Moving device description utils from AudioManager[Base] into AudioDeviceDescription; to be shared b…

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"
 #include "base/trace_event/trace_event.h"
+#include "media/audio/audio_device_description.h"
 #include "media/audio/win/audio_manager_win.h"
 #include "media/audio/win/avrt_wrapper_win.h"
 #include "media/audio/win/core_audio_util_win.h"
 #include "media/base/audio_bus.h"
 
 using base::win::ScopedComPtr;
 using base::win::ScopedCOMInitializer;
 
 namespace media {
 
@@ skipping 432 matching lines @@
 
   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 (device_id_ == AudioManagerBase::kDefaultDeviceId) {
+  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_ == AudioManagerBase::kCommunicationsDeviceId) {
+  } else if (device_id_ == AudioDeviceDescription::kCommunicationsDeviceId) {
     hr = enumerator->GetDefaultAudioEndpoint(eCapture, eCommunications,
                                              endpoint_device_.Receive());
-  } else if (device_id_ == AudioManagerBase::kLoopbackInputDeviceId) {
+  } 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))
     return hr;
@@ skipping 80 matching lines @@
   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) {
+  if (device_id_ == AudioDeviceDescription::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,
+      AUDCLNT_SHAREMODE_SHARED, flags,
       0,  // hnsBufferDuration
-      0,
-      &format_,
-      device_id_ == AudioManagerBase::kCommunicationsDeviceId ?
-          &kCommunicationsSessionId : nullptr);
+      0, &format_, device_id_ == AudioDeviceDescription::kCommunicationsDeviceId
+                       ? &kCommunicationsSessionId
+                       : nullptr);
 
   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_);
@@ skipping 36 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_ == AudioManagerBase::kLoopbackInputDeviceId) {
+  if (device_id_ == AudioDeviceDescription::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);
@@ skipping 17 matching lines @@
     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