Low-latency AudioOutputStream implementation based on WASAPI for Windows.

media/audio/audio_util.cc

Index: media/audio/audio_util.cc
===================================================================
--- media/audio/audio_util.cc	(revision 108984)
+++ media/audio/audio_util.cc	(working copy)
@@ -24,6 +24,7 @@
 #endif
 #if defined(OS_WIN)
 #include "media/audio/win/audio_low_latency_input_win.h"
+#include "media/audio/win/audio_low_latency_output_win.h"
 #endif
 
 using base::subtle::Atomic32;
@@ -241,8 +242,19 @@
 #if defined(OS_MACOSX)
     // Hardware sample-rate on the Mac can be configured, so we must query.
     return AUAudioOutputStream::HardwareSampleRate();
+#elif defined(OS_WIN)
+  if (base::win::GetVersion() <= base::win::VERSION_XP) {
+    // Fall back to Windows Wave implementation on Windows XP or lower
+    // and use 48kHz as default input sample rate.
+    return 48000.0;
+  } else {

[scherkus (not reviewing), 2011/11/08 23:52:21]

nit: no need for else statement if you return

[henrika (OOO until Aug 14), 2011/11/09 13:59:22]

Done.

+    // Hardware sample-rate on Windows can be configured, so we must query.
+    // TODO(henrika): improve possibility to specify audio endpoint.
+    // Use the default device (same as for Wave) for now to be compatible.
+    return WASAPIAudioOutputStream::HardwareSampleRate(eConsole);
+  }
 #else
-    // Hardware for Windows and Linux is nearly always 48KHz.
+    // Hardware for Linux is nearly always 48KHz.
     // TODO(crogers) : return correct value in rare non-48KHz cases.
     return 48000.0;
 #endif
@@ -275,13 +287,21 @@
   // the lowest value (for low latency) that still allowed glitch-free
   // audio under high loads.
   //
-  // For Mac OS X the chromium audio backend uses a low-latency
-  // CoreAudio API, so a low buffer size is possible.  For other OSes,
-  // further tuning may be needed.
+  // For Mac OS X and Windows the chromium audio backend uses a low-latency
+  // Core Audio API, so a low buffer size is possible. For Linux, further
+  // tuning may be needed.
+  // TODO(crogers): figure out the best possible buffer sizes for WebAudio.
 #if defined(OS_MACOSX)
   return 128;
-#elif defined(OS_LINUX)
-  return 2048;
+#elif defined(OS_WIN)
+  int mixing_sample_rate =
+      static_cast<int>(WASAPIAudioOutputStream::HardwareSampleRate(eConsole));
+  if (mixing_sample_rate == 48000 || mixing_sample_rate == 44100)
+    return 1024;
+  else if (mixing_sample_rate == 96000)

[scherkus (not reviewing), 2011/11/08 23:52:21]

ditto

[henrika (OOO until Aug 14), 2011/11/09 13:59:22]

Done.

+    return 2048;
+  else
+    return 2048;
 #else
   return 2048;
 #endif