Refactor the AudioDevice for iOS and improve the performance and stability

webrtc/modules/audio_device/android/fine_audio_buffer.h

-/*
- *  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.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_FINE_AUDIO_BUFFER_H_
-#define WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_FINE_AUDIO_BUFFER_H_
-
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-class AudioDeviceBuffer;
-
-// FineAudioBuffer takes an AudioDeviceBuffer which delivers audio data
-// corresponding to 10ms of data. It then allows for this data to be pulled in
-// a finer or coarser granularity. I.e. interacting with this class instead of
-// directly with the AudioDeviceBuffer one can ask for any number of audio data
-// samples.
-class FineAudioBuffer {
- public:
-  // |device_buffer| is a buffer that provides 10ms of audio data.
-  // |desired_frame_size_bytes| is the number of bytes of audio data
-  // (not samples) |GetBufferData| should return on success.
-  // |sample_rate| is the sample rate of the audio data. This is needed because
-  // |device_buffer| delivers 10ms of data. Given the sample rate the number
-  // of samples can be calculated.
-  FineAudioBuffer(AudioDeviceBuffer* device_buffer,
-                  size_t desired_frame_size_bytes,
-                  int sample_rate);
-  ~FineAudioBuffer();
-
-  // Returns the required size of |buffer| when calling GetBufferData. If the
-  // buffer is smaller memory trampling will happen.
-  // |desired_frame_size_bytes| and |samples_rate| are as described in the
-  // constructor.
-  size_t RequiredBufferSizeBytes();
-
-  // |buffer| must be of equal or greater size than what is returned by
-  // RequiredBufferSize. This is to avoid unnecessary memcpy.
-  void GetBufferData(int8_t* buffer);
-
- private:
-  // Device buffer that provides 10ms chunks of data.
-  AudioDeviceBuffer* device_buffer_;
-  // Number of bytes delivered per GetBufferData
-  size_t desired_frame_size_bytes_;
-  int sample_rate_;
-  size_t samples_per_10_ms_;
-  // Convenience parameter to avoid converting from samples
-  size_t bytes_per_10_ms_;
-
-  // Storage for samples that are not yet asked for.
-  rtc::scoped_ptr<int8_t[]> cache_buffer_;
-  // Location of first unread sample.
-  size_t cached_buffer_start_;
-  // Number of bytes stored in cache.
-  size_t cached_bytes_;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_FINE_AUDIO_BUFFER_H_