我正在尝试使用WASPAI接口通过默认的音频端点渲染器播放噪音。我正在尝试使用Microsoft在此页面上提供的代码：https://learn.microsoft.com/en-us/windows/win32/coreaudio/rendering-a-stream.我想为这个代码示例编写一个可以生成噪声的类。
我已经尝试将有符号和无符号整数值写入默认音频端点渲染器的缓冲区，并看到值正在写入缓冲区，但没有声音播放。
首先，我用所需的方法和一个随机数生成器制作了一个标头。
#pragma once
// RNG
#include <random>
template <typename T>
class Random {
public:
Random(T low, T high) : mLow(low), mHigh(high), function(std::mt19937_64(__rdtsc())) {};
T operator()() {
signed __int64 f = function();
return ((f % ((signed __int64) mHigh + (signed __int64) mLow)) + (signed __int64) mLow); }
private:
T mLow;
T mHigh;
std::mt19937_64 function;
};
class Noise_Gen {
public:
Noise_Gen() : nChannels(NULL), nSamplesPerSec(NULL), nAvgBytesPerSec(NULL), nByteAlign(NULL), wBitsPerSample(NULL),
wValidBitsPerSample(NULL), wSamplesPerBlock(NULL), dwChannelMask(NULL), rd(NULL) {};
~Noise_Gen() {
if(rd != NULL) {
delete rd;
}
};
HRESULT SetFormat(WAVEFORMATEX*);
HRESULT LoadData(UINT32 bufferFrameCount, BYTE* pData, DWORD* flags);
private:
void* rd;
// WAVEFORMATEX
WORD nChannels;
DWORD nSamplesPerSec;
DWORD nAvgBytesPerSec;
WORD nByteAlign;
WORD wBitsPerSample;
// WAVEFORMATEXTENSIBLE
WORD wValidBitsPerSample;
WORD wSamplesPerBlock;
DWORD dwChannelMask;
};
然后我添加了定义：
// WASAPI
#include <Audiopolicy.h>
#include <Audioclient.h>
#include <time.h>
#include "Noise_Gen.h"
HRESULT Noise_Gen::SetFormat(WAVEFORMATEX* format) {
nChannels = format->nChannels;
nSamplesPerSec = format->nSamplesPerSec;
nAvgBytesPerSec = format->nAvgBytesPerSec;
nByteAlign = format->nBlockAlign;
wBitsPerSample = format->wBitsPerSample;
WORD wFormatTag = format->wFormatTag;
if(wFormatTag == WAVE_FORMAT_EXTENSIBLE) {
WAVEFORMATEXTENSIBLE* pWFE = reinterpret_cast<WAVEFORMATEXTENSIBLE*>(format);
wValidBitsPerSample = pWFE->Samples.wValidBitsPerSample;
wSamplesPerBlock = pWFE->Samples.wSamplesPerBlock;
dwChannelMask = pWFE->dwChannelMask;
} else {
wValidBitsPerSample = wBitsPerSample;
}
double amplitude = std::pow(2.0, wValidBitsPerSample) - 1;
switch(wBitsPerSample / 8) {
case(1):
rd = new Random<unsigned __int8>(0.0, amplitude);
break;
case(2):
rd = new Random<unsigned __int16>(0.0, amplitude);
break;
case(3):
rd = new Random<unsigned __int32>(0.0, amplitude);
break;
case(4):
rd = new Random<signed __int32>(-amplitude, amplitude);
break;
case(5):
rd = new Random<unsigned __int64>(0.0, amplitude);
break;
case(6):
rd = new Random<unsigned __int64>(0.0, amplitude);
break;
case(7):
rd = new Random<unsigned __int64>(0.0, amplitude);
break;
case(8):
rd = new Random<unsigned __int64>(0.0, amplitude);
break;
default:
return E_NOTIMPL;
}
return S_OK;
}
// (The size of an audio frame = nChannels * wBitsPerSample)
HRESULT Noise_Gen::LoadData(UINT32 bufferFrameCount, BYTE* pData, DWORD* flags) {
for(UINT32 i = 0; i < nChannels *bufferFrameCount; i++) {
switch(wBitsPerSample / 8) {
case(1):
pData[i] = (((Random<unsigned __int8>*)rd)->operator()());
break;
case(2):{
unsigned __int16* pData2 = (unsigned __int16*) pData;
pData2[i] = (((Random<unsigned __int16>*)rd)->operator()());
break;
}
case(3): {
__int32 data = ((Random<unsigned __int32>*)rd)->operator()();
unsigned char* cp = (unsigned char*) (&data);
pData[(3 * i)] = cp[0];
pData[1 + (3 * i)] = cp[1];
pData[2 + (3 * i)] = cp[2];
break;
}
case(4):{
signed __int32* pData2 = (signed __int32*) pData;
pData2[i] = (((Random<signed __int32>*)rd)->operator()());
break;
}
case(5): {
__int64 data = ((Random<unsigned __int64>*)rd)->operator()();
unsigned char* cp = (unsigned char*) &data;
pData[(5 * i)] = cp[0];
pData[1 + (5 * i)] = cp[1];
pData[2 + (5 * i)] = cp[2];
pData[3 + (5 * i)] = cp[3];
pData[4 + (5 * i)] = cp[4];
break;
}
case(6): {
__int64 data = ((Random<unsigned __int64>*)rd)->operator()();
unsigned char* cp = (unsigned char*) &data;
pData[(6 * i)] = cp[0];
pData[1 + (6 * i)] = cp[1];
pData[2 + (6 * i)] = cp[2];
pData[3 + (6 * i)] = cp[3];
pData[4 + (6 * i)] = cp[4];
pData[5 + (6 * i)] = cp[5];
break;
}
case(7): {
__int64 data = ((Random<unsigned __int64>*)rd)->operator()();
unsigned char* cp = (unsigned char*) &data;
pData[(7 * i)] = cp[0];
pData[1 + (7 * i)] = cp[1];
pData[2 + (7 * i)] = cp[2];
pData[3 + (7 * i)] = cp[3];
pData[4 + (7 * i)] = cp[4];
pData[5 + (7 * i)] = cp[5];
pData[6 + (7 * i)] = cp[6];
break;
}
case(8): {
unsigned __int64* pData2 = (unsigned __int64*) pData;
pData2[i] = (((Random<unsigned __int64>*)rd)->operator()());
break;
}
default:
// For stopping playback
(*flags) = AUDCLNT_BUFFERFLAGS_SILENT;
return E_NOTIMPL;
}
}
return S_OK;
}
然后，我将我的类添加到Microsoft提供的模板中，并将默认的音频端点渲染器打印到控制台中。
#include <InitGuid.h>
#include <iostream>
#include <Windows.h>
#include <dshow.h>
// Windows multimedia device
#include <Mmdeviceapi.h>
#include <Functiondiscoverykeys_devpkey.h>
// WASAPI
#include <Audiopolicy.h>
#include <Audioclient.h>
#include "Noise_Gen.h"
//-----------------------------------------------------------
// Play an audio stream on the default audio rendering
// device. The PlayAudioStream function allocates a shared
// buffer big enough to hold one second of PCM audio data.
// The function uses this buffer to stream data to the
// rendering device. The inner loop runs every 1/2 second.
//-----------------------------------------------------------
// REFERENCE_TIME time units per second and per millisecond
#define REFTIMES_PER_SEC 10000000
#define REFTIMES_PER_MILLISEC 10000
#define EXIT_ON_ERROR(hres)
if (FAILED(hres)) { goto Exit; }
#define SAFE_RELEASE(punk)
if ((punk) != NULL)
{ (punk)->Release(); (punk) = NULL; }
const CLSID CLSID_MMDeviceEnumerator = __uuidof(MMDeviceEnumerator);
const IID IID_IMMDeviceEnumerator = __uuidof(IMMDeviceEnumerator);
const IID IID_IAudioClient = __uuidof(IAudioClient);
const IID IID_IAudioRenderClient = __uuidof(IAudioRenderClient);
HRESULT PlayAudioStream(Noise_Gen* pMySource) {
HRESULT hr;
REFERENCE_TIME hnsRequestedDuration = REFTIMES_PER_SEC;
REFERENCE_TIME hnsActualDuration;
IMMDeviceEnumerator* pEnumerator = NULL;
IMMDevice* pDevice = NULL;
IAudioClient* pAudioClient = NULL;
IAudioRenderClient* pRenderClient = NULL;
WAVEFORMATEX* pwfx = NULL;
UINT32 bufferFrameCount;
UINT32 numFramesAvailable;
UINT32 numFramesPadding;
BYTE* pData;
DWORD flags = 0;
IPropertyStore* pPropertyStore = NULL;
PROPVARIANT name;
hr = CoCreateInstance(CLSID_MMDeviceEnumerator, NULL,
CLSCTX_ALL, IID_IMMDeviceEnumerator,
(void**) &pEnumerator);
EXIT_ON_ERROR(hr);
hr = pEnumerator->GetDefaultAudioEndpoint(
eRender, eConsole, &pDevice);
hr = pDevice->OpenPropertyStore(STGM_READ, &pPropertyStore);
PropVariantInit(&name);
hr = pPropertyStore->GetValue(PKEY_Device_FriendlyName, &name);
printf("%S", name.pwszVal);
printf("n");
EXIT_ON_ERROR(hr);
hr = pDevice->Activate(IID_IAudioClient, CLSCTX_ALL,
NULL, (void**) &pAudioClient);
EXIT_ON_ERROR(hr);
hr = pAudioClient->GetMixFormat(&pwfx);
EXIT_ON_ERROR(hr);
hr = pAudioClient->Initialize(AUDCLNT_SHAREMODE_SHARED,
0, hnsRequestedDuration,
0, pwfx, NULL);
EXIT_ON_ERROR(hr);
// Tell the audio source which format to use.
hr = pMySource->SetFormat(pwfx);
EXIT_ON_ERROR(hr);
// Get the actual size of the allocated buffer.
hr = pAudioClient->GetBufferSize(&bufferFrameCount);
EXIT_ON_ERROR(hr);
hr = pAudioClient->GetService(IID_IAudioRenderClient,
(void**) &pRenderClient);
EXIT_ON_ERROR(hr);
// Grab the entire buffer for the initial fill operation.
hr = pRenderClient->GetBuffer(bufferFrameCount, &pData);
EXIT_ON_ERROR(hr);
// Load the initial data into the shared buffer.
hr = pMySource->LoadData(bufferFrameCount, pData, &flags);
EXIT_ON_ERROR(hr);
hr = pRenderClient->ReleaseBuffer(bufferFrameCount, flags);
EXIT_ON_ERROR(hr);
// Calculate the actual duration of the allocated buffer.
hnsActualDuration = (double) REFTIMES_PER_SEC * bufferFrameCount / pwfx->nSamplesPerSec;
hr = pAudioClient->Start(); // Start playing.
EXIT_ON_ERROR(hr);
// Each loop fills about half of the shared buffer.
while(flags != AUDCLNT_BUFFERFLAGS_SILENT) {
// Sleep for half the buffer duration.
Sleep((DWORD) (hnsActualDuration / REFTIMES_PER_MILLISEC / 2));
// See how much buffer space is available.
hr = pAudioClient->GetCurrentPadding(&numFramesPadding);
EXIT_ON_ERROR(hr);
numFramesAvailable = bufferFrameCount - numFramesPadding;
// Grab all the available space in the shared buffer.
hr = pRenderClient->GetBuffer(numFramesAvailable, &pData);
EXIT_ON_ERROR(hr);
// Get next 1/2-second of data from the audio source.
hr = pMySource->LoadData(numFramesAvailable, pData, &flags);
EXIT_ON_ERROR(hr);
hr = pRenderClient->ReleaseBuffer(numFramesAvailable, flags);
EXIT_ON_ERROR(hr);
}
// Wait for last data in buffer to play before stopping.
Sleep((DWORD) (hnsActualDuration / REFTIMES_PER_MILLISEC / 2));
hr = pAudioClient->Stop(); // Stop playing.
EXIT_ON_ERROR(hr);
Exit:
CoTaskMemFree(pwfx);
SAFE_RELEASE(pEnumerator);
SAFE_RELEASE(pDevice);
SAFE_RELEASE(pAudioClient);
SAFE_RELEASE(pRenderClient);
return hr;
}
int main() {
HRESULT hr = CoInitialize(nullptr);
if(FAILED(hr)) { return hr; }
Noise_Gen* ng = new Noise_Gen();
PlayAudioStream(ng);
delete ng;
CoUninitialize();
}
我的系统上默认的音频端点渲染器使用32位值，因此代码是通过将未签名的32位值写入缓冲区开始的。然后我尝试使用带符号的值，这可以在上面的代码中看到。在这两种情况下都没有播放任何声音。我在调试时检查了缓冲区的内容，它们确实发生了变化。我将默认的音频端点渲染器打印到控制台，它是我系统的扬声器。Windows甚至在音量混合器中显示我的应用程序，但即使音量一直调高，也没有声音显示。然后我检查了睡眠时间，以确保它处于睡眠状态，这样系统就可以访问缓冲区，并且在写入缓冲区之间会睡眠500毫秒。
更新：我发现我使用的是KSDATAFORMAT_SUBTYPE_IEEE_FLOAT子格式，并尝试在-幅度到幅度、0到幅度、-1到1和0到1范围内馈送缓冲浮动。
我错过了什么？
您的随机数分布代码无法正确处理浮点格式(据我所知，浮点格式基本上总是共享模式下的混合格式(。
即使是整数也是错误的。我想你是想写
((f % ((signed __int64) mHigh - (signed __int64) mLow)) + (signed __int64) mLow);
(注意负号(，但无论如何都不应该使用原始模量，因为它有点偏。
对于浮点格式，您总是使用-1到1的范围。
我已经调整了您的代码以使用std:：uniform_real_distribution，我的扬声器上会播放噪音。
#include <cstdio>
#include <Windows.h>
// Windows multimedia device
#include <Mmdeviceapi.h>
#include <Functiondiscoverykeys_devpkey.h>
// WASAPI
#include <Audiopolicy.h>
#include <Audioclient.h>
#include <random>
class Noise_Gen {
public:
Noise_Gen() : format(), engine(__rdtsc()), float_dist(-1.f, 1.f) {};
void SetFormat(WAVEFORMATEX* wfex) {
if(wfex->wFormatTag == WAVE_FORMAT_EXTENSIBLE) {
format = *reinterpret_cast<WAVEFORMATEXTENSIBLE*>(wfex);
} else {
format.Format = *wfex;
format.Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
INIT_WAVEFORMATEX_GUID(&format.SubFormat, wfex->wFormatTag);
format.Samples.wValidBitsPerSample = format.Format.wBitsPerSample;
format.dwChannelMask = 0;
}
}
// (The size of an audio frame = nChannels * wBitsPerSample)
void FillBuffer(UINT32 bufferFrameCount, BYTE* pData, DWORD* flags) {
const UINT16 formatTag = EXTRACT_WAVEFORMATEX_ID(&format.SubFormat);
if(formatTag == WAVE_FORMAT_IEEE_FLOAT) {
float* fData = (float*)pData;
for(UINT32 i = 0; i < format.Format.nChannels * bufferFrameCount; i++) {
fData[i] = float_dist(engine);
}
} else if(formatTag == WAVE_FORMAT_PCM) {
using rndT = decltype(engine)::result_type;
UINT32 iterations = format.Format.nBlockAlign * bufferFrameCount / sizeof(rndT);
UINT32 leftoverBytes = format.Format.nBlockAlign * bufferFrameCount % sizeof(rndT);
rndT* iData = (rndT*)pData;
UINT32 i = 0;
for(; i < iterations; i++) {
iData[i] = engine();
}
if(leftoverBytes != 0) {
rndT lastRnd = engine();
BYTE* pLastBytes = pData + i * sizeof(rndT);
for(UINT32 j = 0; j < leftoverBytes; ++j) {
pLastBytes[j] = lastRnd >> (j * 8) & 0xFF;
}
}
} else {
//memset(pData, 0, wfex.Format.nBlockAlign * bufferFrameCount);
*flags = AUDCLNT_BUFFERFLAGS_SILENT;
}
}
private:
WAVEFORMATEXTENSIBLE format;
std::mt19937_64 engine;
std::uniform_real_distribution<float> float_dist;
};
// REFERENCE_TIME time units per second and per millisecond
#define REFTIMES_PER_SEC 10000000ll
#define REFTIMES_PER_MILLISEC 10000
#define EXIT_ON_ERROR(hres)
if (FAILED(hres)) { goto Exit; }
#define SAFE_RELEASE(punk)
if ((punk) != NULL)
{ (punk)->Release(); (punk) = NULL; }
HRESULT PlayAudioStream(Noise_Gen* pMySource) {
HRESULT hr;
REFERENCE_TIME hnsRequestedDuration = REFTIMES_PER_SEC;
REFERENCE_TIME hnsActualDuration;
IMMDeviceEnumerator* pEnumerator = NULL;
IPropertyStore* pPropertyStore = NULL;
IMMDevice* pDevice = NULL;
IAudioClient* pAudioClient = NULL;
IAudioRenderClient* pRenderClient = NULL;
WAVEFORMATEX* pwfx = NULL;
UINT32 bufferFrameCount;
BYTE* pData;
DWORD flags = 0;
PROPVARIANT name;
hr = CoCreateInstance(__uuidof(MMDeviceEnumerator), NULL,
CLSCTX_ALL, IID_PPV_ARGS(&pEnumerator));
EXIT_ON_ERROR(hr);
hr = pEnumerator->GetDefaultAudioEndpoint(
eRender, eConsole, &pDevice);
EXIT_ON_ERROR(hr);
hr = pDevice->OpenPropertyStore(STGM_READ, &pPropertyStore);
EXIT_ON_ERROR(hr);
PropVariantInit(&name);
hr = pPropertyStore->GetValue(PKEY_Device_FriendlyName, &name);
EXIT_ON_ERROR(hr);
printf("%S", name.pwszVal);
printf("n");
hr = pDevice->Activate(__uuidof(pAudioClient), CLSCTX_ALL,
NULL, (void**) &pAudioClient);
EXIT_ON_ERROR(hr);
hr = pAudioClient->GetMixFormat(&pwfx);
EXIT_ON_ERROR(hr);
hr = pAudioClient->Initialize(AUDCLNT_SHAREMODE_SHARED,
0, hnsRequestedDuration,
0, pwfx, NULL);
EXIT_ON_ERROR(hr);
// Tell the audio source which format to use.
pMySource->SetFormat(pwfx);
// Get the actual size of the allocated buffer.
hr = pAudioClient->GetBufferSize(&bufferFrameCount);
EXIT_ON_ERROR(hr);
hr = pAudioClient->GetService(IID_PPV_ARGS(&pRenderClient));
EXIT_ON_ERROR(hr);
// Grab the entire buffer for the initial fill operation.
hr = pRenderClient->GetBuffer(bufferFrameCount, &pData);
EXIT_ON_ERROR(hr);
// Load the initial data into the shared buffer.
pMySource->FillBuffer(bufferFrameCount, pData, &flags);
hr = pRenderClient->ReleaseBuffer(bufferFrameCount, flags);
EXIT_ON_ERROR(hr);
// Calculate the actual duration of the allocated buffer.
hnsActualDuration = REFTIMES_PER_SEC * bufferFrameCount / pwfx->nSamplesPerSec;
hr = pAudioClient->Start(); // Start playing.
EXIT_ON_ERROR(hr);
// Each loop fills about half of the shared buffer.
DWORD sleepTime;
while(flags != AUDCLNT_BUFFERFLAGS_SILENT) {
// Sleep for half the buffer duration.
sleepTime = (DWORD) (hnsActualDuration / REFTIMES_PER_MILLISEC / 2);
if(sleepTime != 0)
Sleep(sleepTime);
// See how much buffer space is available.
UINT32 numFramesPadding;
hr = pAudioClient->GetCurrentPadding(&numFramesPadding);
EXIT_ON_ERROR(hr);
UINT32 numFramesAvailable = bufferFrameCount - numFramesPadding;
// Grab all the available space in the shared buffer.
hr = pRenderClient->GetBuffer(numFramesAvailable, &pData);
EXIT_ON_ERROR(hr);
// Get next 1/2-second of data from the audio source.
pMySource->FillBuffer(numFramesAvailable, pData, &flags);
hr = pRenderClient->ReleaseBuffer(numFramesAvailable, flags);
EXIT_ON_ERROR(hr);
}
// Wait for last data in buffer to play before stopping.
sleepTime = (DWORD) (hnsActualDuration / REFTIMES_PER_MILLISEC / 2);
if(sleepTime != 0)
Sleep(sleepTime);
hr = pAudioClient->Stop(); // Stop playing.
EXIT_ON_ERROR(hr);
Exit:
CoTaskMemFree(pwfx);
SAFE_RELEASE(pRenderClient);
SAFE_RELEASE(pAudioClient);
SAFE_RELEASE(pDevice);
SAFE_RELEASE(pPropertyStore); // you forgot to free the property store
SAFE_RELEASE(pEnumerator);
return hr;
}
int main() {
HRESULT hr = CoInitialize(nullptr);
if(FAILED(hr)) { return hr; }
Noise_Gen ng;
PlayAudioStream(&ng);
CoUninitialize();
}