2013-10-03 18:23:13 +00:00
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/*
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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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2017-09-15 06:47:31 +02:00
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#include "modules/utility/include/helpers_android.h"
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#include "rtc_base/checks.h"
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2013-10-03 18:23:13 +00:00
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2015-02-11 08:38:35 +00:00
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#include <android/log.h>
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2013-10-03 18:23:13 +00:00
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#include <assert.h>
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2015-02-11 08:38:35 +00:00
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#include <pthread.h>
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2013-10-03 18:23:13 +00:00
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#include <stddef.h>
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2015-02-11 08:38:35 +00:00
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#include <unistd.h>
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#define TAG "HelpersAndroid"
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#define ALOGD(...) __android_log_print(ANDROID_LOG_DEBUG, TAG, __VA_ARGS__)
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2013-10-03 18:23:13 +00:00
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namespace webrtc {
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2015-02-11 08:38:35 +00:00
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JNIEnv* GetEnv(JavaVM* jvm) {
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void* env = NULL;
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jint status = jvm->GetEnv(&env, JNI_VERSION_1_6);
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2015-09-17 00:24:34 -07:00
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RTC_CHECK(((env != NULL) && (status == JNI_OK)) ||
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((env == NULL) && (status == JNI_EDETACHED)))
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2015-02-11 08:38:35 +00:00
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<< "Unexpected GetEnv return: " << status << ":" << env;
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return reinterpret_cast<JNIEnv*>(env);
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}
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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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// Return a |jlong| that will correctly convert back to |ptr|. This is needed
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// because the alternative (of silently passing a 32-bit pointer to a vararg
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// function expecting a 64-bit param) picks up garbage in the high 32 bits.
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jlong PointerTojlong(void* ptr) {
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static_assert(sizeof(intptr_t) <= sizeof(jlong),
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"Time to rethink the use of jlongs");
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// Going through intptr_t to be obvious about the definedness of the
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// conversion from pointer to integral type. intptr_t to jlong is a standard
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// widening by the static_assert above.
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jlong ret = reinterpret_cast<intptr_t>(ptr);
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2015-09-17 00:24:34 -07:00
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RTC_DCHECK(reinterpret_cast<void*>(ret) == ptr);
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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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return ret;
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}
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2016-11-25 11:45:05 -08:00
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jmethodID GetMethodID (
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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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JNIEnv* jni, jclass c, const char* name, const char* signature) {
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jmethodID m = jni->GetMethodID(c, name, signature);
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2015-02-11 08:38:35 +00:00
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CHECK_EXCEPTION(jni) << "Error during GetMethodID: " << name << ", "
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<< signature;
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2015-09-17 00:24:34 -07:00
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RTC_CHECK(m) << name << ", " << signature;
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2015-02-11 08:38:35 +00:00
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return m;
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}
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2016-11-25 11:45:05 -08:00
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jmethodID GetStaticMethodID (
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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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JNIEnv* jni, jclass c, const char* name, const char* signature) {
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jmethodID m = jni->GetStaticMethodID(c, name, signature);
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CHECK_EXCEPTION(jni) << "Error during GetStaticMethodID: " << name << ", "
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<< signature;
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2015-09-17 00:24:34 -07:00
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RTC_CHECK(m) << name << ", " << signature;
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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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return m;
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}
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jclass FindClass(JNIEnv* jni, const char* name) {
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jclass c = jni->FindClass(name);
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2015-02-11 08:38:35 +00:00
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CHECK_EXCEPTION(jni) << "Error during FindClass: " << name;
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2015-09-17 00:24:34 -07:00
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RTC_CHECK(c) << name;
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2015-02-11 08:38:35 +00:00
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return c;
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}
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jobject NewGlobalRef(JNIEnv* jni, jobject o) {
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jobject ret = jni->NewGlobalRef(o);
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CHECK_EXCEPTION(jni) << "Error during NewGlobalRef";
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2015-09-17 00:24:34 -07:00
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RTC_CHECK(ret);
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2015-02-11 08:38:35 +00:00
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return ret;
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}
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void DeleteGlobalRef(JNIEnv* jni, jobject o) {
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jni->DeleteGlobalRef(o);
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CHECK_EXCEPTION(jni) << "Error during DeleteGlobalRef";
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}
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std::string GetThreadId() {
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char buf[21]; // Big enough to hold a kuint64max plus terminating NULL.
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int thread_id = gettid();
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2015-09-17 00:24:34 -07:00
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RTC_CHECK_LT(snprintf(buf, sizeof(buf), "%i", thread_id),
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static_cast<int>(sizeof(buf)))
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<< "Thread id is bigger than uint64??";
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2015-02-11 08:38:35 +00:00
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return std::string(buf);
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}
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std::string GetThreadInfo() {
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return "@[tid=" + GetThreadId() + "]";
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}
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2013-10-03 18:23:13 +00:00
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AttachThreadScoped::AttachThreadScoped(JavaVM* jvm)
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: attached_(false), jvm_(jvm), env_(NULL) {
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2015-02-11 08:38:35 +00:00
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env_ = GetEnv(jvm);
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if (!env_) {
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// Adding debug log here so we can track down potential leaks and figure
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// out why we sometimes see "Native thread exiting without having called
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// DetachCurrentThread" in logcat outputs.
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ALOGD("Attaching thread to JVM%s", GetThreadInfo().c_str());
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jint res = jvm->AttachCurrentThread(&env_, NULL);
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attached_ = (res == JNI_OK);
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2015-09-17 00:24:34 -07:00
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RTC_CHECK(attached_) << "AttachCurrentThread failed: " << res;
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2013-10-03 18:23:13 +00:00
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}
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}
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AttachThreadScoped::~AttachThreadScoped() {
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2015-02-11 08:38:35 +00:00
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if (attached_) {
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ALOGD("Detaching thread from JVM%s", GetThreadInfo().c_str());
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jint res = jvm_->DetachCurrentThread();
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2015-09-17 00:24:34 -07:00
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RTC_CHECK(res == JNI_OK) << "DetachCurrentThread failed: " << res;
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RTC_CHECK(!GetEnv(jvm_));
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2013-10-03 18:23:13 +00:00
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}
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}
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JNIEnv* AttachThreadScoped::env() { return env_; }
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} // namespace webrtc
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