Support AudioContextOptions latencyHint as double.

media/base/audio_latency.h

 // Copyright 2016 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.
 
 #ifndef MEDIA_BASE_AUDIO_LATENCY_H_
 #define MEDIA_BASE_AUDIO_LATENCY_H_
 
 #include "media/base/media_export.h"
 
+namespace base {
+class TimeDelta;
+}
+
 namespace media {
 
 class MEDIA_EXPORT AudioLatency {
  public:
   // Categories of expected latencies for input/output audio. Do not change
   // existing values, they are used for UMA histogram reporting.
   enum LatencyType {
     // Specific latency in milliseconds.
     LATENCY_EXACT_MS = 0,
     // Lowest possible latency which does not cause glitches.
     LATENCY_INTERACTIVE = 1,
     // Latency optimized for real time communication.
     LATENCY_RTC = 2,
     // Latency optimized for continuous playback and power saving.
     LATENCY_PLAYBACK = 3,
     // For validation only.
     LATENCY_LAST = LATENCY_PLAYBACK,
     LATENCY_COUNT = LATENCY_LAST + 1
   };
 
   // |preferred_buffer_size| should be set to 0 if a client has no preference.
   static int GetHighLatencyBufferSize(int sample_rate,
                                       int preferred_buffer_size);
 
   // |hardware_buffer_size| should be set to 0 if unknown/invalid/not preferred.
   static int GetRtcBufferSize(int sample_rate, int hardware_buffer_size);
 
   static int GetInteractiveBufferSize(int hardware_buffer_size);
+
+  static int GetExactBufferSize(base::TimeDelta duration,
+                                int sample_rate,
+                                int hardware_buffer_size);
 };
 
 }  // namespace media
 
 #endif  // MEDIA_BASE_AUDIO_LATENCY_H_