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00b8462eb7
The app showcased the ability to send real-time voice data between two endpoints using the VoIP API. Users can also configure session parameters such as the endpoint information and codec used. Bug: webrtc:11723 Change-Id: I682f4aa743b707759536bce59e598789a77b7ec6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178467 Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> Reviewed-by: Tim Na <natim@webrtc.org> Commit-Queue: Tim Na <natim@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31775}
28 lines
910 B
C++
28 lines
910 B
C++
/*
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* Copyright 2020 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 <jni.h>
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#include "rtc_base/ssl_adapter.h"
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#include "sdk/android/native_api/base/init.h"
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namespace webrtc_examples {
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extern "C" jint JNIEXPORT JNICALL JNI_OnLoad(JavaVM* jvm, void* reserved) {
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webrtc::InitAndroid(jvm);
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RTC_CHECK(rtc::InitializeSSL()) << "Failed to InitializeSSL()";
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return JNI_VERSION_1_6;
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}
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extern "C" void JNIEXPORT JNICALL JNI_OnUnLoad(JavaVM* jvm, void* reserved) {
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RTC_CHECK(rtc::CleanupSSL()) << "Failed to CleanupSSL()";
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}
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} // namespace webrtc_examples
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