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 <[email protected]> Reviewed-by: Paulina Hensman <[email protected]> Cr-Commit-Position: refs/heads/master@{#22660}
darkGoldC · Mar 28, 2018 · 1a18e0a · 1a18e0a
1 parent 85eef49
commit 1a18e0a
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Showing 21 changed files with 321 additions and 286 deletions.
diff --git a/examples/androidapp/src/org/appspot/apprtc/PeerConnectionClient.java b/examples/androidapp/src/org/appspot/apprtc/PeerConnectionClient.java
@@ -595,10 +595,9 @@ void setupAudioDevice() {
     // 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
@@ -164,7 +164,8 @@ rtc_source_set("audio_device_jni") {
 
   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
@@ -24,10 +24,6 @@ public class AudioDeviceModule {
   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
@@ -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) {
+  return CreateAudioDeviceModuleTemplate<android_adm::AudioRecordJni,
+                                         android_adm::AudioTrackJni>(
+      AudioDeviceModule::kAndroidJavaAudio, env, application_context);
+}
+
+rtc::scoped_refptr<AudioDeviceModule> CreateOpenSLESAudioDeviceModule(
     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::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
@@ -23,11 +23,17 @@ rtc::scoped_refptr<AudioDeviceModule> CreateAAudioAudioDeviceModule(
     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
@@ -42,20 +42,6 @@ class WebRtcAudioManager {
   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.
@@ -204,27 +190,14 @@ private boolean isCommunicationModeEnabled() {
     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);
   }
 
   // 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
@@ -193,7 +193,7 @@ public void stopThread() {
 
   @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
@@ -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 @@ public static boolean runningOnEmulator() {
     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
@@ -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 @@ class FineAudioBuffer;
 
 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 @@ class AudioManager;
 // 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
@@ -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 @@ class AudioDeviceBuffer;
 
 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 @@ class AudioManager;
 //
 // 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 InitRecording();
-  bool RecordingIsInitialized() const { return initialized_; }
+  int Init() override;
+  int Terminate() override;
 
-  int StartRecording();
-  int StopRecording();
-  bool Recording() const { return recording_; }
+  int InitRecording() override;
+  bool RecordingIsInitialized() const override { return initialized_; }
 
-  void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
+  int StartRecording() override;
+  int StopRecording() override;
+  bool Recording() const override { return recording_; }
 
-  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.