Add API to query for the audio buffer size.

This is essentially identical to the GetAudioHardwareSampleRate API, except
we get the "optimal" hardware buffer size to be used.

BUG=None
TEST=None


Review URL: http://codereview.chromium.org/7837030

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@101249 0039d316-1c4b-4281-b951-d872f2087c98

6 files changed, 38 insertions, 0 deletions

diff --git a/content/renderer/media/audio_device.cc b/content/renderer/media/audio_device.cc
index 7b2843f..68a8756 100644
--- a/content/renderer/media/audio_device.cc
+++ b/content/renderer/media/audio_device.cc
@@ -243,3 +243,13 @@ double AudioDevice::GetAudioHardwareSampleRate() {
   }
   return hardware_sample_rate;
 }
+
+size_t AudioDevice::GetAudioHardwareBufferSize() {
+  // Uses cached value if possible.
+  static size_t buffer_size = 0;
+
+  if (!buffer_size)
+    buffer_size = media::GetAudioHardwareBufferSize();
+
+  return buffer_size;
+}
diff --git a/content/renderer/media/audio_device.h b/content/renderer/media/audio_device.h
index eee2513..54f5dc7 100644
--- a/content/renderer/media/audio_device.h
+++ b/content/renderer/media/audio_device.h
@@ -101,6 +101,7 @@ class AudioDevice
   size_t buffer_size() const { return buffer_size_; }
 
   static double GetAudioHardwareSampleRate();
+  static size_t GetAudioHardwareBufferSize();
 
   // Methods called on IO thread ----------------------------------------------
   // AudioMessageFilter::Delegate methods, called by AudioMessageFilter.
diff --git a/content/renderer/renderer_webkitplatformsupport_impl.cc b/content/renderer/renderer_webkitplatformsupport_impl.cc
index b4eb45f..6d459e9 100644
--- a/content/renderer/renderer_webkitplatformsupport_impl.cc
+++ b/content/renderer/renderer_webkitplatformsupport_impl.cc
@@ -567,6 +567,10 @@ double RendererWebKitPlatformSupportImpl::audioHardwareSampleRate() {
   return AudioDevice::GetAudioHardwareSampleRate();
 }
 
+size_t RendererWebKitPlatformSupportImpl::audioHardwareBufferSize() {
+  return AudioDevice::GetAudioHardwareBufferSize();
+}
+
 WebAudioDevice*
 RendererWebKitPlatformSupportImpl::createAudioDevice(
     size_t buffer_size,
diff --git a/content/renderer/renderer_webkitplatformsupport_impl.h b/content/renderer/renderer_webkitplatformsupport_impl.h
index 83b3235..9517693 100644
--- a/content/renderer/renderer_webkitplatformsupport_impl.h
+++ b/content/renderer/renderer_webkitplatformsupport_impl.h
@@ -78,6 +78,7 @@ class RendererWebKitPlatformSupportImpl
   virtual WebKit::WebSharedWorkerRepository* sharedWorkerRepository();
   virtual WebKit::WebGraphicsContext3D* createGraphicsContext3D();
   virtual double audioHardwareSampleRate();
+  virtual size_t audioHardwareBufferSize();
   virtual WebKit::WebAudioDevice* createAudioDevice(
       size_t buffer_size, unsigned channels, double sample_rate,
       WebKit::WebAudioDevice::RenderCallback* callback);
diff --git a/media/audio/audio_util.cc b/media/audio/audio_util.cc
index 2b790d8..120e2ed 100644
--- a/media/audio/audio_util.cc
+++ b/media/audio/audio_util.cc
@@ -242,6 +242,23 @@ double GetAudioHardwareSampleRate()
 #endif
 }
 
+size_t GetAudioHardwareBufferSize() {
+  // The sizes here were determined by experimentation and are roughly
+  // 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.
+#if defined(OS_MACOSX)
+  return 128;
+#elif defined(OS_LINUX)
+  return 2048;
+#else
+  return 2048;
+#endif
+}
+
 // When transferring data in the shared memory, first word is size of data
 // in bytes. Actual data starts immediately after it.
 
diff --git a/media/audio/audio_util.h b/media/audio/audio_util.h
index e9452b2..6ea697b 100644
--- a/media/audio/audio_util.h
+++ b/media/audio/audio_util.h
@@ -82,6 +82,11 @@ MEDIA_EXPORT void InterleaveFloatToInt16(const std::vector<float*>& source,
 // Returns the default audio hardware sample-rate.
 MEDIA_EXPORT double GetAudioHardwareSampleRate();
 
+// Returns the optimal low-latency buffer size for the audio hardware.
+// This is the smallest buffer size the system can comfortably render
+// at without glitches.  The buffer size is in sample-frames.
+MEDIA_EXPORT size_t GetAudioHardwareBufferSize();
+
 // Functions that handle data buffer passed between processes in the shared
 // memory. Called on both IPC sides.