MuteSource Audio During Full Screen Cast

media/audio/win/audio_low_latency_input_win.cc

 // 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 <memory>
 
 #include "base/logging.h"
 #include "base/strings/utf_string_conversions.h"
@@ skipping 17 matching lines @@
       opened_(false),
       started_(false),
       frame_size_(0),
       packet_size_frames_(0),
       packet_size_bytes_(0),
       endpoint_buffer_size_frames_(0),
       device_id_(device_id),
       perf_count_to_100ns_units_(0.0),
       ms_to_frame_count_(0.0),
       sink_(NULL),
-      audio_bus_(media::AudioBus::Create(params)) {
+      audio_bus_(media::AudioBus::Create(params)),
+      mute_done_(false) {
   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();
@@ skipping 49 matching lines @@
   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.
+// 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
@@ skipping 37 matching lines @@
 
   started_ = SUCCEEDED(hr);
 }
 
 void WASAPIAudioInputStream::Stop() {
   DCHECK(CalledOnValidThread());
   DVLOG(1) << "WASAPIAudioInputStream::Stop()";
   if (!started_)
     return;
 
+  // We have muted system audio for capturing, so we need to unmute it when
+  // capturing stops.
+  if (device_id_ == AudioDeviceDescription::kLoopbackWithMuteDeviceId &&
+      mute_done_) {
+    DCHECK(system_audio_volume_);
+    if (system_audio_volume_) {
+      system_audio_volume_->SetMute(false, NULL);
+      mute_done_ = false;
+    }
+  }
+
   // 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();
@@ skipping 88 matching lines @@
 
   return is_muted != FALSE;
 }
 
 void WASAPIAudioInputStream::Run() {
   ScopedCOMInitializer com_init(ScopedCOMInitializer::kMTA);
 
   // 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);
+  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_);
+  size_t capture_buffer_size =
+      std::max(2 * endpoint_buffer_size_frames_ * frame_size_,
+               2 * packet_size_frames_ * frame_size_);
   std::unique_ptr<uint8_t[]> capture_buffer(new uint8_t[capture_buffer_size]);
 
   LARGE_INTEGER now_count = {};
   bool recording = true;
   bool error = false;
   double volume = GetVolume();
-  HANDLE wait_array[2] =
-      { stop_capture_event_.Get(), audio_samples_ready_event_.Get() };
+  HANDLE wait_array[2] = {stop_capture_event_.Get(),
+                          audio_samples_ready_event_.Get()};
 
   base::win::ScopedComPtr<IAudioClock> audio_clock;
   audio_client_->GetService(__uuidof(IAudioClock), audio_clock.ReceiveVoid());
 
   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:
-        {
-          TRACE_EVENT0("audio", "WASAPIAudioInputStream::Run_0");
-          // |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;
+      case WAIT_OBJECT_0 + 1: {
+        TRACE_EVENT0("audio", "WASAPIAudioInputStream::Run_0");
+        // |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 (audio_clock) {
+          // The reported timestamp from GetBuffer is not as reliable as the
+          // clock from the client.  We've seen timestamps reported for
+          // USB audio devices, be off by several days.  Furthermore we've
+          // seen them jump back in time every 2 seconds or so.
+          audio_clock->GetPosition(&device_position,
+                                   &first_audio_frame_timestamp);
+        }
+
+        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);
           }
 
-          if (audio_clock) {
-            // The reported timestamp from GetBuffer is not as reliable as the
-            // clock from the client.  We've seen timestamps reported for
-            // USB audio devices, be off by several days.  Furthermore we've
-            // seen them jump back in time every 2 seconds or so.
-            audio_clock->GetPosition(
-                &device_position, &first_audio_frame_timestamp);
-          }
-
-
-          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);
-          // first_audio_frame_timestamp will be 0 if we didn't get a timestamp.
-          double audio_delay_frames = first_audio_frame_timestamp == 0 ?
-              num_frames_to_read :
-              ((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_t delay_frames =
-              static_cast<uint32_t>(audio_delay_frames + 0.5);
-          while (buffer_frame_index >= packet_size_frames_) {
-            // Copy data to audio bus to match the OnData interface.
-            uint8_t* audio_data =
-                reinterpret_cast<uint8_t*>(capture_buffer.get());
-            audio_bus_->FromInterleaved(
-                audio_data, audio_bus_->frames(), format_.wBitsPerSample / 8);
-
-            // Deliver data packet, delay estimation and volume level to
-            // the user.
-            sink_->OnData(
-                this, audio_bus_.get(), 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.
-            // TODO(tommi): If this data will be used in the next capture
-            // event, we will report incorrect delay estimates because
-            // we'll use the one for the captured data that time around
-            // (i.e. in the future).
-            memmove(&capture_buffer[0],
-                    &capture_buffer[packet_size_bytes_],
-                    (buffer_frame_index - packet_size_frames_) * frame_size_);
-
-            DCHECK_GE(buffer_frame_index, packet_size_frames_);
-            buffer_frame_index -= packet_size_frames_;
-            if (delay_frames > packet_size_frames_) {
-              delay_frames -= packet_size_frames_;
-            } else {
-              delay_frames = 0;
-            }
+          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);
+        // first_audio_frame_timestamp will be 0 if we didn't get a timestamp.
+        double audio_delay_frames =
+            first_audio_frame_timestamp == 0
+                ? num_frames_to_read
+                : ((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_t delay_frames = static_cast<uint32_t>(audio_delay_frames + 0.5);
+        while (buffer_frame_index >= packet_size_frames_) {
+          // Copy data to audio bus to match the OnData interface.
+          uint8_t* audio_data =
+              reinterpret_cast<uint8_t*>(capture_buffer.get());
+          audio_bus_->FromInterleaved(audio_data, audio_bus_->frames(),
+                                      format_.wBitsPerSample / 8);
+
+          // Deliver data packet, delay estimation and volume level to
+          // the user.
+          sink_->OnData(this, audio_bus_.get(), 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.
+          // TODO(tommi): If this data will be used in the next capture
+          // event, we will report incorrect delay estimates because
+          // we'll use the one for the captured data that time around
+          // (i.e. in the future).
+          memmove(&capture_buffer[0], &capture_buffer[packet_size_bytes_],
+                  (buffer_frame_index - packet_size_frames_) * frame_size_);
+
+          DCHECK_GE(buffer_frame_index, packet_size_frames_);
+          buffer_frame_index -= packet_size_frames_;
+          if (delay_frames > packet_size_frames_) {
+            delay_frames -= packet_size_frames_;
+          } else {
+            delay_frames = 0;
           }
         }
-        break;
+      } 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_.get());
 
   ScopedComPtr<IMMDeviceEnumerator> enumerator;
-  HRESULT hr = enumerator.CreateInstance(__uuidof(MMDeviceEnumerator),
-                                         NULL, CLSCTX_INPROC_SERVER);
+  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 (device_id_ == AudioDeviceDescription::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_ == AudioDeviceDescription::kCommunicationsDeviceId) {
     hr = enumerator->GetDefaultAudioEndpoint(eCapture, eCommunications,
                                              endpoint_device_.Receive());
+  } else if (device_id_ == AudioDeviceDescription::kLoopbackWithMuteDeviceId) {
+    // Capture the default playback stream.
+    hr = enumerator->GetDefaultAudioEndpoint(eRender, eConsole,
+                                             endpoint_device_.Receive());
+
+    endpoint_device_->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL, NULL,
+                               system_audio_volume_.ReceiveVoid());
+    if (system_audio_volume_) {
+      BOOL muted = false;
+      system_audio_volume_->GetMute(&muted);
+
+      // If the system audio is muted at the time of capturing, then no need to
+      // mute it again, and later we do not unmute system audio when stopping
+      // capturing.
+      if (!muted) {
+        system_audio_volume_->SetMute(true, NULL);
+        mute_done_ = true;
+      }
+    }
   } else if (device_id_ == AudioDeviceDescription::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))
@@ skipping 10 matching lines @@
     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());
+  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));
+  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;
+  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);
+                                                &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_ == AudioDeviceDescription::kLoopbackInputDeviceId) {
+  if (device_id_ == AudioDeviceDescription::kLoopbackInputDeviceId ||
+      device_id_ == AudioDeviceDescription::kLoopbackWithMuteDeviceId) {
     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.
@@ skipping 53 matching lines @@
   // 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_ == AudioDeviceDescription::kLoopbackInputDeviceId) {
+  if (device_id_ == AudioDeviceDescription::kLoopbackInputDeviceId ||
+      device_id_ == AudioDeviceDescription::kLoopbackWithMuteDeviceId) {
     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);
+        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