OLD | NEW |
---|---|
(Empty) | |
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 | |
Henrik Grunell
2016/06/29 12:20:54
"Categories" (spelling).
o1ka
2016/06/29 13:57:31
Done.
| |
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_ | |
OLD | NEW |