Make OS audio buffer size limits visible.

media/audio/mac/audio_manager_mac.cc

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "media/audio/mac/audio_manager_mac.h"
 
 #include <algorithm>
 #include <limits>
 #include <vector>
 
@@ skipping 23 matching lines @@
 // 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;
 
+static int GetDefaultBufferSize(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;
+  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;
+}
+
 // Helper method to construct AudioObjectPropertyAddress structure given
 // property selector and scope. The property element is always set to
 // kAudioObjectPropertyElementMaster.
 static AudioObjectPropertyAddress GetAudioObjectPropertyAddress(
     AudioObjectPropertySelector selector,
     bool is_input) {
   AudioObjectPropertyScope scope = is_input ? kAudioObjectPropertyScopeInput
                                             : kAudioObjectPropertyScopeOutput;
   AudioObjectPropertyAddress property_address = {
       selector, scope, kAudioObjectPropertyElementMaster};
@@ skipping 779 matching lines @@
   const bool has_valid_input_params = input_params.IsValid();
   const int hardware_sample_rate = HardwareSampleRateForDevice(device);
 
   // Allow pass through buffer sizes.  If concurrent input and output streams
   // exist, they will use the smallest buffer size amongst them.  As such, each
   // 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));
+                 std::max(input_params.frames_per_buffer(),
+                          GetMinimumAudioBufferSize(hardware_sample_rate)));
   }
 
   int hardware_channels;
   if (!GetDeviceChannels(device, AUElement::OUTPUT, &hardware_channels))
     hardware_channels = 2;
 
   // Use the input channel count and channel layout if possible.  Let OSX take
   // care of remapping the channels; this lets user specified channel layouts
   // work correctly.
   int output_channels = input_params.channels();
@@ skipping 26 matching lines @@
       current_output_device_ == new_output_device) {
     return;
   }
 
   current_sample_rate_ = new_sample_rate;
   current_output_device_ = new_output_device;
   NotifyAllOutputDeviceChangeListeners();
 }
 
 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 GetDefaultBufferSize(is_input, sample_rate);
 }
 
 bool AudioManagerMac::IsSuspending() const {
   DCHECK(GetTaskRunner()->BelongsToCurrentThread());
   return power_observer_->IsSuspending();
 }
 
 bool AudioManagerMac::ShouldDeferStreamStart() const {
   DCHECK(GetTaskRunner()->BelongsToCurrentThread());
   return power_observer_->ShouldDeferStreamStart();
@@ skipping 259 matching lines @@
   AudioManagerBase::ReleaseInputStream(stream);
 }
 
 std::unique_ptr<AudioManager> CreateAudioManager(
     std::unique_ptr<AudioThread> audio_thread,
     AudioLogFactory* audio_log_factory) {
   return base::MakeUnique<AudioManagerMac>(std::move(audio_thread),
                                            audio_log_factory);
 }
 
+int AudioManager::GetMinimumAudioBufferSize(int sample_rate) {
+  return GetDefaultBufferSize(false, sample_rate);
+}
+
+int AudioManager::GetMaximumAudioBufferSize(int sample_rate) {
+  return kMaximumInputOutputBufferSize;
+}
+
 }  // namespace media