728d9037c0
This includes changes like:
* Attempt to break lines at better positions
* Use "override" in more places, don't use "virtual" with it
* Use {} where the body is more than one line
* Make declaration and definition arg names match
* Eliminate unused code
* EXPECT_EQ(expected, actual) (but use (actual, expected) for e.g. _GT)
* Correct #include order
* Use anonymous namespaces in preference to "static" for file-scoping
* Eliminate unnecessary casts
* Update reference code in comments of ARM assembly sources to match actual current C code
* Fix indenting to be more style-guide compliant
* Use arraysize() in more places
* Use bool instead of int for "boolean" values (0/1)
* Shorten and simplify code
* Spaces around operators
* 80 column limit
* Use const more consistently
* Space goes after '*' in type name, not before
* Remove unnecessary return values
* Use "(var == const)", not "(const == var)"
* Spelling
* Prefer true, typed constants to "enum hack" constants
* Avoid "virtual" on non-overridden functions
* ASSERT(x == y) -> ASSERT_EQ(y, x)
BUG=none
R=andrew@webrtc.org, asapersson@webrtc.org, henrika@webrtc.org, juberti@webrtc.org, kjellander@webrtc.org, kwiberg@webrtc.org
Review URL: https://codereview.webrtc.org/1172163004
Cr-Commit-Position: refs/heads/master@{#9420}
36 lines
1.4 KiB
C++
36 lines
1.4 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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enum {
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kDefaultSampleRate = 44100,
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kNumChannels = 1,
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// Number of bytes per audio frame.
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// Example: 16-bit PCM in mono => 1*(16/8)=2 [bytes/frame]
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kBytesPerFrame = kNumChannels * (16 / 8),
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// Delay estimates for the two different supported modes. These values
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// are based on real-time round-trip delay estimates on a large set of
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// devices and they are lower bounds since the filter length is 128 ms,
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// so the AEC works for delays in the range [50, ~170] ms and [150, ~270] ms.
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// Note that, in most cases, the lowest delay estimate will not be utilized
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// since devices that support low-latency output audio often supports
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// HW AEC as well.
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kLowLatencyModeDelayEstimateInMilliseconds = 50,
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kHighLatencyModeDelayEstimateInMilliseconds = 150,
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_COMMON_H_
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