UWP 的 AvSetMmThreadCharacteristicsW
AvSetMmThreadCharacteristicsW for UWP
我正在使用 cpp/winrt 开发 WASAPI UWP 音频应用程序,它需要从输入中获取音频并在处理后将其发送到输出。
我想用 AvSetMmThreadCharacteristicsW(L"Pro Audio", &taskIndex)
设置我的音频线程特性,但我只是注意到这个功能(以及 avrt.h
的大部分功能)仅限于 WINAPI_PARTITION_DESKTOP
和 WINAPI_PARTITION_GAMES
.
我想我需要这个,因为当我的代码集成到我的 UWP 应用程序中时,音频输入充满了不连续性,而且我使用 avrt
[=37= 的测试代码没有问题].
还有其他方法可以配置我的音频处理线程吗?
编辑:这是我的测试程序https://github.com/loics2/test-wasapi。有趣的部分发生在 AudioStream
class 中。我无法共享我的 UWP 应用程序,但我可以将这些 classes 复制到 Windows 运行时组件中。
编辑 2:这是音频线程代码:
void AudioStream::StreamWorker()
{
WAVEFORMATEX* captureFormat = nullptr;
WAVEFORMATEX* renderFormat = nullptr;
RingBuffer<float> captureBuffer;
RingBuffer<float> renderBuffer;
BYTE* streamBuffer = nullptr;
unsigned int streamBufferSize = 0;
unsigned int bufferFrameCount = 0;
unsigned int numFramesPadding = 0;
unsigned int inputBufferSize = 0;
unsigned int outputBufferSize = 0;
DWORD captureFlags = 0;
winrt::hresult hr = S_OK;
// m_inputClient is a winrt::com_ptr<IAudioClient3>
if (m_inputClient) {
hr = m_inputClient->GetMixFormat(&captureFormat);
// m_audioCaptureClient is a winrt::com_ptr<IAudioCaptureClient>
if (!m_audioCaptureClient) {
hr = m_inputClient->Initialize(
AUDCLNT_SHAREMODE_SHARED,
AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
0,
0,
captureFormat,
nullptr);
hr = m_inputClient->GetService(__uuidof(IAudioCaptureClient), m_audioCaptureClient.put_void());
hr = m_inputClient->SetEventHandle(m_inputReadyEvent.get());
hr = m_inputClient->Reset();
hr = m_inputClient->Start();
}
}
hr = m_inputClient->GetBufferSize(&inputBufferSize);
// multiplying the buffer size by the number of channels
inputBufferSize *= 2;
// m_outputClient is a winrt::com_ptr<IAudioClient3>
if (m_outputClient) {
hr = m_outputClient->GetMixFormat(&renderFormat);
// m_audioRenderClientis a winrt::com_ptr<IAudioRenderClient>
if (!m_audioRenderClient) {
hr = m_outputClient->Initialize(
AUDCLNT_SHAREMODE_SHARED,
AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
0,
0,
captureFormat,
nullptr);
hr = m_outputClient->GetService(__uuidof(IAudioRenderClient), m_audioRenderClient.put_void());
hr = m_outputClient->SetEventHandle(m_outputReadyEvent.get());
hr = m_outputClient->Reset();
hr = m_outputClient->Start();
}
}
hr = m_outputClient->GetBufferSize(&outputBufferSize);
// multiplying the buffer size by the number of channels
outputBufferSize *= 2;
while (m_isRunning)
{
// ===== INPUT =====
// waiting for the capture event
WaitForSingleObject(m_inputReadyEvent.get(), INFINITE);
// getting the input buffer data
hr = m_audioCaptureClient->GetNextPacketSize(&bufferFrameCount);
while (SUCCEEDED(hr) && bufferFrameCount > 0) {
m_audioCaptureClient->GetBuffer(&streamBuffer, &bufferFrameCount, &captureFlags, nullptr, nullptr);
if (bufferFrameCount != 0) {
captureBuffer.write(reinterpret_cast<float*>(streamBuffer), bufferFrameCount * 2);
hr = m_audioCaptureClient->ReleaseBuffer(bufferFrameCount);
if (FAILED(hr)) {
m_audioCaptureClient->ReleaseBuffer(0);
}
}
else
{
m_audioCaptureClient->ReleaseBuffer(0);
}
hr = m_audioCaptureClient->GetNextPacketSize(&bufferFrameCount);
}
// ===== CALLBACK =====
auto size = captureBuffer.size();
float* userInputData = (float*)calloc(size, sizeof(float));
float* userOutputData = (float*)calloc(size, sizeof(float));
captureBuffer.read(userInputData, size);
OnData(userInputData, userOutputData, size / 2, 2, 48000);
renderBuffer.write(userOutputData, size);
free(userInputData);
free(userOutputData);
// ===== OUTPUT =====
// waiting for the render event
WaitForSingleObject(m_outputReadyEvent.get(), INFINITE);
// getting information about the output buffer
hr = m_outputClient->GetBufferSize(&bufferFrameCount);
hr = m_outputClient->GetCurrentPadding(&numFramesPadding);
// adjust the frame count with the padding
bufferFrameCount -= numFramesPadding;
if (bufferFrameCount != 0) {
hr = m_audioRenderClient->GetBuffer(bufferFrameCount, &streamBuffer);
auto count = (bufferFrameCount * 2);
if (renderBuffer.read(reinterpret_cast<float*>(streamBuffer), count) < count) {
// captureBuffer is not full enough, we should fill the remainder with 0
}
hr = m_audioRenderClient->ReleaseBuffer(bufferFrameCount, 0);
if (FAILED(hr)) {
m_audioRenderClient->ReleaseBuffer(0, 0);
}
}
else
{
m_audioRenderClient->ReleaseBuffer(0, 0);
}
}
exit:
// Cleanup code
}
为了清楚起见,我删除了错误处理代码,其中大部分是:
if (FAILED(hr))
goto exit;
@IInspectable 是对的,我的代码有问题:音频处理由一个库完成,然后调用回调函数获得一些结果。
在我的回调中,我尝试引发一个 winrt::event
,但有时需要超过 50 毫秒。当它发生时,它会阻塞音频线程,并造成不连续性...
我正在使用 cpp/winrt 开发 WASAPI UWP 音频应用程序,它需要从输入中获取音频并在处理后将其发送到输出。
我想用 AvSetMmThreadCharacteristicsW(L"Pro Audio", &taskIndex)
设置我的音频线程特性,但我只是注意到这个功能(以及 avrt.h
的大部分功能)仅限于 WINAPI_PARTITION_DESKTOP
和 WINAPI_PARTITION_GAMES
.
我想我需要这个,因为当我的代码集成到我的 UWP 应用程序中时,音频输入充满了不连续性,而且我使用 avrt
[=37= 的测试代码没有问题].
还有其他方法可以配置我的音频处理线程吗?
编辑:这是我的测试程序https://github.com/loics2/test-wasapi。有趣的部分发生在 AudioStream
class 中。我无法共享我的 UWP 应用程序,但我可以将这些 classes 复制到 Windows 运行时组件中。
编辑 2:这是音频线程代码:
void AudioStream::StreamWorker()
{
WAVEFORMATEX* captureFormat = nullptr;
WAVEFORMATEX* renderFormat = nullptr;
RingBuffer<float> captureBuffer;
RingBuffer<float> renderBuffer;
BYTE* streamBuffer = nullptr;
unsigned int streamBufferSize = 0;
unsigned int bufferFrameCount = 0;
unsigned int numFramesPadding = 0;
unsigned int inputBufferSize = 0;
unsigned int outputBufferSize = 0;
DWORD captureFlags = 0;
winrt::hresult hr = S_OK;
// m_inputClient is a winrt::com_ptr<IAudioClient3>
if (m_inputClient) {
hr = m_inputClient->GetMixFormat(&captureFormat);
// m_audioCaptureClient is a winrt::com_ptr<IAudioCaptureClient>
if (!m_audioCaptureClient) {
hr = m_inputClient->Initialize(
AUDCLNT_SHAREMODE_SHARED,
AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
0,
0,
captureFormat,
nullptr);
hr = m_inputClient->GetService(__uuidof(IAudioCaptureClient), m_audioCaptureClient.put_void());
hr = m_inputClient->SetEventHandle(m_inputReadyEvent.get());
hr = m_inputClient->Reset();
hr = m_inputClient->Start();
}
}
hr = m_inputClient->GetBufferSize(&inputBufferSize);
// multiplying the buffer size by the number of channels
inputBufferSize *= 2;
// m_outputClient is a winrt::com_ptr<IAudioClient3>
if (m_outputClient) {
hr = m_outputClient->GetMixFormat(&renderFormat);
// m_audioRenderClientis a winrt::com_ptr<IAudioRenderClient>
if (!m_audioRenderClient) {
hr = m_outputClient->Initialize(
AUDCLNT_SHAREMODE_SHARED,
AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
0,
0,
captureFormat,
nullptr);
hr = m_outputClient->GetService(__uuidof(IAudioRenderClient), m_audioRenderClient.put_void());
hr = m_outputClient->SetEventHandle(m_outputReadyEvent.get());
hr = m_outputClient->Reset();
hr = m_outputClient->Start();
}
}
hr = m_outputClient->GetBufferSize(&outputBufferSize);
// multiplying the buffer size by the number of channels
outputBufferSize *= 2;
while (m_isRunning)
{
// ===== INPUT =====
// waiting for the capture event
WaitForSingleObject(m_inputReadyEvent.get(), INFINITE);
// getting the input buffer data
hr = m_audioCaptureClient->GetNextPacketSize(&bufferFrameCount);
while (SUCCEEDED(hr) && bufferFrameCount > 0) {
m_audioCaptureClient->GetBuffer(&streamBuffer, &bufferFrameCount, &captureFlags, nullptr, nullptr);
if (bufferFrameCount != 0) {
captureBuffer.write(reinterpret_cast<float*>(streamBuffer), bufferFrameCount * 2);
hr = m_audioCaptureClient->ReleaseBuffer(bufferFrameCount);
if (FAILED(hr)) {
m_audioCaptureClient->ReleaseBuffer(0);
}
}
else
{
m_audioCaptureClient->ReleaseBuffer(0);
}
hr = m_audioCaptureClient->GetNextPacketSize(&bufferFrameCount);
}
// ===== CALLBACK =====
auto size = captureBuffer.size();
float* userInputData = (float*)calloc(size, sizeof(float));
float* userOutputData = (float*)calloc(size, sizeof(float));
captureBuffer.read(userInputData, size);
OnData(userInputData, userOutputData, size / 2, 2, 48000);
renderBuffer.write(userOutputData, size);
free(userInputData);
free(userOutputData);
// ===== OUTPUT =====
// waiting for the render event
WaitForSingleObject(m_outputReadyEvent.get(), INFINITE);
// getting information about the output buffer
hr = m_outputClient->GetBufferSize(&bufferFrameCount);
hr = m_outputClient->GetCurrentPadding(&numFramesPadding);
// adjust the frame count with the padding
bufferFrameCount -= numFramesPadding;
if (bufferFrameCount != 0) {
hr = m_audioRenderClient->GetBuffer(bufferFrameCount, &streamBuffer);
auto count = (bufferFrameCount * 2);
if (renderBuffer.read(reinterpret_cast<float*>(streamBuffer), count) < count) {
// captureBuffer is not full enough, we should fill the remainder with 0
}
hr = m_audioRenderClient->ReleaseBuffer(bufferFrameCount, 0);
if (FAILED(hr)) {
m_audioRenderClient->ReleaseBuffer(0, 0);
}
}
else
{
m_audioRenderClient->ReleaseBuffer(0, 0);
}
}
exit:
// Cleanup code
}
为了清楚起见,我删除了错误处理代码,其中大部分是:
if (FAILED(hr))
goto exit;
@IInspectable 是对的,我的代码有问题:音频处理由一个库完成,然后调用回调函数获得一些结果。
在我的回调中,我尝试引发一个 winrt::event
,但有时需要超过 50 毫秒。当它发生时,它会阻塞音频线程,并造成不连续性...