Make OS audio buffer size limits visible.

media/audio/mac/audio_manager_mac.cc

Index: media/audio/mac/audio_manager_mac.cc
diff --git a/media/audio/mac/audio_manager_mac.cc b/media/audio/mac/audio_manager_mac.cc
index 09b6433e631d819fc6bc87ab72f4c55b5cc672d1..c837442d3ff809ff89af370e5069fb078226d9ae 100644
--- a/media/audio/mac/audio_manager_mac.cc
+++ b/media/audio/mac/audio_manager_mac.cc
@@ -34,10 +34,6 @@ namespace media {
 // Maximum number of output streams that can be open simultaneously.
 static const int kMaxOutputStreams = 50;
 
-// Define bounds for for low-latency input and output streams.
-static const int kMinimumInputOutputBufferSize = 128;
-static const int kMaximumInputOutputBufferSize = 4096;
-
 // Default sample-rate on most Apple hardware.
 static const int kFallbackSampleRate = 44100;
 
@@ -838,9 +834,11 @@ AudioParameters AudioManagerMac::GetPreferredOutputStreamParameters(
   // stream must be able to FIFO requests appropriately when this happens.
   int buffer_size = ChooseBufferSize(false, hardware_sample_rate);
   if (has_valid_input_params) {
-    buffer_size =
-        std::min(kMaximumInputOutputBufferSize,
-                 std::max(input_params.frames_per_buffer(), buffer_size));
+    buffer_size = std::min(
+        AudioLatency::GetMaximumAudioBufferSize(hardware_sample_rate),
+        std::max(
+            input_params.frames_per_buffer(),
+            AudioLatency::GetMinimumAudioBufferSize(hardware_sample_rate)));
   }
 
   int hardware_channels;
@@ -887,29 +885,10 @@ void AudioManagerMac::HandleDeviceChanges() {
 }
 
 int AudioManagerMac::ChooseBufferSize(bool is_input, int sample_rate) {
-  // kMinimumInputOutputBufferSize is too small for the output side because
-  // CoreAudio can get into under-run if the renderer fails delivering data
-  // to the browser within the allowed time by the OS. The workaround is to
-  // use 256 samples as the default output buffer size for sample rates
-  // smaller than 96KHz.
-  // TODO(xians): Remove this workaround after WebAudio supports user defined
-  // buffer size.  See https://github.com/WebAudio/web-audio-api/issues/348
-  // for details.
-  int buffer_size = is_input ?
-      kMinimumInputOutputBufferSize : 2 * kMinimumInputOutputBufferSize;
   const int user_buffer_size = GetUserBufferSize();
-  if (user_buffer_size) {
-    buffer_size = user_buffer_size;
-  } else if (sample_rate > 48000) {
-    // The default buffer size is too small for higher sample rates and may lead
-    // to glitching.  Adjust upwards by multiples of the default size.
-    if (sample_rate <= 96000)
-      buffer_size = 2 * kMinimumInputOutputBufferSize;
-    else if (sample_rate <= 192000)
-      buffer_size = 4 * kMinimumInputOutputBufferSize;
-  }
-
-  return buffer_size;
+  if (user_buffer_size)
+    return user_buffer_size;
+  return AudioLatency::GetDefaultAudioBufferSize(is_input, sample_rate);
 }
 
 bool AudioManagerMac::IsSuspending() const {