Modified Windows buffer size to avoid audio glitches.

remoting/host/audio_capturer_win.cc

Index: remoting/host/audio_capturer_win.cc
diff --git a/remoting/host/audio_capturer_win.cc b/remoting/host/audio_capturer_win.cc
index c648a0ab0727c5c776e51a9bf66db3c4100bc8e4..50b6ee06214f55389a8acc744cd929adc5bef8a6 100644
--- a/remoting/host/audio_capturer_win.cc
+++ b/remoting/host/audio_capturer_win.cc
@@ -9,6 +9,7 @@
 #include <mmreg.h>
 #include <mmsystem.h>
 
+#include <algorithm>
 #include <stdlib.h>
 
 #include "base/basictypes.h"
@@ -34,6 +35,12 @@ const int kHnsToMs = 10000;
 // silence. A value of 2 was chosen, because Windows can give samples of 1 and
 // -1, even when no audio is playing.
 const int kSilenceThreshold = 2;
+
+// Lower bound for timer intervals, in milliseconds.
+const int kMinTimerInteveral = 30;

[Wez, 2012/08/16 18:49:36]

typo: kMinTimerInterval

[kxing, 2012/08/16 20:17:21]

Done.

+
+// Upper bound for the timer precision error, in milliseconds.

[Wez, 2012/08/16 18:49:36]

nit: Clarify where this value comes from, even if it's just a an estimate based
on experimentation.

[kxing, 2012/08/16 20:17:21]

Done.

+const int kMaxTimerLag = 30;

[Wez, 2012/08/16 18:49:36]

nit: Consider calling this kMaxExpectedTimerLag; it's possible for the timer to
lag by longer but we don't expect that in normal usage!

[kxing, 2012/08/16 20:17:21]

Done.

 }  // namespace
 
 namespace remoting {
@@ -129,7 +136,7 @@ bool AudioCapturerWin::Start(const PacketCapturedCallback& callback) {
     return false;
   }
   audio_device_period_ = base::TimeDelta::FromMilliseconds(
-      device_period / kChannels / kHnsToMs);
+      std::max(static_cast<int>(device_period / kHnsToMs), kMinTimerInteveral));

[Wez, 2012/08/16 18:49:36]

nit: If |device_period| is not an integer number of milliseconds then this will
round-down; is that a problem?

[kxing, 2012/08/16 20:17:21]

Done.

 
   // Get the wave format.
   hr = audio_client_->GetMixFormat(&wave_format_ex_);
@@ -193,12 +200,13 @@ bool AudioCapturerWin::Start(const PacketCapturedCallback& callback) {
   }
 
   // Initialize the IAudioClient.
-  hr = audio_client_->Initialize(AUDCLNT_SHAREMODE_SHARED,
-                                 AUDCLNT_STREAMFLAGS_LOOPBACK,
-                                 0,
-                                 0,
-                                 wave_format_ex_,
-                                 NULL);
+  hr = audio_client_->Initialize(
+      AUDCLNT_SHAREMODE_SHARED,
+      AUDCLNT_STREAMFLAGS_LOOPBACK,
+      (kMaxTimerLag + audio_device_period_.InMilliseconds()) * kHnsToMs,

[Wez, 2012/08/16 18:49:36]

You're multiplying a value in milliseconds by the 100ns->ms conversion ratio;
surely you want to multiply by the ms->100ns ratio?

[kxing, 2012/08/16 20:17:21]

Done.

+      0,
+      wave_format_ex_,
+      NULL);
   if (FAILED(hr)) {
     LOG(ERROR) << "Failed to initialize IAudioClient. Error " << hr;
     return false;