OpenSL ES stability improvements.

This CL does two things:

1) Improves stability in the existing OpenSL ES implementation for devices that
supports OpenSL ES. The cost is a slight increase in latency since the focus here
has been on avoiding audio glitches.

2) Adds a new Java API to exclude usage of OpenSL ES to enable comparisons between
OpenSL ES and Java based audio backends.

BUG=b/22452539

Review URL: https://codereview.webrtc.org/1440623002

Cr-Commit-Position: refs/heads/master@{#10618}

webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java

@@ -33,11 +33,24 @@ import java.lang.Math;
 // recommended to always use AudioManager.MODE_IN_COMMUNICATION.
 // This class also adds support for output volume control of the
 // STREAM_VOICE_CALL-type stream.
-class WebRtcAudioManager {
+public class WebRtcAudioManager {
   private static final boolean DEBUG = false;
 
   private static final String TAG = "WebRtcAudioManager";
 
+  private static boolean blacklistDeviceForOpenSLESUsage = false;
+  private static boolean blacklistDeviceForOpenSLESUsageIsOverridden = false;
+
+  // Call this method to override the deault list of blacklisted devices
+  // specified in WebRtcAudioUtils.BLACKLISTED_OPEN_SL_ES_MODELS.
+  // Allows an app to take control over which devices to exlude from using
+  // the OpenSL ES audio output path
+  public static synchronized void setBlacklistDeviceForOpenSLESUsage(
+      boolean enable) {
+    blacklistDeviceForOpenSLESUsageIsOverridden = true;
+    blacklistDeviceForOpenSLESUsage = enable;
+  }
+
   // Default audio data format is PCM 16 bit per sample.
   // Guaranteed to be supported by all devices.
   private static final int BITS_PER_SAMPLE = 16;
@@ -110,7 +123,8 @@ class WebRtcAudioManager {
   }
 
    private boolean isDeviceBlacklistedForOpenSLESUsage() {
-    boolean blacklisted =
+    boolean blacklisted = blacklistDeviceForOpenSLESUsageIsOverridden ?
+        blacklistDeviceForOpenSLESUsage :
         WebRtcAudioUtils.deviceIsBlacklistedForOpenSLESUsage();
     if (blacklisted) {
       Logging.e(TAG, Build.MODEL + " is blacklisted for OpenSL ES usage!");

webrtc/modules/audio_device/android/opensles_player.cc

@@ -15,6 +15,7 @@
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/format_macros.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/modules/audio_device/android/audio_manager.h"
 #include "webrtc/modules/audio_device/fine_audio_buffer.h"
 
@@ -46,7 +47,8 @@ OpenSLESPlayer::OpenSLESPlayer(AudioManager* audio_manager)
       engine_(nullptr),
       player_(nullptr),
       simple_buffer_queue_(nullptr),
-      volume_(nullptr) {
+      volume_(nullptr),
+      last_play_time_(0) {
   ALOGD("ctor%s", GetThreadInfo().c_str());
   // Use native audio output parameters provided by the audio manager and
   // define the PCM format structure.
@@ -95,6 +97,7 @@ int OpenSLESPlayer::InitPlayout() {
   CreateMix();
   initialized_ = true;
   buffer_index_ = 0;
+  last_play_time_ = rtc::Time();
   return 0;
 }
 
@@ -233,7 +236,16 @@ void OpenSLESPlayer::AllocateDataBuffers() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   RTC_DCHECK(!simple_buffer_queue_);
   RTC_CHECK(audio_device_buffer_);
-  bytes_per_buffer_ = audio_parameters_.GetBytesPerBuffer();
+  // Don't use the lowest possible size as native buffer size. Instead,
+  // use 10ms to better match the frame size that WebRTC uses. It will result
+  // in a reduced risk for audio glitches and also in a more "clean" sequence
+  // of callbacks from the OpenSL ES thread in to WebRTC when asking for audio
+  // to render.
+  ALOGD("lowest possible buffer size: %" PRIuS,
+      audio_parameters_.GetBytesPerBuffer());
+  bytes_per_buffer_ = audio_parameters_.GetBytesPerFrame() *
+      audio_parameters_.frames_per_10ms_buffer();
+  RTC_DCHECK_GT(bytes_per_buffer_, audio_parameters_.GetBytesPerBuffer());
   ALOGD("native buffer size: %" PRIuS, bytes_per_buffer_);
   // Create a modified audio buffer class which allows us to ask for any number
   // of samples (and not only multiple of 10ms) to match the native OpenSL ES
@@ -418,6 +430,15 @@ void OpenSLESPlayer::FillBufferQueue() {
 }
 
 void OpenSLESPlayer::EnqueuePlayoutData() {
+  // Check delta time between two successive callbacks and provide a warning
+  // if it becomes very large.
+  // TODO(henrika): using 100ms as upper limit but this value is rather random.
+  const uint32_t current_time = rtc::Time();
+  const uint32_t diff = current_time - last_play_time_;
+  if (diff > 100) {
+    ALOGW("Bad OpenSL ES playout timing, dT=%u [ms]", diff);
+  }
+  last_play_time_ = current_time;
   // Read audio data from the WebRTC source using the FineAudioBuffer object
   // to adjust for differences in buffer size between WebRTC (10ms) and native
   // OpenSL ES.

webrtc/modules/audio_device/android/opensles_player.h

@@ -52,7 +52,7 @@ class OpenSLESPlayer {
   // buffer count of 2 or more, and a buffer size and sample rate that are
   // compatible with the device's native output configuration provided via the
   // audio manager at construction.
-  static const int kNumOfOpenSLESBuffers = 2;
+  static const int kNumOfOpenSLESBuffers = 4;
 
   // There is no need for this class to use JNI.
   static int32_t SetAndroidAudioDeviceObjects(void* javaVM, void* context) {
@@ -195,6 +195,9 @@ class OpenSLESPlayer {
   // This interface exposes controls for manipulating the object’s audio volume
   // properties. This interface is supported on the Audio Player object.
   SLVolumeItf volume_;
+
+  // Last time the OpenSL ES layer asked for audio data to play out.
+  uint32_t last_play_time_;
 };
 
 }  // namespace webrtc