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platform-external-webrtc/webrtc/modules/audio_device/android/audio_manager.cc
Henrik Kjellander ff761fba82 modules: more interface -> include renames 
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface

To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
                "use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)

BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc

R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1417683006 .

Cr-Commit-Position: refs/heads/master@{#10500}
2015-11-04 07:32:04 +00:00

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/*
* 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 <android/log.h>
#include "webrtc/base/arraysize.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/scoped_ptr.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, rtc::scoped_ptr<GlobalRef> audio_manager)
: audio_manager_(audio_manager.Pass()),
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),
delay_estimate_in_milliseconds_(0) {
ALOGD("ctor%s", GetThreadInfo().c_str());
RTC_CHECK(j_environment_);
JNINativeMethod native_methods[] = {
{"nativeCacheAudioParameters",
"(IIZZZZIIJ)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 currenttly 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_);
}
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_;
}
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,
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, 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,
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("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;
// TODO(henrika): add support for stereo output.
playout_parameters_.reset(sample_rate, channels,
static_cast<size_t>(output_buffer_size));
record_parameters_.reset(sample_rate, 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
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