基于C++实现的扩音器音频捕捉器+将PCM文件转为WAV文件源码+sln.zip

#include "VedioCatcher.h" bool g_output_to_console_as_well = false; bool g_loopFlag = false; std::ofstream g_log; WAVEFORMATEXTENSIBLE g_formatex; void InitLog(bool output_to_console) { time_t rawtime; struct tm *timeinfo; char buffer[32] = { 0 }; time(&rawtime); timeinfo = localtime(&rawtime); strftime(buffer, 32, "%Y-%m-%d_%H-%M-%S", timeinfo); std::string logName(buffer); logName = "logs/" + logName + ".log"; g_log.open(logName, std::ios::app); } HRESULT CreateDeviceEnumerator(IMMDeviceEnumerator **enumerator) { CoInitializeEx(nullptr, COINIT_MULTITHREADED); return CoCreateInstance(__uuidof(MMDeviceEnumerator), NULL, CLSCTX_ALL, __uuidof(IMMDeviceEnumerator), reinterpret_cast<void **>(enumerator)); } HRESULT CreateDevice(IMMDeviceEnumerator *enumerator, IMMDevice **device) { EDataFlow enDataFlow = eRender; // 表示获取扬声器的audio_endpoint ERole enRole = eConsole; return enumerator->GetDefaultAudioEndpoint(enDataFlow, enRole, device); } HRESULT CreateAudioClient(IMMDevice *device, IAudioClient **audioClient) { return device->Activate(__uuidof(IAudioClient), CLSCTX_ALL, NULL, (void **)audioClient); } HRESULT IsFormatSupported(IAudioClient *audioClient) { WAVEFORMATEX *format = &g_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 { logd() << "Epxected format: " << "sample_rate[" << format->nSamplesPerSec << "] " << "bits_per_sample[" << format->wBitsPerSample << "] " << "channels[" << format->nChannels << "] " << "\n" << "Supported format: " << "sample_rate[" << closestMatch->nSamplesPerSec << "] " << "bits_per_sample[" << closestMatch->wBitsPerSample << "] " << "channels[" << closestMatch->nChannels << "] " << "\n"; format->nSamplesPerSec = closestMatch->nSamplesPerSec; format->wBitsPerSample = closestMatch->wBitsPerSample; format->nChannels = closestMatch->nChannels; return S_OK; } } return hr; } HRESULT GetPreferFormat(IAudioClient *audioClient, WAVEFORMATEXTENSIBLE *formatex) { WAVEFORMATEX *format = nullptr; HRESULT hr = audioClient->GetMixFormat(&format); if (FAILED(hr)) { return hr; } logd() << "Prefer format: " << "sample_rate[" << format->nSamplesPerSec << "] " << "bits_per_sample[" << format->wBitsPerSample << "] " << "channels[" << format->nChannels << "] " << "\n"; formatex->Format.nSamplesPerSec = format->nSamplesPerSec; formatex->Format.wBitsPerSample = format->wBitsPerSample; formatex->Format.nChannels = format->nChannels; return hr; } HRESULT InitAudioClient(IAudioClient *audioClient, WAVEFORMATEXTENSIBLE *formatex) { AUDCLNT_SHAREMODE shareMode = AUDCLNT_SHAREMODE_SHARED; // share Audio Engine with other applications DWORD streamFlags = AUDCLNT_STREAMFLAGS_LOOPBACK; // loopback speaker // 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; return audioClient->Initialize(shareMode, streamFlags, hnsBufferDuration, 0, format, nullptr); } HRESULT CreateAudioCaptureClient(IAudioClient *audioClient, IAudioCaptureClient **audioCaptureClient) { HRESULT hr = audioClient->GetService(IID_PPV_ARGS(audioCaptureClient)); if (FAILED(hr)) { *audioCaptureClient = nullptr; } return hr; } void ThreadRun(IAudioClient *audio_client, IAudioCaptureClient *audio_capture_client) { HRESULT hr = S_OK; UINT32 num_success = 0; char pcmName[128] = { 0 }; sprintf(pcmName, "echo_speaker-%dHz_%db_%dc.pcm", g_formatex.Format.nSamplesPerSec, g_formatex.Format.wBitsPerSample, g_formatex.Format.nChannels); logd() << "pcmName: " << pcmName; BYTE *p_audio_data = nullptr; UINT32 num_frames_to_read = 0; DWORD dw_flag = 0; UINT32 num_frames_in_next_packet = 0; UINT32 num_loop = 0; audio_client->Start(); while (g_loopFlag) { std::this_thread::sleep_for(std::chrono::milliseconds(0)); while (true) { hr = audio_capture_client->GetNextPacketSize(&num_frames_in_next_packet); if (FAILED(hr)) { throw std::exception(); } if (num_frames_in_next_packet == 0) { break; } log() << "Loop[" << num_loop << "] " << "GetNextPacketSize: " << "num_frames_in_next_packet[" << num_frames_in_next_packet << "] " << "\n"; hr = audio_capture_client->GetBuffer(&p_audio_data, &num_frames_to_read, &dw_flag, nullptr, nullptr); if (FAILED(hr)) { throw std::exception(); } log() << "Loop[" << num_loop << "] " << "GetBuffer: " << "num_frames_to_read[" << num_frames_to_read << "] " << "\n"; savePCM(pcmName, p_audio_data, g_formatex.Format.wBitsPerSample, g_formatex.Format.nChannels, num_frames_to_read); log() << "Loop[" << num_loop << "] " << "savePCM: " << "bits_per_sample[" << g_formatex.Format.wBitsPerSample << "] " << "channels[" << g_formatex.Format.nChannels << "] " << "num_frames_to_read[" << num_frames_to_read << "] " << "\n"; if (num_success++ % 500 == 0) { std::cout << "Have already cpatured [" << num_success << "] times." << std::endl; } hr = audio_capture_client->ReleaseBuffer(num_frames_to_read); if (FAILED(hr)) { throw std::exception(); } num_loop++; } } audio_client->Stop(); } void savePCM(const char *name, void *data, int bitsPerSample, int channels, int frames) { FILE *fp = fopen(name, "ab+"); fwrite(data, bitsPerSample >> 3, channels * frames, fp); fclose(fp); // split_PCM(name, DEFAULT_CHANNELS, DEFAULT_BITS_PER_SAMPLE, DEFAULT_SAMPLE_RATE); fp = NULL; }