Files
platform-external-webrtc/webrtc/modules/audio_device/android/ensure_initialized.cc
henrika b26198972c 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 14:49:04 +00:00

50 lines
1.8 KiB
C++

/*
* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/audio_device/android/ensure_initialized.h"
#include <pthread.h>
#include "base/android/jni_android.h"
#include "webrtc/base/checks.h"
#include "webrtc/modules/audio_device/android/audio_device_template.h"
#include "webrtc/modules/audio_device/android/audio_record_jni.h"
#include "webrtc/modules/audio_device/android/audio_track_jni.h"
#include "webrtc/modules/utility/interface/jvm_android.h"
namespace webrtc {
namespace audiodevicemodule {
static pthread_once_t g_initialize_once = PTHREAD_ONCE_INIT;
void EnsureInitializedOnce() {
CHECK(::base::android::IsVMInitialized());
JNIEnv* jni = ::base::android::AttachCurrentThread();
JavaVM* jvm = NULL;
CHECK_EQ(0, jni->GetJavaVM(&jvm));
jobject context = ::base::android::GetApplicationContext();
// Initialize the Java environment (currently only used by the audio manager).
webrtc::JVM::Initialize(jvm, context);
// TODO(henrika): remove this call when AudioRecordJni and AudioTrackJni
// are modified to use the same sort of Java initialization as the audio
// manager.
using AudioDeviceJava = AudioDeviceTemplate<AudioRecordJni, AudioTrackJni>;
AudioDeviceJava::SetAndroidAudioDeviceObjects(jvm, context);
}
void EnsureInitialized() {
CHECK_EQ(0, pthread_once(&g_initialize_once, &EnsureInitializedOnce));
}
} // namespace audiodevicemodule
} // namespace webrtc