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}
37 lines
1.4 KiB
C++
37 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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kDefaultBufSizeInSamples = kDefaultSampleRate * 10 / 1000,
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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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