Android audio code: Replace C++ template with input/output interface
Bug: webrtc:7452
Change-Id: Id816500051e065918bba5c2235d38ad8eb50a8eb
Reviewed-on: https://webrtc-review.googlesource.com/64442
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22660}diff --git a/examples/androidapp/src/org/appspot/apprtc/PeerConnectionClient.java b/examples/androidapp/src/org/appspot/apprtc/PeerConnectionClient.java
index 93343a0..fab9201 100644
--- a/examples/androidapp/src/org/appspot/apprtc/PeerConnectionClient.java
+++ b/examples/androidapp/src/org/appspot/apprtc/PeerConnectionClient.java
@@ -595,10 +595,9 @@
// Enable/disable OpenSL ES playback.
if (!peerConnectionParameters.useOpenSLES) {
Log.d(TAG, "Disable OpenSL ES audio even if device supports it");
- AudioDeviceModule.setBlacklistDeviceForOpenSLESUsage(true /* enable */);
} else {
Log.d(TAG, "Allow OpenSL ES audio if device supports it");
- AudioDeviceModule.setBlacklistDeviceForOpenSLESUsage(false);
+ // TODO(magjed): Add support for external OpenSLES ADM.
}
if (peerConnectionParameters.disableBuiltInAEC) {
diff --git a/sdk/android/BUILD.gn b/sdk/android/BUILD.gn
index 8b6c0d2..3061170 100644
--- a/sdk/android/BUILD.gn
+++ b/sdk/android/BUILD.gn
@@ -164,7 +164,8 @@
sources = [
"src/jni/audio_device/audio_common.h",
- "src/jni/audio_device/audio_device_template_android.h",
+ "src/jni/audio_device/audio_device_module.cc",
+ "src/jni/audio_device/audio_device_module.h",
"src/jni/audio_device/audio_manager.cc",
"src/jni/audio_device/audio_manager.h",
"src/jni/audio_device/audio_record_jni.cc",
diff --git a/sdk/android/api/org/webrtc/audio/AudioDeviceModule.java b/sdk/android/api/org/webrtc/audio/AudioDeviceModule.java
index cc6f37f..a9c75b8 100644
--- a/sdk/android/api/org/webrtc/audio/AudioDeviceModule.java
+++ b/sdk/android/api/org/webrtc/audio/AudioDeviceModule.java
@@ -24,10 +24,6 @@
public AudioDeviceModule() {}
/* AudioManager */
- public static void setBlacklistDeviceForOpenSLESUsage(boolean enable) {
- WebRtcAudioManager.setBlacklistDeviceForOpenSLESUsage(enable);
- }
-
public static void setStereoInput(boolean enable) {
WebRtcAudioManager.setStereoInput(enable);
}
diff --git a/sdk/android/native_api/audio_device_module/audio_device_android.cc b/sdk/android/native_api/audio_device_module/audio_device_android.cc
index 9821f3f6..2649aa7 100644
--- a/sdk/android/native_api/audio_device_module/audio_device_android.cc
+++ b/sdk/android/native_api/audio_device_module/audio_device_android.cc
@@ -11,31 +11,76 @@
#include "sdk/android/native_api/audio_device_module/audio_device_android.h"
#include <stdlib.h>
+#include <utility>
+
#include "rtc_base/logging.h"
+#include "rtc_base/ptr_util.h"
#include "rtc_base/refcount.h"
#include "rtc_base/refcountedobject.h"
-#include "system_wrappers/include/metrics.h"
+#include "sdk/android/src/jni/audio_device/aaudio_player.h"
+#include "sdk/android/src/jni/audio_device/aaudio_recorder.h"
#include "sdk/android/src/jni/audio_device/audio_manager.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 "system_wrappers/include/metrics.h"
namespace webrtc {
+namespace {
+
+// This template function takes care of some boiler plate.
+template <typename AudioInputT, typename AudioOutputT>
+rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceModuleTemplate(
+ AudioDeviceModule::AudioLayer audio_layer,
+ JNIEnv* env,
+ jobject application_context) {
+ auto audio_manager = rtc::MakeUnique<android_adm::AudioManager>(
+ env, audio_layer, JavaParamRef<jobject>(application_context));
+ auto audio_input = rtc::MakeUnique<AudioInputT>(audio_manager.get());
+ auto audio_output = rtc::MakeUnique<AudioOutputT>(audio_manager.get());
+ return CreateAudioDeviceModuleFromInputAndOutput(
+ audio_layer, std::move(audio_manager), std::move(audio_input),
+ std::move(audio_output));
+}
+
+} // namespace
+
#if defined(AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO)
rtc::scoped_refptr<AudioDeviceModule> CreateAAudioAudioDeviceModule(
JNIEnv* env,
jobject application_context) {
- return android_adm::AudioManager::CreateAAudioAudioDeviceModule(
- env, JavaParamRef<jobject>(application_context));
+ RTC_LOG(INFO) << __FUNCTION__;
+ return CreateAudioDeviceModuleTemplate<android_adm::AAudioRecorder,
+ android_adm::AAudioPlayer>(
+ AudioDeviceModule::kAndroidAAudioAudio, env, application_context);
}
#endif
-rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceModule(
+rtc::scoped_refptr<AudioDeviceModule> CreateJavaAudioDeviceModule(
JNIEnv* env,
- jobject application_context,
- bool use_opensles_input,
- bool use_opensles_output) {
- return android_adm::AudioManager::CreateAudioDeviceModule(
- env, JavaParamRef<jobject>(application_context), use_opensles_input,
- use_opensles_output);
+ jobject application_context) {
+ return CreateAudioDeviceModuleTemplate<android_adm::AudioRecordJni,
+ android_adm::AudioTrackJni>(
+ AudioDeviceModule::kAndroidJavaAudio, env, application_context);
+}
+
+rtc::scoped_refptr<AudioDeviceModule> CreateOpenSLESAudioDeviceModule(
+ JNIEnv* env,
+ jobject application_context) {
+ return CreateAudioDeviceModuleTemplate<android_adm::OpenSLESRecorder,
+ android_adm::OpenSLESPlayer>(
+ AudioDeviceModule::kAndroidJavaAudio, env, application_context);
+}
+
+rtc::scoped_refptr<AudioDeviceModule>
+CreateJavaInputAndOpenSLESOutputAudioDeviceModule(JNIEnv* env,
+ jobject application_context) {
+ return CreateAudioDeviceModuleTemplate<android_adm::AudioRecordJni,
+ android_adm::OpenSLESPlayer>(
+ AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio, env,
+ application_context);
}
} // namespace webrtc
diff --git a/sdk/android/native_api/audio_device_module/audio_device_android.h b/sdk/android/native_api/audio_device_module/audio_device_android.h
index 5f2561d..ef404fd 100644
--- a/sdk/android/native_api/audio_device_module/audio_device_android.h
+++ b/sdk/android/native_api/audio_device_module/audio_device_android.h
@@ -23,11 +23,17 @@
jobject application_context);
#endif
-rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceModule(
+rtc::scoped_refptr<AudioDeviceModule> CreateJavaAudioDeviceModule(
JNIEnv* env,
- jobject application_context,
- bool use_opensles_input,
- bool use_opensles_output);
+ jobject application_context);
+
+rtc::scoped_refptr<AudioDeviceModule> CreateOpenSLESAudioDeviceModule(
+ JNIEnv* env,
+ jobject application_context);
+
+rtc::scoped_refptr<AudioDeviceModule>
+CreateJavaInputAndOpenSLESOutputAudioDeviceModule(JNIEnv* env,
+ jobject application_context);
} // namespace webrtc
diff --git a/sdk/android/src/java/org/webrtc/audio/WebRtcAudioManager.java b/sdk/android/src/java/org/webrtc/audio/WebRtcAudioManager.java
index aa83656..e75b3ed 100644
--- a/sdk/android/src/java/org/webrtc/audio/WebRtcAudioManager.java
+++ b/sdk/android/src/java/org/webrtc/audio/WebRtcAudioManager.java
@@ -42,20 +42,6 @@
private static boolean useStereoOutput = false;
private static boolean useStereoInput = false;
- private static boolean blacklistDeviceForOpenSLESUsage = false;
- private static boolean blacklistDeviceForOpenSLESUsageIsOverridden = false;
-
- // Call this method to override the default list of blacklisted devices
- // specified in WebRtcAudioUtils.BLACKLISTED_OPEN_SL_ES_MODELS.
- // Allows an app to take control over which devices to exclude from using
- // the OpenSL ES audio output path
- // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
- @SuppressWarnings("NoSynchronizedMethodCheck")
- public static synchronized void setBlacklistDeviceForOpenSLESUsage(boolean enable) {
- blacklistDeviceForOpenSLESUsageIsOverridden = true;
- blacklistDeviceForOpenSLESUsage = enable;
- }
-
// Call these methods to override the default mono audio modes for the specified direction(s)
// (input and/or output).
// TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
@@ -204,19 +190,7 @@
return (audioManager.getMode() == AudioManager.MODE_IN_COMMUNICATION);
}
- @CalledByNative
- private static boolean isDeviceBlacklistedForOpenSLESUsage() {
- boolean blacklisted = blacklistDeviceForOpenSLESUsageIsOverridden
- ? blacklistDeviceForOpenSLESUsage
- : WebRtcAudioUtils.deviceIsBlacklistedForOpenSLESUsage();
- if (blacklisted) {
- Logging.d(TAG, Build.MODEL + " is blacklisted for OpenSL ES usage!");
- }
- return blacklisted;
- }
-
// Returns true if low-latency audio output is supported.
- @CalledByNative
public static boolean isLowLatencyOutputSupported(Context context) {
return context.getPackageManager().hasSystemFeature(PackageManager.FEATURE_AUDIO_LOW_LATENCY);
}
@@ -224,7 +198,6 @@
// Returns true if low-latency audio input is supported.
// TODO(henrika): remove the hardcoded false return value when OpenSL ES
// input performance has been evaluated and tested more.
- @CalledByNative
public static boolean isLowLatencyInputSupported(Context context) {
// TODO(henrika): investigate if some sort of device list is needed here
// as well. The NDK doc states that: "As of API level 21, lower latency
diff --git a/sdk/android/src/java/org/webrtc/audio/WebRtcAudioTrack.java b/sdk/android/src/java/org/webrtc/audio/WebRtcAudioTrack.java
index d3304ce..5a72300 100644
--- a/sdk/android/src/java/org/webrtc/audio/WebRtcAudioTrack.java
+++ b/sdk/android/src/java/org/webrtc/audio/WebRtcAudioTrack.java
@@ -193,7 +193,7 @@
@CalledByNative
WebRtcAudioTrack(long nativeAudioTrack) {
- threadChecker.checkIsOnValidThread();
+ threadChecker.detachThread();
Logging.d(TAG, "ctor" + WebRtcAudioUtils.getThreadInfo());
this.nativeAudioTrack = nativeAudioTrack;
audioManager =
diff --git a/sdk/android/src/java/org/webrtc/audio/WebRtcAudioUtils.java b/sdk/android/src/java/org/webrtc/audio/WebRtcAudioUtils.java
index 0693922..2be436a 100644
--- a/sdk/android/src/java/org/webrtc/audio/WebRtcAudioUtils.java
+++ b/sdk/android/src/java/org/webrtc/audio/WebRtcAudioUtils.java
@@ -35,16 +35,6 @@
final class WebRtcAudioUtils {
private static final String TAG = "WebRtcAudioUtils";
- // List of devices where we have seen issues (e.g. bad audio quality) using
- // the low latency output mode in combination with OpenSL ES.
- // The device name is given by Build.MODEL.
- private static final String[] BLACKLISTED_OPEN_SL_ES_MODELS = new String[] {
- // It is recommended to maintain a list of blacklisted models outside
- // this package and instead call
- // WebRtcAudioManager.setBlacklistDeviceForOpenSLESUsage(true)
- // from the client for devices where OpenSL ES shall be disabled.
- };
-
// List of devices where it has been verified that the built-in effect
// bad and where it makes sense to avoid using it and instead rely on the
// native WebRTC version instead. The device name is given by Build.MODEL.
@@ -207,12 +197,6 @@
return Build.HARDWARE.equals("goldfish") && Build.BRAND.startsWith("generic_");
}
- // Returns true if the device is blacklisted for OpenSL ES usage.
- public static boolean deviceIsBlacklistedForOpenSLESUsage() {
- List<String> blackListedModels = Arrays.asList(BLACKLISTED_OPEN_SL_ES_MODELS);
- return blackListedModels.contains(Build.MODEL);
- }
-
// Information about the current build, taken from system properties.
static void logDeviceInfo(String tag) {
Logging.d(tag,
diff --git a/sdk/android/src/jni/audio_device/aaudio_player.h b/sdk/android/src/jni/audio_device/aaudio_player.h
index c23081a..0770b46 100644
--- a/sdk/android/src/jni/audio_device/aaudio_player.h
+++ b/sdk/android/src/jni/audio_device/aaudio_player.h
@@ -22,6 +22,7 @@
#include "rtc_base/thread_annotations.h"
#include "rtc_base/thread_checker.h"
#include "sdk/android/src/jni/audio_device/aaudio_wrapper.h"
+#include "sdk/android/src/jni/audio_device/audio_device_module.h"
namespace webrtc {
@@ -30,8 +31,6 @@
namespace android_adm {
-class AudioManager;
-
// Implements low-latency 16-bit mono PCM audio output support for Android
// using the C based AAudio API.
//
@@ -52,30 +51,31 @@
// where the internal AAudio buffer can be increased when needed. It will
// reduce the risk of underruns (~glitches) at the expense of an increased
// latency.
-class AAudioPlayer final : public AAudioObserverInterface,
+class AAudioPlayer final : public AudioOutput,
+ public AAudioObserverInterface,
public rtc::MessageHandler {
public:
explicit AAudioPlayer(AudioManager* audio_manager);
~AAudioPlayer();
- int Init();
- int Terminate();
+ int Init() override;
+ int Terminate() override;
- int InitPlayout();
- bool PlayoutIsInitialized() const;
+ int InitPlayout() override;
+ bool PlayoutIsInitialized() const override;
- int StartPlayout();
- int StopPlayout();
- bool Playing() const;
+ int StartPlayout() override;
+ int StopPlayout() override;
+ bool Playing() const override;
- void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
+ void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
// Not implemented in AAudio.
- bool SpeakerVolumeIsAvailable();
- int SetSpeakerVolume(uint32_t volume);
- rtc::Optional<uint32_t> SpeakerVolume() const;
- rtc::Optional<uint32_t> MaxSpeakerVolume() const;
- rtc::Optional<uint32_t> MinSpeakerVolume() const;
+ bool SpeakerVolumeIsAvailable() override;
+ int SetSpeakerVolume(uint32_t volume) override;
+ rtc::Optional<uint32_t> SpeakerVolume() const override;
+ rtc::Optional<uint32_t> MaxSpeakerVolume() const override;
+ rtc::Optional<uint32_t> MinSpeakerVolume() const override;
protected:
// AAudioObserverInterface implementation.
diff --git a/sdk/android/src/jni/audio_device/aaudio_recorder.h b/sdk/android/src/jni/audio_device/aaudio_recorder.h
index 9924b75..a424c47 100644
--- a/sdk/android/src/jni/audio_device/aaudio_recorder.h
+++ b/sdk/android/src/jni/audio_device/aaudio_recorder.h
@@ -20,6 +20,7 @@
#include "rtc_base/thread.h"
#include "rtc_base/thread_checker.h"
#include "sdk/android/src/jni/audio_device/aaudio_wrapper.h"
+#include "sdk/android/src/jni/audio_device/audio_device_module.h"
namespace webrtc {
@@ -28,8 +29,6 @@
namespace android_adm {
-class AudioManager;
-
// Implements low-latency 16-bit mono PCM audio input support for Android
// using the C based AAudio API.
//
@@ -44,30 +43,29 @@
//
// TODO(henrika): add comments about device changes and adaptive buffer
// management.
-class AAudioRecorder : public AAudioObserverInterface,
+class AAudioRecorder : public AudioInput,
+ public AAudioObserverInterface,
public rtc::MessageHandler {
public:
explicit AAudioRecorder(AudioManager* audio_manager);
~AAudioRecorder();
- int Init();
- int Terminate();
+ int Init() override;
+ int Terminate() override;
- int InitRecording();
- bool RecordingIsInitialized() const { return initialized_; }
+ int InitRecording() override;
+ bool RecordingIsInitialized() const override { return initialized_; }
- int StartRecording();
- int StopRecording();
- bool Recording() const { return recording_; }
+ int StartRecording() override;
+ int StopRecording() override;
+ bool Recording() const override { return recording_; }
- void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
-
- double latency_millis() const { return latency_millis_; }
+ void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
// TODO(henrika): add support using AAudio APIs when available.
- int EnableBuiltInAEC(bool enable);
- int EnableBuiltInAGC(bool enable);
- int EnableBuiltInNS(bool enable);
+ int EnableBuiltInAEC(bool enable) override;
+ int EnableBuiltInAGC(bool enable) override;
+ int EnableBuiltInNS(bool enable) override;
protected:
// AAudioObserverInterface implementation.
diff --git a/sdk/android/src/jni/audio_device/audio_device_template_android.h b/sdk/android/src/jni/audio_device/audio_device_module.cc
similarity index 89%
rename from sdk/android/src/jni/audio_device/audio_device_template_android.h
rename to sdk/android/src/jni/audio_device/audio_device_module.cc
index 938a57b..e57d7c6 100644
--- a/sdk/android/src/jni/audio_device/audio_device_template_android.h
+++ b/sdk/android/src/jni/audio_device/audio_device_module.cc
@@ -8,16 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef SDK_ANDROID_SRC_JNI_AUDIO_DEVICE_AUDIO_DEVICE_TEMPLATE_ANDROID_H_
-#define SDK_ANDROID_SRC_JNI_AUDIO_DEVICE_AUDIO_DEVICE_TEMPLATE_ANDROID_H_
+#include "sdk/android/src/jni/audio_device/audio_device_module.h"
-#include <memory>
+#include <utility>
-#include "modules/audio_device/audio_device_buffer.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
+#include "rtc_base/refcountedobject.h"
#include "rtc_base/thread_checker.h"
-#include "sdk/android/src/jni/audio_device/audio_manager.h"
#include "system_wrappers/include/metrics.h"
#define CHECKinitialized_() \
@@ -38,6 +36,8 @@
namespace android_adm {
+namespace {
+
// InputType/OutputType can be any class that implements the capturing/rendering
// part of the AudioDeviceGeneric API.
// Construction and destruction must be done on one and the same thread. Each
@@ -47,9 +47,7 @@
// It is possible to call the two static methods (SetAndroidAudioDeviceObjects
// and ClearAndroidAudioDeviceObjects) from a different thread but both will
// RTC_CHECK that the calling thread is attached to a Java VM.
-
-template <class InputType, class OutputType>
-class AudioDeviceTemplateAndroid : public AudioDeviceModule {
+class AndroidAudioDeviceModule : public AudioDeviceModule {
public:
// For use with UMA logging. Must be kept in sync with histograms.xml in
// Chrome, located at
@@ -62,17 +60,23 @@
NUM_STATUSES = 4
};
- AudioDeviceTemplateAndroid(JNIEnv* env,
- const JavaParamRef<jobject>& application_context,
- AudioDeviceModule::AudioLayer audio_layer)
+ AndroidAudioDeviceModule(AudioDeviceModule::AudioLayer audio_layer,
+ std::unique_ptr<AudioManager> audio_manager,
+ std::unique_ptr<AudioInput> audio_input,
+ std::unique_ptr<AudioOutput> audio_output)
: audio_layer_(audio_layer),
- audio_manager_(env, audio_layer, application_context),
+ audio_manager_(std::move(audio_manager)),
+ input_(std::move(audio_input)),
+ output_(std::move(audio_output)),
initialized_(false) {
+ RTC_CHECK(input_);
+ RTC_CHECK(output_);
+ RTC_CHECK(audio_manager_);
RTC_LOG(INFO) << __FUNCTION__;
thread_checker_.DetachFromThread();
}
- virtual ~AudioDeviceTemplateAndroid() { RTC_LOG(INFO) << __FUNCTION__; }
+ virtual ~AndroidAudioDeviceModule() { RTC_LOG(INFO) << __FUNCTION__; }
int32_t ActiveAudioLayer(
AudioDeviceModule::AudioLayer* audioLayer) const override {
@@ -89,22 +93,20 @@
int32_t Init() override {
RTC_LOG(INFO) << __FUNCTION__;
RTC_DCHECK(thread_checker_.CalledOnValidThread());
- output_ = rtc::MakeUnique<OutputType>(&audio_manager_);
- input_ = rtc::MakeUnique<InputType>(&audio_manager_);
audio_device_buffer_ = rtc::MakeUnique<AudioDeviceBuffer>();
AttachAudioBuffer();
if (initialized_) {
return 0;
}
InitStatus status;
- if (!audio_manager_.Init()) {
+ if (!audio_manager_->Init()) {
status = InitStatus::OTHER_ERROR;
} else if (output_->Init() != 0) {
- audio_manager_.Close();
+ audio_manager_->Close();
status = InitStatus::PLAYOUT_ERROR;
} else if (input_->Init() != 0) {
output_->Terminate();
- audio_manager_.Close();
+ audio_manager_->Close();
status = InitStatus::RECORDING_ERROR;
} else {
initialized_ = true;
@@ -127,7 +129,7 @@
RTC_DCHECK(thread_checker_.CalledOnValidThread());
int32_t err = input_->Terminate();
err |= output_->Terminate();
- err |= !audio_manager_.Close();
+ err |= !audio_manager_->Close();
initialized_ = false;
RTC_DCHECK_EQ(err, 0);
return err;
@@ -250,7 +252,7 @@
return 0;
}
audio_device_buffer_->StartPlayout();
- if (!audio_manager_.IsCommunicationModeEnabled()) {
+ if (!audio_manager_->IsCommunicationModeEnabled()) {
RTC_LOG(WARNING)
<< "The application should use MODE_IN_COMMUNICATION audio mode!";
}
@@ -288,7 +290,7 @@
if (Recording()) {
return 0;
}
- if (!audio_manager_.IsCommunicationModeEnabled()) {
+ if (!audio_manager_->IsCommunicationModeEnabled()) {
RTC_LOG(WARNING)
<< "The application should use MODE_IN_COMMUNICATION audio mode!";
}
@@ -474,7 +476,7 @@
int32_t StereoPlayoutIsAvailable(bool* available) const override {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
- *available = audio_manager_.IsStereoPlayoutSupported();
+ *available = audio_manager_->IsStereoPlayoutSupported();
RTC_LOG(INFO) << "output: " << *available;
return 0;
}
@@ -486,7 +488,7 @@
RTC_LOG(WARNING) << "recording in stereo is not supported";
return -1;
}
- bool available = audio_manager_.IsStereoPlayoutSupported();
+ bool available = audio_manager_->IsStereoPlayoutSupported();
// Android does not support changes between mono and stero on the fly.
// Instead, the native audio layer is configured via the audio manager
// to either support mono or stereo. It is allowed to call this method
@@ -506,7 +508,7 @@
int32_t StereoPlayout(bool* enabled) const override {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
- *enabled = audio_manager_.IsStereoPlayoutSupported();
+ *enabled = audio_manager_->IsStereoPlayoutSupported();
RTC_LOG(INFO) << "output: " << *enabled;
return 0;
}
@@ -514,7 +516,7 @@
int32_t StereoRecordingIsAvailable(bool* available) const override {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
- *available = audio_manager_.IsStereoRecordSupported();
+ *available = audio_manager_->IsStereoRecordSupported();
RTC_LOG(INFO) << "output: " << *available;
return 0;
}
@@ -526,7 +528,7 @@
RTC_LOG(WARNING) << "recording in stereo is not supported";
return -1;
}
- bool available = audio_manager_.IsStereoRecordSupported();
+ bool available = audio_manager_->IsStereoRecordSupported();
// Android does not support changes between mono and stero on the fly.
// Instead, the native audio layer is configured via the audio manager
// to either support mono or stereo. It is allowed to call this method
@@ -546,7 +548,7 @@
int32_t StereoRecording(bool* enabled) const override {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
- *enabled = audio_manager_.IsStereoRecordSupported();
+ *enabled = audio_manager_->IsStereoRecordSupported();
RTC_LOG(INFO) << "output: " << *enabled;
return 0;
}
@@ -554,7 +556,7 @@
int32_t PlayoutDelay(uint16_t* delay_ms) const override {
CHECKinitialized_();
// Best guess we can do is to use half of the estimated total delay.
- *delay_ms = audio_manager_.GetDelayEstimateInMilliseconds() / 2;
+ *delay_ms = audio_manager_->GetDelayEstimateInMilliseconds() / 2;
RTC_DCHECK_GT(*delay_ms, 0);
return 0;
}
@@ -574,7 +576,7 @@
bool BuiltInAECIsAvailable() const override {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized__BOOL();
- bool isAvailable = audio_manager_.IsAcousticEchoCancelerSupported();
+ bool isAvailable = audio_manager_->IsAcousticEchoCancelerSupported();
RTC_LOG(INFO) << "output: " << isAvailable;
return isAvailable;
}
@@ -598,7 +600,7 @@
bool BuiltInNSIsAvailable() const override {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized__BOOL();
- bool isAvailable = audio_manager_.IsNoiseSuppressorSupported();
+ bool isAvailable = audio_manager_->IsNoiseSuppressorSupported();
RTC_LOG(INFO) << "output: " << isAvailable;
return isAvailable;
}
@@ -644,17 +646,26 @@
rtc::ThreadChecker thread_checker_;
const AudioDeviceModule::AudioLayer audio_layer_;
-
- AudioManager audio_manager_;
- std::unique_ptr<OutputType> output_;
- std::unique_ptr<InputType> input_;
+ const std::unique_ptr<AudioManager> audio_manager_;
+ const std::unique_ptr<AudioInput> input_;
+ const std::unique_ptr<AudioOutput> output_;
std::unique_ptr<AudioDeviceBuffer> audio_device_buffer_;
bool initialized_;
};
+} // namespace
+
+rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceModuleFromInputAndOutput(
+ AudioDeviceModule::AudioLayer audio_layer,
+ std::unique_ptr<AudioManager> audio_manager,
+ std::unique_ptr<AudioInput> audio_input,
+ std::unique_ptr<AudioOutput> audio_output) {
+ return new rtc::RefCountedObject<AndroidAudioDeviceModule>(
+ audio_layer, std::move(audio_manager), std::move(audio_input),
+ std::move(audio_output));
+}
+
} // namespace android_adm
} // namespace webrtc
-
-#endif // SDK_ANDROID_SRC_JNI_AUDIO_DEVICE_AUDIO_DEVICE_TEMPLATE_ANDROID_H_
diff --git a/sdk/android/src/jni/audio_device/audio_device_module.h b/sdk/android/src/jni/audio_device/audio_device_module.h
new file mode 100644
index 0000000..b11beb4
--- /dev/null
+++ b/sdk/android/src/jni/audio_device/audio_device_module.h
@@ -0,0 +1,77 @@
+/*
+ * 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.
+ */
+
+#ifndef SDK_ANDROID_SRC_JNI_AUDIO_DEVICE_AUDIO_DEVICE_MODULE_H_
+#define SDK_ANDROID_SRC_JNI_AUDIO_DEVICE_AUDIO_DEVICE_MODULE_H_
+
+#include <memory>
+
+#include "api/optional.h"
+#include "modules/audio_device/audio_device_buffer.h"
+#include "sdk/android/native_api/jni/scoped_java_ref.h"
+#include "sdk/android/src/jni/audio_device/audio_manager.h"
+
+namespace webrtc {
+
+namespace android_adm {
+
+class AudioManager;
+
+class AudioInput {
+ public:
+ virtual ~AudioInput() {}
+
+ virtual int32_t Init() = 0;
+ virtual int32_t Terminate() = 0;
+
+ virtual int32_t InitRecording() = 0;
+ virtual bool RecordingIsInitialized() const = 0;
+
+ virtual int32_t StartRecording() = 0;
+ virtual int32_t StopRecording() = 0;
+ virtual bool Recording() const = 0;
+
+ virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) = 0;
+
+ virtual int32_t EnableBuiltInAEC(bool enable) = 0;
+ virtual int32_t EnableBuiltInAGC(bool enable) = 0;
+ virtual int32_t EnableBuiltInNS(bool enable) = 0;
+};
+
+class AudioOutput {
+ public:
+ virtual ~AudioOutput() {}
+
+ virtual int32_t Init() = 0;
+ virtual int32_t Terminate() = 0;
+ virtual int32_t InitPlayout() = 0;
+ virtual bool PlayoutIsInitialized() const = 0;
+ virtual int32_t StartPlayout() = 0;
+ virtual int32_t StopPlayout() = 0;
+ virtual bool Playing() const = 0;
+ virtual bool SpeakerVolumeIsAvailable() = 0;
+ virtual int SetSpeakerVolume(uint32_t volume) = 0;
+ virtual rtc::Optional<uint32_t> SpeakerVolume() const = 0;
+ virtual rtc::Optional<uint32_t> MaxSpeakerVolume() const = 0;
+ virtual rtc::Optional<uint32_t> MinSpeakerVolume() const = 0;
+ virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) = 0;
+};
+
+rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceModuleFromInputAndOutput(
+ AudioDeviceModule::AudioLayer audio_layer,
+ std::unique_ptr<AudioManager> audio_manager,
+ std::unique_ptr<AudioInput> audio_input,
+ std::unique_ptr<AudioOutput> audio_output);
+
+} // namespace android_adm
+
+} // namespace webrtc
+
+#endif // SDK_ANDROID_SRC_JNI_AUDIO_DEVICE_AUDIO_DEVICE_MODULE_H_
diff --git a/sdk/android/src/jni/audio_device/audio_manager.cc b/sdk/android/src/jni/audio_device/audio_manager.cc
index 8e09d8a..41943dc 100644
--- a/sdk/android/src/jni/audio_device/audio_manager.cc
+++ b/sdk/android/src/jni/audio_device/audio_manager.cc
@@ -28,7 +28,7 @@
#include "sdk/android/src/jni/audio_device/aaudio_player.h"
#include "sdk/android/src/jni/audio_device/aaudio_recorder.h"
#endif
-#include "sdk/android/src/jni/audio_device/audio_device_template_android.h"
+#include "sdk/android/src/jni/audio_device/audio_device_module.h"
#include "sdk/android/src/jni/audio_device/audio_manager.h"
#include "sdk/android/src/jni/audio_device/audio_record_jni.h"
#include "sdk/android/src/jni/audio_device/audio_track_jni.h"
@@ -39,64 +39,6 @@
namespace android_adm {
-#if defined(AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO)
-rtc::scoped_refptr<AudioDeviceModule>
-AudioManager::CreateAAudioAudioDeviceModule(
- JNIEnv* env,
- const JavaParamRef<jobject>& application_context) {
- RTC_LOG(INFO) << __FUNCTION__;
- return new rtc::RefCountedObject<android_adm::AudioDeviceTemplateAndroid<
- android_adm::AAudioRecorder, android_adm::AAudioPlayer>>(
- env, application_context, AudioDeviceModule::kAndroidAAudioAudio);
-}
-#endif
-
-rtc::scoped_refptr<AudioDeviceModule> AudioManager::CreateAudioDeviceModule(
- JNIEnv* env,
- const JavaParamRef<jobject>& application_context) {
- const bool use_opensles_output =
- !Java_WebRtcAudioManager_isDeviceBlacklistedForOpenSLESUsage(env) &&
- Java_WebRtcAudioManager_isLowLatencyOutputSupported(env,
- application_context);
- const bool use_opensles_input =
- use_opensles_output && Java_WebRtcAudioManager_isLowLatencyInputSupported(
- env, application_context);
- return CreateAudioDeviceModule(env, application_context, use_opensles_input,
- use_opensles_output);
-}
-
-rtc::scoped_refptr<AudioDeviceModule> AudioManager::CreateAudioDeviceModule(
- JNIEnv* env,
- const JavaParamRef<jobject>& application_context,
- bool use_opensles_input,
- bool use_opensles_output) {
- RTC_LOG(INFO) << __FUNCTION__;
-
- if (use_opensles_output) {
- if (use_opensles_input) {
- // Use OpenSL ES for both playout and recording.
- return new rtc::RefCountedObject<android_adm::AudioDeviceTemplateAndroid<
- android_adm::OpenSLESRecorder, android_adm::OpenSLESPlayer>>(
- env, application_context, AudioDeviceModule::kAndroidOpenSLESAudio);
- } else {
- // Use OpenSL ES for output and AudioRecord API for input. This
- // combination provides low-latency output audio and at the same
- // time support for HW AEC using the AudioRecord Java API.
- return new rtc::RefCountedObject<android_adm::AudioDeviceTemplateAndroid<
- android_adm::AudioRecordJni, android_adm::OpenSLESPlayer>>(
- env, application_context,
- AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio);
- }
- } else {
- RTC_DCHECK(!use_opensles_input)
- << "Combination of OpenSLES input and Java-based output not supported";
- // Use Java-based audio in both directions.
- return new rtc::RefCountedObject<android_adm::AudioDeviceTemplateAndroid<
- android_adm::AudioRecordJni, android_adm::AudioTrackJni>>(
- env, application_context, AudioDeviceModule::kAndroidJavaAudio);
- }
-}
-
// AudioManager implementation
AudioManager::AudioManager(JNIEnv* env,
AudioDeviceModule::AudioLayer audio_layer,
@@ -120,6 +62,8 @@
static_cast<size_t>(output_buffer_size));
record_parameters_.reset(sample_rate, static_cast<size_t>(input_channels),
static_cast<size_t>(input_buffer_size));
+ RTC_CHECK(playout_parameters_.is_valid());
+ RTC_CHECK(record_parameters_.is_valid());
thread_checker_.DetachFromThread();
}
@@ -217,12 +161,10 @@
}
bool AudioManager::IsStereoPlayoutSupported() const {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
return (playout_parameters_.channels() == 2);
}
bool AudioManager::IsStereoRecordSupported() const {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
return (record_parameters_.channels() == 2);
}
@@ -234,13 +176,11 @@
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_;
}
diff --git a/sdk/android/src/jni/audio_device/audio_manager.h b/sdk/android/src/jni/audio_device/audio_manager.h
index 201b363..5f32e2a 100644
--- a/sdk/android/src/jni/audio_device/audio_manager.h
+++ b/sdk/android/src/jni/audio_device/audio_manager.h
@@ -33,24 +33,6 @@
// unless Init() is called.
class AudioManager {
public:
-#if defined(AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO)
- static rtc::scoped_refptr<AudioDeviceModule> CreateAAudioAudioDeviceModule(
- JNIEnv* env,
- const JavaParamRef<jobject>& application_context);
-#endif
-
- static rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceModule(
- JNIEnv* env,
- const JavaParamRef<jobject>& application_context,
- bool use_opensles_input,
- bool use_opensles_output);
-
- // This function has internal logic checking if OpenSLES is blacklisted and
- // whether it's supported.
- static rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceModule(
- JNIEnv* env,
- const JavaParamRef<jobject>& application_context);
-
AudioManager(JNIEnv* env,
AudioDeviceModule::AudioLayer audio_layer,
const JavaParamRef<jobject>& application_context);
diff --git a/sdk/android/src/jni/audio_device/audio_record_jni.cc b/sdk/android/src/jni/audio_device/audio_record_jni.cc
index 08c54a9..93675f7 100644
--- a/sdk/android/src/jni/audio_device/audio_record_jni.cc
+++ b/sdk/android/src/jni/audio_device/audio_record_jni.cc
@@ -45,13 +45,13 @@
const std::string histogram_name_;
int64_t start_time_ms_;
};
+
} // namespace
// AudioRecordJni implementation.
AudioRecordJni::AudioRecordJni(AudioManager* audio_manager)
- : env_(AttachCurrentThreadIfNeeded()),
- j_audio_record_(
- Java_WebRtcAudioRecord_Constructor(env_,
+ : j_audio_record_(
+ Java_WebRtcAudioRecord_Constructor(AttachCurrentThreadIfNeeded(),
jni::jlongFromPointer(this))),
audio_manager_(audio_manager),
audio_parameters_(audio_manager->GetRecordAudioParameters()),
@@ -64,8 +64,9 @@
audio_device_buffer_(nullptr) {
RTC_LOG(INFO) << "ctor";
RTC_DCHECK(audio_parameters_.is_valid());
- // Detach from this thread since we want to use the checker to verify calls
- // from the Java based audio thread.
+ // Detach from this thread since construction is allowed to happen on a
+ // different thread.
+ thread_checker_.DetachFromThread();
thread_checker_java_.DetachFromThread();
}
@@ -77,6 +78,7 @@
int32_t AudioRecordJni::Init() {
RTC_LOG(INFO) << "Init";
+ env_ = AttachCurrentThreadIfNeeded();
RTC_DCHECK(thread_checker_.CalledOnValidThread());
return 0;
}
diff --git a/sdk/android/src/jni/audio_device/audio_record_jni.h b/sdk/android/src/jni/audio_device/audio_record_jni.h
index a8a3177..40cfa0b 100644
--- a/sdk/android/src/jni/audio_device/audio_record_jni.h
+++ b/sdk/android/src/jni/audio_device/audio_record_jni.h
@@ -17,6 +17,7 @@
#include "modules/audio_device/audio_device_buffer.h"
#include "modules/audio_device/include/audio_device_defines.h"
#include "rtc_base/thread_checker.h"
+#include "sdk/android/src/jni/audio_device/audio_device_module.h"
#include "sdk/android/src/jni/audio_device/audio_manager.h"
namespace webrtc {
@@ -41,26 +42,26 @@
// This class uses AttachCurrentThreadIfNeeded to attach to a Java VM if needed
// and detach when the object goes out of scope. Additional thread checking
// guarantees that no other (possibly non attached) thread is used.
-class AudioRecordJni {
+class AudioRecordJni : public AudioInput {
public:
explicit AudioRecordJni(AudioManager* audio_manager);
- ~AudioRecordJni();
+ ~AudioRecordJni() override;
- int32_t Init();
- int32_t Terminate();
+ int32_t Init() override;
+ int32_t Terminate() override;
- int32_t InitRecording();
- bool RecordingIsInitialized() const { return initialized_; }
+ int32_t InitRecording() override;
+ bool RecordingIsInitialized() const override { return initialized_; }
- int32_t StartRecording();
- int32_t StopRecording();
- bool Recording() const { return recording_; }
+ int32_t StartRecording() override;
+ int32_t StopRecording() override;
+ bool Recording() const override { return recording_; }
- void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
+ void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
- int32_t EnableBuiltInAEC(bool enable);
- int32_t EnableBuiltInAGC(bool enable);
- int32_t EnableBuiltInNS(bool enable);
+ int32_t EnableBuiltInAEC(bool enable) override;
+ int32_t EnableBuiltInAGC(bool enable) override;
+ int32_t EnableBuiltInNS(bool enable) override;
// Called from Java side so we can cache the address of the Java-manged
// |byte_buffer| in |direct_buffer_address_|. The size of the buffer
@@ -90,7 +91,7 @@
rtc::ThreadChecker thread_checker_java_;
// Wraps the Java specific parts of the AudioRecordJni class.
- JNIEnv* const env_;
+ JNIEnv* env_ = nullptr;
ScopedJavaGlobalRef<jobject> j_audio_record_;
// Raw pointer to the audio manger.
diff --git a/sdk/android/src/jni/audio_device/audio_track_jni.cc b/sdk/android/src/jni/audio_device/audio_track_jni.cc
index 48aab44..c5ab6b3 100644
--- a/sdk/android/src/jni/audio_device/audio_track_jni.cc
+++ b/sdk/android/src/jni/audio_device/audio_track_jni.cc
@@ -27,9 +27,9 @@
// TODO(henrika): possible extend usage of AudioManager and add it as member.
AudioTrackJni::AudioTrackJni(AudioManager* audio_manager)
- : env_(AttachCurrentThreadIfNeeded()),
- j_audio_track_(
- Java_WebRtcAudioTrack_Constructor(env_, jni::jlongFromPointer(this))),
+ : j_audio_track_(
+ Java_WebRtcAudioTrack_Constructor(AttachCurrentThreadIfNeeded(),
+ jni::jlongFromPointer(this))),
audio_parameters_(audio_manager->GetPlayoutAudioParameters()),
direct_buffer_address_(nullptr),
direct_buffer_capacity_in_bytes_(0),
@@ -39,8 +39,9 @@
audio_device_buffer_(nullptr) {
RTC_LOG(INFO) << "ctor";
RTC_DCHECK(audio_parameters_.is_valid());
- // Detach from this thread since we want to use the checker to verify calls
- // from the Java based audio thread.
+ // Detach from this thread since construction is allowed to happen on a
+ // different thread.
+ thread_checker_.DetachFromThread();
thread_checker_java_.DetachFromThread();
}
@@ -52,6 +53,7 @@
int32_t AudioTrackJni::Init() {
RTC_LOG(INFO) << "Init";
+ env_ = AttachCurrentThreadIfNeeded();
RTC_DCHECK(thread_checker_.CalledOnValidThread());
return 0;
}
diff --git a/sdk/android/src/jni/audio_device/audio_track_jni.h b/sdk/android/src/jni/audio_device/audio_track_jni.h
index c73bd70..d56f30c 100644
--- a/sdk/android/src/jni/audio_device/audio_track_jni.h
+++ b/sdk/android/src/jni/audio_device/audio_track_jni.h
@@ -19,6 +19,7 @@
#include "modules/audio_device/include/audio_device_defines.h"
#include "rtc_base/thread_checker.h"
#include "sdk/android/src/jni/audio_device/audio_common.h"
+#include "sdk/android/src/jni/audio_device/audio_device_module.h"
#include "sdk/android/src/jni/audio_device/audio_manager.h"
namespace webrtc {
@@ -38,28 +39,28 @@
// This class uses AttachCurrentThreadIfNeeded to attach to a Java VM if needed
// and detach when the object goes out of scope. Additional thread checking
// guarantees that no other (possibly non attached) thread is used.
-class AudioTrackJni {
+class AudioTrackJni : public AudioOutput {
public:
explicit AudioTrackJni(AudioManager* audio_manager);
- ~AudioTrackJni();
+ ~AudioTrackJni() override;
- int32_t Init();
- int32_t Terminate();
+ int32_t Init() override;
+ int32_t Terminate() override;
- int32_t InitPlayout();
- bool PlayoutIsInitialized() const { return initialized_; }
+ int32_t InitPlayout() override;
+ bool PlayoutIsInitialized() const override { return initialized_; }
- int32_t StartPlayout();
- int32_t StopPlayout();
- bool Playing() const { return playing_; }
+ int32_t StartPlayout() override;
+ int32_t StopPlayout() override;
+ bool Playing() const override { return playing_; }
- bool SpeakerVolumeIsAvailable();
- int SetSpeakerVolume(uint32_t volume);
- rtc::Optional<uint32_t> SpeakerVolume() const;
- rtc::Optional<uint32_t> MaxSpeakerVolume() const;
- rtc::Optional<uint32_t> MinSpeakerVolume() const;
+ bool SpeakerVolumeIsAvailable() override;
+ int SetSpeakerVolume(uint32_t volume) override;
+ rtc::Optional<uint32_t> SpeakerVolume() const override;
+ rtc::Optional<uint32_t> MaxSpeakerVolume() const override;
+ rtc::Optional<uint32_t> MinSpeakerVolume() const override;
- void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
+ void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
// Called from Java side so we can cache the address of the Java-manged
// |byte_buffer| in |direct_buffer_address_|. The size of the buffer
@@ -86,7 +87,7 @@
rtc::ThreadChecker thread_checker_java_;
// Wraps the Java specific parts of the AudioTrackJni class.
- JNIEnv* const env_;
+ JNIEnv* env_ = nullptr;
ScopedJavaGlobalRef<jobject> j_audio_track_;
// Contains audio parameters provided to this class at construction by the
diff --git a/sdk/android/src/jni/audio_device/opensles_player.h b/sdk/android/src/jni/audio_device/opensles_player.h
index 70629ea..34d3c53 100644
--- a/sdk/android/src/jni/audio_device/opensles_player.h
+++ b/sdk/android/src/jni/audio_device/opensles_player.h
@@ -22,6 +22,7 @@
#include "modules/audio_device/include/audio_device_defines.h"
#include "rtc_base/thread_checker.h"
#include "sdk/android/src/jni/audio_device/audio_common.h"
+#include "sdk/android/src/jni/audio_device/audio_device_module.h"
#include "sdk/android/src/jni/audio_device/audio_manager.h"
#include "sdk/android/src/jni/audio_device/opensles_common.h"
@@ -49,7 +50,7 @@
// If the device doesn't claim this feature but supports API level 9 (Android
// platform version 2.3) or later, then we can still use the OpenSL ES APIs but
// the output latency may be higher.
-class OpenSLESPlayer {
+class OpenSLESPlayer : public AudioOutput {
public:
// Beginning with API level 17 (Android 4.2), a buffer count of 2 or more is
// required for lower latency. Beginning with API level 18 (Android 4.3), a
@@ -60,25 +61,25 @@
static const int kNumOfOpenSLESBuffers = 2;
explicit OpenSLESPlayer(AudioManager* audio_manager);
- ~OpenSLESPlayer();
+ ~OpenSLESPlayer() override;
- int Init();
- int Terminate();
+ int Init() override;
+ int Terminate() override;
- int InitPlayout();
- bool PlayoutIsInitialized() const { return initialized_; }
+ int InitPlayout() override;
+ bool PlayoutIsInitialized() const override { return initialized_; }
- int StartPlayout();
- int StopPlayout();
- bool Playing() const { return playing_; }
+ int StartPlayout() override;
+ int StopPlayout() override;
+ bool Playing() const override { return playing_; }
- bool SpeakerVolumeIsAvailable();
- int SetSpeakerVolume(uint32_t volume);
- rtc::Optional<uint32_t> SpeakerVolume() const;
- rtc::Optional<uint32_t> MaxSpeakerVolume() const;
- rtc::Optional<uint32_t> MinSpeakerVolume() const;
+ bool SpeakerVolumeIsAvailable() override;
+ int SetSpeakerVolume(uint32_t volume) override;
+ rtc::Optional<uint32_t> SpeakerVolume() const override;
+ rtc::Optional<uint32_t> MaxSpeakerVolume() const override;
+ rtc::Optional<uint32_t> MinSpeakerVolume() const override;
- void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
+ void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
private:
// These callback methods are called when data is required for playout.
diff --git a/sdk/android/src/jni/audio_device/opensles_recorder.h b/sdk/android/src/jni/audio_device/opensles_recorder.h
index 556e578..81efcb8 100644
--- a/sdk/android/src/jni/audio_device/opensles_recorder.h
+++ b/sdk/android/src/jni/audio_device/opensles_recorder.h
@@ -22,6 +22,7 @@
#include "modules/audio_device/include/audio_device_defines.h"
#include "rtc_base/thread_checker.h"
#include "sdk/android/src/jni/audio_device/audio_common.h"
+#include "sdk/android/src/jni/audio_device/audio_device_module.h"
#include "sdk/android/src/jni/audio_device/audio_manager.h"
#include "sdk/android/src/jni/audio_device/opensles_common.h"
@@ -52,7 +53,7 @@
// for input effects preclude the lower latency path.
// See https://developer.android.com/ndk/guides/audio/opensl-prog-notes.html
// for more details.
-class OpenSLESRecorder {
+class OpenSLESRecorder : public AudioInput {
public:
// Beginning with API level 17 (Android 4.2), a buffer count of 2 or more is
// required for lower latency. Beginning with API level 18 (Android 4.3), a
@@ -63,24 +64,24 @@
static const int kNumOfOpenSLESBuffers = 2;
explicit OpenSLESRecorder(AudioManager* audio_manager);
- ~OpenSLESRecorder();
+ ~OpenSLESRecorder() override;
- int Init();
- int Terminate();
+ int Init() override;
+ int Terminate() override;
- int InitRecording();
- bool RecordingIsInitialized() const { return initialized_; }
+ int InitRecording() override;
+ bool RecordingIsInitialized() const override { return initialized_; }
- int StartRecording();
- int StopRecording();
- bool Recording() const { return recording_; }
+ int StartRecording() override;
+ int StopRecording() override;
+ bool Recording() const override { return recording_; }
- void AttachAudioBuffer(AudioDeviceBuffer* audio_buffer);
+ void AttachAudioBuffer(AudioDeviceBuffer* audio_buffer) override;
// TODO(henrika): add support using OpenSL ES APIs when available.
- int EnableBuiltInAEC(bool enable);
- int EnableBuiltInAGC(bool enable);
- int EnableBuiltInNS(bool enable);
+ int EnableBuiltInAEC(bool enable) override;
+ int EnableBuiltInAGC(bool enable) override;
+ int EnableBuiltInNS(bool enable) override;
private:
// Obtaines the SL Engine Interface from the existing global Engine object.
diff --git a/sdk/android/src/jni/pc/peerconnectionfactory.cc b/sdk/android/src/jni/pc/peerconnectionfactory.cc
index 62873bd..04874bb 100644
--- a/sdk/android/src/jni/pc/peerconnectionfactory.cc
+++ b/sdk/android/src/jni/pc/peerconnectionfactory.cc
@@ -27,6 +27,8 @@
#include "sdk/android/generated_peerconnection_jni/jni/PeerConnectionFactory_jni.h"
#include "sdk/android/native_api/jni/java_types.h"
#include "sdk/android/src/jni/audio_device/audio_manager.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/jni_helpers.h"
#include "sdk/android/src/jni/pc/androidnetworkmonitor.h"
#include "sdk/android/src/jni/pc/audio.h"
@@ -238,10 +240,23 @@
rtc::NetworkMonitorFactory::SetFactory(network_monitor_factory);
}
- rtc::scoped_refptr<AudioDeviceModule> adm =
- field_trial::IsEnabled(kExternalAndroidAudioDeviceFieldTrialName)
- ? android_adm::AudioManager::CreateAudioDeviceModule(jni, jcontext)
- : nullptr;
+ rtc::scoped_refptr<AudioDeviceModule> adm = nullptr;
+ if (field_trial::IsEnabled(kExternalAndroidAudioDeviceFieldTrialName)) {
+ // Only Java AudioDeviceModule is supported as an external ADM at the
+ // moment.
+ const AudioDeviceModule::AudioLayer audio_layer =
+ AudioDeviceModule::kAndroidJavaAudio;
+ auto audio_manager =
+ rtc::MakeUnique<android_adm::AudioManager>(jni, audio_layer, jcontext);
+ auto audio_input =
+ rtc::MakeUnique<android_adm::AudioRecordJni>(audio_manager.get());
+ auto audio_output =
+ rtc::MakeUnique<android_adm::AudioTrackJni>(audio_manager.get());
+ adm = CreateAudioDeviceModuleFromInputAndOutput(
+ audio_layer, std::move(audio_manager), std::move(audio_input),
+ std::move(audio_output));
+ }
+
rtc::scoped_refptr<AudioMixer> audio_mixer = nullptr;
std::unique_ptr<CallFactoryInterface> call_factory(CreateCallFactory());
std::unique_ptr<RtcEventLogFactoryInterface> rtc_event_log_factory(
