| // 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 COBALT_MEDIA_BASE_AUDIO_LATENCY_H_ |
| #define COBALT_MEDIA_BASE_AUDIO_LATENCY_H_ |
| |
| #include "cobalt/media/base/media_export.h" |
| |
| namespace cobalt { |
| 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 |
| } // namespace cobalt |
| |
| #endif // COBALT_MEDIA_BASE_AUDIO_LATENCY_H_ |