[media/audio] Convert VLOGs to DVLOGs

media/audio/win/audio_low_latency_output_win.cc

Index: media/audio/win/audio_low_latency_output_win.cc
diff --git a/media/audio/win/audio_low_latency_output_win.cc b/media/audio/win/audio_low_latency_output_win.cc
index 0988eab5db26b78ef94041a4d730c6b59d2f548e..c3cf9c0967fa53802677c8c22ac3028c5672caf8 100644
--- a/media/audio/win/audio_low_latency_output_win.cc
+++ b/media/audio/win/audio_low_latency_output_win.cc
@@ -73,9 +73,9 @@ WASAPIAudioOutputStream::WASAPIAudioOutputStream(AudioManagerWin* manager,
       audio_bus_(AudioBus::Create(params)) {
   DCHECK(manager_);
 
-  VLOG(1) << "WASAPIAudioOutputStream::WASAPIAudioOutputStream()";
-  VLOG_IF(1, share_mode_ == AUDCLNT_SHAREMODE_EXCLUSIVE)
-      << "Core Audio (WASAPI) EXCLUSIVE MODE is enabled.";
+  DVLOG(1) << "WASAPIAudioOutputStream::WASAPIAudioOutputStream()";
+  DVLOG_IF(1, share_mode_ == AUDCLNT_SHAREMODE_EXCLUSIVE)
+       << "Core Audio (WASAPI) EXCLUSIVE MODE is enabled.";
 
   // Load the Avrt DLL if not already loaded. Required to support MMCSS.
   bool avrt_init = avrt::Initialize();
@@ -104,10 +104,10 @@ WASAPIAudioOutputStream::WASAPIAudioOutputStream(AudioManagerWin* manager,
   // get from the audio endpoint device in each render event.
   packet_size_frames_ = params.frames_per_buffer();
   packet_size_bytes_ = params.GetBytesPerBuffer();
-  VLOG(1) << "Number of bytes per audio frame  : " << format->nBlockAlign;
-  VLOG(1) << "Number of audio frames per packet: " << packet_size_frames_;
-  VLOG(1) << "Number of bytes per packet       : " << packet_size_bytes_;
-  VLOG(1) << "Number of milliseconds per packet: "
+  DVLOG(1) << "Number of bytes per audio frame  : " << format->nBlockAlign;
+  DVLOG(1) << "Number of audio frames per packet: " << packet_size_frames_;
+  DVLOG(1) << "Number of bytes per packet       : " << packet_size_bytes_;
+  DVLOG(1) << "Number of milliseconds per packet: "
           << params.GetBufferDuration().InMillisecondsF();
 
   // All events are auto-reset events and non-signaled initially.
@@ -127,7 +127,7 @@ WASAPIAudioOutputStream::~WASAPIAudioOutputStream() {
 }
 
 bool WASAPIAudioOutputStream::Open() {
-  VLOG(1) << "WASAPIAudioOutputStream::Open()";
+  DVLOG(1) << "WASAPIAudioOutputStream::Open()";
   DCHECK_EQ(GetCurrentThreadId(), creating_thread_id_);
   if (opened_)
     return true;
@@ -226,7 +226,7 @@ bool WASAPIAudioOutputStream::Open() {
 }
 
 void WASAPIAudioOutputStream::Start(AudioSourceCallback* callback) {
-  VLOG(1) << "WASAPIAudioOutputStream::Start()";
+  DVLOG(1) << "WASAPIAudioOutputStream::Start()";
   DCHECK_EQ(GetCurrentThreadId(), creating_thread_id_);
   CHECK(callback);
   CHECK(opened_);
@@ -271,7 +271,7 @@ void WASAPIAudioOutputStream::Start(AudioSourceCallback* callback) {
 }
 
 void WASAPIAudioOutputStream::Stop() {
-  VLOG(1) << "WASAPIAudioOutputStream::Stop()";
+  DVLOG(1) << "WASAPIAudioOutputStream::Stop()";
   DCHECK_EQ(GetCurrentThreadId(), creating_thread_id_);
   if (!render_thread_)
     return;
@@ -306,7 +306,7 @@ void WASAPIAudioOutputStream::Stop() {
 }
 
 void WASAPIAudioOutputStream::Close() {
-  VLOG(1) << "WASAPIAudioOutputStream::Close()";
+  DVLOG(1) << "WASAPIAudioOutputStream::Close()";
   DCHECK_EQ(GetCurrentThreadId(), creating_thread_id_);
 
   // It is valid to call Close() before calling open or Start().
@@ -319,7 +319,7 @@ void WASAPIAudioOutputStream::Close() {
 }
 
 void WASAPIAudioOutputStream::SetVolume(double volume) {
-  VLOG(1) << "SetVolume(volume=" << volume << ")";
+  DVLOG(1) << "SetVolume(volume=" << volume << ")";
   float volume_float = static_cast<float>(volume);
   if (volume_float < 0.0f || volume_float > 1.0f) {
     return;
@@ -328,7 +328,7 @@ void WASAPIAudioOutputStream::SetVolume(double volume) {
 }
 
 void WASAPIAudioOutputStream::GetVolume(double* volume) {
-  VLOG(1) << "GetVolume()";
+  DVLOG(1) << "GetVolume()";
   *volume = static_cast<double>(volume_);
 }
 
@@ -538,7 +538,7 @@ HRESULT WASAPIAudioOutputStream::ExclusiveModeInitialization(
                     event_handle != INVALID_HANDLE_VALUE);
   if (use_event)
     stream_flags |= AUDCLNT_STREAMFLAGS_EVENTCALLBACK;
-  VLOG(2) << "stream_flags: 0x" << std::hex << stream_flags;
+  DVLOG(2) << "stream_flags: 0x" << std::hex << stream_flags;
 
   // Initialize the audio stream between the client and the device.
   // For an exclusive-mode stream that uses event-driven buffering, the
@@ -561,7 +561,7 @@ HRESULT WASAPIAudioOutputStream::ExclusiveModeInitialization(
 
       UINT32 aligned_buffer_size = 0;
       client->GetBufferSize(&aligned_buffer_size);
-      VLOG(1) << "Use aligned buffer size instead: " << aligned_buffer_size;
+      DVLOG(1) << "Use aligned buffer size instead: " << aligned_buffer_size;
 
       // Calculate new aligned periodicity. Each unit of reference time
       // is 100 nanoseconds.
@@ -573,9 +573,9 @@ HRESULT WASAPIAudioOutputStream::ExclusiveModeInitialization(
       // at this stage but we bail out with an error code instead and
       // combine it with a log message which informs about the suggested
       // aligned buffer size which should be used instead.
-      VLOG(1) << "aligned_buffer_duration: "
-              << static_cast<double>(aligned_buffer_duration / 10000.0)
-              << " [ms]";
+      DVLOG(1) << "aligned_buffer_duration: "
+               << static_cast<double>(aligned_buffer_duration / 10000.0)
+               << " [ms]";
     } else if (hr == AUDCLNT_E_INVALID_DEVICE_PERIOD) {
       // We will get this error if we try to use a smaller buffer size than
       // the minimum supported size (usually ~3ms on Windows 7).
@@ -587,7 +587,7 @@ HRESULT WASAPIAudioOutputStream::ExclusiveModeInitialization(
   if (use_event) {
     hr = client->SetEventHandle(event_handle);
     if (FAILED(hr)) {
-      VLOG(1) << "IAudioClient::SetEventHandle: " << std::hex << hr;
+      DVLOG(1) << "IAudioClient::SetEventHandle: " << std::hex << hr;
       return hr;
     }
   }
@@ -595,12 +595,12 @@ HRESULT WASAPIAudioOutputStream::ExclusiveModeInitialization(
   UINT32 buffer_size_in_frames = 0;
   hr = client->GetBufferSize(&buffer_size_in_frames);
   if (FAILED(hr)) {
-    VLOG(1) << "IAudioClient::GetBufferSize: " << std::hex << hr;
+    DVLOG(1) << "IAudioClient::GetBufferSize: " << std::hex << hr;
     return hr;
   }
 
   *endpoint_buffer_size = buffer_size_in_frames;
-  VLOG(2) << "endpoint buffer size: " << buffer_size_in_frames;
+  DVLOG(2) << "endpoint buffer size: " << buffer_size_in_frames;
   return hr;
 }