use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
33 lines
1.4 KiB
C++
33 lines
1.4 KiB
C++
/*
|
|
* 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;
|
|
const int kNumChannels = 1;
|
|
// Number of bytes per audio frame.
|
|
// Example: 16-bit PCM in mono => 1*(16/8)=2 [bytes/frame]
|
|
const size_t kBytesPerFrame = kNumChannels * (16 / 8);
|
|
// 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_
|