#include "audio_capturer.h" #include "basic/basic.h" #define DEFAULT_SAMPLE_RATE 48000 // 默认采样率:48kHz #define DEFAULT_BITS_PER_SAMPLE 16 // 默认位深:16bit #define DEFAULT_CHANNELS 1 // 默认音频通道数:1 #define DEFAULT_AUDIO_PACKET_INTERVAL 10 // 默认音频包发送间隔:10ms bool AudioCapturer::Init(Type deviceType, CallBack callback, void* userInfo) { Stop(); _userInfo = userInfo; _callback = callback; _deviceType = deviceType; __CheckBool(_CreateDeviceEnumerator(&_pDeviceEnumerator)); __CheckBool(_CreateDevice(_pDeviceEnumerator, &_pDevice)); __CheckBool(_CreateAudioClient(_pDevice, &_pAudioClient)); if (!_IsFormatSupported(_pAudioClient)) { __CheckBool(_GetPreferFormat(_pAudioClient, &_formatex)); } __CheckBool(_InitAudioClient(_pAudioClient, &_formatex)); __CheckBool(_CreateAudioCaptureClient(_pAudioClient, &_pAudioCaptureClient)); _isInit = true; return true; } bool AudioCapturer::Start() { __CheckBool(_isInit); _loopFlag = true; // 用于强制打开扬声器 PlaySoundA("./rc/mute.wav", nullptr, SND_FILENAME | SND_ASYNC | SND_LOOP); _captureThread = new std::thread( [this] { _ThreadRun(_pAudioClient, _pAudioCaptureClient); }); return true; } void AudioCapturer::Stop() { // CoUninitialize(); _isInit = false; _loopFlag = false; Free(_captureThread, [this] { _captureThread->join(); delete _captureThread; }); Free(_pAudioCaptureClient, [this] { _pAudioCaptureClient->Release(); }); if (_pAudioClient != nullptr) { _pAudioClient->Stop(); } PlaySoundA(nullptr, nullptr, SND_FILENAME | SND_ASYNC | SND_LOOP); Free(_pAudioClient, [this] { _pAudioClient->Release(); }); Free(_pDevice, [this] { _pDevice->Release(); }); Free(_pDeviceEnumerator, [this] { _pDeviceEnumerator->Release(); }); } bool AudioCapturer::_CreateDeviceEnumerator(IMMDeviceEnumerator** enumerator) { // __CheckBool(SUCCEEDED(CoInitializeEx(nullptr, COINIT_MULTITHREADED))); // __CheckBool(SUCCEEDED(CoInitializeEx(nullptr, COINIT_APARTMENTTHREADED))); __CheckBool(SUCCEEDED(CoCreateInstance(__uuidof(MMDeviceEnumerator), NULL, CLSCTX_ALL, __uuidof(IMMDeviceEnumerator), reinterpret_cast(enumerator)))); return true; } bool AudioCapturer::_CreateDevice(IMMDeviceEnumerator* enumerator, IMMDevice** device) { EDataFlow enDataFlow = _deviceType == Microphone ? eCapture : eRender; ERole enRole = eConsole; __CheckBool(SUCCEEDED(enumerator->GetDefaultAudioEndpoint(enDataFlow, enRole, device))); return true; } bool AudioCapturer::_CreateAudioClient(IMMDevice* device, IAudioClient** audioClient) { __CheckBool(SUCCEEDED(device->Activate(__uuidof(IAudioClient), CLSCTX_ALL, NULL, (void**)audioClient))); return true; } bool AudioCapturer::_IsFormatSupported(IAudioClient* audioClient) { memset(&_formatex, 0, sizeof(_formatex)); WAVEFORMATEX* format = &_formatex.Format; format->nSamplesPerSec = DEFAULT_SAMPLE_RATE; format->wBitsPerSample = DEFAULT_BITS_PER_SAMPLE; format->nChannels = DEFAULT_CHANNELS; WAVEFORMATEX* closestMatch = nullptr; HRESULT hr = audioClient->IsFormatSupported(AUDCLNT_SHAREMODE_SHARED, format, &closestMatch); if (hr == AUDCLNT_E_UNSUPPORTED_FORMAT) // 0x88890008 { if (closestMatch == nullptr) // 如果找不到最相近的格式,closestMatch可能为nullptr { return false; } format->nSamplesPerSec = closestMatch->nSamplesPerSec; format->wBitsPerSample = closestMatch->wBitsPerSample; format->nChannels = closestMatch->nChannels; return true; } return false; } bool AudioCapturer::_GetPreferFormat(IAudioClient* audioClient, WAVEFORMATEXTENSIBLE* formatex) { WAVEFORMATEX* format = nullptr; __CheckBool(SUCCEEDED(audioClient->GetMixFormat(&format))); formatex->Format.nSamplesPerSec = format->nSamplesPerSec; formatex->Format.wBitsPerSample = format->wBitsPerSample; formatex->Format.nChannels = format->nChannels; return true; } bool AudioCapturer::_InitAudioClient(IAudioClient* audioClient, WAVEFORMATEXTENSIBLE* formatex) { AUDCLNT_SHAREMODE shareMode = AUDCLNT_SHAREMODE_SHARED; // share Audio Engine with other applications DWORD streamFlags = _deviceType == Microphone ? 0 : AUDCLNT_STREAMFLAGS_LOOPBACK; streamFlags |= AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM; // A channel matrixer and a sample // rate converter are inserted streamFlags |= AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY; // a sample rate converter // with better quality than // the default conversion but // with a higher performance // cost is used REFERENCE_TIME hnsBufferDuration = 0; WAVEFORMATEX* format = &formatex->Format; format->wFormatTag = WAVE_FORMAT_EXTENSIBLE; format->nBlockAlign = (format->wBitsPerSample >> 3) * format->nChannels; format->nAvgBytesPerSec = format->nBlockAlign * format->nSamplesPerSec; format->cbSize = sizeof(WAVEFORMATEXTENSIBLE) - sizeof(WAVEFORMATEX); formatex->Samples.wValidBitsPerSample = format->wBitsPerSample; formatex->dwChannelMask = format->nChannels == 1 ? KSAUDIO_SPEAKER_MONO : KSAUDIO_SPEAKER_STEREO; formatex->SubFormat = KSDATAFORMAT_SUBTYPE_PCM; __CheckBool(SUCCEEDED(audioClient->Initialize(shareMode, streamFlags, hnsBufferDuration, 0, format, nullptr))); return true; } bool AudioCapturer::_CreateAudioCaptureClient(IAudioClient* audioClient, IAudioCaptureClient** audioCaptureClient) { __CheckBool(SUCCEEDED(audioClient->GetService(IID_PPV_ARGS(audioCaptureClient)))); return true; } bool AudioCapturer::_ThreadRun(IAudioClient* audio_client, IAudioCaptureClient* audio_capture_client) { UINT32 num_success = 0; BYTE* p_audio_data = nullptr; UINT32 num_frames_to_read = 0; DWORD dw_flag = 0; UINT32 num_frames_in_next_packet = 0; audio_client->Start(); while (_loopFlag) { SleepMs(5); while (true) { __CheckBool(SUCCEEDED(audio_capture_client->GetNextPacketSize(&num_frames_in_next_packet))); if (num_frames_in_next_packet == 0) { break; } __CheckBool(SUCCEEDED(audio_capture_client->GetBuffer(&p_audio_data, &num_frames_to_read, &dw_flag, nullptr, nullptr))); size_t size = (_formatex.Format.wBitsPerSample >> 3) * _formatex.Format.nChannels * num_frames_to_read; _callback(p_audio_data, size, _userInfo); __CheckBool(SUCCEEDED(audio_capture_client->ReleaseBuffer(num_frames_to_read))); } } audio_client->Stop(); return true; }