Refactor the AudioDevice for iOS and improve the performance and stability

webrtc/modules/audio_device/android/fine_audio_buffer.cc

-/*
- *  Copyright (c) 2013 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/fine_audio_buffer.h"
-
-#include <memory.h>
-#include <stdio.h>
-#include <algorithm>
-
-#include "webrtc/base/checks.h"
-#include "webrtc/modules/audio_device/audio_device_buffer.h"
-
-namespace webrtc {
-
-FineAudioBuffer::FineAudioBuffer(AudioDeviceBuffer* device_buffer,
-                                 size_t desired_frame_size_bytes,
-                                 int sample_rate)
-    : device_buffer_(device_buffer),
-      desired_frame_size_bytes_(desired_frame_size_bytes),
-      sample_rate_(sample_rate),
-      samples_per_10_ms_(static_cast<size_t>(sample_rate_ * 10 / 1000)),
-      bytes_per_10_ms_(samples_per_10_ms_ * sizeof(int16_t)),
-      cached_buffer_start_(0),
-      cached_bytes_(0) {
-  cache_buffer_.reset(new int8_t[bytes_per_10_ms_]);
-}
-
-FineAudioBuffer::~FineAudioBuffer() {
-}
-
-size_t FineAudioBuffer::RequiredBufferSizeBytes() {
-  // It is possible that we store the desired frame size - 1 samples. Since new
-  // audio frames are pulled in chunks of 10ms we will need a buffer that can
-  // hold desired_frame_size - 1 + 10ms of data. We omit the - 1.
-  return desired_frame_size_bytes_ + bytes_per_10_ms_;
-}
-
-void FineAudioBuffer::GetBufferData(int8_t* buffer) {
-  if (desired_frame_size_bytes_ <= cached_bytes_) {
-    memcpy(buffer, &cache_buffer_.get()[cached_buffer_start_],
-           desired_frame_size_bytes_);
-    cached_buffer_start_ += desired_frame_size_bytes_;
-    cached_bytes_ -= desired_frame_size_bytes_;
-    CHECK_LT(cached_buffer_start_ + cached_bytes_, bytes_per_10_ms_);
-    return;
-  }
-  memcpy(buffer, &cache_buffer_.get()[cached_buffer_start_], cached_bytes_);
-  // Push another n*10ms of audio to |buffer|. n > 1 if
-  // |desired_frame_size_bytes_| is greater than 10ms of audio. Note that we
-  // write the audio after the cached bytes copied earlier.
-  int8_t* unwritten_buffer = &buffer[cached_bytes_];
-  int bytes_left = static_cast<int>(desired_frame_size_bytes_ - cached_bytes_);
-  // Ceiling of integer division: 1 + ((x - 1) / y)
-  size_t number_of_requests = 1 + (bytes_left - 1) / (bytes_per_10_ms_);
-  for (size_t i = 0; i < number_of_requests; ++i) {
-    device_buffer_->RequestPlayoutData(samples_per_10_ms_);
-    int num_out = device_buffer_->GetPlayoutData(unwritten_buffer);
-    if (static_cast<size_t>(num_out) != samples_per_10_ms_) {
-      CHECK_EQ(num_out, 0);
-      cached_bytes_ = 0;
-      return;
-    }
-    unwritten_buffer += bytes_per_10_ms_;
-    CHECK_GE(bytes_left, 0);
-    bytes_left -= bytes_per_10_ms_;
-  }
-  CHECK_LE(bytes_left, 0);
-  // Put the samples that were written to |buffer| but are not used in the
-  // cache.
-  size_t cache_location = desired_frame_size_bytes_;
-  int8_t* cache_ptr = &buffer[cache_location];
-  cached_bytes_ = number_of_requests * bytes_per_10_ms_ -
-      (desired_frame_size_bytes_ - cached_bytes_);
-  // If cached_bytes_ is larger than the cache buffer, uninitialized memory
-  // will be read.
-  CHECK_LE(cached_bytes_, bytes_per_10_ms_);
-  CHECK_EQ(static_cast<size_t>(-bytes_left), cached_bytes_);
-  cached_buffer_start_ = 0;
-  memcpy(cache_buffer_.get(), cache_ptr, cached_bytes_);
-}
-
-}  // namespace webrtc