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| 1 // Copyright 2016 The Chromium Authors. All rights reserved. |
| 2 // Use of this source code is governed by a BSD-style license that can be |
| 3 // found in the LICENSE file. |
| 4 |
| 5 #ifndef MEDIA_BASE_AUDIO_LATENCY_H_ |
| 6 #define MEDIA_BASE_AUDIO_LATENCY_H_ |
| 7 |
| 8 #include "media/base/media_export.h" |
| 9 |
| 10 namespace media { |
| 11 |
| 12 class MEDIA_EXPORT AudioLatency { |
| 13 public: |
| 14 // Categoties of expected latencies for input/output audio. Do not change |
| 15 // existing values, they are used for UMA histogram reporting. |
| 16 enum LatencyType { |
| 17 // Specific latency in milliseconds. |
| 18 LATENCY_EXACT_MS = 0, |
| 19 // Lowest possible latency which does not cause glitches. |
| 20 LATENCY_INTERACTIVE = 1, |
| 21 // Latency optimized for real time communication. |
| 22 LATENCY_RTC = 2, |
| 23 // Latency optimized for continuous playback and power saving. |
| 24 LATENCY_PLAYBACK = 3, |
| 25 // For validation only. |
| 26 LATENCY_LAST = LATENCY_PLAYBACK, |
| 27 LATENCY_COUNT = LATENCY_LAST + 1 |
| 28 }; |
| 29 |
| 30 // |preferred_buffer_size| should be set to 0 if a client has no preference. |
| 31 static int GetHighLatencyBufferSize(int sample_rate, |
| 32 int preferred_buffer_size); |
| 33 |
| 34 // |hardware_buffer_size| should be set to 0 if unknown/invalid/not preferred. |
| 35 static int GetRtcBufferSize(int sample_rate, int hardware_buffer_size); |
| 36 |
| 37 static int GetInteractiveBufferSize(int hardware_buffer_size); |
| 38 }; |
| 39 |
| 40 } // namespace media |
| 41 |
| 42 #endif // MEDIA_BASE_AUDIO_LATENCY_H_ |
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