9ee75e9c77
BUG=N/A R=fischman@webrtc.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/4189004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5270 4adac7df-926f-26a2-2b94-8c16560cd09d
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_opensl {
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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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