OpenSL ES stability improvements

webrtc/modules/audio_device/android/opensles_player.cc

Index: webrtc/modules/audio_device/android/opensles_player.cc
diff --git a/webrtc/modules/audio_device/android/opensles_player.cc b/webrtc/modules/audio_device/android/opensles_player.cc
index b9ccfd594d36dad0d7b85fbe1f8dbd4f16249f09..d1edef227e652c11161250197d1aecc06cc0e1f7 100644
--- a/webrtc/modules/audio_device/android/opensles_player.cc
+++ b/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.