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.
5 #include "media/audio/win/audio_low_latency_input_win.h"
7 #include "base/logging.h"
8 #include "base/memory/scoped_ptr.h"
9 #include "base/strings/utf_string_conversions.h"
10 #include "media/audio/win/audio_manager_win.h"
11 #include "media/audio/win/avrt_wrapper_win.h"
12 #include "media/audio/win/core_audio_util_win.h"
13 #include "media/base/audio_bus.h"
15 using base::win::ScopedComPtr;
16 using base::win::ScopedCOMInitializer;
21 // Returns true if |device| represents the default communication capture device.
22 bool IsDefaultCommunicationDevice(IMMDeviceEnumerator* enumerator,
24 ScopedComPtr<IMMDevice> communications;
25 if (FAILED(enumerator->GetDefaultAudioEndpoint(eCapture, eCommunications,
26 communications.Receive()))) {
30 base::win::ScopedCoMem<WCHAR> communications_id, device_id;
31 device->GetId(&device_id);
32 communications->GetId(&communications_id);
33 return lstrcmpW(communications_id, device_id) == 0;
38 WASAPIAudioInputStream::WASAPIAudioInputStream(AudioManagerWin* manager,
39 const AudioParameters& params,
40 const std::string& device_id)
42 capture_thread_(NULL),
46 packet_size_frames_(0),
47 packet_size_bytes_(0),
48 endpoint_buffer_size_frames_(0),
49 effects_(params.effects()),
50 device_id_(device_id),
51 perf_count_to_100ns_units_(0.0),
52 ms_to_frame_count_(0.0),
54 audio_bus_(media::AudioBus::Create(params)) {
57 // Load the Avrt DLL if not already loaded. Required to support MMCSS.
58 bool avrt_init = avrt::Initialize();
59 DCHECK(avrt_init) << "Failed to load the Avrt.dll";
61 // Set up the desired capture format specified by the client.
62 format_.nSamplesPerSec = params.sample_rate();
63 format_.wFormatTag = WAVE_FORMAT_PCM;
64 format_.wBitsPerSample = params.bits_per_sample();
65 format_.nChannels = params.channels();
66 format_.nBlockAlign = (format_.wBitsPerSample / 8) * format_.nChannels;
67 format_.nAvgBytesPerSec = format_.nSamplesPerSec * format_.nBlockAlign;
70 // Size in bytes of each audio frame.
71 frame_size_ = format_.nBlockAlign;
72 // Store size of audio packets which we expect to get from the audio
73 // endpoint device in each capture event.
74 packet_size_frames_ = params.GetBytesPerBuffer() / format_.nBlockAlign;
75 packet_size_bytes_ = params.GetBytesPerBuffer();
76 DVLOG(1) << "Number of bytes per audio frame : " << frame_size_;
77 DVLOG(1) << "Number of audio frames per packet: " << packet_size_frames_;
79 // All events are auto-reset events and non-signaled initially.
81 // Create the event which the audio engine will signal each time
82 // a buffer becomes ready to be processed by the client.
83 audio_samples_ready_event_.Set(CreateEvent(NULL, FALSE, FALSE, NULL));
84 DCHECK(audio_samples_ready_event_.IsValid());
86 // Create the event which will be set in Stop() when capturing shall stop.
87 stop_capture_event_.Set(CreateEvent(NULL, FALSE, FALSE, NULL));
88 DCHECK(stop_capture_event_.IsValid());
90 ms_to_frame_count_ = static_cast<double>(params.sample_rate()) / 1000.0;
92 LARGE_INTEGER performance_frequency;
93 if (QueryPerformanceFrequency(&performance_frequency)) {
94 perf_count_to_100ns_units_ =
95 (10000000.0 / static_cast<double>(performance_frequency.QuadPart));
97 DLOG(ERROR) << "High-resolution performance counters are not supported.";
101 WASAPIAudioInputStream::~WASAPIAudioInputStream() {
102 DCHECK(CalledOnValidThread());
105 bool WASAPIAudioInputStream::Open() {
106 DCHECK(CalledOnValidThread());
107 // Verify that we are not already opened.
111 // Obtain a reference to the IMMDevice interface of the capturing
112 // device with the specified unique identifier or role which was
113 // set at construction.
114 HRESULT hr = SetCaptureDevice();
118 // Obtain an IAudioClient interface which enables us to create and initialize
119 // an audio stream between an audio application and the audio engine.
120 hr = ActivateCaptureDevice();
124 // Retrieve the stream format which the audio engine uses for its internal
125 // processing/mixing of shared-mode streams. This function call is for
126 // diagnostic purposes only and only in debug mode.
128 hr = GetAudioEngineStreamFormat();
131 // Verify that the selected audio endpoint supports the specified format
132 // set during construction.
133 if (!DesiredFormatIsSupported())
136 // Initialize the audio stream between the client and the device using
137 // shared mode and a lowest possible glitch-free latency.
138 hr = InitializeAudioEngine();
140 opened_ = SUCCEEDED(hr);
144 void WASAPIAudioInputStream::Start(AudioInputCallback* callback) {
145 DCHECK(CalledOnValidThread());
147 DLOG_IF(ERROR, !opened_) << "Open() has not been called successfully";
157 // Starts periodic AGC microphone measurements if the AGC has been enabled
158 // using SetAutomaticGainControl().
161 // Create and start the thread that will drive the capturing by waiting for
164 new base::DelegateSimpleThread(this, "wasapi_capture_thread");
165 capture_thread_->Start();
167 // Start streaming data between the endpoint buffer and the audio engine.
168 HRESULT hr = audio_client_->Start();
169 DLOG_IF(ERROR, FAILED(hr)) << "Failed to start input streaming.";
171 if (SUCCEEDED(hr) && audio_render_client_for_loopback_)
172 hr = audio_render_client_for_loopback_->Start();
174 started_ = SUCCEEDED(hr);
177 void WASAPIAudioInputStream::Stop() {
178 DCHECK(CalledOnValidThread());
179 DVLOG(1) << "WASAPIAudioInputStream::Stop()";
183 // Stops periodic AGC microphone measurements.
186 // Shut down the capture thread.
187 if (stop_capture_event_.IsValid()) {
188 SetEvent(stop_capture_event_.Get());
191 // Stop the input audio streaming.
192 HRESULT hr = audio_client_->Stop();
194 LOG(ERROR) << "Failed to stop input streaming.";
197 // Wait until the thread completes and perform cleanup.
198 if (capture_thread_) {
199 SetEvent(stop_capture_event_.Get());
200 capture_thread_->Join();
201 capture_thread_ = NULL;
208 void WASAPIAudioInputStream::Close() {
209 DVLOG(1) << "WASAPIAudioInputStream::Close()";
210 // It is valid to call Close() before calling open or Start().
211 // It is also valid to call Close() after Start() has been called.
214 // Inform the audio manager that we have been closed. This will cause our
216 manager_->ReleaseInputStream(this);
219 double WASAPIAudioInputStream::GetMaxVolume() {
220 // Verify that Open() has been called succesfully, to ensure that an audio
221 // session exists and that an ISimpleAudioVolume interface has been created.
222 DLOG_IF(ERROR, !opened_) << "Open() has not been called successfully";
226 // The effective volume value is always in the range 0.0 to 1.0, hence
227 // we can return a fixed value (=1.0) here.
231 void WASAPIAudioInputStream::SetVolume(double volume) {
232 DVLOG(1) << "SetVolume(volume=" << volume << ")";
233 DCHECK(CalledOnValidThread());
234 DCHECK_GE(volume, 0.0);
235 DCHECK_LE(volume, 1.0);
237 DLOG_IF(ERROR, !opened_) << "Open() has not been called successfully";
241 // Set a new master volume level. Valid volume levels are in the range
242 // 0.0 to 1.0. Ignore volume-change events.
243 HRESULT hr = simple_audio_volume_->SetMasterVolume(static_cast<float>(volume),
245 DLOG_IF(WARNING, FAILED(hr)) << "Failed to set new input master volume.";
247 // Update the AGC volume level based on the last setting above. Note that,
248 // the volume-level resolution is not infinite and it is therefore not
249 // possible to assume that the volume provided as input parameter can be
250 // used directly. Instead, a new query to the audio hardware is required.
251 // This method does nothing if AGC is disabled.
255 double WASAPIAudioInputStream::GetVolume() {
256 DCHECK(opened_) << "Open() has not been called successfully";
260 // Retrieve the current volume level. The value is in the range 0.0 to 1.0.
262 HRESULT hr = simple_audio_volume_->GetMasterVolume(&level);
263 DLOG_IF(WARNING, FAILED(hr)) << "Failed to get input master volume.";
265 return static_cast<double>(level);
268 bool WASAPIAudioInputStream::IsMuted() {
269 DCHECK(opened_) << "Open() has not been called successfully";
270 DCHECK(CalledOnValidThread());
274 // Retrieves the current muting state for the audio session.
275 BOOL is_muted = FALSE;
276 HRESULT hr = simple_audio_volume_->GetMute(&is_muted);
277 DLOG_IF(WARNING, FAILED(hr)) << "Failed to get input master volume.";
279 return is_muted != FALSE;
283 AudioParameters WASAPIAudioInputStream::GetInputStreamParameters(
284 const std::string& device_id) {
285 int sample_rate = 48000;
286 ChannelLayout channel_layout = CHANNEL_LAYOUT_STEREO;
288 base::win::ScopedCoMem<WAVEFORMATEX> audio_engine_mix_format;
289 int effects = AudioParameters::NO_EFFECTS;
290 if (SUCCEEDED(GetMixFormat(device_id, &audio_engine_mix_format, &effects))) {
291 sample_rate = static_cast<int>(audio_engine_mix_format->nSamplesPerSec);
292 channel_layout = audio_engine_mix_format->nChannels == 1 ?
293 CHANNEL_LAYOUT_MONO : CHANNEL_LAYOUT_STEREO;
296 // Use 10ms frame size as default.
297 int frames_per_buffer = sample_rate / 100;
298 return AudioParameters(
299 AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout, sample_rate,
300 16, frames_per_buffer, effects);
304 HRESULT WASAPIAudioInputStream::GetMixFormat(const std::string& device_id,
305 WAVEFORMATEX** device_format,
309 // It is assumed that this static method is called from a COM thread, i.e.,
310 // CoInitializeEx() is not called here to avoid STA/MTA conflicts.
311 ScopedComPtr<IMMDeviceEnumerator> enumerator;
312 HRESULT hr = enumerator.CreateInstance(__uuidof(MMDeviceEnumerator), NULL,
313 CLSCTX_INPROC_SERVER);
317 ScopedComPtr<IMMDevice> endpoint_device;
318 if (device_id == AudioManagerBase::kDefaultDeviceId) {
319 // Retrieve the default capture audio endpoint.
320 hr = enumerator->GetDefaultAudioEndpoint(eCapture, eConsole,
321 endpoint_device.Receive());
322 } else if (device_id == AudioManagerBase::kLoopbackInputDeviceId) {
323 // Get the mix format of the default playback stream.
324 hr = enumerator->GetDefaultAudioEndpoint(eRender, eConsole,
325 endpoint_device.Receive());
327 // Retrieve a capture endpoint device that is specified by an endpoint
328 // device-identification string.
329 hr = enumerator->GetDevice(base::UTF8ToUTF16(device_id).c_str(),
330 endpoint_device.Receive());
336 *effects = IsDefaultCommunicationDevice(enumerator, endpoint_device) ?
337 AudioParameters::DUCKING : AudioParameters::NO_EFFECTS;
339 ScopedComPtr<IAudioClient> audio_client;
340 hr = endpoint_device->Activate(__uuidof(IAudioClient),
341 CLSCTX_INPROC_SERVER,
343 audio_client.ReceiveVoid());
344 return SUCCEEDED(hr) ? audio_client->GetMixFormat(device_format) : hr;
347 void WASAPIAudioInputStream::Run() {
348 ScopedCOMInitializer com_init(ScopedCOMInitializer::kMTA);
350 // Increase the thread priority.
351 capture_thread_->SetThreadPriority(base::kThreadPriority_RealtimeAudio);
353 // Enable MMCSS to ensure that this thread receives prioritized access to
355 DWORD task_index = 0;
356 HANDLE mm_task = avrt::AvSetMmThreadCharacteristics(L"Pro Audio",
359 (mm_task && avrt::AvSetMmThreadPriority(mm_task, AVRT_PRIORITY_CRITICAL));
361 // Failed to enable MMCSS on this thread. It is not fatal but can lead
362 // to reduced QoS at high load.
363 DWORD err = GetLastError();
364 LOG(WARNING) << "Failed to enable MMCSS (error code=" << err << ").";
367 // Allocate a buffer with a size that enables us to take care of cases like:
368 // 1) The recorded buffer size is smaller, or does not match exactly with,
369 // the selected packet size used in each callback.
370 // 2) The selected buffer size is larger than the recorded buffer size in
372 size_t buffer_frame_index = 0;
373 size_t capture_buffer_size = std::max(
374 2 * endpoint_buffer_size_frames_ * frame_size_,
375 2 * packet_size_frames_ * frame_size_);
376 scoped_ptr<uint8[]> capture_buffer(new uint8[capture_buffer_size]);
378 LARGE_INTEGER now_count;
379 bool recording = true;
381 double volume = GetVolume();
382 HANDLE wait_array[2] =
383 { stop_capture_event_.Get(), audio_samples_ready_event_.Get() };
385 while (recording && !error) {
386 HRESULT hr = S_FALSE;
388 // Wait for a close-down event or a new capture event.
389 DWORD wait_result = WaitForMultipleObjects(2, wait_array, FALSE, INFINITE);
390 switch (wait_result) {
394 case WAIT_OBJECT_0 + 0:
395 // |stop_capture_event_| has been set.
398 case WAIT_OBJECT_0 + 1:
400 // |audio_samples_ready_event_| has been set.
401 BYTE* data_ptr = NULL;
402 UINT32 num_frames_to_read = 0;
404 UINT64 device_position = 0;
405 UINT64 first_audio_frame_timestamp = 0;
407 // Retrieve the amount of data in the capture endpoint buffer,
408 // replace it with silence if required, create callbacks for each
409 // packet and store non-delivered data for the next event.
410 hr = audio_capture_client_->GetBuffer(&data_ptr,
414 &first_audio_frame_timestamp);
416 DLOG(ERROR) << "Failed to get data from the capture buffer";
420 if (num_frames_to_read != 0) {
421 size_t pos = buffer_frame_index * frame_size_;
422 size_t num_bytes = num_frames_to_read * frame_size_;
423 DCHECK_GE(capture_buffer_size, pos + num_bytes);
425 if (flags & AUDCLNT_BUFFERFLAGS_SILENT) {
426 // Clear out the local buffer since silence is reported.
427 memset(&capture_buffer[pos], 0, num_bytes);
429 // Copy captured data from audio engine buffer to local buffer.
430 memcpy(&capture_buffer[pos], data_ptr, num_bytes);
433 buffer_frame_index += num_frames_to_read;
436 hr = audio_capture_client_->ReleaseBuffer(num_frames_to_read);
437 DLOG_IF(ERROR, FAILED(hr)) << "Failed to release capture buffer";
439 // Derive a delay estimate for the captured audio packet.
440 // The value contains two parts (A+B), where A is the delay of the
441 // first audio frame in the packet and B is the extra delay
442 // contained in any stored data. Unit is in audio frames.
443 QueryPerformanceCounter(&now_count);
444 double audio_delay_frames =
445 ((perf_count_to_100ns_units_ * now_count.QuadPart -
446 first_audio_frame_timestamp) / 10000.0) * ms_to_frame_count_ +
447 buffer_frame_index - num_frames_to_read;
449 // Get a cached AGC volume level which is updated once every second
450 // on the audio manager thread. Note that, |volume| is also updated
451 // each time SetVolume() is called through IPC by the render-side AGC.
452 GetAgcVolume(&volume);
454 // Deliver captured data to the registered consumer using a packet
455 // size which was specified at construction.
456 uint32 delay_frames = static_cast<uint32>(audio_delay_frames + 0.5);
457 while (buffer_frame_index >= packet_size_frames_) {
458 // Copy data to audio bus to match the OnData interface.
459 uint8* audio_data = reinterpret_cast<uint8*>(capture_buffer.get());
460 audio_bus_->FromInterleaved(
461 audio_data, audio_bus_->frames(), format_.wBitsPerSample / 8);
463 // Deliver data packet, delay estimation and volume level to
466 this, audio_bus_.get(), delay_frames * frame_size_, volume);
468 // Store parts of the recorded data which can't be delivered
469 // using the current packet size. The stored section will be used
470 // either in the next while-loop iteration or in the next
472 memmove(&capture_buffer[0],
473 &capture_buffer[packet_size_bytes_],
474 (buffer_frame_index - packet_size_frames_) * frame_size_);
476 buffer_frame_index -= packet_size_frames_;
477 delay_frames -= packet_size_frames_;
487 if (recording && error) {
488 // TODO(henrika): perhaps it worth improving the cleanup here by e.g.
489 // stopping the audio client, joining the thread etc.?
490 NOTREACHED() << "WASAPI capturing failed with error code "
495 if (mm_task && !avrt::AvRevertMmThreadCharacteristics(mm_task)) {
496 PLOG(WARNING) << "Failed to disable MMCSS";
500 void WASAPIAudioInputStream::HandleError(HRESULT err) {
501 NOTREACHED() << "Error code: " << err;
503 sink_->OnError(this);
506 HRESULT WASAPIAudioInputStream::SetCaptureDevice() {
507 DCHECK(!endpoint_device_);
509 ScopedComPtr<IMMDeviceEnumerator> enumerator;
510 HRESULT hr = enumerator.CreateInstance(__uuidof(MMDeviceEnumerator),
511 NULL, CLSCTX_INPROC_SERVER);
515 // Retrieve the IMMDevice by using the specified role or the specified
516 // unique endpoint device-identification string.
518 if (effects_ & AudioParameters::DUCKING) {
519 // Ducking has been requested and it is only supported for the default
520 // communication device. So, let's open up the communication device and
521 // see if the ID of that device matches the requested ID.
522 // We consider a kDefaultDeviceId as well as an explicit device id match,
523 // to be valid matches.
524 hr = enumerator->GetDefaultAudioEndpoint(eCapture, eCommunications,
525 endpoint_device_.Receive());
526 if (endpoint_device_ && device_id_ != AudioManagerBase::kDefaultDeviceId) {
527 base::win::ScopedCoMem<WCHAR> communications_id;
528 endpoint_device_->GetId(&communications_id);
530 base::WideToUTF8(static_cast<WCHAR*>(communications_id))) {
531 DLOG(WARNING) << "Ducking has been requested for a non-default device."
533 // We can't honor the requested effect flag, so turn it off and
534 // continue. We'll check this flag later to see if we've actually
535 // opened up the communications device, so it's important that it
536 // reflects the active state.
537 effects_ &= ~AudioParameters::DUCKING;
538 endpoint_device_.Release(); // Fall back on code below.
543 if (!endpoint_device_) {
544 if (device_id_ == AudioManagerBase::kDefaultDeviceId) {
545 // Retrieve the default capture audio endpoint for the specified role.
546 // Note that, in Windows Vista, the MMDevice API supports device roles
547 // but the system-supplied user interface programs do not.
548 hr = enumerator->GetDefaultAudioEndpoint(eCapture, eConsole,
549 endpoint_device_.Receive());
550 } else if (device_id_ == AudioManagerBase::kLoopbackInputDeviceId) {
551 // Capture the default playback stream.
552 hr = enumerator->GetDefaultAudioEndpoint(eRender, eConsole,
553 endpoint_device_.Receive());
555 hr = enumerator->GetDevice(base::UTF8ToUTF16(device_id_).c_str(),
556 endpoint_device_.Receive());
563 // Verify that the audio endpoint device is active, i.e., the audio
564 // adapter that connects to the endpoint device is present and enabled.
565 DWORD state = DEVICE_STATE_DISABLED;
566 hr = endpoint_device_->GetState(&state);
570 if (!(state & DEVICE_STATE_ACTIVE)) {
571 DLOG(ERROR) << "Selected capture device is not active.";
578 HRESULT WASAPIAudioInputStream::ActivateCaptureDevice() {
579 // Creates and activates an IAudioClient COM object given the selected
580 // capture endpoint device.
581 HRESULT hr = endpoint_device_->Activate(__uuidof(IAudioClient),
582 CLSCTX_INPROC_SERVER,
584 audio_client_.ReceiveVoid());
588 HRESULT WASAPIAudioInputStream::GetAudioEngineStreamFormat() {
591 // The GetMixFormat() method retrieves the stream format that the
592 // audio engine uses for its internal processing of shared-mode streams.
593 // The method always uses a WAVEFORMATEXTENSIBLE structure, instead
594 // of a stand-alone WAVEFORMATEX structure, to specify the format.
595 // An WAVEFORMATEXTENSIBLE structure can specify both the mapping of
596 // channels to speakers and the number of bits of precision in each sample.
597 base::win::ScopedCoMem<WAVEFORMATEXTENSIBLE> format_ex;
598 hr = audio_client_->GetMixFormat(
599 reinterpret_cast<WAVEFORMATEX**>(&format_ex));
601 // See http://msdn.microsoft.com/en-us/windows/hardware/gg463006#EFH
602 // for details on the WAVE file format.
603 WAVEFORMATEX format = format_ex->Format;
604 DVLOG(2) << "WAVEFORMATEX:";
605 DVLOG(2) << " wFormatTags : 0x" << std::hex << format.wFormatTag;
606 DVLOG(2) << " nChannels : " << format.nChannels;
607 DVLOG(2) << " nSamplesPerSec : " << format.nSamplesPerSec;
608 DVLOG(2) << " nAvgBytesPerSec: " << format.nAvgBytesPerSec;
609 DVLOG(2) << " nBlockAlign : " << format.nBlockAlign;
610 DVLOG(2) << " wBitsPerSample : " << format.wBitsPerSample;
611 DVLOG(2) << " cbSize : " << format.cbSize;
613 DVLOG(2) << "WAVEFORMATEXTENSIBLE:";
614 DVLOG(2) << " wValidBitsPerSample: " <<
615 format_ex->Samples.wValidBitsPerSample;
616 DVLOG(2) << " dwChannelMask : 0x" << std::hex <<
617 format_ex->dwChannelMask;
618 if (format_ex->SubFormat == KSDATAFORMAT_SUBTYPE_PCM)
619 DVLOG(2) << " SubFormat : KSDATAFORMAT_SUBTYPE_PCM";
620 else if (format_ex->SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT)
621 DVLOG(2) << " SubFormat : KSDATAFORMAT_SUBTYPE_IEEE_FLOAT";
622 else if (format_ex->SubFormat == KSDATAFORMAT_SUBTYPE_WAVEFORMATEX)
623 DVLOG(2) << " SubFormat : KSDATAFORMAT_SUBTYPE_WAVEFORMATEX";
628 bool WASAPIAudioInputStream::DesiredFormatIsSupported() {
629 // An application that uses WASAPI to manage shared-mode streams can rely
630 // on the audio engine to perform only limited format conversions. The audio
631 // engine can convert between a standard PCM sample size used by the
632 // application and the floating-point samples that the engine uses for its
633 // internal processing. However, the format for an application stream
634 // typically must have the same number of channels and the same sample
635 // rate as the stream format used by the device.
636 // Many audio devices support both PCM and non-PCM stream formats. However,
637 // the audio engine can mix only PCM streams.
638 base::win::ScopedCoMem<WAVEFORMATEX> closest_match;
639 HRESULT hr = audio_client_->IsFormatSupported(AUDCLNT_SHAREMODE_SHARED,
642 DLOG_IF(ERROR, hr == S_FALSE) << "Format is not supported "
643 << "but a closest match exists.";
647 HRESULT WASAPIAudioInputStream::InitializeAudioEngine() {
649 // Use event-driven mode only fo regular input devices. For loopback the
650 // EVENTCALLBACK flag is specified when intializing
651 // |audio_render_client_for_loopback_|.
652 if (device_id_ == AudioManagerBase::kLoopbackInputDeviceId) {
653 flags = AUDCLNT_STREAMFLAGS_LOOPBACK | AUDCLNT_STREAMFLAGS_NOPERSIST;
656 AUDCLNT_STREAMFLAGS_EVENTCALLBACK | AUDCLNT_STREAMFLAGS_NOPERSIST;
659 // Initialize the audio stream between the client and the device.
660 // We connect indirectly through the audio engine by using shared mode.
661 // Note that, |hnsBufferDuration| is set of 0, which ensures that the
662 // buffer is never smaller than the minimum buffer size needed to ensure
663 // that glitches do not occur between the periodic processing passes.
664 // This setting should lead to lowest possible latency.
665 HRESULT hr = audio_client_->Initialize(
666 AUDCLNT_SHAREMODE_SHARED,
668 0, // hnsBufferDuration
671 (effects_ & AudioParameters::DUCKING) ? &kCommunicationsSessionId : NULL);
676 // Retrieve the length of the endpoint buffer shared between the client
677 // and the audio engine. The buffer length determines the maximum amount
678 // of capture data that the audio engine can read from the endpoint buffer
679 // during a single processing pass.
680 // A typical value is 960 audio frames <=> 20ms @ 48kHz sample rate.
681 hr = audio_client_->GetBufferSize(&endpoint_buffer_size_frames_);
685 DVLOG(1) << "endpoint buffer size: " << endpoint_buffer_size_frames_
689 // The period between processing passes by the audio engine is fixed for a
690 // particular audio endpoint device and represents the smallest processing
691 // quantum for the audio engine. This period plus the stream latency between
692 // the buffer and endpoint device represents the minimum possible latency
693 // that an audio application can achieve.
694 // TODO(henrika): possibly remove this section when all parts are ready.
695 REFERENCE_TIME device_period_shared_mode = 0;
696 REFERENCE_TIME device_period_exclusive_mode = 0;
697 HRESULT hr_dbg = audio_client_->GetDevicePeriod(
698 &device_period_shared_mode, &device_period_exclusive_mode);
699 if (SUCCEEDED(hr_dbg)) {
700 DVLOG(1) << "device period: "
701 << static_cast<double>(device_period_shared_mode / 10000.0)
705 REFERENCE_TIME latency = 0;
706 hr_dbg = audio_client_->GetStreamLatency(&latency);
707 if (SUCCEEDED(hr_dbg)) {
708 DVLOG(1) << "stream latency: " << static_cast<double>(latency / 10000.0)
713 // Set the event handle that the audio engine will signal each time a buffer
714 // becomes ready to be processed by the client.
716 // In loopback case the capture device doesn't receive any events, so we
717 // need to create a separate playback client to get notifications. According
720 // A pull-mode capture client does not receive any events when a stream is
721 // initialized with event-driven buffering and is loopback-enabled. To
722 // work around this, initialize a render stream in event-driven mode. Each
723 // time the client receives an event for the render stream, it must signal
724 // the capture client to run the capture thread that reads the next set of
725 // samples from the capture endpoint buffer.
727 // http://msdn.microsoft.com/en-us/library/windows/desktop/dd316551(v=vs.85).aspx
728 if (device_id_ == AudioManagerBase::kLoopbackInputDeviceId) {
729 hr = endpoint_device_->Activate(
730 __uuidof(IAudioClient), CLSCTX_INPROC_SERVER, NULL,
731 audio_render_client_for_loopback_.ReceiveVoid());
735 hr = audio_render_client_for_loopback_->Initialize(
736 AUDCLNT_SHAREMODE_SHARED,
737 AUDCLNT_STREAMFLAGS_EVENTCALLBACK | AUDCLNT_STREAMFLAGS_NOPERSIST,
738 0, 0, &format_, NULL);
742 hr = audio_render_client_for_loopback_->SetEventHandle(
743 audio_samples_ready_event_.Get());
745 hr = audio_client_->SetEventHandle(audio_samples_ready_event_.Get());
751 // Get access to the IAudioCaptureClient interface. This interface
752 // enables us to read input data from the capture endpoint buffer.
753 hr = audio_client_->GetService(__uuidof(IAudioCaptureClient),
754 audio_capture_client_.ReceiveVoid());
758 // Obtain a reference to the ISimpleAudioVolume interface which enables
759 // us to control the master volume level of an audio session.
760 hr = audio_client_->GetService(__uuidof(ISimpleAudioVolume),
761 simple_audio_volume_.ReceiveVoid());