root/media/audio/win/audio_low_latency_input_win.cc

DEFINITIONS

1. IsDefaultCommunicationDevice
2. sink_
3. Open
4. Start
5. Stop
6. Close
7. GetMaxVolume
8. SetVolume
9. GetVolume
10. GetInputStreamParameters
11. GetMixFormat
12. Run
13. HandleError
14. SetCaptureDevice
15. ActivateCaptureDevice
16. GetAudioEngineStreamFormat
17. DesiredFormatIsSupported
18. InitializeAudioEngine

// Copyright (c) 2012 The Chromium Authors. All rights reserved.
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.

#include "media/audio/win/audio_low_latency_input_win.h"

#include "base/logging.h"
#include "base/memory/scoped_ptr.h"
#include "base/strings/utf_string_conversions.h"
#include "media/audio/win/audio_manager_win.h"
#include "media/audio/win/avrt_wrapper_win.h"

using base::win::ScopedComPtr;
using base::win::ScopedCOMInitializer;

namespace media {
namespace {

// Returns true if |device| represents the default communication capture device.
bool IsDefaultCommunicationDevice(IMMDeviceEnumerator* enumerator,
                                  IMMDevice* device) {
  ScopedComPtr<IMMDevice> communications;
  if (FAILED(enumerator->GetDefaultAudioEndpoint(eCapture, eCommunications,
                                                 communications.Receive()))) {
    return false;
  }

  base::win::ScopedCoMem<WCHAR> communications_id, device_id;
  device->GetId(&device_id);
  communications->GetId(&communications_id);
  return lstrcmpW(communications_id, device_id) == 0;
}

}  // namespace

WASAPIAudioInputStream::WASAPIAudioInputStream(
    AudioManagerWin* manager,
    const AudioParameters& params,
    const std::string& device_id)
    : manager_(manager),
      capture_thread_(NULL),
      opened_(false),
      started_(false),
      frame_size_(0),
      packet_size_frames_(0),
      packet_size_bytes_(0),
      endpoint_buffer_size_frames_(0),
      effects_(params.effects()),
      device_id_(device_id),
      perf_count_to_100ns_units_(0.0),
      ms_to_frame_count_(0.0),
      sink_(NULL) {
  DCHECK(manager_);

  // Load the Avrt DLL if not already loaded. Required to support MMCSS.
  bool avrt_init = avrt::Initialize();
  DCHECK(avrt_init) << "Failed to load the Avrt.dll";

  // Set up the desired capture format specified by the client.
  format_.nSamplesPerSec = params.sample_rate();
  format_.wFormatTag = WAVE_FORMAT_PCM;
  format_.wBitsPerSample = params.bits_per_sample();
  format_.nChannels = params.channels();
  format_.nBlockAlign = (format_.wBitsPerSample / 8) * format_.nChannels;
  format_.nAvgBytesPerSec = format_.nSamplesPerSec * format_.nBlockAlign;
  format_.cbSize = 0;

  // Size in bytes of each audio frame.
  frame_size_ = format_.nBlockAlign;
  // Store size of audio packets which we expect to get from the audio
  // endpoint device in each capture event.
  packet_size_frames_ = params.GetBytesPerBuffer() / format_.nBlockAlign;
  packet_size_bytes_ = params.GetBytesPerBuffer();
  DVLOG(1) << "Number of bytes per audio frame  : " << frame_size_;
  DVLOG(1) << "Number of audio frames per packet: " << packet_size_frames_;

  // All events are auto-reset events and non-signaled initially.

  // Create the event which the audio engine will signal each time
  // a buffer becomes ready to be processed by the client.
  audio_samples_ready_event_.Set(CreateEvent(NULL, FALSE, FALSE, NULL));
  DCHECK(audio_samples_ready_event_.IsValid());

  // Create the event which will be set in Stop() when capturing shall stop.
  stop_capture_event_.Set(CreateEvent(NULL, FALSE, FALSE, NULL));
  DCHECK(stop_capture_event_.IsValid());

  ms_to_frame_count_ = static_cast<double>(params.sample_rate()) / 1000.0;

  LARGE_INTEGER performance_frequency;
  if (QueryPerformanceFrequency(&performance_frequency)) {
    perf_count_to_100ns_units_ =
        (10000000.0 / static_cast<double>(performance_frequency.QuadPart));
  } else {
    DLOG(ERROR) << "High-resolution performance counters are not supported.";
  }
}

WASAPIAudioInputStream::~WASAPIAudioInputStream() {}

bool WASAPIAudioInputStream::Open() {
  DCHECK(CalledOnValidThread());
  // Verify that we are not already opened.
  if (opened_)
    return false;

  // Obtain a reference to the IMMDevice interface of the capturing
  // device with the specified unique identifier or role which was
  // set at construction.
  HRESULT hr = SetCaptureDevice();
  if (FAILED(hr))
    return false;

  // Obtain an IAudioClient interface which enables us to create and initialize
  // an audio stream between an audio application and the audio engine.
  hr = ActivateCaptureDevice();
  if (FAILED(hr))
    return false;

  // Retrieve the stream format which the audio engine uses for its internal
  // processing/mixing of shared-mode streams. This function call is for
  // diagnostic purposes only and only in debug mode.
#ifndef NDEBUG
  hr = GetAudioEngineStreamFormat();
#endif

  // Verify that the selected audio endpoint supports the specified format
  // set during construction.
  if (!DesiredFormatIsSupported())
    return false;

  // Initialize the audio stream between the client and the device using
  // shared mode and a lowest possible glitch-free latency.
  hr = InitializeAudioEngine();

  opened_ = SUCCEEDED(hr);
  return opened_;
}

void WASAPIAudioInputStream::Start(AudioInputCallback* callback) {
  DCHECK(CalledOnValidThread());
  DCHECK(callback);
  DLOG_IF(ERROR, !opened_) << "Open() has not been called successfully";
  if (!opened_)
    return;

  if (started_)
    return;

  DCHECK(!sink_);
  sink_ = callback;

  // Starts periodic AGC microphone measurements if the AGC has been enabled
  // using SetAutomaticGainControl().
  StartAgc();

  // Create and start the thread that will drive the capturing by waiting for
  // capture events.
  capture_thread_ =
      new base::DelegateSimpleThread(this, "wasapi_capture_thread");
  capture_thread_->Start();

  // Start streaming data between the endpoint buffer and the audio engine.
  HRESULT hr = audio_client_->Start();
  DLOG_IF(ERROR, FAILED(hr)) << "Failed to start input streaming.";

  if (SUCCEEDED(hr) && audio_render_client_for_loopback_)
    hr = audio_render_client_for_loopback_->Start();

  started_ = SUCCEEDED(hr);
}

void WASAPIAudioInputStream::Stop() {
  DCHECK(CalledOnValidThread());
  DVLOG(1) << "WASAPIAudioInputStream::Stop()";
  if (!started_)
    return;

  // Stops periodic AGC microphone measurements.
  StopAgc();

  // Shut down the capture thread.
  if (stop_capture_event_.IsValid()) {
    SetEvent(stop_capture_event_.Get());
  }

  // Stop the input audio streaming.
  HRESULT hr = audio_client_->Stop();
  if (FAILED(hr)) {
    LOG(ERROR) << "Failed to stop input streaming.";
  }

  // Wait until the thread completes and perform cleanup.
  if (capture_thread_) {
    SetEvent(stop_capture_event_.Get());
    capture_thread_->Join();
    capture_thread_ = NULL;
  }

  started_ = false;
  sink_ = NULL;
}

void WASAPIAudioInputStream::Close() {
  DVLOG(1) << "WASAPIAudioInputStream::Close()";
  // It is valid to call Close() before calling open or Start().
  // It is also valid to call Close() after Start() has been called.
  Stop();

  // Inform the audio manager that we have been closed. This will cause our
  // destruction.
  manager_->ReleaseInputStream(this);
}

double WASAPIAudioInputStream::GetMaxVolume() {
  // Verify that Open() has been called succesfully, to ensure that an audio
  // session exists and that an ISimpleAudioVolume interface has been created.
  DLOG_IF(ERROR, !opened_) << "Open() has not been called successfully";
  if (!opened_)
    return 0.0;

  // The effective volume value is always in the range 0.0 to 1.0, hence
  // we can return a fixed value (=1.0) here.
  return 1.0;
}

void WASAPIAudioInputStream::SetVolume(double volume) {
  DVLOG(1) << "SetVolume(volume=" << volume << ")";
  DCHECK(CalledOnValidThread());
  DCHECK_GE(volume, 0.0);
  DCHECK_LE(volume, 1.0);

  DLOG_IF(ERROR, !opened_) << "Open() has not been called successfully";
  if (!opened_)
    return;

  // Set a new master volume level. Valid volume levels are in the range
  // 0.0 to 1.0. Ignore volume-change events.
  HRESULT hr = simple_audio_volume_->SetMasterVolume(static_cast<float>(volume),
      NULL);
  DLOG_IF(WARNING, FAILED(hr)) << "Failed to set new input master volume.";

  // Update the AGC volume level based on the last setting above. Note that,
  // the volume-level resolution is not infinite and it is therefore not
  // possible to assume that the volume provided as input parameter can be
  // used directly. Instead, a new query to the audio hardware is required.
  // This method does nothing if AGC is disabled.
  UpdateAgcVolume();
}

double WASAPIAudioInputStream::GetVolume() {
  DLOG_IF(ERROR, !opened_) << "Open() has not been called successfully";
  if (!opened_)
    return 0.0;

  // Retrieve the current volume level. The value is in the range 0.0 to 1.0.
  float level = 0.0f;
  HRESULT hr = simple_audio_volume_->GetMasterVolume(&level);
  DLOG_IF(WARNING, FAILED(hr)) << "Failed to get input master volume.";

  return static_cast<double>(level);
}

// static
AudioParameters WASAPIAudioInputStream::GetInputStreamParameters(
    const std::string& device_id) {
  int sample_rate = 48000;
  ChannelLayout channel_layout = CHANNEL_LAYOUT_STEREO;

  base::win::ScopedCoMem<WAVEFORMATEX> audio_engine_mix_format;
  int effects = AudioParameters::NO_EFFECTS;
  if (SUCCEEDED(GetMixFormat(device_id, &audio_engine_mix_format, &effects))) {
    sample_rate = static_cast<int>(audio_engine_mix_format->nSamplesPerSec);
    channel_layout = audio_engine_mix_format->nChannels == 1 ?
        CHANNEL_LAYOUT_MONO : CHANNEL_LAYOUT_STEREO;
  }

  // Use 10ms frame size as default.
  int frames_per_buffer = sample_rate / 100;
  return AudioParameters(
      AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout, 0, sample_rate,
      16, frames_per_buffer, effects);
}

// static
HRESULT WASAPIAudioInputStream::GetMixFormat(const std::string& device_id,
                                             WAVEFORMATEX** device_format,
                                             int* effects) {
  DCHECK(effects);

  // It is assumed that this static method is called from a COM thread, i.e.,
  // CoInitializeEx() is not called here to avoid STA/MTA conflicts.
  ScopedComPtr<IMMDeviceEnumerator> enumerator;
  HRESULT hr = enumerator.CreateInstance(__uuidof(MMDeviceEnumerator), NULL,
                                         CLSCTX_INPROC_SERVER);
  if (FAILED(hr))
    return hr;

  ScopedComPtr<IMMDevice> endpoint_device;
  if (device_id == AudioManagerBase::kDefaultDeviceId) {
    // Retrieve the default capture audio endpoint.
    hr = enumerator->GetDefaultAudioEndpoint(eCapture, eConsole,
                                             endpoint_device.Receive());
  } else if (device_id == AudioManagerBase::kLoopbackInputDeviceId) {
    // Get the mix format of the default playback stream.
    hr = enumerator->GetDefaultAudioEndpoint(eRender, eConsole,
                                             endpoint_device.Receive());
  } else {
    // Retrieve a capture endpoint device that is specified by an endpoint
    // device-identification string.
    hr = enumerator->GetDevice(base::UTF8ToUTF16(device_id).c_str(),
                               endpoint_device.Receive());
  }

  if (FAILED(hr))
    return hr;

  *effects = IsDefaultCommunicationDevice(enumerator, endpoint_device) ?
      AudioParameters::DUCKING : AudioParameters::NO_EFFECTS;

  ScopedComPtr<IAudioClient> audio_client;
  hr = endpoint_device->Activate(__uuidof(IAudioClient),
                                 CLSCTX_INPROC_SERVER,
                                 NULL,
                                 audio_client.ReceiveVoid());
  return SUCCEEDED(hr) ? audio_client->GetMixFormat(device_format) : hr;
}

void WASAPIAudioInputStream::Run() {
  ScopedCOMInitializer com_init(ScopedCOMInitializer::kMTA);

  // Increase the thread priority.
  capture_thread_->SetThreadPriority(base::kThreadPriority_RealtimeAudio);

  // Enable MMCSS to ensure that this thread receives prioritized access to
  // CPU resources.
  DWORD task_index = 0;
  HANDLE mm_task = avrt::AvSetMmThreadCharacteristics(L"Pro Audio",
                                                      &task_index);
  bool mmcss_is_ok =
      (mm_task && avrt::AvSetMmThreadPriority(mm_task, AVRT_PRIORITY_CRITICAL));
  if (!mmcss_is_ok) {
    // Failed to enable MMCSS on this thread. It is not fatal but can lead
    // to reduced QoS at high load.
    DWORD err = GetLastError();
    LOG(WARNING) << "Failed to enable MMCSS (error code=" << err << ").";
  }

  // Allocate a buffer with a size that enables us to take care of cases like:
  // 1) The recorded buffer size is smaller, or does not match exactly with,
  //    the selected packet size used in each callback.
  // 2) The selected buffer size is larger than the recorded buffer size in
  //    each event.
  size_t buffer_frame_index = 0;
  size_t capture_buffer_size = std::max(
      2 * endpoint_buffer_size_frames_ * frame_size_,
      2 * packet_size_frames_ * frame_size_);
  scoped_ptr<uint8[]> capture_buffer(new uint8[capture_buffer_size]);

  LARGE_INTEGER now_count;
  bool recording = true;
  bool error = false;
  double volume = GetVolume();
  HANDLE wait_array[2] = {stop_capture_event_, audio_samples_ready_event_};

  while (recording && !error) {
    HRESULT hr = S_FALSE;

    // Wait for a close-down event or a new capture event.
    DWORD wait_result = WaitForMultipleObjects(2, wait_array, FALSE, INFINITE);
    switch (wait_result) {
      case WAIT_FAILED:
        error = true;
        break;
      case WAIT_OBJECT_0 + 0:
        // |stop_capture_event_| has been set.
        recording = false;
        break;
      case WAIT_OBJECT_0 + 1:
        {
          // |audio_samples_ready_event_| has been set.
          BYTE* data_ptr = NULL;
          UINT32 num_frames_to_read = 0;
          DWORD flags = 0;
          UINT64 device_position = 0;
          UINT64 first_audio_frame_timestamp = 0;

          // Retrieve the amount of data in the capture endpoint buffer,
          // replace it with silence if required, create callbacks for each
          // packet and store non-delivered data for the next event.
          hr = audio_capture_client_->GetBuffer(&data_ptr,
                                                &num_frames_to_read,
                                                &flags,
                                                &device_position,
                                                &first_audio_frame_timestamp);
          if (FAILED(hr)) {
            DLOG(ERROR) << "Failed to get data from the capture buffer";
            continue;
          }

          if (num_frames_to_read != 0) {
            size_t pos = buffer_frame_index * frame_size_;
            size_t num_bytes = num_frames_to_read * frame_size_;
            DCHECK_GE(capture_buffer_size, pos + num_bytes);

            if (flags & AUDCLNT_BUFFERFLAGS_SILENT) {
              // Clear out the local buffer since silence is reported.
              memset(&capture_buffer[pos], 0, num_bytes);
            } else {
              // Copy captured data from audio engine buffer to local buffer.
              memcpy(&capture_buffer[pos], data_ptr, num_bytes);
            }

            buffer_frame_index += num_frames_to_read;
          }

          hr = audio_capture_client_->ReleaseBuffer(num_frames_to_read);
          DLOG_IF(ERROR, FAILED(hr)) << "Failed to release capture buffer";

          // Derive a delay estimate for the captured audio packet.
          // The value contains two parts (A+B), where A is the delay of the
          // first audio frame in the packet and B is the extra delay
          // contained in any stored data. Unit is in audio frames.
          QueryPerformanceCounter(&now_count);
          double audio_delay_frames =
              ((perf_count_to_100ns_units_ * now_count.QuadPart -
                first_audio_frame_timestamp) / 10000.0) * ms_to_frame_count_ +
                buffer_frame_index - num_frames_to_read;

          // Get a cached AGC volume level which is updated once every second
          // on the audio manager thread. Note that, |volume| is also updated
          // each time SetVolume() is called through IPC by the render-side AGC.
          GetAgcVolume(&volume);

          // Deliver captured data to the registered consumer using a packet
          // size which was specified at construction.
          uint32 delay_frames = static_cast<uint32>(audio_delay_frames + 0.5);
          while (buffer_frame_index >= packet_size_frames_) {
            uint8* audio_data =
                reinterpret_cast<uint8*>(capture_buffer.get());

            // Deliver data packet, delay estimation and volume level to
            // the user.
            sink_->OnData(this,
                          audio_data,
                          packet_size_bytes_,
                          delay_frames * frame_size_,
                          volume);

            // Store parts of the recorded data which can't be delivered
            // using the current packet size. The stored section will be used
            // either in the next while-loop iteration or in the next
            // capture event.
            memmove(&capture_buffer[0],
                    &capture_buffer[packet_size_bytes_],
                    (buffer_frame_index - packet_size_frames_) * frame_size_);

            buffer_frame_index -= packet_size_frames_;
            delay_frames -= packet_size_frames_;
          }
        }
        break;
      default:
        error = true;
        break;
    }
  }

  if (recording && error) {
    // TODO(henrika): perhaps it worth improving the cleanup here by e.g.
    // stopping the audio client, joining the thread etc.?
    NOTREACHED() << "WASAPI capturing failed with error code "
                 << GetLastError();
  }

  // Disable MMCSS.
  if (mm_task && !avrt::AvRevertMmThreadCharacteristics(mm_task)) {
    PLOG(WARNING) << "Failed to disable MMCSS";
  }
}

void WASAPIAudioInputStream::HandleError(HRESULT err) {
  NOTREACHED() << "Error code: " << err;
  if (sink_)
    sink_->OnError(this);
}

HRESULT WASAPIAudioInputStream::SetCaptureDevice() {
  DCHECK(!endpoint_device_);

  ScopedComPtr<IMMDeviceEnumerator> enumerator;
  HRESULT hr = enumerator.CreateInstance(__uuidof(MMDeviceEnumerator),
                                         NULL, CLSCTX_INPROC_SERVER);
  if (FAILED(hr))
    return hr;

  // Retrieve the IMMDevice by using the specified role or the specified
  // unique endpoint device-identification string.

  if (effects_ & AudioParameters::DUCKING) {
    // Ducking has been requested and it is only supported for the default
    // communication device.  So, let's open up the communication device and
    // see if the ID of that device matches the requested ID.
    // We consider a kDefaultDeviceId as well as an explicit device id match,
    // to be valid matches.
    hr = enumerator->GetDefaultAudioEndpoint(eCapture, eCommunications,
                                             endpoint_device_.Receive());
    if (endpoint_device_ && device_id_ != AudioManagerBase::kDefaultDeviceId) {
      base::win::ScopedCoMem<WCHAR> communications_id;
      endpoint_device_->GetId(&communications_id);
      if (device_id_ !=
          base::WideToUTF8(static_cast<WCHAR*>(communications_id))) {
        DLOG(WARNING) << "Ducking has been requested for a non-default device."
                         "Not supported.";
        endpoint_device_.Release();  // Fall back on code below.
      }
    }
  }

  if (!endpoint_device_) {
    if (device_id_ == AudioManagerBase::kDefaultDeviceId) {
      // Retrieve the default capture audio endpoint for the specified role.
      // Note that, in Windows Vista, the MMDevice API supports device roles
      // but the system-supplied user interface programs do not.
      hr = enumerator->GetDefaultAudioEndpoint(eCapture, eConsole,
                                               endpoint_device_.Receive());
    } else if (device_id_ == AudioManagerBase::kLoopbackInputDeviceId) {
      // Capture the default playback stream.
      hr = enumerator->GetDefaultAudioEndpoint(eRender, eConsole,
                                               endpoint_device_.Receive());
    } else {
      hr = enumerator->GetDevice(base::UTF8ToUTF16(device_id_).c_str(),
                                 endpoint_device_.Receive());
    }
  }

  if (FAILED(hr))
    return hr;

  // Verify that the audio endpoint device is active, i.e., the audio
  // adapter that connects to the endpoint device is present and enabled.
  DWORD state = DEVICE_STATE_DISABLED;
  hr = endpoint_device_->GetState(&state);
  if (FAILED(hr))
    return hr;

  if (!(state & DEVICE_STATE_ACTIVE)) {
    DLOG(ERROR) << "Selected capture device is not active.";
    hr = E_ACCESSDENIED;
  }

  return hr;
}

HRESULT WASAPIAudioInputStream::ActivateCaptureDevice() {
  // Creates and activates an IAudioClient COM object given the selected
  // capture endpoint device.
  HRESULT hr = endpoint_device_->Activate(__uuidof(IAudioClient),
                                          CLSCTX_INPROC_SERVER,
                                          NULL,
                                          audio_client_.ReceiveVoid());
  return hr;
}

HRESULT WASAPIAudioInputStream::GetAudioEngineStreamFormat() {
  HRESULT hr = S_OK;
#ifndef NDEBUG
  // The GetMixFormat() method retrieves the stream format that the
  // audio engine uses for its internal processing of shared-mode streams.
  // The method always uses a WAVEFORMATEXTENSIBLE structure, instead
  // of a stand-alone WAVEFORMATEX structure, to specify the format.
  // An WAVEFORMATEXTENSIBLE structure can specify both the mapping of
  // channels to speakers and the number of bits of precision in each sample.
  base::win::ScopedCoMem<WAVEFORMATEXTENSIBLE> format_ex;
  hr = audio_client_->GetMixFormat(
      reinterpret_cast<WAVEFORMATEX**>(&format_ex));

  // See http://msdn.microsoft.com/en-us/windows/hardware/gg463006#EFH
  // for details on the WAVE file format.
  WAVEFORMATEX format = format_ex->Format;
  DVLOG(2) << "WAVEFORMATEX:";
  DVLOG(2) << "  wFormatTags    : 0x" << std::hex << format.wFormatTag;
  DVLOG(2) << "  nChannels      : " << format.nChannels;
  DVLOG(2) << "  nSamplesPerSec : " << format.nSamplesPerSec;
  DVLOG(2) << "  nAvgBytesPerSec: " << format.nAvgBytesPerSec;
  DVLOG(2) << "  nBlockAlign    : " << format.nBlockAlign;
  DVLOG(2) << "  wBitsPerSample : " << format.wBitsPerSample;
  DVLOG(2) << "  cbSize         : " << format.cbSize;

  DVLOG(2) << "WAVEFORMATEXTENSIBLE:";
  DVLOG(2) << " wValidBitsPerSample: " <<
      format_ex->Samples.wValidBitsPerSample;
  DVLOG(2) << " dwChannelMask      : 0x" << std::hex <<
      format_ex->dwChannelMask;
  if (format_ex->SubFormat == KSDATAFORMAT_SUBTYPE_PCM)
    DVLOG(2) << " SubFormat          : KSDATAFORMAT_SUBTYPE_PCM";
  else if (format_ex->SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT)
    DVLOG(2) << " SubFormat          : KSDATAFORMAT_SUBTYPE_IEEE_FLOAT";
  else if (format_ex->SubFormat == KSDATAFORMAT_SUBTYPE_WAVEFORMATEX)
    DVLOG(2) << " SubFormat          : KSDATAFORMAT_SUBTYPE_WAVEFORMATEX";
#endif
  return hr;
}

bool WASAPIAudioInputStream::DesiredFormatIsSupported() {
  // An application that uses WASAPI to manage shared-mode streams can rely
  // on the audio engine to perform only limited format conversions. The audio
  // engine can convert between a standard PCM sample size used by the
  // application and the floating-point samples that the engine uses for its
  // internal processing. However, the format for an application stream
  // typically must have the same number of channels and the same sample
  // rate as the stream format used by the device.
  // Many audio devices support both PCM and non-PCM stream formats. However,
  // the audio engine can mix only PCM streams.
  base::win::ScopedCoMem<WAVEFORMATEX> closest_match;
  HRESULT hr = audio_client_->IsFormatSupported(AUDCLNT_SHAREMODE_SHARED,
                                                &format_,
                                                &closest_match);
  DLOG_IF(ERROR, hr == S_FALSE) << "Format is not supported "
                                << "but a closest match exists.";
  return (hr == S_OK);
}

HRESULT WASAPIAudioInputStream::InitializeAudioEngine() {
  DWORD flags;
  // Use event-driven mode only fo regular input devices. For loopback the
  // EVENTCALLBACK flag is specified when intializing
  // |audio_render_client_for_loopback_|.
  if (device_id_ == AudioManagerBase::kLoopbackInputDeviceId) {
    flags = AUDCLNT_STREAMFLAGS_LOOPBACK | AUDCLNT_STREAMFLAGS_NOPERSIST;
  } else {
    flags =
      AUDCLNT_STREAMFLAGS_EVENTCALLBACK | AUDCLNT_STREAMFLAGS_NOPERSIST;
  }

  // Initialize the audio stream between the client and the device.
  // We connect indirectly through the audio engine by using shared mode.
  // Note that, |hnsBufferDuration| is set of 0, which ensures that the
  // buffer is never smaller than the minimum buffer size needed to ensure
  // that glitches do not occur between the periodic processing passes.
  // This setting should lead to lowest possible latency.
  HRESULT hr = audio_client_->Initialize(AUDCLNT_SHAREMODE_SHARED,
                                         flags,
                                         0,  // hnsBufferDuration
                                         0,
                                         &format_,
                                         NULL);
  if (FAILED(hr))
    return hr;

  // Retrieve the length of the endpoint buffer shared between the client
  // and the audio engine. The buffer length determines the maximum amount
  // of capture data that the audio engine can read from the endpoint buffer
  // during a single processing pass.
  // A typical value is 960 audio frames <=> 20ms @ 48kHz sample rate.
  hr = audio_client_->GetBufferSize(&endpoint_buffer_size_frames_);
  if (FAILED(hr))
    return hr;

  DVLOG(1) << "endpoint buffer size: " << endpoint_buffer_size_frames_
           << " [frames]";

#ifndef NDEBUG
  // The period between processing passes by the audio engine is fixed for a
  // particular audio endpoint device and represents the smallest processing
  // quantum for the audio engine. This period plus the stream latency between
  // the buffer and endpoint device represents the minimum possible latency
  // that an audio application can achieve.
  // TODO(henrika): possibly remove this section when all parts are ready.
  REFERENCE_TIME device_period_shared_mode = 0;
  REFERENCE_TIME device_period_exclusive_mode = 0;
  HRESULT hr_dbg = audio_client_->GetDevicePeriod(
      &device_period_shared_mode, &device_period_exclusive_mode);
  if (SUCCEEDED(hr_dbg)) {
    DVLOG(1) << "device period: "
             << static_cast<double>(device_period_shared_mode / 10000.0)
             << " [ms]";
  }

  REFERENCE_TIME latency = 0;
  hr_dbg = audio_client_->GetStreamLatency(&latency);
  if (SUCCEEDED(hr_dbg)) {
    DVLOG(1) << "stream latency: " << static_cast<double>(latency / 10000.0)
             << " [ms]";
  }
#endif

  // Set the event handle that the audio engine will signal each time a buffer
  // becomes ready to be processed by the client.
  //
  // In loopback case the capture device doesn't receive any events, so we
  // need to create a separate playback client to get notifications. According
  // to MSDN:
  //
  //   A pull-mode capture client does not receive any events when a stream is
  //   initialized with event-driven buffering and is loopback-enabled. To
  //   work around this, initialize a render stream in event-driven mode. Each
  //   time the client receives an event for the render stream, it must signal
  //   the capture client to run the capture thread that reads the next set of
  //   samples from the capture endpoint buffer.
  //
  // http://msdn.microsoft.com/en-us/library/windows/desktop/dd316551(v=vs.85).aspx
  if (device_id_ == AudioManagerBase::kLoopbackInputDeviceId) {
    hr = endpoint_device_->Activate(
        __uuidof(IAudioClient), CLSCTX_INPROC_SERVER, NULL,
        audio_render_client_for_loopback_.ReceiveVoid());
    if (FAILED(hr))
      return hr;

    hr = audio_render_client_for_loopback_->Initialize(
        AUDCLNT_SHAREMODE_SHARED,
        AUDCLNT_STREAMFLAGS_EVENTCALLBACK | AUDCLNT_STREAMFLAGS_NOPERSIST,
        0, 0, &format_, NULL);
    if (FAILED(hr))
      return hr;

    hr = audio_render_client_for_loopback_->SetEventHandle(
        audio_samples_ready_event_.Get());
  } else {
    hr = audio_client_->SetEventHandle(audio_samples_ready_event_.Get());
  }

  if (FAILED(hr))
    return hr;

  // Get access to the IAudioCaptureClient interface. This interface
  // enables us to read input data from the capture endpoint buffer.
  hr = audio_client_->GetService(__uuidof(IAudioCaptureClient),
                                 audio_capture_client_.ReceiveVoid());
  if (FAILED(hr))
    return hr;

  // Obtain a reference to the ISimpleAudioVolume interface which enables
  // us to control the master volume level of an audio session.
  hr = audio_client_->GetService(__uuidof(ISimpleAudioVolume),
                                 simple_audio_volume_.ReceiveVoid());
  return hr;
}

}  // namespace media