OpenSL (not default): Enables low latency audio on Android.

BUG=1669
R=andrew@webrtc.org, fischman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2032004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4719 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/audio_device/android/fine_audio_buffer.h

@@ -0,0 +1,66 @@
+/*
+ *  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/system_wrappers/interface/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,
+                  int 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.
+  int 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_;
+  int desired_frame_size_bytes_;  // Number of bytes delivered per GetBufferData
+  int sample_rate_;
+  int samples_per_10_ms_;
+  // Convenience parameter to avoid converting from samples
+  int bytes_per_10_ms_;
+
+  // Storage for samples that are not yet asked for.
+  scoped_array<int8_t> cache_buffer_;
+  int cached_buffer_start_;  // Location of first unread sample.
+  int cached_bytes_;  // Number of bytes stored in cache.
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_FINE_AUDIO_BUFFER_H_