b26198972c
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}
42 lines
1.5 KiB
C++
42 lines
1.5 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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#include "webrtc/modules/audio_device/android/opensles_common.h"
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#include <assert.h>
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#include "webrtc/modules/audio_device/android/audio_common.h"
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using webrtc::kNumChannels;
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namespace webrtc {
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SLDataFormat_PCM CreatePcmConfiguration(int sample_rate) {
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SLDataFormat_PCM configuration;
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configuration.formatType = SL_DATAFORMAT_PCM;
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configuration.numChannels = kNumChannels;
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// According to the opensles documentation in the ndk:
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// samplesPerSec is actually in units of milliHz, despite the misleading name.
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// It further recommends using constants. However, this would lead to a lot
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// of boilerplate code so it is not done here.
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configuration.samplesPerSec = sample_rate * 1000;
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configuration.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16;
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configuration.containerSize = SL_PCMSAMPLEFORMAT_FIXED_16;
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configuration.channelMask = SL_SPEAKER_FRONT_CENTER;
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if (2 == configuration.numChannels) {
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configuration.channelMask =
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SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT;
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}
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configuration.endianness = SL_BYTEORDER_LITTLEENDIAN;
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return configuration;
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}
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} // namespace webrtc_opensl
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