blob: 6438f086af9ff9549c17c4e6e690c40af913f258 [file] [log] [blame]
/*
* Copyright (c) 2018 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 "sdk/android/native_api/audio_device_module/audio_device_android.h"
#include <stdlib.h>
#include <memory>
#include <utility>
#include "api/scoped_refptr.h"
#include "rtc_base/logging.h"
#include "rtc_base/ref_count.h"
#if defined(WEBRTC_AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO)
#include "sdk/android/src/jni/audio_device/aaudio_player.h"
#include "sdk/android/src/jni/audio_device/aaudio_recorder.h"
#endif
#include "sdk/android/native_api/jni/application_context_provider.h"
#include "sdk/android/src/jni/audio_device/audio_record_jni.h"
#include "sdk/android/src/jni/audio_device/audio_track_jni.h"
#include "sdk/android/src/jni/audio_device/opensles_player.h"
#include "sdk/android/src/jni/audio_device/opensles_recorder.h"
#include "sdk/android/src/jni/jvm.h"
#include "system_wrappers/include/metrics.h"
#include "third_party/jni_zero/jni_zero.h"
namespace webrtc {
namespace {
void GetDefaultAudioParameters(JNIEnv* env,
jobject application_context,
AudioParameters* input_parameters,
AudioParameters* output_parameters) {
const jni_zero::JavaParamRef<jobject> j_context(env, application_context);
const jni_zero::ScopedJavaLocalRef<jobject> j_audio_manager =
jni::GetAudioManager(env, j_context);
const int input_sample_rate = jni::GetDefaultSampleRate(env, j_audio_manager);
const int output_sample_rate =
jni::GetDefaultSampleRate(env, j_audio_manager);
jni::GetAudioParameters(env, j_context, j_audio_manager, input_sample_rate,
output_sample_rate, false /* use_stereo_input */,
false /* use_stereo_output */, input_parameters,
output_parameters);
}
} // namespace
#if defined(WEBRTC_AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO)
rtc::scoped_refptr<AudioDeviceModule> CreateAAudioAudioDeviceModule(
JNIEnv* env,
jobject application_context) {
RTC_DLOG(LS_INFO) << __FUNCTION__;
// Get default audio input/output parameters.
AudioParameters input_parameters;
AudioParameters output_parameters;
GetDefaultAudioParameters(env, application_context, &input_parameters,
&output_parameters);
// Create ADM from AAudioRecorder and AAudioPlayer.
return CreateAudioDeviceModuleFromInputAndOutput(
AudioDeviceModule::kAndroidAAudioAudio, false /* use_stereo_input */,
false /* use_stereo_output */,
jni::kLowLatencyModeDelayEstimateInMilliseconds,
std::make_unique<jni::AAudioRecorder>(input_parameters),
std::make_unique<jni::AAudioPlayer>(output_parameters));
}
rtc::scoped_refptr<AudioDeviceModule>
CreateJavaInputAndAAudioOutputAudioDeviceModule(JNIEnv* env,
jobject application_context) {
RTC_DLOG(LS_INFO) << __FUNCTION__;
// Get default audio input/output parameters.
const jni_zero::JavaParamRef<jobject> j_context(env, application_context);
const jni_zero::ScopedJavaLocalRef<jobject> j_audio_manager =
jni::GetAudioManager(env, j_context);
AudioParameters input_parameters;
AudioParameters output_parameters;
GetDefaultAudioParameters(env, application_context, &input_parameters,
&output_parameters);
// Create ADM from AudioRecord and OpenSLESPlayer.
auto audio_input = std::make_unique<jni::AudioRecordJni>(
env, input_parameters, jni::kLowLatencyModeDelayEstimateInMilliseconds,
jni::AudioRecordJni::CreateJavaWebRtcAudioRecord(env, j_context,
j_audio_manager));
return CreateAudioDeviceModuleFromInputAndOutput(
AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio,
false /* use_stereo_input */, false /* use_stereo_output */,
jni::kLowLatencyModeDelayEstimateInMilliseconds, std::move(audio_input),
std::make_unique<jni::AAudioPlayer>(output_parameters));
}
#endif
rtc::scoped_refptr<AudioDeviceModule> CreateJavaAudioDeviceModule(
JNIEnv* env,
jobject application_context) {
RTC_DLOG(LS_INFO) << __FUNCTION__;
// Get default audio input/output parameters.
const jni_zero::JavaParamRef<jobject> j_context(env, application_context);
const jni_zero::ScopedJavaLocalRef<jobject> j_audio_manager =
jni::GetAudioManager(env, j_context);
AudioParameters input_parameters;
AudioParameters output_parameters;
GetDefaultAudioParameters(env, application_context, &input_parameters,
&output_parameters);
// Create ADM from AudioRecord and AudioTrack.
auto audio_input = std::make_unique<jni::AudioRecordJni>(
env, input_parameters, jni::kHighLatencyModeDelayEstimateInMilliseconds,
jni::AudioRecordJni::CreateJavaWebRtcAudioRecord(env, j_context,
j_audio_manager));
auto audio_output = std::make_unique<jni::AudioTrackJni>(
env, output_parameters,
jni::AudioTrackJni::CreateJavaWebRtcAudioTrack(env, j_context,
j_audio_manager));
return CreateAudioDeviceModuleFromInputAndOutput(
AudioDeviceModule::kAndroidJavaAudio, false /* use_stereo_input */,
false /* use_stereo_output */,
jni::kHighLatencyModeDelayEstimateInMilliseconds, std::move(audio_input),
std::move(audio_output));
}
rtc::scoped_refptr<AudioDeviceModule> CreateOpenSLESAudioDeviceModule(
JNIEnv* env,
jobject application_context) {
RTC_DLOG(LS_INFO) << __FUNCTION__;
// Get default audio input/output parameters.
AudioParameters input_parameters;
AudioParameters output_parameters;
GetDefaultAudioParameters(env, application_context, &input_parameters,
&output_parameters);
// Create ADM from OpenSLESRecorder and OpenSLESPlayer.
rtc::scoped_refptr<jni::OpenSLEngineManager> engine_manager(
new jni::OpenSLEngineManager());
auto audio_input =
std::make_unique<jni::OpenSLESRecorder>(input_parameters, engine_manager);
auto audio_output = std::make_unique<jni::OpenSLESPlayer>(
output_parameters, std::move(engine_manager));
return CreateAudioDeviceModuleFromInputAndOutput(
AudioDeviceModule::kAndroidOpenSLESAudio, false /* use_stereo_input */,
false /* use_stereo_output */,
jni::kLowLatencyModeDelayEstimateInMilliseconds, std::move(audio_input),
std::move(audio_output));
}
rtc::scoped_refptr<AudioDeviceModule>
CreateJavaInputAndOpenSLESOutputAudioDeviceModule(JNIEnv* env,
jobject application_context) {
RTC_DLOG(LS_INFO) << __FUNCTION__;
// Get default audio input/output parameters.
const jni_zero::JavaParamRef<jobject> j_context(env, application_context);
const jni_zero::ScopedJavaLocalRef<jobject> j_audio_manager =
jni::GetAudioManager(env, j_context);
AudioParameters input_parameters;
AudioParameters output_parameters;
GetDefaultAudioParameters(env, application_context, &input_parameters,
&output_parameters);
// Create ADM from AudioRecord and OpenSLESPlayer.
auto audio_input = std::make_unique<jni::AudioRecordJni>(
env, input_parameters, jni::kLowLatencyModeDelayEstimateInMilliseconds,
jni::AudioRecordJni::CreateJavaWebRtcAudioRecord(env, j_context,
j_audio_manager));
rtc::scoped_refptr<jni::OpenSLEngineManager> engine_manager(
new jni::OpenSLEngineManager());
auto audio_output = std::make_unique<jni::OpenSLESPlayer>(
output_parameters, std::move(engine_manager));
return CreateAudioDeviceModuleFromInputAndOutput(
AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio,
false /* use_stereo_input */, false /* use_stereo_output */,
jni::kLowLatencyModeDelayEstimateInMilliseconds, std::move(audio_input),
std::move(audio_output));
}
rtc::scoped_refptr<AudioDeviceModule> CreateAndroidAudioDeviceModule(
AudioDeviceModule::AudioLayer audio_layer) {
auto env = AttachCurrentThreadIfNeeded();
auto j_context = webrtc::GetAppContext(env);
// Select best possible combination of audio layers.
if (audio_layer == AudioDeviceModule::kPlatformDefaultAudio) {
#if defined(WEBRTC_AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO)
// AAudio based audio for both input and output.
audio_layer = AudioDeviceModule::kAndroidAAudioAudio;
#else
if (jni::IsLowLatencyInputSupported(env, j_context) &&
jni::IsLowLatencyOutputSupported(env, j_context)) {
// Use OpenSL ES for both playout and recording.
audio_layer = AudioDeviceModule::kAndroidOpenSLESAudio;
} else if (jni::IsLowLatencyOutputSupported(env, j_context) &&
!jni::IsLowLatencyInputSupported(env, j_context)) {
// Use OpenSL ES for output on devices that only supports the
// low-latency output audio path.
audio_layer = AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio;
} else {
// Use Java-based audio in both directions when low-latency output is
// not supported.
audio_layer = AudioDeviceModule::kAndroidJavaAudio;
}
#endif
}
switch (audio_layer) {
case AudioDeviceModule::kAndroidJavaAudio:
// Java audio for both input and output audio.
return CreateJavaAudioDeviceModule(env, j_context.obj());
case AudioDeviceModule::kAndroidOpenSLESAudio:
// OpenSL ES based audio for both input and output audio.
return CreateOpenSLESAudioDeviceModule(env, j_context.obj());
case AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio:
// Java audio for input and OpenSL ES for output audio (i.e. mixed APIs).
// This combination provides low-latency output audio and at the same
// time support for HW AEC using the AudioRecord Java API.
return CreateJavaInputAndOpenSLESOutputAudioDeviceModule(
env, j_context.obj());
#if defined(WEBRTC_AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO)
case AudioDeviceModule::kAndroidAAudioAudio:
// AAudio based audio for both input and output.
return CreateAAudioAudioDeviceModule(env, j_context.obj());
case AudioDeviceModule::kAndroidJavaInputAndAAudioOutputAudio:
// Java audio for input and AAudio for output audio (i.e. mixed APIs).
return CreateJavaInputAndAAudioOutputAudioDeviceModule(
env, j_context.obj());
#endif
default:
return nullptr;
}
}
} // namespace webrtc