Files
platform-external-webrtc/webrtc/modules/audio_device/android/audio_common.h
henrika b26198972c Adding support for OpenSL ES output in native WebRTC
BUG=4573,2982,2175,3590
TEST=modules_unittests --gtest_filter=AudioDevice*, AppRTCDemo and WebRTCDemo

Summary:

- Removes dependency of the 'enable_android_opensl' compiler flag.
  Instead, OpenSL ES is always supported, and will enabled for devices that
  supports low-latency output.
- WebRTC no longer supports OpenSL ES for the input/recording side.
- Removes old code and demos using OpenSL ES for audio input.
- Improves accuracy of total delay estimates (better AEC performance).
- Reduces roundtrip audio latency; especially when OpenSL can be used.

Performance verified on: Nexus 5, 6, 7 and 9. Samsung Galaxy S4 and S6.
Android One device.

R=magjed@webrtc.org, phoglund@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51759004

Cr-Commit-Position: refs/heads/master@{#9208}
2015-05-18 14:49:04 +00:00

37 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 {
enum {
kDefaultSampleRate = 44100,
kNumChannels = 1,
kDefaultBufSizeInSamples = kDefaultSampleRate * 10 / 1000,
// Number of bytes per audio frame.
// Example: 16-bit PCM in mono => 1*(16/8)=2 [bytes/frame]
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.
kLowLatencyModeDelayEstimateInMilliseconds = 50,
kHighLatencyModeDelayEstimateInMilliseconds = 150,
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_COMMON_H_