Simplifies FineAudioBuffer by using rtc::Buffer

webrtc/modules/audio_device/android/opensles_player.cc

 /*
  *  Copyright (c) 2015 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.
  */
 
@@ skipping 187 matching lines @@
   RTC_DCHECK(!simple_buffer_queue_);
   RTC_CHECK(audio_device_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
   // buffer size. The native buffer size corresponds to the
   // PROPERTY_OUTPUT_FRAMES_PER_BUFFER property which is the number of audio
   // frames that the HAL (Hardware Abstraction Layer) buffer can hold. It is
   // 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]);
   }
 }
 
 bool OpenSLESPlayer::ObtainEngineInterface() {
   ALOGD("ObtainEngineInterface");
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   if (engine_)
     return true;
   // Get access to (or create if not already existing) the global OpenSL Engine
   // object.
@@ skipping 186 matching lines @@
   RTC_DCHECK(player_);
   SLuint32 state;
   SLresult err = (*player_)->GetPlayState(player_, &state);
   if (SL_RESULT_SUCCESS != err) {
     ALOGE("GetPlayState failed: %d", err);
   }
   return state;
 }
 
 }  // namespace webrtc