audio_low_latency_input_win.cc revision 5821806d5e7f356e8fa4b058a389a808ea183019
1// Copyright (c) 2012 The Chromium Authors. All rights reserved. 2// Use of this source code is governed by a BSD-style license that can be 3// found in the LICENSE file. 4 5#include "media/audio/win/audio_low_latency_input_win.h" 6 7#include "base/logging.h" 8#include "base/memory/scoped_ptr.h" 9#include "base/utf_string_conversions.h" 10#include "media/audio/audio_util.h" 11#include "media/audio/win/audio_manager_win.h" 12#include "media/audio/win/avrt_wrapper_win.h" 13 14using base::win::ScopedComPtr; 15using base::win::ScopedCOMInitializer; 16 17namespace media { 18 19WASAPIAudioInputStream::WASAPIAudioInputStream( 20 AudioManagerWin* manager, const AudioParameters& params, 21 const std::string& device_id) 22 : manager_(manager), 23 capture_thread_(NULL), 24 opened_(false), 25 started_(false), 26 endpoint_buffer_size_frames_(0), 27 device_id_(device_id), 28 sink_(NULL) { 29 DCHECK(manager_); 30 31 // Load the Avrt DLL if not already loaded. Required to support MMCSS. 32 bool avrt_init = avrt::Initialize(); 33 DCHECK(avrt_init) << "Failed to load the Avrt.dll"; 34 35 // Set up the desired capture format specified by the client. 36 format_.nSamplesPerSec = params.sample_rate(); 37 format_.wFormatTag = WAVE_FORMAT_PCM; 38 format_.wBitsPerSample = params.bits_per_sample(); 39 format_.nChannels = params.channels(); 40 format_.nBlockAlign = (format_.wBitsPerSample / 8) * format_.nChannels; 41 format_.nAvgBytesPerSec = format_.nSamplesPerSec * format_.nBlockAlign; 42 format_.cbSize = 0; 43 44 // Size in bytes of each audio frame. 45 frame_size_ = format_.nBlockAlign; 46 // Store size of audio packets which we expect to get from the audio 47 // endpoint device in each capture event. 48 packet_size_frames_ = params.GetBytesPerBuffer() / format_.nBlockAlign; 49 packet_size_bytes_ = params.GetBytesPerBuffer(); 50 DVLOG(1) << "Number of bytes per audio frame : " << frame_size_; 51 DVLOG(1) << "Number of audio frames per packet: " << packet_size_frames_; 52 53 // All events are auto-reset events and non-signaled initially. 54 55 // Create the event which the audio engine will signal each time 56 // a buffer becomes ready to be processed by the client. 57 audio_samples_ready_event_.Set(CreateEvent(NULL, FALSE, FALSE, NULL)); 58 DCHECK(audio_samples_ready_event_.IsValid()); 59 60 // Create the event which will be set in Stop() when capturing shall stop. 61 stop_capture_event_.Set(CreateEvent(NULL, FALSE, FALSE, NULL)); 62 DCHECK(stop_capture_event_.IsValid()); 63 64 ms_to_frame_count_ = static_cast<double>(params.sample_rate()) / 1000.0; 65 66 LARGE_INTEGER performance_frequency; 67 if (QueryPerformanceFrequency(&performance_frequency)) { 68 perf_count_to_100ns_units_ = 69 (10000000.0 / static_cast<double>(performance_frequency.QuadPart)); 70 } else { 71 LOG(ERROR) << "High-resolution performance counters are not supported."; 72 perf_count_to_100ns_units_ = 0.0; 73 } 74} 75 76WASAPIAudioInputStream::~WASAPIAudioInputStream() {} 77 78bool WASAPIAudioInputStream::Open() { 79 DCHECK(CalledOnValidThread()); 80 // Verify that we are not already opened. 81 if (opened_) 82 return false; 83 84 // Obtain a reference to the IMMDevice interface of the capturing 85 // device with the specified unique identifier or role which was 86 // set at construction. 87 HRESULT hr = SetCaptureDevice(); 88 if (FAILED(hr)) 89 return false; 90 91 // Obtain an IAudioClient interface which enables us to create and initialize 92 // an audio stream between an audio application and the audio engine. 93 hr = ActivateCaptureDevice(); 94 if (FAILED(hr)) 95 return false; 96 97 // Retrieve the stream format which the audio engine uses for its internal 98 // processing/mixing of shared-mode streams. This function call is for 99 // diagnostic purposes only and only in debug mode. 100#ifndef NDEBUG 101 hr = GetAudioEngineStreamFormat(); 102#endif 103 104 // Verify that the selected audio endpoint supports the specified format 105 // set during construction. 106 if (!DesiredFormatIsSupported()) { 107 return false; 108 } 109 110 // Initialize the audio stream between the client and the device using 111 // shared mode and a lowest possible glitch-free latency. 112 hr = InitializeAudioEngine(); 113 114 opened_ = SUCCEEDED(hr); 115 return opened_; 116} 117 118void WASAPIAudioInputStream::Start(AudioInputCallback* callback) { 119 DCHECK(CalledOnValidThread()); 120 DCHECK(callback); 121 DLOG_IF(ERROR, !opened_) << "Open() has not been called successfully"; 122 if (!opened_) 123 return; 124 125 if (started_) 126 return; 127 128 sink_ = callback; 129 130 // Create and start the thread that will drive the capturing by waiting for 131 // capture events. 132 capture_thread_ = 133 new base::DelegateSimpleThread(this, "wasapi_capture_thread"); 134 capture_thread_->Start(); 135 136 // Start streaming data between the endpoint buffer and the audio engine. 137 HRESULT hr = audio_client_->Start(); 138 DLOG_IF(ERROR, FAILED(hr)) << "Failed to start input streaming."; 139 140 started_ = SUCCEEDED(hr); 141} 142 143void WASAPIAudioInputStream::Stop() { 144 DCHECK(CalledOnValidThread()); 145 if (!started_) 146 return; 147 148 // Shut down the capture thread. 149 if (stop_capture_event_.IsValid()) { 150 SetEvent(stop_capture_event_.Get()); 151 } 152 153 // Stop the input audio streaming. 154 HRESULT hr = audio_client_->Stop(); 155 if (FAILED(hr)) { 156 LOG(ERROR) << "Failed to stop input streaming."; 157 } 158 159 // Wait until the thread completes and perform cleanup. 160 if (capture_thread_) { 161 SetEvent(stop_capture_event_.Get()); 162 capture_thread_->Join(); 163 capture_thread_ = NULL; 164 } 165 166 started_ = false; 167} 168 169void WASAPIAudioInputStream::Close() { 170 // It is valid to call Close() before calling open or Start(). 171 // It is also valid to call Close() after Start() has been called. 172 Stop(); 173 if (sink_) { 174 sink_->OnClose(this); 175 sink_ = NULL; 176 } 177 178 // Inform the audio manager that we have been closed. This will cause our 179 // destruction. 180 manager_->ReleaseInputStream(this); 181} 182 183double WASAPIAudioInputStream::GetMaxVolume() { 184 // Verify that Open() has been called succesfully, to ensure that an audio 185 // session exists and that an ISimpleAudioVolume interface has been created. 186 DLOG_IF(ERROR, !opened_) << "Open() has not been called successfully"; 187 if (!opened_) 188 return 0.0; 189 190 // The effective volume value is always in the range 0.0 to 1.0, hence 191 // we can return a fixed value (=1.0) here. 192 return 1.0; 193} 194 195void WASAPIAudioInputStream::SetVolume(double volume) { 196 DVLOG(1) << "SetVolume(volume=" << volume << ")"; 197 DCHECK(CalledOnValidThread()); 198 DCHECK_GE(volume, 0.0); 199 DCHECK_LE(volume, 1.0); 200 201 DLOG_IF(ERROR, !opened_) << "Open() has not been called successfully"; 202 if (!opened_) 203 return; 204 205 // Set a new master volume level. Valid volume levels are in the range 206 // 0.0 to 1.0. Ignore volume-change events. 207 HRESULT hr = simple_audio_volume_->SetMasterVolume(static_cast<float>(volume), 208 NULL); 209 DLOG_IF(WARNING, FAILED(hr)) << "Failed to set new input master volume."; 210 211 // Update the AGC volume level based on the last setting above. Note that, 212 // the volume-level resolution is not infinite and it is therefore not 213 // possible to assume that the volume provided as input parameter can be 214 // used directly. Instead, a new query to the audio hardware is required. 215 // This method does nothing if AGC is disabled. 216 UpdateAgcVolume(); 217} 218 219double WASAPIAudioInputStream::GetVolume() { 220 DLOG_IF(ERROR, !opened_) << "Open() has not been called successfully"; 221 if (!opened_) 222 return 0.0; 223 224 // Retrieve the current volume level. The value is in the range 0.0 to 1.0. 225 float level = 0.0f; 226 HRESULT hr = simple_audio_volume_->GetMasterVolume(&level); 227 DLOG_IF(WARNING, FAILED(hr)) << "Failed to get input master volume."; 228 229 return static_cast<double>(level); 230} 231 232// static 233int WASAPIAudioInputStream::HardwareSampleRate( 234 const std::string& device_id) { 235 base::win::ScopedCoMem<WAVEFORMATEX> audio_engine_mix_format; 236 HRESULT hr = GetMixFormat(device_id, &audio_engine_mix_format); 237 if (FAILED(hr)) 238 return 0; 239 240 return static_cast<int>(audio_engine_mix_format->nSamplesPerSec); 241} 242 243// static 244uint32 WASAPIAudioInputStream::HardwareChannelCount( 245 const std::string& device_id) { 246 base::win::ScopedCoMem<WAVEFORMATEX> audio_engine_mix_format; 247 HRESULT hr = GetMixFormat(device_id, &audio_engine_mix_format); 248 if (FAILED(hr)) 249 return 0; 250 251 return static_cast<uint32>(audio_engine_mix_format->nChannels); 252} 253 254// static 255HRESULT WASAPIAudioInputStream::GetMixFormat(const std::string& device_id, 256 WAVEFORMATEX** device_format) { 257 // It is assumed that this static method is called from a COM thread, i.e., 258 // CoInitializeEx() is not called here to avoid STA/MTA conflicts. 259 ScopedComPtr<IMMDeviceEnumerator> enumerator; 260 HRESULT hr = CoCreateInstance(__uuidof(MMDeviceEnumerator), 261 NULL, 262 CLSCTX_INPROC_SERVER, 263 __uuidof(IMMDeviceEnumerator), 264 enumerator.ReceiveVoid()); 265 if (FAILED(hr)) 266 return hr; 267 268 ScopedComPtr<IMMDevice> endpoint_device; 269 if (device_id == AudioManagerBase::kDefaultDeviceId) { 270 // Retrieve the default capture audio endpoint. 271 hr = enumerator->GetDefaultAudioEndpoint(eCapture, eConsole, 272 endpoint_device.Receive()); 273 } else { 274 // Retrieve a capture endpoint device that is specified by an endpoint 275 // device-identification string. 276 hr = enumerator->GetDevice(UTF8ToUTF16(device_id).c_str(), 277 endpoint_device.Receive()); 278 } 279 if (FAILED(hr)) 280 return hr; 281 282 ScopedComPtr<IAudioClient> audio_client; 283 hr = endpoint_device->Activate(__uuidof(IAudioClient), 284 CLSCTX_INPROC_SERVER, 285 NULL, 286 audio_client.ReceiveVoid()); 287 return SUCCEEDED(hr) ? audio_client->GetMixFormat(device_format) : hr; 288} 289 290void WASAPIAudioInputStream::Run() { 291 ScopedCOMInitializer com_init(ScopedCOMInitializer::kMTA); 292 293 // Increase the thread priority. 294 capture_thread_->SetThreadPriority(base::kThreadPriority_RealtimeAudio); 295 296 // Enable MMCSS to ensure that this thread receives prioritized access to 297 // CPU resources. 298 DWORD task_index = 0; 299 HANDLE mm_task = avrt::AvSetMmThreadCharacteristics(L"Pro Audio", 300 &task_index); 301 bool mmcss_is_ok = 302 (mm_task && avrt::AvSetMmThreadPriority(mm_task, AVRT_PRIORITY_CRITICAL)); 303 if (!mmcss_is_ok) { 304 // Failed to enable MMCSS on this thread. It is not fatal but can lead 305 // to reduced QoS at high load. 306 DWORD err = GetLastError(); 307 LOG(WARNING) << "Failed to enable MMCSS (error code=" << err << ")."; 308 } 309 310 // Allocate a buffer with a size that enables us to take care of cases like: 311 // 1) The recorded buffer size is smaller, or does not match exactly with, 312 // the selected packet size used in each callback. 313 // 2) The selected buffer size is larger than the recorded buffer size in 314 // each event. 315 size_t buffer_frame_index = 0; 316 size_t capture_buffer_size = std::max( 317 2 * endpoint_buffer_size_frames_ * frame_size_, 318 2 * packet_size_frames_ * frame_size_); 319 scoped_array<uint8> capture_buffer(new uint8[capture_buffer_size]); 320 321 LARGE_INTEGER now_count; 322 bool recording = true; 323 bool error = false; 324 double volume = GetVolume(); 325 HANDLE wait_array[2] = {stop_capture_event_, audio_samples_ready_event_}; 326 327 while (recording && !error) { 328 HRESULT hr = S_FALSE; 329 330 // Wait for a close-down event or a new capture event. 331 DWORD wait_result = WaitForMultipleObjects(2, wait_array, FALSE, INFINITE); 332 switch (wait_result) { 333 case WAIT_FAILED: 334 error = true; 335 break; 336 case WAIT_OBJECT_0 + 0: 337 // |stop_capture_event_| has been set. 338 recording = false; 339 break; 340 case WAIT_OBJECT_0 + 1: 341 { 342 // |audio_samples_ready_event_| has been set. 343 BYTE* data_ptr = NULL; 344 UINT32 num_frames_to_read = 0; 345 DWORD flags = 0; 346 UINT64 device_position = 0; 347 UINT64 first_audio_frame_timestamp = 0; 348 349 // Retrieve the amount of data in the capture endpoint buffer, 350 // replace it with silence if required, create callbacks for each 351 // packet and store non-delivered data for the next event. 352 hr = audio_capture_client_->GetBuffer(&data_ptr, 353 &num_frames_to_read, 354 &flags, 355 &device_position, 356 &first_audio_frame_timestamp); 357 if (FAILED(hr)) { 358 DLOG(ERROR) << "Failed to get data from the capture buffer"; 359 continue; 360 } 361 362 if (num_frames_to_read != 0) { 363 size_t pos = buffer_frame_index * frame_size_; 364 size_t num_bytes = num_frames_to_read * frame_size_; 365 DCHECK_GE(capture_buffer_size, pos + num_bytes); 366 367 if (flags & AUDCLNT_BUFFERFLAGS_SILENT) { 368 // Clear out the local buffer since silence is reported. 369 memset(&capture_buffer[pos], 0, num_bytes); 370 } else { 371 // Copy captured data from audio engine buffer to local buffer. 372 memcpy(&capture_buffer[pos], data_ptr, num_bytes); 373 } 374 375 buffer_frame_index += num_frames_to_read; 376 } 377 378 hr = audio_capture_client_->ReleaseBuffer(num_frames_to_read); 379 DLOG_IF(ERROR, FAILED(hr)) << "Failed to release capture buffer"; 380 381 // Derive a delay estimate for the captured audio packet. 382 // The value contains two parts (A+B), where A is the delay of the 383 // first audio frame in the packet and B is the extra delay 384 // contained in any stored data. Unit is in audio frames. 385 QueryPerformanceCounter(&now_count); 386 double audio_delay_frames = 387 ((perf_count_to_100ns_units_ * now_count.QuadPart - 388 first_audio_frame_timestamp) / 10000.0) * ms_to_frame_count_ + 389 buffer_frame_index - num_frames_to_read; 390 391 // Update the AGC volume level once every second. Note that, 392 // |volume| is also updated each time SetVolume() is called 393 // through IPC by the render-side AGC. 394 QueryAgcVolume(&volume); 395 396 // Deliver captured data to the registered consumer using a packet 397 // size which was specified at construction. 398 uint32 delay_frames = static_cast<uint32>(audio_delay_frames + 0.5); 399 while (buffer_frame_index >= packet_size_frames_) { 400 uint8* audio_data = 401 reinterpret_cast<uint8*>(capture_buffer.get()); 402 403 // Deliver data packet, delay estimation and volume level to 404 // the user. 405 sink_->OnData(this, 406 audio_data, 407 packet_size_bytes_, 408 delay_frames * frame_size_, 409 volume); 410 411 // Store parts of the recorded data which can't be delivered 412 // using the current packet size. The stored section will be used 413 // either in the next while-loop iteration or in the next 414 // capture event. 415 memmove(&capture_buffer[0], 416 &capture_buffer[packet_size_bytes_], 417 (buffer_frame_index - packet_size_frames_) * frame_size_); 418 419 buffer_frame_index -= packet_size_frames_; 420 delay_frames -= packet_size_frames_; 421 } 422 } 423 break; 424 default: 425 error = true; 426 break; 427 } 428 } 429 430 if (recording && error) { 431 // TODO(henrika): perhaps it worth improving the cleanup here by e.g. 432 // stopping the audio client, joining the thread etc.? 433 NOTREACHED() << "WASAPI capturing failed with error code " 434 << GetLastError(); 435 } 436 437 // Disable MMCSS. 438 if (mm_task && !avrt::AvRevertMmThreadCharacteristics(mm_task)) { 439 PLOG(WARNING) << "Failed to disable MMCSS"; 440 } 441} 442 443void WASAPIAudioInputStream::HandleError(HRESULT err) { 444 NOTREACHED() << "Error code: " << err; 445 if (sink_) 446 sink_->OnError(this, static_cast<int>(err)); 447} 448 449HRESULT WASAPIAudioInputStream::SetCaptureDevice() { 450 ScopedComPtr<IMMDeviceEnumerator> enumerator; 451 HRESULT hr = CoCreateInstance(__uuidof(MMDeviceEnumerator), 452 NULL, 453 CLSCTX_INPROC_SERVER, 454 __uuidof(IMMDeviceEnumerator), 455 enumerator.ReceiveVoid()); 456 if (SUCCEEDED(hr)) { 457 // Retrieve the IMMDevice by using the specified role or the specified 458 // unique endpoint device-identification string. 459 // TODO(henrika): possibly add support for the eCommunications as well. 460 if (device_id_ == AudioManagerBase::kDefaultDeviceId) { 461 // Retrieve the default capture audio endpoint for the specified role. 462 // Note that, in Windows Vista, the MMDevice API supports device roles 463 // but the system-supplied user interface programs do not. 464 hr = enumerator->GetDefaultAudioEndpoint(eCapture, 465 eConsole, 466 endpoint_device_.Receive()); 467 } else { 468 // Retrieve a capture endpoint device that is specified by an endpoint 469 // device-identification string. 470 hr = enumerator->GetDevice(UTF8ToUTF16(device_id_).c_str(), 471 endpoint_device_.Receive()); 472 } 473 474 if (FAILED(hr)) 475 return hr; 476 477 // Verify that the audio endpoint device is active, i.e., the audio 478 // adapter that connects to the endpoint device is present and enabled. 479 DWORD state = DEVICE_STATE_DISABLED; 480 hr = endpoint_device_->GetState(&state); 481 if (SUCCEEDED(hr)) { 482 if (!(state & DEVICE_STATE_ACTIVE)) { 483 DLOG(ERROR) << "Selected capture device is not active."; 484 hr = E_ACCESSDENIED; 485 } 486 } 487 } 488 489 return hr; 490} 491 492HRESULT WASAPIAudioInputStream::ActivateCaptureDevice() { 493 // Creates and activates an IAudioClient COM object given the selected 494 // capture endpoint device. 495 HRESULT hr = endpoint_device_->Activate(__uuidof(IAudioClient), 496 CLSCTX_INPROC_SERVER, 497 NULL, 498 audio_client_.ReceiveVoid()); 499 return hr; 500} 501 502HRESULT WASAPIAudioInputStream::GetAudioEngineStreamFormat() { 503 HRESULT hr = S_OK; 504#ifndef NDEBUG 505 // The GetMixFormat() method retrieves the stream format that the 506 // audio engine uses for its internal processing of shared-mode streams. 507 // The method always uses a WAVEFORMATEXTENSIBLE structure, instead 508 // of a stand-alone WAVEFORMATEX structure, to specify the format. 509 // An WAVEFORMATEXTENSIBLE structure can specify both the mapping of 510 // channels to speakers and the number of bits of precision in each sample. 511 base::win::ScopedCoMem<WAVEFORMATEXTENSIBLE> format_ex; 512 hr = audio_client_->GetMixFormat( 513 reinterpret_cast<WAVEFORMATEX**>(&format_ex)); 514 515 // See http://msdn.microsoft.com/en-us/windows/hardware/gg463006#EFH 516 // for details on the WAVE file format. 517 WAVEFORMATEX format = format_ex->Format; 518 DVLOG(2) << "WAVEFORMATEX:"; 519 DVLOG(2) << " wFormatTags : 0x" << std::hex << format.wFormatTag; 520 DVLOG(2) << " nChannels : " << format.nChannels; 521 DVLOG(2) << " nSamplesPerSec : " << format.nSamplesPerSec; 522 DVLOG(2) << " nAvgBytesPerSec: " << format.nAvgBytesPerSec; 523 DVLOG(2) << " nBlockAlign : " << format.nBlockAlign; 524 DVLOG(2) << " wBitsPerSample : " << format.wBitsPerSample; 525 DVLOG(2) << " cbSize : " << format.cbSize; 526 527 DVLOG(2) << "WAVEFORMATEXTENSIBLE:"; 528 DVLOG(2) << " wValidBitsPerSample: " << 529 format_ex->Samples.wValidBitsPerSample; 530 DVLOG(2) << " dwChannelMask : 0x" << std::hex << 531 format_ex->dwChannelMask; 532 if (format_ex->SubFormat == KSDATAFORMAT_SUBTYPE_PCM) 533 DVLOG(2) << " SubFormat : KSDATAFORMAT_SUBTYPE_PCM"; 534 else if (format_ex->SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT) 535 DVLOG(2) << " SubFormat : KSDATAFORMAT_SUBTYPE_IEEE_FLOAT"; 536 else if (format_ex->SubFormat == KSDATAFORMAT_SUBTYPE_WAVEFORMATEX) 537 DVLOG(2) << " SubFormat : KSDATAFORMAT_SUBTYPE_WAVEFORMATEX"; 538#endif 539 return hr; 540} 541 542bool WASAPIAudioInputStream::DesiredFormatIsSupported() { 543 // An application that uses WASAPI to manage shared-mode streams can rely 544 // on the audio engine to perform only limited format conversions. The audio 545 // engine can convert between a standard PCM sample size used by the 546 // application and the floating-point samples that the engine uses for its 547 // internal processing. However, the format for an application stream 548 // typically must have the same number of channels and the same sample 549 // rate as the stream format used by the device. 550 // Many audio devices support both PCM and non-PCM stream formats. However, 551 // the audio engine can mix only PCM streams. 552 base::win::ScopedCoMem<WAVEFORMATEX> closest_match; 553 HRESULT hr = audio_client_->IsFormatSupported(AUDCLNT_SHAREMODE_SHARED, 554 &format_, 555 &closest_match); 556 DLOG_IF(ERROR, hr == S_FALSE) << "Format is not supported " 557 << "but a closest match exists."; 558 return (hr == S_OK); 559} 560 561HRESULT WASAPIAudioInputStream::InitializeAudioEngine() { 562 // Initialize the audio stream between the client and the device. 563 // We connect indirectly through the audio engine by using shared mode 564 // and WASAPI is initialized in an event driven mode. 565 // Note that, |hnsBufferDuration| is set of 0, which ensures that the 566 // buffer is never smaller than the minimum buffer size needed to ensure 567 // that glitches do not occur between the periodic processing passes. 568 // This setting should lead to lowest possible latency. 569 HRESULT hr = audio_client_->Initialize(AUDCLNT_SHAREMODE_SHARED, 570 AUDCLNT_STREAMFLAGS_EVENTCALLBACK | 571 AUDCLNT_STREAMFLAGS_NOPERSIST, 572 0, // hnsBufferDuration 573 0, 574 &format_, 575 NULL); 576 if (FAILED(hr)) 577 return hr; 578 579 // Retrieve the length of the endpoint buffer shared between the client 580 // and the audio engine. The buffer length determines the maximum amount 581 // of capture data that the audio engine can read from the endpoint buffer 582 // during a single processing pass. 583 // A typical value is 960 audio frames <=> 20ms @ 48kHz sample rate. 584 hr = audio_client_->GetBufferSize(&endpoint_buffer_size_frames_); 585 if (FAILED(hr)) 586 return hr; 587 DVLOG(1) << "endpoint buffer size: " << endpoint_buffer_size_frames_ 588 << " [frames]"; 589 590#ifndef NDEBUG 591 // The period between processing passes by the audio engine is fixed for a 592 // particular audio endpoint device and represents the smallest processing 593 // quantum for the audio engine. This period plus the stream latency between 594 // the buffer and endpoint device represents the minimum possible latency 595 // that an audio application can achieve. 596 // TODO(henrika): possibly remove this section when all parts are ready. 597 REFERENCE_TIME device_period_shared_mode = 0; 598 REFERENCE_TIME device_period_exclusive_mode = 0; 599 HRESULT hr_dbg = audio_client_->GetDevicePeriod( 600 &device_period_shared_mode, &device_period_exclusive_mode); 601 if (SUCCEEDED(hr_dbg)) { 602 DVLOG(1) << "device period: " 603 << static_cast<double>(device_period_shared_mode / 10000.0) 604 << " [ms]"; 605 } 606 607 REFERENCE_TIME latency = 0; 608 hr_dbg = audio_client_->GetStreamLatency(&latency); 609 if (SUCCEEDED(hr_dbg)) { 610 DVLOG(1) << "stream latency: " << static_cast<double>(latency / 10000.0) 611 << " [ms]"; 612 } 613#endif 614 615 // Set the event handle that the audio engine will signal each time 616 // a buffer becomes ready to be processed by the client. 617 hr = audio_client_->SetEventHandle(audio_samples_ready_event_.Get()); 618 if (FAILED(hr)) 619 return hr; 620 621 // Get access to the IAudioCaptureClient interface. This interface 622 // enables us to read input data from the capture endpoint buffer. 623 hr = audio_client_->GetService(__uuidof(IAudioCaptureClient), 624 audio_capture_client_.ReceiveVoid()); 625 if (FAILED(hr)) 626 return hr; 627 628 // Obtain a reference to the ISimpleAudioVolume interface which enables 629 // us to control the master volume level of an audio session. 630 hr = audio_client_->GetService(__uuidof(ISimpleAudioVolume), 631 simple_audio_volume_.ReceiveVoid()); 632 return hr; 633} 634 635} // namespace media 636