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 16:49:16 +02:00
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/*
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* Copyright (c) 2015 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_BUILD_INFO_H_
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#define WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_BUILD_INFO_H_
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#include <jni.h>
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2016-02-24 05:00:36 -08:00
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#include <memory>
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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 16:49:16 +02:00
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#include <string>
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2015-11-04 08:31:52 +01:00
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#include "webrtc/modules/utility/include/jvm_android.h"
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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 16:49:16 +02:00
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namespace webrtc {
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// Utility class used to query the Java class (org/webrtc/voiceengine/BuildInfo)
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// for device and Android build information.
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// The calling thread is attached to the JVM at construction if needed and a
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// valid Java environment object is also created.
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// All Get methods must be called on the creating thread. If not, the code will
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2015-09-17 00:24:34 -07:00
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// hit RTC_DCHECKs when calling JNIEnvironment::JavaToStdString().
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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 16:49:16 +02:00
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class BuildInfo {
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public:
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BuildInfo();
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~BuildInfo() {}
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// End-user-visible name for the end product (e.g. "Nexus 6").
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std::string GetDeviceModel();
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// Consumer-visible brand (e.g. "google").
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std::string GetBrand();
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// Manufacturer of the product/hardware (e.g. "motorola").
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std::string GetDeviceManufacturer();
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// Android build ID (e.g. LMY47D).
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std::string GetAndroidBuildId();
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// The type of build (e.g. "user" or "eng").
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std::string GetBuildType();
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// The user-visible version string (e.g. "5.1").
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std::string GetBuildRelease();
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// The user-visible SDK version of the framework (e.g. 21).
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std::string GetSdkVersion();
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private:
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// Helper method which calls a static getter method with |name| and returns
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// a string from Java.
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std::string GetStringFromJava(const char* name);
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// Ensures that this class can access a valid JNI interface pointer even
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// if the creating thread was not attached to the JVM.
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AttachCurrentThreadIfNeeded attach_thread_if_needed_;
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// Provides access to the JNIEnv interface pointer and the JavaToStdString()
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// method which is used to translate Java strings to std strings.
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2016-02-24 05:00:36 -08:00
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std::unique_ptr<JNIEnvironment> j_environment_;
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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 16:49:16 +02:00
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// Holds the jclass object and provides access to CallStaticObjectMethod().
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// Used by GetStringFromJava() during construction only.
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JavaClass j_build_info_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_BUILD_INFO_H_
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