audio_low_latency_input_win.cc revision 2a99a7e74a7f215066514fe81d2bfa6639d9eddd
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 DVLOG(1) << "WASAPIAudioInputStream::Stop()"; 146 if (!started_) 147 return; 148 149 // Shut down the capture thread. 150 if (stop_capture_event_.IsValid()) { 151 SetEvent(stop_capture_event_.Get()); 152 } 153 154 // Stop the input audio streaming. 155 HRESULT hr = audio_client_->Stop(); 156 if (FAILED(hr)) { 157 LOG(ERROR) << "Failed to stop input streaming."; 158 } 159 160 // Wait until the thread completes and perform cleanup. 161 if (capture_thread_) { 162 SetEvent(stop_capture_event_.Get()); 163 capture_thread_->Join(); 164 capture_thread_ = NULL; 165 } 166 167 started_ = false; 168} 169 170void WASAPIAudioInputStream::Close() { 171 DVLOG(1) << "WASAPIAudioInputStream::Close()"; 172 // It is valid to call Close() before calling open or Start(). 173 // It is also valid to call Close() after Start() has been called. 174 Stop(); 175 if (sink_) { 176 sink_->OnClose(this); 177 sink_ = NULL; 178 } 179 180 // Inform the audio manager that we have been closed. This will cause our 181 // destruction. 182 manager_->ReleaseInputStream(this); 183} 184 185double WASAPIAudioInputStream::GetMaxVolume() { 186 // Verify that Open() has been called succesfully, to ensure that an audio 187 // session exists and that an ISimpleAudioVolume interface has been created. 188 DLOG_IF(ERROR, !opened_) << "Open() has not been called successfully"; 189 if (!opened_) 190 return 0.0; 191 192 // The effective volume value is always in the range 0.0 to 1.0, hence 193 // we can return a fixed value (=1.0) here. 194 return 1.0; 195} 196 197void WASAPIAudioInputStream::SetVolume(double volume) { 198 DVLOG(1) << "SetVolume(volume=" << volume << ")"; 199 DCHECK(CalledOnValidThread()); 200 DCHECK_GE(volume, 0.0); 201 DCHECK_LE(volume, 1.0); 202 203 DLOG_IF(ERROR, !opened_) << "Open() has not been called successfully"; 204 if (!opened_) 205 return; 206 207 // Set a new master volume level. Valid volume levels are in the range 208 // 0.0 to 1.0. Ignore volume-change events. 209 HRESULT hr = simple_audio_volume_->SetMasterVolume(static_cast<float>(volume), 210 NULL); 211 DLOG_IF(WARNING, FAILED(hr)) << "Failed to set new input master volume."; 212 213 // Update the AGC volume level based on the last setting above. Note that, 214 // the volume-level resolution is not infinite and it is therefore not 215 // possible to assume that the volume provided as input parameter can be 216 // used directly. Instead, a new query to the audio hardware is required. 217 // This method does nothing if AGC is disabled. 218 UpdateAgcVolume(); 219} 220 221double WASAPIAudioInputStream::GetVolume() { 222 DLOG_IF(ERROR, !opened_) << "Open() has not been called successfully"; 223 if (!opened_) 224 return 0.0; 225 226 // Retrieve the current volume level. The value is in the range 0.0 to 1.0. 227 float level = 0.0f; 228 HRESULT hr = simple_audio_volume_->GetMasterVolume(&level); 229 DLOG_IF(WARNING, FAILED(hr)) << "Failed to get input master volume."; 230 231 return static_cast<double>(level); 232} 233 234// static 235int WASAPIAudioInputStream::HardwareSampleRate( 236 const std::string& device_id) { 237 base::win::ScopedCoMem<WAVEFORMATEX> audio_engine_mix_format; 238 HRESULT hr = GetMixFormat(device_id, &audio_engine_mix_format); 239 if (FAILED(hr)) 240 return 0; 241 242 return static_cast<int>(audio_engine_mix_format->nSamplesPerSec); 243} 244 245// static 246uint32 WASAPIAudioInputStream::HardwareChannelCount( 247 const std::string& device_id) { 248 base::win::ScopedCoMem<WAVEFORMATEX> audio_engine_mix_format; 249 HRESULT hr = GetMixFormat(device_id, &audio_engine_mix_format); 250 if (FAILED(hr)) 251 return 0; 252 253 return static_cast<uint32>(audio_engine_mix_format->nChannels); 254} 255 256// static 257HRESULT WASAPIAudioInputStream::GetMixFormat(const std::string& device_id, 258 WAVEFORMATEX** device_format) { 259 // It is assumed that this static method is called from a COM thread, i.e., 260 // CoInitializeEx() is not called here to avoid STA/MTA conflicts. 261 ScopedComPtr<IMMDeviceEnumerator> enumerator; 262 HRESULT hr = CoCreateInstance(__uuidof(MMDeviceEnumerator), 263 NULL, 264 CLSCTX_INPROC_SERVER, 265 __uuidof(IMMDeviceEnumerator), 266 enumerator.ReceiveVoid()); 267 if (FAILED(hr)) 268 return hr; 269 270 ScopedComPtr<IMMDevice> endpoint_device; 271 if (device_id == AudioManagerBase::kDefaultDeviceId) { 272 // Retrieve the default capture audio endpoint. 273 hr = enumerator->GetDefaultAudioEndpoint(eCapture, eConsole, 274 endpoint_device.Receive()); 275 } else { 276 // Retrieve a capture endpoint device that is specified by an endpoint 277 // device-identification string. 278 hr = enumerator->GetDevice(UTF8ToUTF16(device_id).c_str(), 279 endpoint_device.Receive()); 280 } 281 if (FAILED(hr)) 282 return hr; 283 284 ScopedComPtr<IAudioClient> audio_client; 285 hr = endpoint_device->Activate(__uuidof(IAudioClient), 286 CLSCTX_INPROC_SERVER, 287 NULL, 288 audio_client.ReceiveVoid()); 289 return SUCCEEDED(hr) ? audio_client->GetMixFormat(device_format) : hr; 290} 291 292void WASAPIAudioInputStream::Run() { 293 ScopedCOMInitializer com_init(ScopedCOMInitializer::kMTA); 294 295 // Increase the thread priority. 296 capture_thread_->SetThreadPriority(base::kThreadPriority_RealtimeAudio); 297 298 // Enable MMCSS to ensure that this thread receives prioritized access to 299 // CPU resources. 300 DWORD task_index = 0; 301 HANDLE mm_task = avrt::AvSetMmThreadCharacteristics(L"Pro Audio", 302 &task_index); 303 bool mmcss_is_ok = 304 (mm_task && avrt::AvSetMmThreadPriority(mm_task, AVRT_PRIORITY_CRITICAL)); 305 if (!mmcss_is_ok) { 306 // Failed to enable MMCSS on this thread. It is not fatal but can lead 307 // to reduced QoS at high load. 308 DWORD err = GetLastError(); 309 LOG(WARNING) << "Failed to enable MMCSS (error code=" << err << ")."; 310 } 311 312 // Allocate a buffer with a size that enables us to take care of cases like: 313 // 1) The recorded buffer size is smaller, or does not match exactly with, 314 // the selected packet size used in each callback. 315 // 2) The selected buffer size is larger than the recorded buffer size in 316 // each event. 317 size_t buffer_frame_index = 0; 318 size_t capture_buffer_size = std::max( 319 2 * endpoint_buffer_size_frames_ * frame_size_, 320 2 * packet_size_frames_ * frame_size_); 321 scoped_array<uint8> capture_buffer(new uint8[capture_buffer_size]); 322 323 LARGE_INTEGER now_count; 324 bool recording = true; 325 bool error = false; 326 double volume = GetVolume(); 327 HANDLE wait_array[2] = {stop_capture_event_, audio_samples_ready_event_}; 328 329 while (recording && !error) { 330 HRESULT hr = S_FALSE; 331 332 // Wait for a close-down event or a new capture event. 333 DWORD wait_result = WaitForMultipleObjects(2, wait_array, FALSE, INFINITE); 334 switch (wait_result) { 335 case WAIT_FAILED: 336 error = true; 337 break; 338 case WAIT_OBJECT_0 + 0: 339 // |stop_capture_event_| has been set. 340 recording = false; 341 break; 342 case WAIT_OBJECT_0 + 1: 343 { 344 // |audio_samples_ready_event_| has been set. 345 BYTE* data_ptr = NULL; 346 UINT32 num_frames_to_read = 0; 347 DWORD flags = 0; 348 UINT64 device_position = 0; 349 UINT64 first_audio_frame_timestamp = 0; 350 351 // Retrieve the amount of data in the capture endpoint buffer, 352 // replace it with silence if required, create callbacks for each 353 // packet and store non-delivered data for the next event. 354 hr = audio_capture_client_->GetBuffer(&data_ptr, 355 &num_frames_to_read, 356 &flags, 357 &device_position, 358 &first_audio_frame_timestamp); 359 if (FAILED(hr)) { 360 DLOG(ERROR) << "Failed to get data from the capture buffer"; 361 continue; 362 } 363 364 if (num_frames_to_read != 0) { 365 size_t pos = buffer_frame_index * frame_size_; 366 size_t num_bytes = num_frames_to_read * frame_size_; 367 DCHECK_GE(capture_buffer_size, pos + num_bytes); 368 369 if (flags & AUDCLNT_BUFFERFLAGS_SILENT) { 370 // Clear out the local buffer since silence is reported. 371 memset(&capture_buffer[pos], 0, num_bytes); 372 } else { 373 // Copy captured data from audio engine buffer to local buffer. 374 memcpy(&capture_buffer[pos], data_ptr, num_bytes); 375 } 376 377 buffer_frame_index += num_frames_to_read; 378 } 379 380 hr = audio_capture_client_->ReleaseBuffer(num_frames_to_read); 381 DLOG_IF(ERROR, FAILED(hr)) << "Failed to release capture buffer"; 382 383 // Derive a delay estimate for the captured audio packet. 384 // The value contains two parts (A+B), where A is the delay of the 385 // first audio frame in the packet and B is the extra delay 386 // contained in any stored data. Unit is in audio frames. 387 QueryPerformanceCounter(&now_count); 388 double audio_delay_frames = 389 ((perf_count_to_100ns_units_ * now_count.QuadPart - 390 first_audio_frame_timestamp) / 10000.0) * ms_to_frame_count_ + 391 buffer_frame_index - num_frames_to_read; 392 393 // Update the AGC volume level once every second. Note that, 394 // |volume| is also updated each time SetVolume() is called 395 // through IPC by the render-side AGC. 396 QueryAgcVolume(&volume); 397 398 // Deliver captured data to the registered consumer using a packet 399 // size which was specified at construction. 400 uint32 delay_frames = static_cast<uint32>(audio_delay_frames + 0.5); 401 while (buffer_frame_index >= packet_size_frames_) { 402 uint8* audio_data = 403 reinterpret_cast<uint8*>(capture_buffer.get()); 404 405 // Deliver data packet, delay estimation and volume level to 406 // the user. 407 sink_->OnData(this, 408 audio_data, 409 packet_size_bytes_, 410 delay_frames * frame_size_, 411 volume); 412 413 // Store parts of the recorded data which can't be delivered 414 // using the current packet size. The stored section will be used 415 // either in the next while-loop iteration or in the next 416 // capture event. 417 memmove(&capture_buffer[0], 418 &capture_buffer[packet_size_bytes_], 419 (buffer_frame_index - packet_size_frames_) * frame_size_); 420 421 buffer_frame_index -= packet_size_frames_; 422 delay_frames -= packet_size_frames_; 423 } 424 } 425 break; 426 default: 427 error = true; 428 break; 429 } 430 } 431 432 if (recording && error) { 433 // TODO(henrika): perhaps it worth improving the cleanup here by e.g. 434 // stopping the audio client, joining the thread etc.? 435 NOTREACHED() << "WASAPI capturing failed with error code " 436 << GetLastError(); 437 } 438 439 // Disable MMCSS. 440 if (mm_task && !avrt::AvRevertMmThreadCharacteristics(mm_task)) { 441 PLOG(WARNING) << "Failed to disable MMCSS"; 442 } 443} 444 445void WASAPIAudioInputStream::HandleError(HRESULT err) { 446 NOTREACHED() << "Error code: " << err; 447 if (sink_) 448 sink_->OnError(this); 449} 450 451HRESULT WASAPIAudioInputStream::SetCaptureDevice() { 452 ScopedComPtr<IMMDeviceEnumerator> enumerator; 453 HRESULT hr = CoCreateInstance(__uuidof(MMDeviceEnumerator), 454 NULL, 455 CLSCTX_INPROC_SERVER, 456 __uuidof(IMMDeviceEnumerator), 457 enumerator.ReceiveVoid()); 458 if (SUCCEEDED(hr)) { 459 // Retrieve the IMMDevice by using the specified role or the specified 460 // unique endpoint device-identification string. 461 // TODO(henrika): possibly add support for the eCommunications as well. 462 if (device_id_ == AudioManagerBase::kDefaultDeviceId) { 463 // Retrieve the default capture audio endpoint for the specified role. 464 // Note that, in Windows Vista, the MMDevice API supports device roles 465 // but the system-supplied user interface programs do not. 466 hr = enumerator->GetDefaultAudioEndpoint(eCapture, 467 eConsole, 468 endpoint_device_.Receive()); 469 } else { 470 // Retrieve a capture endpoint device that is specified by an endpoint 471 // device-identification string. 472 hr = enumerator->GetDevice(UTF8ToUTF16(device_id_).c_str(), 473 endpoint_device_.Receive()); 474 } 475 476 if (FAILED(hr)) 477 return hr; 478 479 // Verify that the audio endpoint device is active, i.e., the audio 480 // adapter that connects to the endpoint device is present and enabled. 481 DWORD state = DEVICE_STATE_DISABLED; 482 hr = endpoint_device_->GetState(&state); 483 if (SUCCEEDED(hr)) { 484 if (!(state & DEVICE_STATE_ACTIVE)) { 485 DLOG(ERROR) << "Selected capture device is not active."; 486 hr = E_ACCESSDENIED; 487 } 488 } 489 } 490 491 return hr; 492} 493 494HRESULT WASAPIAudioInputStream::ActivateCaptureDevice() { 495 // Creates and activates an IAudioClient COM object given the selected 496 // capture endpoint device. 497 HRESULT hr = endpoint_device_->Activate(__uuidof(IAudioClient), 498 CLSCTX_INPROC_SERVER, 499 NULL, 500 audio_client_.ReceiveVoid()); 501 return hr; 502} 503 504HRESULT WASAPIAudioInputStream::GetAudioEngineStreamFormat() { 505 HRESULT hr = S_OK; 506#ifndef NDEBUG 507 // The GetMixFormat() method retrieves the stream format that the 508 // audio engine uses for its internal processing of shared-mode streams. 509 // The method always uses a WAVEFORMATEXTENSIBLE structure, instead 510 // of a stand-alone WAVEFORMATEX structure, to specify the format. 511 // An WAVEFORMATEXTENSIBLE structure can specify both the mapping of 512 // channels to speakers and the number of bits of precision in each sample. 513 base::win::ScopedCoMem<WAVEFORMATEXTENSIBLE> format_ex; 514 hr = audio_client_->GetMixFormat( 515 reinterpret_cast<WAVEFORMATEX**>(&format_ex)); 516 517 // See http://msdn.microsoft.com/en-us/windows/hardware/gg463006#EFH 518 // for details on the WAVE file format. 519 WAVEFORMATEX format = format_ex->Format; 520 DVLOG(2) << "WAVEFORMATEX:"; 521 DVLOG(2) << " wFormatTags : 0x" << std::hex << format.wFormatTag; 522 DVLOG(2) << " nChannels : " << format.nChannels; 523 DVLOG(2) << " nSamplesPerSec : " << format.nSamplesPerSec; 524 DVLOG(2) << " nAvgBytesPerSec: " << format.nAvgBytesPerSec; 525 DVLOG(2) << " nBlockAlign : " << format.nBlockAlign; 526 DVLOG(2) << " wBitsPerSample : " << format.wBitsPerSample; 527 DVLOG(2) << " cbSize : " << format.cbSize; 528 529 DVLOG(2) << "WAVEFORMATEXTENSIBLE:"; 530 DVLOG(2) << " wValidBitsPerSample: " << 531 format_ex->Samples.wValidBitsPerSample; 532 DVLOG(2) << " dwChannelMask : 0x" << std::hex << 533 format_ex->dwChannelMask; 534 if (format_ex->SubFormat == KSDATAFORMAT_SUBTYPE_PCM) 535 DVLOG(2) << " SubFormat : KSDATAFORMAT_SUBTYPE_PCM"; 536 else if (format_ex->SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT) 537 DVLOG(2) << " SubFormat : KSDATAFORMAT_SUBTYPE_IEEE_FLOAT"; 538 else if (format_ex->SubFormat == KSDATAFORMAT_SUBTYPE_WAVEFORMATEX) 539 DVLOG(2) << " SubFormat : KSDATAFORMAT_SUBTYPE_WAVEFORMATEX"; 540#endif 541 return hr; 542} 543 544bool WASAPIAudioInputStream::DesiredFormatIsSupported() { 545 // An application that uses WASAPI to manage shared-mode streams can rely 546 // on the audio engine to perform only limited format conversions. The audio 547 // engine can convert between a standard PCM sample size used by the 548 // application and the floating-point samples that the engine uses for its 549 // internal processing. However, the format for an application stream 550 // typically must have the same number of channels and the same sample 551 // rate as the stream format used by the device. 552 // Many audio devices support both PCM and non-PCM stream formats. However, 553 // the audio engine can mix only PCM streams. 554 base::win::ScopedCoMem<WAVEFORMATEX> closest_match; 555 HRESULT hr = audio_client_->IsFormatSupported(AUDCLNT_SHAREMODE_SHARED, 556 &format_, 557 &closest_match); 558 DLOG_IF(ERROR, hr == S_FALSE) << "Format is not supported " 559 << "but a closest match exists."; 560 return (hr == S_OK); 561} 562 563HRESULT WASAPIAudioInputStream::InitializeAudioEngine() { 564 // Initialize the audio stream between the client and the device. 565 // We connect indirectly through the audio engine by using shared mode 566 // and WASAPI is initialized in an event driven mode. 567 // Note that, |hnsBufferDuration| is set of 0, which ensures that the 568 // buffer is never smaller than the minimum buffer size needed to ensure 569 // that glitches do not occur between the periodic processing passes. 570 // This setting should lead to lowest possible latency. 571 HRESULT hr = audio_client_->Initialize(AUDCLNT_SHAREMODE_SHARED, 572 AUDCLNT_STREAMFLAGS_EVENTCALLBACK | 573 AUDCLNT_STREAMFLAGS_NOPERSIST, 574 0, // hnsBufferDuration 575 0, 576 &format_, 577 NULL); 578 if (FAILED(hr)) 579 return hr; 580 581 // Retrieve the length of the endpoint buffer shared between the client 582 // and the audio engine. The buffer length determines the maximum amount 583 // of capture data that the audio engine can read from the endpoint buffer 584 // during a single processing pass. 585 // A typical value is 960 audio frames <=> 20ms @ 48kHz sample rate. 586 hr = audio_client_->GetBufferSize(&endpoint_buffer_size_frames_); 587 if (FAILED(hr)) 588 return hr; 589 DVLOG(1) << "endpoint buffer size: " << endpoint_buffer_size_frames_ 590 << " [frames]"; 591 592#ifndef NDEBUG 593 // The period between processing passes by the audio engine is fixed for a 594 // particular audio endpoint device and represents the smallest processing 595 // quantum for the audio engine. This period plus the stream latency between 596 // the buffer and endpoint device represents the minimum possible latency 597 // that an audio application can achieve. 598 // TODO(henrika): possibly remove this section when all parts are ready. 599 REFERENCE_TIME device_period_shared_mode = 0; 600 REFERENCE_TIME device_period_exclusive_mode = 0; 601 HRESULT hr_dbg = audio_client_->GetDevicePeriod( 602 &device_period_shared_mode, &device_period_exclusive_mode); 603 if (SUCCEEDED(hr_dbg)) { 604 DVLOG(1) << "device period: " 605 << static_cast<double>(device_period_shared_mode / 10000.0) 606 << " [ms]"; 607 } 608 609 REFERENCE_TIME latency = 0; 610 hr_dbg = audio_client_->GetStreamLatency(&latency); 611 if (SUCCEEDED(hr_dbg)) { 612 DVLOG(1) << "stream latency: " << static_cast<double>(latency / 10000.0) 613 << " [ms]"; 614 } 615#endif 616 617 // Set the event handle that the audio engine will signal each time 618 // a buffer becomes ready to be processed by the client. 619 hr = audio_client_->SetEventHandle(audio_samples_ready_event_.Get()); 620 if (FAILED(hr)) 621 return hr; 622 623 // Get access to the IAudioCaptureClient interface. This interface 624 // enables us to read input data from the capture endpoint buffer. 625 hr = audio_client_->GetService(__uuidof(IAudioCaptureClient), 626 audio_capture_client_.ReceiveVoid()); 627 if (FAILED(hr)) 628 return hr; 629 630 // Obtain a reference to the ISimpleAudioVolume interface which enables 631 // us to control the master volume level of an audio session. 632 hr = audio_client_->GetService(__uuidof(ISimpleAudioVolume), 633 simple_audio_volume_.ReceiveVoid()); 634 return hr; 635} 636 637} // namespace media 638