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tempBuf_ = std::make_unique<float[]>(reportedBufferSize_ * 2);
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}
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return true;
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}
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bool WASAPIContext::InitOutputDevice(std::string_view uniqueId, LatencyMode latencyMode, bool *revertedToDefault) {
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Stop();
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*revertedToDefault = false;
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ComPtr<IMMDevice> device;
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if (uniqueId.empty()) {
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// Use the default device.
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HRESULT hr = enumerator_->GetDefaultAudioEndpoint(eRender, eConsole, &device);
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if (FAILED(hr)) {
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SetErrorString("Failed to get the default endpoint", hr);
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return false;
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}
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} else {
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// Use whatever device.
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std::wstring wId = ConvertUTF8ToWString(uniqueId);
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HRESULT hr = enumerator_->GetDevice(wId.c_str(), &device);
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if (FAILED(hr)) {
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// Fallback to default device
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INFO_LOG(Log::Audio, "Falling back to default device...\n");
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*revertedToDefault = true;
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hr = enumerator_->GetDefaultAudioEndpoint(eRender, eConsole, &device);
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if (FAILED(hr)) {
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SetErrorString("Failed to fallback", hr);
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return false;
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}
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}
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}
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AudioDeviceDesc desc{};
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GetDeviceDesc(device.Get(), &desc);
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INFO_LOG(Log::Audio, "Activating audio device: %s : %s", desc.name.c_str(), desc.uniqueId.c_str());
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{
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std::lock_guard<std::mutex> guard(deviceLock_);
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curDeviceId_ = desc.uniqueId;
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curDeviceName_ = desc.name;
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}
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// Get rid of any old tempBuf_.
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tempBuf_.reset();
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// This is used by both paths.
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audioEvent_ = CreateEvent(nullptr, FALSE, FALSE, nullptr);
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if (!TryInitAudioClient3(device.Get(), latencyMode)) {
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if (!TryInitAudioClient(device.Get(), latencyMode)) {
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// Failed both client types.
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CloseHandle(audioEvent_);
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audioEvent_ = nullptr;
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return false;
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}
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}
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latencyMode_ = latencyMode;
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_dbg_assert_(audioClient_ || audioClient3_);
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Start();
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return true;
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}
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void WASAPIContext::Start() {
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if (audioThread_.joinable()) {
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_dbg_assert_(false);
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ERROR_LOG(Log::Audio, "Audio thread already running!");
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return;
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}
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running_ = true;
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audioThread_ = std::thread([this]() { AudioLoop(); });
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}
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void WASAPIContext::Stop() {
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running_ = false;
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if (audioEvent_) SetEvent(audioEvent_);
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// Stop is actually called on the audioclient in the thread, while exiting.
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if (audioThread_.joinable()) audioThread_.join();
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renderClient_.Reset();
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audioClient_.Reset();
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audioClient3_.Reset();
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if (audioEvent_) {
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CloseHandle(audioEvent_);
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audioEvent_ = nullptr;
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}
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if (format_) {
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CoTaskMemFree(format_);
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format_ = nullptr;
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}
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{
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std::lock_guard<std::mutex> guard(deviceLock_);
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curDeviceId_.clear();
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curDeviceName_.clear();
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}
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}
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void WASAPIContext::FrameUpdate(bool allowAutoChange) {
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std::string deviceIdToInit;
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{
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std::lock_guard<std::mutex> guard(deviceLock_);
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if (!defaultDeviceChanged_) {
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return;
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}
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if (allowAutoChange) {
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// Check if there actually was a change, we ignore false positives.
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{
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if (newDeviceId_ == curDeviceId_) {
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// False positive, ignore.
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defaultDeviceChanged_ = false;
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return;
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}
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deviceIdToInit = newDeviceId_;
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newDeviceId_.clear();
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}
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defaultDeviceChanged_ = false;
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}
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}
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bool reverted;
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InitOutputDevice(deviceIdToInit, latencyMode_, &reverted);
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}
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void WASAPIContext::AudioLoop() {
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SetCurrentThreadName("WASAPIAudioLoop");
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DWORD taskID = 0;
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HANDLE mmcssHandle = nullptr;
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if (latencyMode_ == LatencyMode::Aggressive) {
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mmcssHandle = AvSetMmThreadCharacteristics(L"Pro Audio", &taskID);
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}
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UINT32 available;
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HRESULT hr;
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if (audioClient3_) {
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hr = audioClient3_->Start();
|
||
if (FAILED(hr)) {
|
||
SetErrorString("AudioClient3::Start failed", hr);
|
||
return;
|
||
}
|
||
hr = audioClient3_->GetBufferSize(&available);
|
||
if (FAILED(hr)) {
|
||
SetErrorString("AudioClient3::GetBufferSize failed", hr);
|
||
audioClient3_->Stop();
|
||
return;
|
||
}
|
||
} else if (audioClient_) {
|
||
hr = audioClient_->Start();
|
||
if (FAILED(hr)) {
|
||
SetErrorString("AudioClient::Start failed", hr);
|
||
return;
|
||
}
|
||
hr = audioClient_->GetBufferSize(&available);
|
||
if (FAILED(hr)) {
|
||
SetErrorString("AudioClient::GetBufferSize failed", hr);
|
||
audioClient_->Stop();
|
||
return;
|
||
}
|
||
} else {
|
||
// No audio client, nothing to do.
|
||
SetErrorString("No audio client in AudioLoop", 0);
|
||
return;
|
||
}
|
||
|
||
if (!format_) {
|
||
ERROR_LOG(Log::Audio, "Can't start audio - no format");
|
||
return;
|
||
}
|
||
|
||
const AudioFormat format = Classify(format_);
|
||
const int nChannels = format_->nChannels;
|
||
|
||
ClearErrorString();
|
||
|
||
while (running_) {
|
||
const DWORD waitResult = WaitForSingleObject(audioEvent_, INFINITE);
|
||
if (waitResult != WAIT_OBJECT_0) {
|
||
// Something bad happened.
|
||
break;
|
||
}
|
||
|
||
UINT32 padding = 0;
|
||
if (audioClient3_) {
|
||
audioClient3_->GetCurrentPadding(&padding);
|
||
} else {
|
||
audioClient_->GetCurrentPadding(&padding);
|
||
}
|
||
|
||
const UINT32 framesToWrite = available - padding;
|
||
BYTE* buffer = nullptr;
|
||
if (framesToWrite > 0 && SUCCEEDED(renderClient_->GetBuffer(framesToWrite, &buffer))) {
|
||
if (!tempBuf_) {
|
||
// Mix directly to the output buffer, avoiding a copy.
|
||
if (buffer) {
|
||
callback_(reinterpret_cast<float *>(buffer), framesToWrite, format_->nSamplesPerSec, userdata_);
|
||
}
|
||
} else {
|
||
// We decided previously that we need conversion, so mix to our temp buffer...
|
||
callback_(tempBuf_.get(), framesToWrite, format_->nSamplesPerSec, userdata_);
|
||
// .. and convert according to format (we support multi-channel float and s16)
|
||
if (format == AudioFormat::PCM16 && buffer) {
|
||
// Need to convert.
|
||
s16 *dest = reinterpret_cast<s16 *>(buffer);
|
||
for (UINT32 i = 0; i < framesToWrite; i++) {
|
||
if (nChannels == 1) {
|
||
// Maybe some bluetooth speakers? Mixdown.
|
||
float sum = 0.5f * (tempBuf_[i * 2] + tempBuf_[i * 2 + 1]);
|
||
dest[i] = ClampFloatToS16(sum);
|
||
} else {
|
||
dest[i * nChannels] = ClampFloatToS16(tempBuf_[i * 2]);
|
||
dest[i * nChannels + 1] = ClampFloatToS16(tempBuf_[i * 2 + 1]);
|
||
// Zero other channels.
|
||
for (int j = 2; j < nChannels; j++) {
|
||
dest[i * nChannels + j] = 0;
|
||
}
|
||
}
|
||
}
|
||
} else if (format == AudioFormat::Float && buffer) {
|
||
// We have a non-2 number of channels (since we're in the tempBuf_ 'if'), so we contract/expand.
|
||
float *dest = reinterpret_cast<float *>(buffer);
|
||
for (UINT32 i = 0; i < framesToWrite; i++) {
|
||
if (nChannels == 1) {
|
||
// Maybe some bluetooth speakers? Mixdown.
|
||
dest[i] = 0.5f * (tempBuf_[i * 2] + tempBuf_[i * 2 + 1]);
|
||
} else {
|
||
dest[i * nChannels] = tempBuf_[i * 2];
|
||
dest[i * nChannels + 1] = tempBuf_[i * 2 + 1];
|
||
// Zero other channels.
|
||
for (int j = 2; j < nChannels; j++) {
|
||
dest[i * nChannels + j] = 0;
|
||
}
|
||
}
|
||
}
|
||
}
|
||
}
|
||
|
||
renderClient_->ReleaseBuffer(framesToWrite, 0);
|
||
}
|
||
|
||
// In the old mode, we just estimate the "actualPeriodFrames_" from the framesToWrite.
|
||
if (audioClient_ && framesToWrite < actualPeriodFrames_) {
|
||
actualPeriodFrames_ = framesToWrite;
|
||
}
|
||
}
|
||
|
||
if (audioClient3_) {
|
||
audioClient3_->Stop();
|
||
}
|
||
if (audioClient_) {
|
||
audioClient_->Stop();
|
||
}
|
||
|
||
if (mmcssHandle) {
|
||
AvRevertMmThreadCharacteristics(mmcssHandle);
|
||
}
|
||
}
|
||
|
||
void WASAPIContext::DescribeOutputFormat(char *buffer, size_t bufferSize) const {
|
||
if (!format_) {
|
||
snprintf(buffer, bufferSize, "No format");
|
||
return;
|
||
}
|
||
const int numChannels = format_->nChannels;
|
||
const int sampleBits = format_->wBitsPerSample;
|
||
const int sampleRateHz = format_->nSamplesPerSec;
|
||
const char *fmt = "N/A";
|
||
if (format_->wFormatTag == WAVE_FORMAT_EXTENSIBLE) {
|
||
const WAVEFORMATEXTENSIBLE *ex = (const WAVEFORMATEXTENSIBLE *)format_;
|
||
if (ex->SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT) {
|
||
fmt = "float";
|
||
} else {
|
||
fmt = "PCM";
|
||
}
|
||
} else {
|
||
fmt = "PCM"; // probably
|
||
}
|
||
snprintf(buffer, bufferSize, "%d Hz %s %d-bit, %d ch%s", sampleRateHz, fmt, sampleBits, numChannels, audioClient3_ ? " (ac3)" : " (ac)");
|
||
}
|
||
|
||
|
||
HRESULT STDMETHODCALLTYPE WASAPIContext::DeviceNotificationClient::OnDefaultDeviceChanged(EDataFlow flow, ERole role, LPCWSTR device) {
|
||
if (flow != eRender) {
|
||
INFO_LOG(Log::Audio, "Default WASAPI audio recording device changed! Currently ignoring.");
|
||
return S_OK;
|
||
}
|
||
INFO_LOG(Log::Audio, "Default device changed to %s! role=%d", ConvertWStringToUTF8(device).c_str(), role);
|
||
if (role == eConsole) {
|
||
// PostMessage(hwnd, WM_APP + 1, 0, 0);
|
||
std::lock_guard<std::mutex> guard(engine_->deviceLock_);
|
||
engine_->defaultDeviceChanged_ = true;
|
||
engine_->newDeviceId_ = ConvertWStringToUTF8(device);
|
||
}
|
||
return S_OK;
|
||
}
|
||
|
||
HRESULT STDMETHODCALLTYPE WASAPIContext::DeviceNotificationClient::OnDeviceAdded(LPCWSTR device) {
|
||
INFO_LOG(Log::Audio, "Audio device added! device=%s", ConvertWStringToUTF8(device).c_str());
|
||
return S_OK;
|
||
}
|
||
|
||
HRESULT STDMETHODCALLTYPE WASAPIContext::DeviceNotificationClient::OnDeviceRemoved(LPCWSTR device) {
|
||
INFO_LOG(Log::Audio, "Audio device removed! device=%s", ConvertWStringToUTF8(device).c_str());
|
||
return S_OK;
|
||
}
|
||
|
||
HRESULT STDMETHODCALLTYPE WASAPIContext::DeviceNotificationClient::OnDeviceStateChanged(LPCWSTR device, DWORD state) {
|
||
INFO_LOG(Log::Audio, "Audio device state changed! device=%s state=%08x", ConvertWStringToUTF8(device).c_str(), state);
|
||
return S_OK;
|
||
}
|
||
|
||
HRESULT STDMETHODCALLTYPE WASAPIContext::DeviceNotificationClient::OnPropertyValueChanged(LPCWSTR device, const PROPERTYKEY key) {
|
||
return S_OK;
|
||
}
|