779017d989
BUG=webrtc:6718 Review-Url: https://codereview.webrtc.org/2499613002 Cr-Commit-Position: refs/heads/master@{#15104}
29 lines
1.2 KiB
C++
29 lines
1.2 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_COMMON_H_
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#define WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_COMMON_H_
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namespace webrtc {
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const int kDefaultSampleRate = 44100;
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// Delay estimates for the two different supported modes. These values are based
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// on real-time round-trip delay estimates on a large set of devices and they
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// are lower bounds since the filter length is 128 ms, so the AEC works for
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// delays in the range [50, ~170] ms and [150, ~270] ms. Note that, in most
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// cases, the lowest delay estimate will not be utilized since devices that
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// support low-latency output audio often supports HW AEC as well.
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const int kLowLatencyModeDelayEstimateInMilliseconds = 50;
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const int kHighLatencyModeDelayEstimateInMilliseconds = 150;
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_COMMON_H_
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