
Also adds a framework for an AudioManager to be used by both sides (playout and recording). This initial implementation only does very simple tasks like setting up the correct audio mode (needed for correct volume behavior). Note that this CL is mainly about modifying the volume. The added AudioManager is only a place holder for future work. I could have done the same parts in the WebRtcAudioTrack class but feel that it is better to move these parts to an AudioManager already at this stage. The AudioManager supports Init() where actual audio changes are done (set audio mode etc.) but it can also be used a simple "construct-and-store-audio-parameters" unit, which is the case here. Hence, the AM now serves as the center for getting audio parameters and then inject these into playout and recording sides. Previously, both sides acquired their own parameters and that is more error prone. BUG=NONE TEST=AudioDeviceTest R=perkj@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/45829004 Cr-Commit-Position: refs/heads/master@{#8875}
174 lines
5.9 KiB
C++
174 lines
5.9 KiB
C++
/*
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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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#include "webrtc/modules/audio_device/android/audio_manager.h"
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#include <android/log.h>
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#include "webrtc/base/arraysize.h"
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#include "webrtc/base/checks.h"
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#include "webrtc/modules/audio_device/android/audio_common.h"
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#include "webrtc/modules/utility/interface/helpers_android.h"
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#define TAG "AudioManager"
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#define ALOGV(...) __android_log_print(ANDROID_LOG_VERBOSE, TAG, __VA_ARGS__)
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#define ALOGD(...) __android_log_print(ANDROID_LOG_DEBUG, TAG, __VA_ARGS__)
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#define ALOGE(...) __android_log_print(ANDROID_LOG_ERROR, TAG, __VA_ARGS__)
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#define ALOGW(...) __android_log_print(ANDROID_LOG_WARN, TAG, __VA_ARGS__)
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#define ALOGI(...) __android_log_print(ANDROID_LOG_INFO, TAG, __VA_ARGS__)
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namespace webrtc {
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static JavaVM* g_jvm = NULL;
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static jobject g_context = NULL;
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static jclass g_audio_manager_class = NULL;
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void AudioManager::SetAndroidAudioDeviceObjects(void* jvm, void* context) {
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ALOGD("SetAndroidAudioDeviceObjects%s", GetThreadInfo().c_str());
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CHECK(jvm);
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CHECK(context);
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g_jvm = reinterpret_cast<JavaVM*>(jvm);
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JNIEnv* jni = GetEnv(g_jvm);
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CHECK(jni) << "AttachCurrentThread must be called on this tread";
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g_context = NewGlobalRef(jni, reinterpret_cast<jobject>(context));
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jclass local_class = FindClass(
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jni, "org/webrtc/voiceengine/WebRtcAudioManager");
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g_audio_manager_class = reinterpret_cast<jclass>(
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NewGlobalRef(jni, local_class));
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CHECK_EXCEPTION(jni);
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// Register native methods with the WebRtcAudioManager class. These methods
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// are declared private native in WebRtcAudioManager.java.
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JNINativeMethod native_methods[] = {
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{"nativeCacheAudioParameters", "(IIJ)V",
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reinterpret_cast<void*>(&webrtc::AudioManager::CacheAudioParameters)}};
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jni->RegisterNatives(g_audio_manager_class,
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native_methods, arraysize(native_methods));
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CHECK_EXCEPTION(jni) << "Error during RegisterNatives";
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}
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void AudioManager::ClearAndroidAudioDeviceObjects() {
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ALOGD("ClearAndroidAudioDeviceObjects%s", GetThreadInfo().c_str());
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JNIEnv* jni = GetEnv(g_jvm);
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CHECK(jni) << "AttachCurrentThread must be called on this tread";
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jni->UnregisterNatives(g_audio_manager_class);
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CHECK_EXCEPTION(jni) << "Error during UnregisterNatives";
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DeleteGlobalRef(jni, g_audio_manager_class);
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g_audio_manager_class = NULL;
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DeleteGlobalRef(jni, g_context);
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g_context = NULL;
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g_jvm = NULL;
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}
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AudioManager::AudioManager()
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: j_audio_manager_(NULL),
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initialized_(false) {
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ALOGD("ctor%s", GetThreadInfo().c_str());
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CHECK(HasDeviceObjects());
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CreateJavaInstance();
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}
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AudioManager::~AudioManager() {
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ALOGD("~dtor%s", GetThreadInfo().c_str());
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DCHECK(thread_checker_.CalledOnValidThread());
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Close();
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AttachThreadScoped ats(g_jvm);
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JNIEnv* jni = ats.env();
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jni->DeleteGlobalRef(j_audio_manager_);
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j_audio_manager_ = NULL;
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DCHECK(!initialized_);
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}
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bool AudioManager::Init() {
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ALOGD("Init%s", GetThreadInfo().c_str());
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DCHECK(thread_checker_.CalledOnValidThread());
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DCHECK(!initialized_);
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AttachThreadScoped ats(g_jvm);
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JNIEnv* jni = ats.env();
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jmethodID initID = GetMethodID(jni, g_audio_manager_class, "init", "()Z");
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jboolean res = jni->CallBooleanMethod(j_audio_manager_, initID);
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CHECK_EXCEPTION(jni);
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if (!res) {
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ALOGE("init failed!");
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return false;
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}
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initialized_ = true;
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return true;
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}
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bool AudioManager::Close() {
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ALOGD("Close%s", GetThreadInfo().c_str());
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DCHECK(thread_checker_.CalledOnValidThread());
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if (!initialized_)
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return true;
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AttachThreadScoped ats(g_jvm);
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JNIEnv* jni = ats.env();
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jmethodID disposeID = GetMethodID(
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jni, g_audio_manager_class, "dispose", "()V");
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jni->CallVoidMethod(j_audio_manager_, disposeID);
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CHECK_EXCEPTION(jni);
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initialized_ = false;
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return true;
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}
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void JNICALL AudioManager::CacheAudioParameters(JNIEnv* env, jobject obj,
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jint sample_rate, jint channels, jlong nativeAudioManager) {
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webrtc::AudioManager* this_object =
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reinterpret_cast<webrtc::AudioManager*> (nativeAudioManager);
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this_object->OnCacheAudioParameters(env, sample_rate, channels);
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}
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void AudioManager::OnCacheAudioParameters(
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JNIEnv* env, jint sample_rate, jint channels) {
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ALOGD("OnCacheAudioParameters%s", GetThreadInfo().c_str());
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ALOGD("sample_rate: %d", sample_rate);
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ALOGD("channels: %d", channels);
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DCHECK(thread_checker_.CalledOnValidThread());
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// TODO(henrika): add support stereo output.
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playout_parameters_.reset(sample_rate, channels);
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record_parameters_.reset(sample_rate, channels);
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}
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AudioParameters AudioManager::GetPlayoutAudioParameters() const {
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CHECK(playout_parameters_.is_valid());
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return playout_parameters_;
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}
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AudioParameters AudioManager::GetRecordAudioParameters() const {
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CHECK(record_parameters_.is_valid());
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return record_parameters_;
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}
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bool AudioManager::HasDeviceObjects() {
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return (g_jvm && g_context && g_audio_manager_class);
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}
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void AudioManager::CreateJavaInstance() {
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ALOGD("CreateJavaInstance");
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AttachThreadScoped ats(g_jvm);
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JNIEnv* jni = ats.env();
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jmethodID constructorID = GetMethodID(
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jni, g_audio_manager_class, "<init>", "(Landroid/content/Context;J)V");
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j_audio_manager_ = jni->NewObject(g_audio_manager_class,
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constructorID,
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g_context,
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reinterpret_cast<intptr_t>(this));
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CHECK_EXCEPTION(jni) << "Error during NewObject";
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CHECK(j_audio_manager_);
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j_audio_manager_ = jni->NewGlobalRef(j_audio_manager_);
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CHECK_EXCEPTION(jni) << "Error during NewGlobalRef";
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CHECK(j_audio_manager_);
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}
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} // namespace webrtc
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