Refactor the AudioDevice for iOS and improve the performance and stability

webrtc/modules/audio_device/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_FINE_AUDIO_BUFFER_H_
+#define WEBRTC_MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class AudioDeviceBuffer;
+
+// FineAudioBuffer takes an AudioDeviceBuffer (ADB) which deals with 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. This class also ensures that audio data can be delivered to the ADB
+// in 10ms chunks when the size of the provided audio buffers differs from 10ms.
+// As an example: calling DeliverRecordedData() with 5ms buffers will deliver
+// accumulated 10ms worth of data to the ADB every second call.
+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
+  // GetPlayoutData() should return on success. It is also the required size of
+  // each recorded buffer used in DeliverRecordedData() calls.
+  // |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 GetPlayoutData(). If
+  // the buffer is smaller memory trampling will happen.
+  size_t RequiredPlayoutBufferSizeBytes();
+
+  // Clears buffers and counters dealing with playour and/or recording.
+  void ResetPlayout();
+  void ResetRecord();
+
+  // |buffer| must be of equal or greater size than what is returned by
+  // RequiredBufferSize(). This is to avoid unnecessary memcpy.
+  void GetPlayoutData(int8_t* buffer);
+
+  // Consumes the audio data in |buffer| and sends it to the WebRTC layer in
+  // chunks of 10ms. The provided delay estimates in |playout_delay_ms| and
+  // |record_delay_ms| are given to the AEC in the audio processing module.
+  // They can be fixed values on most platforms and they are ignored if an
+  // external (hardware/built-in) AEC is used.
+  // The size of |buffer| is given by |size_in_bytes| and must be equal to
+  // |desired_frame_size_bytes_|. A CHECK will be hit if this is not the case.
+  // Example: buffer size is 5ms => call #1 stores 5ms of data, call #2 stores
+  // 5ms of data and sends a total of 10ms to WebRTC and clears the intenal
+  // cache. Call #3 restarts the scheme above.
+  void DeliverRecordedData(const int8_t* buffer,
+                           size_t size_in_bytes,
+                           int playout_delay_ms,
+                           int record_delay_ms);
+
+ private:
+  // Device buffer that works with 10ms chunks of data both for playout and
+  // for recording. I.e., the WebRTC side will always be asked for audio to be
+  // played out in 10ms chunks and recorded audio will be sent to WebRTC in
+  // 10ms chunks as well. This pointer is owned by the constructor of this
+  // class and the owner must ensure that the pointer is valid during the life-
+  // time of this object.
+  AudioDeviceBuffer* const device_buffer_;
+  // Number of bytes delivered by GetPlayoutData() call and provided to
+  // DeliverRecordedData().
+  const size_t desired_frame_size_bytes_;
+  // Sample rate in Hertz.
+  const int sample_rate_;
+  // Number of audio samples per 10ms.
+  const size_t samples_per_10_ms_;
+  // Number of audio bytes per 10ms.
+  const size_t bytes_per_10_ms_;
+  // Storage for output samples that are not yet asked for.
+  rtc::scoped_ptr<int8_t[]> playout_cache_buffer_;
+  // Location of first unread output sample.
+  size_t playout_cached_buffer_start_;
+  // Number of bytes stored in output (contain samples to be played out) cache.
+  size_t playout_cached_bytes_;
+  // Storage for input samples that are about to be delivered to the WebRTC
+  // ADB or remains from the last successful delivery of a 10ms audio buffer.
+  rtc::scoped_ptr<int8_t[]> record_cache_buffer_;
+  // Required (max) size in bytes of the |record_cache_buffer_|.
+  const size_t required_record_buffer_size_bytes_;
+  // Number of bytes in input (contains recorded samples) cache.
+  size_t record_cached_bytes_;
+  // Read and write pointers used in the buffering scheme on the recording side.
+  size_t record_read_pos_;
+  size_t record_write_pos_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_