Make OS audio buffer size limits visible.

media/audio/cras/audio_manager_cras.cc

Index: media/audio/cras/audio_manager_cras.cc
diff --git a/media/audio/cras/audio_manager_cras.cc b/media/audio/cras/audio_manager_cras.cc
index 2e6d346e73bb1d47d5c343911e4bcb6417d079ac..50f3abdf9d0aca01a8a35600a71924de6c62e995 100644
--- a/media/audio/cras/audio_manager_cras.cc
+++ b/media/audio/cras/audio_manager_cras.cc
@@ -16,7 +16,6 @@
 #include "base/nix/xdg_util.h"
 #include "base/stl_util.h"
 #include "base/strings/string_number_conversions.h"
-#include "base/sys_info.h"
 #include "chromeos/audio/audio_device.h"
 #include "chromeos/audio/cras_audio_handler.h"
 #include "media/audio/audio_device_description.h"
@@ -40,10 +39,6 @@ const int kMaxOutputStreams = 50;
 // Default sample rate for input and output streams.
 const int kDefaultSampleRate = 48000;
 
-// Define bounds for the output buffer size.
-const int kMinimumOutputBufferSize = 512;
-const int kMaximumOutputBufferSize = 8192;
-
 // Default input buffer size.
 const int kDefaultInputBufferSize = 1024;
 
@@ -300,30 +295,19 @@ AudioInputStream* AudioManagerCras::MakeLowLatencyInputStream(
   return MakeInputStream(params, device_id);
 }
 
-int AudioManagerCras::GetMinimumOutputBufferSizePerBoard() {
-  // On faster boards we can use smaller buffer size for lower latency.
-  // On slower boards we should use larger buffer size to prevent underrun.
-  std::string board = base::SysInfo::GetLsbReleaseBoard();
-  if (board == "kevin")
-    return 768;
-  else if (board == "samus")
-    return 256;
-  return kMinimumOutputBufferSize;
-}
-
 AudioParameters AudioManagerCras::GetPreferredOutputStreamParameters(
     const std::string& output_device_id,
     const AudioParameters& input_params) {
   ChannelLayout channel_layout = CHANNEL_LAYOUT_STEREO;
   int sample_rate = kDefaultSampleRate;
-  int buffer_size = GetMinimumOutputBufferSizePerBoard();
+  int buffer_size = AudioLatency::GetDefaultAudioBufferSize(false, sample_rate);

[o1ka, 2017/05/30 16:04:33]

Sorry, now I lost the track of what is going on.
Why is the need to modify GetPreferredOutputStreamParameters? They are not
WebAudio-only.

For WebAudio GetPreferredOutputStreamParameters() result is returned to
AudioManagerBase::GetDefaultOutputStreamParameters() and ends up in
AudioOutputDevice::GetOutputDeviceInfo() on the renderer side. So HW buffer size
will be minimum buffer size both on ChromeOS and Mac if I'm not mistaken.


Why not to follow dalecurtis@'s recommendation (which I've just noticed)
described in the bug the CL is referencing
(https://bugs.chromium.org/p/chromium/issues/detail?id=708917)?

AudioLatency is renderer-side application-aware concept, and AudioManager is
browser-side platform audio system abstraction; none of them can depend on the
other.

   int bits_per_sample = 16;
   if (input_params.IsValid()) {
     sample_rate = input_params.sample_rate();
     bits_per_sample = input_params.bits_per_sample();
     channel_layout = input_params.channel_layout();
     buffer_size =
-        std::min(kMaximumOutputBufferSize,
+        std::min(AudioLatency::GetMaximumAudioBufferSize(sample_rate),
                  std::max(buffer_size, input_params.frames_per_buffer()));
   }