| /* |
| * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #ifndef WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_COMMON_H_ |
| #define WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_COMMON_H_ |
| |
| namespace webrtc { |
| |
| const int kDefaultSampleRate = 44100; |
| // Delay estimates for the two different supported modes. These values are based |
| // on real-time round-trip delay estimates on a large set of devices and they |
| // are lower bounds since the filter length is 128 ms, so the AEC works for |
| // delays in the range [50, ~170] ms and [150, ~270] ms. Note that, in most |
| // cases, the lowest delay estimate will not be utilized since devices that |
| // support low-latency output audio often supports HW AEC as well. |
| const int kLowLatencyModeDelayEstimateInMilliseconds = 50; |
| const int kHighLatencyModeDelayEstimateInMilliseconds = 150; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_COMMON_H_ |