Refactor the AudioDevice for iOS and improve the performance and stability

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.
  */
 
 #include "webrtc/modules/audio_device/android/opensles_player.h"
 
 #include <android/log.h>
 
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/format_macros.h"
 #include "webrtc/modules/audio_device/android/audio_manager.h"
-#include "webrtc/modules/audio_device/android/fine_audio_buffer.h"
+#include "webrtc/modules/audio_device/fine_audio_buffer.h"
 
 #define TAG "OpenSLESPlayer"
 #define ALOGV(...) __android_log_print(ANDROID_LOG_VERBOSE, TAG, __VA_ARGS__)
 #define ALOGD(...) __android_log_print(ANDROID_LOG_DEBUG, TAG, __VA_ARGS__)
 #define ALOGE(...) __android_log_print(ANDROID_LOG_ERROR, TAG, __VA_ARGS__)
 #define ALOGW(...) __android_log_print(ANDROID_LOG_WARN, TAG, __VA_ARGS__)
 #define ALOGI(...) __android_log_print(ANDROID_LOG_INFO, TAG, __VA_ARGS__)
 
 #define RETURN_ON_ERROR(op, ...)        \
   do {                                  \
@@ skipping 205 matching lines @@
   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
   // buffer size.
   fine_buffer_.reset(new FineAudioBuffer(audio_device_buffer_,
                                          bytes_per_buffer_,
                                          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_buffer_->RequiredBufferSizeBytes();
+  const size_t required_buffer_size =
+      fine_buffer_->RequiredPlayoutBufferSizeBytes();
   ALOGD("required buffer size: %" PRIuS, required_buffer_size);
   for (int i = 0; i < kNumOfOpenSLESBuffers; ++i) {
     audio_buffers_[i].reset(new SLint8[required_buffer_size]);
   }
 }
 
 bool OpenSLESPlayer::CreateEngine() {
   ALOGD("CreateEngine");
   DCHECK(thread_checker_.CalledOnValidThread());
   if (engine_object_.Get())
@@ skipping 157 matching lines @@
     return;
   }
   EnqueuePlayoutData();
 }
 
 void OpenSLESPlayer::EnqueuePlayoutData() {
   // 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.
   SLint8* audio_ptr = audio_buffers_[buffer_index_].get();
-  fine_buffer_->GetBufferData(audio_ptr);
+  fine_buffer_->GetPlayoutData(audio_ptr);
   // Enqueue the decoded audio buffer for playback.
   SLresult err =
       (*simple_buffer_queue_)
           ->Enqueue(simple_buffer_queue_, audio_ptr, bytes_per_buffer_);
   if (SL_RESULT_SUCCESS != err) {
     ALOGE("Enqueue failed: %d", err);
   }
   buffer_index_ = (buffer_index_ + 1) % kNumOfOpenSLESBuffers;
 }
 
 SLuint32 OpenSLESPlayer::GetPlayState() const {
   DCHECK(player_);
   SLuint32 state;
   SLresult err = (*player_)->GetPlayState(player_, &state);
   if (SL_RESULT_SUCCESS != err) {
     ALOGE("GetPlayState failed: %d", err);
   }
   return state;
 }
 
 }  // namespace webrtc