918b554789
NOTE: the new code is disabled by default in the WebRtcAudioManager to ensure that OpenSL ES is not accidentally activated in existing clients. There are still some unresolved issues to sort out before it can be utilized. Enables possibility to use OpenSL ES based audio in both directions for WebRTC. All unit tests and demo clients have been tested with the new implementation but the new support is behind a flag (see above). More testing is needed before it can be used in the field and additional support for hardware effects is still missing. BUG=webrtc:5925 R=tommi@webrtc.org Review URL: https://codereview.webrtc.org/2119633004 . Cr-Commit-Position: refs/heads/master@{#14290}
298 lines
12 KiB
C++
298 lines
12 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/audio_manager.h"
|
|
|
|
#include <utility>
|
|
|
|
#include <android/log.h>
|
|
|
|
#include "webrtc/base/arraysize.h"
|
|
#include "webrtc/base/checks.h"
|
|
#include "webrtc/modules/audio_device/android/audio_common.h"
|
|
#include "webrtc/modules/utility/include/helpers_android.h"
|
|
|
|
#define TAG "AudioManager"
|
|
#define ALOGV(...) __android_log_print(ANDROID_LOG_VERBOSE, TAG, __VA_ARGS__)
|
|
#define ALOGD(...) __android_log_print(ANDROID_LOG_DEBUG, TAG, __VA_ARGS__)
|
|
#define ALOGE(...) __android_log_print(ANDROID_LOG_ERROR, TAG, __VA_ARGS__)
|
|
#define ALOGW(...) __android_log_print(ANDROID_LOG_WARN, TAG, __VA_ARGS__)
|
|
#define ALOGI(...) __android_log_print(ANDROID_LOG_INFO, TAG, __VA_ARGS__)
|
|
|
|
namespace webrtc {
|
|
|
|
// AudioManager::JavaAudioManager implementation
|
|
AudioManager::JavaAudioManager::JavaAudioManager(
|
|
NativeRegistration* native_reg,
|
|
std::unique_ptr<GlobalRef> audio_manager)
|
|
: audio_manager_(std::move(audio_manager)),
|
|
init_(native_reg->GetMethodId("init", "()Z")),
|
|
dispose_(native_reg->GetMethodId("dispose", "()V")),
|
|
is_communication_mode_enabled_(
|
|
native_reg->GetMethodId("isCommunicationModeEnabled", "()Z")),
|
|
is_device_blacklisted_for_open_sles_usage_(
|
|
native_reg->GetMethodId("isDeviceBlacklistedForOpenSLESUsage",
|
|
"()Z")) {
|
|
ALOGD("JavaAudioManager::ctor%s", GetThreadInfo().c_str());
|
|
}
|
|
|
|
AudioManager::JavaAudioManager::~JavaAudioManager() {
|
|
ALOGD("JavaAudioManager::dtor%s", GetThreadInfo().c_str());
|
|
}
|
|
|
|
bool AudioManager::JavaAudioManager::Init() {
|
|
return audio_manager_->CallBooleanMethod(init_);
|
|
}
|
|
|
|
void AudioManager::JavaAudioManager::Close() {
|
|
audio_manager_->CallVoidMethod(dispose_);
|
|
}
|
|
|
|
bool AudioManager::JavaAudioManager::IsCommunicationModeEnabled() {
|
|
return audio_manager_->CallBooleanMethod(is_communication_mode_enabled_);
|
|
}
|
|
|
|
bool AudioManager::JavaAudioManager::IsDeviceBlacklistedForOpenSLESUsage() {
|
|
return audio_manager_->CallBooleanMethod(
|
|
is_device_blacklisted_for_open_sles_usage_);
|
|
}
|
|
|
|
// AudioManager implementation
|
|
AudioManager::AudioManager()
|
|
: j_environment_(JVM::GetInstance()->environment()),
|
|
audio_layer_(AudioDeviceModule::kPlatformDefaultAudio),
|
|
initialized_(false),
|
|
hardware_aec_(false),
|
|
hardware_agc_(false),
|
|
hardware_ns_(false),
|
|
low_latency_playout_(false),
|
|
low_latency_record_(false),
|
|
delay_estimate_in_milliseconds_(0) {
|
|
ALOGD("ctor%s", GetThreadInfo().c_str());
|
|
RTC_CHECK(j_environment_);
|
|
JNINativeMethod native_methods[] = {
|
|
{"nativeCacheAudioParameters", "(IIZZZZZZIIJ)V",
|
|
reinterpret_cast<void*>(&webrtc::AudioManager::CacheAudioParameters)}};
|
|
j_native_registration_ = j_environment_->RegisterNatives(
|
|
"org/webrtc/voiceengine/WebRtcAudioManager", native_methods,
|
|
arraysize(native_methods));
|
|
j_audio_manager_.reset(new JavaAudioManager(
|
|
j_native_registration_.get(),
|
|
j_native_registration_->NewObject(
|
|
"<init>", "(Landroid/content/Context;J)V",
|
|
JVM::GetInstance()->context(), PointerTojlong(this))));
|
|
}
|
|
|
|
AudioManager::~AudioManager() {
|
|
ALOGD("~dtor%s", GetThreadInfo().c_str());
|
|
RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
|
Close();
|
|
}
|
|
|
|
void AudioManager::SetActiveAudioLayer(
|
|
AudioDeviceModule::AudioLayer audio_layer) {
|
|
ALOGD("SetActiveAudioLayer(%d)%s", audio_layer, GetThreadInfo().c_str());
|
|
RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
|
RTC_DCHECK(!initialized_);
|
|
// Store the currently utilized audio layer.
|
|
audio_layer_ = audio_layer;
|
|
// The delay estimate can take one of two fixed values depending on if the
|
|
// device supports low-latency output or not. However, it is also possible
|
|
// that the user explicitly selects the high-latency audio path, hence we use
|
|
// the selected |audio_layer| here to set the delay estimate.
|
|
delay_estimate_in_milliseconds_ =
|
|
(audio_layer == AudioDeviceModule::kAndroidJavaAudio) ?
|
|
kHighLatencyModeDelayEstimateInMilliseconds :
|
|
kLowLatencyModeDelayEstimateInMilliseconds;
|
|
ALOGD("delay_estimate_in_milliseconds: %d", delay_estimate_in_milliseconds_);
|
|
}
|
|
|
|
SLObjectItf AudioManager::GetOpenSLEngine() {
|
|
ALOGD("GetOpenSLEngine%s", GetThreadInfo().c_str());
|
|
RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
|
// Only allow usage of OpenSL ES if such an audio layer has been specified.
|
|
if (audio_layer_ != AudioDeviceModule::kAndroidOpenSLESAudio &&
|
|
audio_layer_ !=
|
|
AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio) {
|
|
ALOGI("Unable to create OpenSL engine for the current audio layer: %d",
|
|
audio_layer_);
|
|
return nullptr;
|
|
}
|
|
// OpenSL ES for Android only supports a single engine per application.
|
|
// If one already has been created, return existing object instead of
|
|
// creating a new.
|
|
if (engine_object_.Get() != nullptr) {
|
|
ALOGI("The OpenSL ES engine object has already been created");
|
|
return engine_object_.Get();
|
|
}
|
|
// Create the engine object in thread safe mode.
|
|
const SLEngineOption option[] = {
|
|
{SL_ENGINEOPTION_THREADSAFE, static_cast<SLuint32>(SL_BOOLEAN_TRUE)}};
|
|
SLresult result =
|
|
slCreateEngine(engine_object_.Receive(), 1, option, 0, NULL, NULL);
|
|
if (result != SL_RESULT_SUCCESS) {
|
|
ALOGE("slCreateEngine() failed: %s", GetSLErrorString(result));
|
|
engine_object_.Reset();
|
|
return nullptr;
|
|
}
|
|
// Realize the SL Engine in synchronous mode.
|
|
result = engine_object_->Realize(engine_object_.Get(), SL_BOOLEAN_FALSE);
|
|
if (result != SL_RESULT_SUCCESS) {
|
|
ALOGE("Realize() failed: %s", GetSLErrorString(result));
|
|
engine_object_.Reset();
|
|
return nullptr;
|
|
}
|
|
// Finally return the SLObjectItf interface of the engine object.
|
|
return engine_object_.Get();
|
|
}
|
|
|
|
bool AudioManager::Init() {
|
|
ALOGD("Init%s", GetThreadInfo().c_str());
|
|
RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
|
RTC_DCHECK(!initialized_);
|
|
RTC_DCHECK_NE(audio_layer_, AudioDeviceModule::kPlatformDefaultAudio);
|
|
if (!j_audio_manager_->Init()) {
|
|
ALOGE("init failed!");
|
|
return false;
|
|
}
|
|
initialized_ = true;
|
|
return true;
|
|
}
|
|
|
|
bool AudioManager::Close() {
|
|
ALOGD("Close%s", GetThreadInfo().c_str());
|
|
RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
|
if (!initialized_)
|
|
return true;
|
|
j_audio_manager_->Close();
|
|
initialized_ = false;
|
|
return true;
|
|
}
|
|
|
|
bool AudioManager::IsCommunicationModeEnabled() const {
|
|
ALOGD("IsCommunicationModeEnabled()");
|
|
RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
|
return j_audio_manager_->IsCommunicationModeEnabled();
|
|
}
|
|
|
|
bool AudioManager::IsAcousticEchoCancelerSupported() const {
|
|
RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
|
return hardware_aec_;
|
|
}
|
|
|
|
bool AudioManager::IsAutomaticGainControlSupported() const {
|
|
RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
|
return hardware_agc_;
|
|
}
|
|
|
|
bool AudioManager::IsNoiseSuppressorSupported() const {
|
|
RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
|
return hardware_ns_;
|
|
}
|
|
|
|
bool AudioManager::IsLowLatencyPlayoutSupported() const {
|
|
RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
|
ALOGD("IsLowLatencyPlayoutSupported()");
|
|
// Some devices are blacklisted for usage of OpenSL ES even if they report
|
|
// that low-latency playout is supported. See b/21485703 for details.
|
|
return j_audio_manager_->IsDeviceBlacklistedForOpenSLESUsage() ?
|
|
false : low_latency_playout_;
|
|
}
|
|
|
|
bool AudioManager::IsLowLatencyRecordSupported() const {
|
|
RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
|
ALOGD("IsLowLatencyRecordSupported()");
|
|
return low_latency_record_;
|
|
}
|
|
|
|
bool AudioManager::IsProAudioSupported() const {
|
|
RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
|
ALOGD("IsProAudioSupported()");
|
|
// TODO(henrika): return the state independently of if OpenSL ES is
|
|
// blacklisted or not for now. We could use the same approach as in
|
|
// IsLowLatencyPlayoutSupported() but I can't see the need for it yet.
|
|
return pro_audio_;
|
|
}
|
|
|
|
int AudioManager::GetDelayEstimateInMilliseconds() const {
|
|
return delay_estimate_in_milliseconds_;
|
|
}
|
|
|
|
void JNICALL AudioManager::CacheAudioParameters(JNIEnv* env,
|
|
jobject obj,
|
|
jint sample_rate,
|
|
jint channels,
|
|
jboolean hardware_aec,
|
|
jboolean hardware_agc,
|
|
jboolean hardware_ns,
|
|
jboolean low_latency_output,
|
|
jboolean low_latency_input,
|
|
jboolean pro_audio,
|
|
jint output_buffer_size,
|
|
jint input_buffer_size,
|
|
jlong native_audio_manager) {
|
|
webrtc::AudioManager* this_object =
|
|
reinterpret_cast<webrtc::AudioManager*>(native_audio_manager);
|
|
this_object->OnCacheAudioParameters(
|
|
env, sample_rate, channels, hardware_aec, hardware_agc, hardware_ns,
|
|
low_latency_output, low_latency_input, pro_audio, output_buffer_size,
|
|
input_buffer_size);
|
|
}
|
|
|
|
void AudioManager::OnCacheAudioParameters(JNIEnv* env,
|
|
jint sample_rate,
|
|
jint channels,
|
|
jboolean hardware_aec,
|
|
jboolean hardware_agc,
|
|
jboolean hardware_ns,
|
|
jboolean low_latency_output,
|
|
jboolean low_latency_input,
|
|
jboolean pro_audio,
|
|
jint output_buffer_size,
|
|
jint input_buffer_size) {
|
|
ALOGD("OnCacheAudioParameters%s", GetThreadInfo().c_str());
|
|
ALOGD("hardware_aec: %d", hardware_aec);
|
|
ALOGD("hardware_agc: %d", hardware_agc);
|
|
ALOGD("hardware_ns: %d", hardware_ns);
|
|
ALOGD("low_latency_output: %d", low_latency_output);
|
|
ALOGD("low_latency_input: %d", low_latency_input);
|
|
ALOGD("pro_audio: %d", pro_audio);
|
|
ALOGD("sample_rate: %d", sample_rate);
|
|
ALOGD("channels: %d", channels);
|
|
ALOGD("output_buffer_size: %d", output_buffer_size);
|
|
ALOGD("input_buffer_size: %d", input_buffer_size);
|
|
RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
|
hardware_aec_ = hardware_aec;
|
|
hardware_agc_ = hardware_agc;
|
|
hardware_ns_ = hardware_ns;
|
|
low_latency_playout_ = low_latency_output;
|
|
low_latency_record_ = low_latency_input;
|
|
pro_audio_ = pro_audio;
|
|
// TODO(henrika): add support for stereo output.
|
|
playout_parameters_.reset(sample_rate, static_cast<size_t>(channels),
|
|
static_cast<size_t>(output_buffer_size));
|
|
record_parameters_.reset(sample_rate, static_cast<size_t>(channels),
|
|
static_cast<size_t>(input_buffer_size));
|
|
}
|
|
|
|
const AudioParameters& AudioManager::GetPlayoutAudioParameters() {
|
|
RTC_CHECK(playout_parameters_.is_valid());
|
|
RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
|
return playout_parameters_;
|
|
}
|
|
|
|
const AudioParameters& AudioManager::GetRecordAudioParameters() {
|
|
RTC_CHECK(record_parameters_.is_valid());
|
|
RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
|
return record_parameters_;
|
|
}
|
|
|
|
} // namespace webrtc
|