1/*
2 *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
3 *
4 *  Use of this source code is governed by a BSD-style license
5 *  that can be found in the LICENSE file in the root of the source
6 *  tree. An additional intellectual property rights grant can be found
7 *  in the file PATENTS.  All contributing project authors may
8 *  be found in the AUTHORS file in the root of the source tree.
9 */
10
11#include "webrtc/modules/audio_device/android/fine_audio_buffer.h"
12
13#include <memory.h>
14#include <stdio.h>
15#include <algorithm>
16
17#include "webrtc/modules/audio_device/audio_device_buffer.h"
18
19namespace webrtc {
20
21FineAudioBuffer::FineAudioBuffer(AudioDeviceBuffer* device_buffer,
22                                 int desired_frame_size_bytes,
23                                 int sample_rate)
24    : device_buffer_(device_buffer),
25      desired_frame_size_bytes_(desired_frame_size_bytes),
26      sample_rate_(sample_rate),
27      samples_per_10_ms_(sample_rate_ * 10 / 1000),
28      bytes_per_10_ms_(samples_per_10_ms_ * sizeof(int16_t)),
29      cached_buffer_start_(0),
30      cached_bytes_(0) {
31  cache_buffer_.reset(new int8_t[bytes_per_10_ms_]);
32}
33
34FineAudioBuffer::~FineAudioBuffer() {
35}
36
37int FineAudioBuffer::RequiredBufferSizeBytes() {
38  // It is possible that we store the desired frame size - 1 samples. Since new
39  // audio frames are pulled in chunks of 10ms we will need a buffer that can
40  // hold desired_frame_size - 1 + 10ms of data. We omit the - 1.
41  return desired_frame_size_bytes_ + bytes_per_10_ms_;
42}
43
44void FineAudioBuffer::GetBufferData(int8_t* buffer) {
45  if (desired_frame_size_bytes_ <= cached_bytes_) {
46    memcpy(buffer, &cache_buffer_.get()[cached_buffer_start_],
47           desired_frame_size_bytes_);
48    cached_buffer_start_ += desired_frame_size_bytes_;
49    cached_bytes_ -= desired_frame_size_bytes_;
50    assert(cached_buffer_start_ + cached_bytes_ < bytes_per_10_ms_);
51    return;
52  }
53  memcpy(buffer, &cache_buffer_.get()[cached_buffer_start_], cached_bytes_);
54  // Push another n*10ms of audio to |buffer|. n > 1 if
55  // |desired_frame_size_bytes_| is greater than 10ms of audio. Note that we
56  // write the audio after the cached bytes copied earlier.
57  int8_t* unwritten_buffer = &buffer[cached_bytes_];
58  int bytes_left = desired_frame_size_bytes_ - cached_bytes_;
59  // Ceiling of integer division: 1 + ((x - 1) / y)
60  int number_of_requests = 1 + (bytes_left - 1) / (bytes_per_10_ms_);
61  for (int i = 0; i < number_of_requests; ++i) {
62    device_buffer_->RequestPlayoutData(samples_per_10_ms_);
63    int num_out = device_buffer_->GetPlayoutData(unwritten_buffer);
64    if (num_out != samples_per_10_ms_) {
65      assert(num_out == 0);
66      cached_bytes_ = 0;
67      return;
68    }
69    unwritten_buffer += bytes_per_10_ms_;
70    assert(bytes_left >= 0);
71    bytes_left -= bytes_per_10_ms_;
72  }
73  assert(bytes_left <= 0);
74  // Put the samples that were written to |buffer| but are not used in the
75  // cache.
76  int cache_location = desired_frame_size_bytes_;
77  int8_t* cache_ptr = &buffer[cache_location];
78  cached_bytes_ = number_of_requests * bytes_per_10_ms_ -
79      (desired_frame_size_bytes_ - cached_bytes_);
80  // If cached_bytes_ is larger than the cache buffer, uninitialized memory
81  // will be read.
82  assert(cached_bytes_ <= bytes_per_10_ms_);
83  assert(-bytes_left == cached_bytes_);
84  cached_buffer_start_ = 0;
85  memcpy(cache_buffer_.get(), cache_ptr, cached_bytes_);
86}
87
88}  // namespace webrtc
89