62f6e75673
This is a big refactoring of the existing C++/JNI/Java support for audio recording in native WebRTC: - Removes unused code and old WEBRTC logging macros - Now uses optimal sample rate and buffer size in Java AudioRecord (used hard-coded sample rate before) - Makes code more inline with the implementation in Chrome - Adds helper methods for JNI handling to improve readability - Changes the threading model (high-prio audio thread now lives in Java-land and C++ only works as proxy) - Adds basic thread checks - Removes all locks in C++ land - Removes all locks in Java - Improves construction/destruction - Additional cleanup Tested using AppRTCDemo and WebRTCDemo APKs on N6, N5, N7, Samsung Galaxy S4 and Samsung Galaxy S4 mini (which uses 44.1kHz as native sample rate). BUG=NONE R=magjed@webrtc.org, perkj@webrtc.org, pthatcher@webrtc.org Review URL: https://webrtc-codereview.appspot.com/33969004 Cr-Commit-Position: refs/heads/master@{#8325} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8325 4adac7df-926f-26a2-2b94-8c16560cd09d
35 lines
943 B
C++
35 lines
943 B
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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kBitsPerSample = 16,
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kNumChannels = 1,
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kDefaultBufSizeInSamples = kDefaultSampleRate * 10 / 1000,
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};
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class PlayoutDelayProvider {
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public:
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virtual int PlayoutDelayMs() = 0;
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protected:
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PlayoutDelayProvider() {}
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virtual ~PlayoutDelayProvider() {}
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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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