Mixing audio with different latency requirements

media/base/audio_latency.h

Index: media/base/audio_latency.h
diff --git a/media/base/audio_latency.h b/media/base/audio_latency.h
new file mode 100644
index 0000000000000000000000000000000000000000..c6c44f9cebd70176766d097adf47db42af19feb2
--- /dev/null
+++ b/media/base/audio_latency.h
@@ -0,0 +1,42 @@
+// 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 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);
+};
+
+}  // namespace media
+
+#endif  // MEDIA_BASE_AUDIO_LATENCY_H_