Simplifies FineAudioBuffer by using rtc::Buffer

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 7dfc5ec891226709a99e6becad6f30206380c131..2d305f0ff7406ae67d1aeb1feee2185c80f025dd 100644
--- a/webrtc/modules/audio_device/android/opensles_player.cc
+++ b/webrtc/modules/audio_device/android/opensles_player.cc
@@ -205,19 +205,16 @@ void OpenSLESPlayer::AllocateDataBuffers() {
   // recommended to construct audio buffers so that they contain an exact
   // multiple of this number. If so, callbacks will occur at regular intervals,
   // which reduces jitter.
-  ALOGD("native buffer size: %" PRIuS, audio_parameters_.GetBytesPerBuffer());
+  const size_t buffer_size_in_bytes = audio_parameters_.GetBytesPerBuffer();
+  ALOGD("native buffer size: %" PRIuS, buffer_size_in_bytes);
   ALOGD("native buffer size in ms: %.2f",
         audio_parameters_.GetBufferSizeInMilliseconds());
-  fine_audio_buffer_.reset(new FineAudioBuffer(
-      audio_device_buffer_, audio_parameters_.GetBytesPerBuffer(),
-      audio_parameters_.sample_rate()));
-  // Each buffer must be of this size to avoid unnecessary memcpy while caching
-  // data between successive callbacks.
-  const size_t required_buffer_size =
-      fine_audio_buffer_->RequiredPlayoutBufferSizeBytes();
-  ALOGD("required buffer size: %" PRIuS, required_buffer_size);
+  fine_audio_buffer_.reset(
+      new FineAudioBuffer(audio_device_buffer_, buffer_size_in_bytes,
+                          audio_parameters_.sample_rate()));
+  // Allocated memory for audio buffers.
   for (int i = 0; i < kNumOfOpenSLESBuffers; ++i) {
-    audio_buffers_[i].reset(new SLint8[required_buffer_size]);
+    audio_buffers_[i].reset(new SLint8[buffer_size_in_bytes]);
   }
 }