填充 WASAPI 提供的音频端点缓冲区不播放声音
Filling audio endpoint buffer provided by WASAPI not playing sound
我正在尝试使用 WASPAI 接口通过默认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://docs.microsoft.com/en-us/windows/win32/coreaudio/rendering-a-stream。我想写一个 class 可以为此代码示例生成噪声。
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 header。
#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 class 添加到 Microsoft 提供的模板中，并将默认音频端点渲染器打印到控制台。
#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();
}
我系统上的默认音频端点渲染器使用 32 位值，因此代码首先将无符号的 32 位值写入缓冲区。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正在睡眠，这样系统就可以访问缓冲区，并且在写入缓冲区之间确实会休眠 500 毫秒。
更新：我发现我正在使用 KSDATAFORMAT_SUBTYPE_IEEE_FLOAT 子格式并尝试在 -amplitude 到振幅范围、0 到振幅范围、-1 到 1 范围以及0 到 1 范围。
我错过了什么？
您的随机数分配代码无法正确处理浮点格式（据我所知，在共享模式下基本上总是混合格式）。
即使是整数也是错误的。我假设你打算写
((f % ((signed __int64) mHigh - (signed __int64) mLow)) + (signed __int64) mLow);
（注意减号），
但是无论如何你都不应该使用原始模数，因为它有点偏差。
对于浮点格式，您始终使用 -1 到 1 范围。
我已调整您的代码以使用 std::uniform_real_distribution，但我的扬声器上播放噪音。
#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();
}
我正在尝试使用 WASPAI 接口通过默认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://docs.microsoft.com/en-us/windows/win32/coreaudio/rendering-a-stream。我想写一个 class 可以为此代码示例生成噪声。
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 header。
#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 class 添加到 Microsoft 提供的模板中，并将默认音频端点渲染器打印到控制台。
#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();
}
我系统上的默认音频端点渲染器使用 32 位值，因此代码首先将无符号的 32 位值写入缓冲区。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正在睡眠，这样系统就可以访问缓冲区，并且在写入缓冲区之间确实会休眠 500 毫秒。
更新：我发现我正在使用 KSDATAFORMAT_SUBTYPE_IEEE_FLOAT 子格式并尝试在 -amplitude 到振幅范围、0 到振幅范围、-1 到 1 范围以及0 到 1 范围。
我错过了什么？
您的随机数分配代码无法正确处理浮点格式（据我所知，在共享模式下基本上总是混合格式）。
即使是整数也是错误的。我假设你打算写
((f % ((signed __int64) mHigh - (signed __int64) mLow)) + (signed __int64) mLow);
（注意减号）， 但是无论如何你都不应该使用原始模数，因为它有点偏差。
对于浮点格式，您始终使用 -1 到 1 范围。
我已调整您的代码以使用 std::uniform_real_distribution，但我的扬声器上播放噪音。
#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();
}