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/fine_audio_buffer.h"
12
13 #include <memory.h>
14 #include <stdio.h>
15 #include <algorithm>
16
17 #include "webrtc/base/checks.h"
18 #include "webrtc/base/logging.h"
19 #include "webrtc/modules/audio_device/audio_device_buffer.h"
20
21 namespace webrtc {
22
FineAudioBuffer(AudioDeviceBuffer * device_buffer,size_t desired_frame_size_bytes,int sample_rate)23 FineAudioBuffer::FineAudioBuffer(AudioDeviceBuffer* device_buffer,
24 size_t desired_frame_size_bytes,
25 int sample_rate)
26 : device_buffer_(device_buffer),
27 desired_frame_size_bytes_(desired_frame_size_bytes),
28 sample_rate_(sample_rate),
29 samples_per_10_ms_(static_cast<size_t>(sample_rate_ * 10 / 1000)),
30 bytes_per_10_ms_(samples_per_10_ms_ * sizeof(int16_t)),
31 playout_cached_buffer_start_(0),
32 playout_cached_bytes_(0),
33 // Allocate extra space on the recording side to reduce the number of
34 // memmove() calls.
35 required_record_buffer_size_bytes_(
36 5 * (desired_frame_size_bytes + bytes_per_10_ms_)),
37 record_cached_bytes_(0),
38 record_read_pos_(0),
39 record_write_pos_(0) {
40 playout_cache_buffer_.reset(new int8_t[bytes_per_10_ms_]);
41 record_cache_buffer_.reset(new int8_t[required_record_buffer_size_bytes_]);
42 memset(record_cache_buffer_.get(), 0, required_record_buffer_size_bytes_);
43 }
44
~FineAudioBuffer()45 FineAudioBuffer::~FineAudioBuffer() {}
46
RequiredPlayoutBufferSizeBytes()47 size_t FineAudioBuffer::RequiredPlayoutBufferSizeBytes() {
48 // It is possible that we store the desired frame size - 1 samples. Since new
49 // audio frames are pulled in chunks of 10ms we will need a buffer that can
50 // hold desired_frame_size - 1 + 10ms of data. We omit the - 1.
51 return desired_frame_size_bytes_ + bytes_per_10_ms_;
52 }
53
ResetPlayout()54 void FineAudioBuffer::ResetPlayout() {
55 playout_cached_buffer_start_ = 0;
56 playout_cached_bytes_ = 0;
57 memset(playout_cache_buffer_.get(), 0, bytes_per_10_ms_);
58 }
59
ResetRecord()60 void FineAudioBuffer::ResetRecord() {
61 record_cached_bytes_ = 0;
62 record_read_pos_ = 0;
63 record_write_pos_ = 0;
64 memset(record_cache_buffer_.get(), 0, required_record_buffer_size_bytes_);
65 }
66
GetPlayoutData(int8_t * buffer)67 void FineAudioBuffer::GetPlayoutData(int8_t* buffer) {
68 if (desired_frame_size_bytes_ <= playout_cached_bytes_) {
69 memcpy(buffer, &playout_cache_buffer_.get()[playout_cached_buffer_start_],
70 desired_frame_size_bytes_);
71 playout_cached_buffer_start_ += desired_frame_size_bytes_;
72 playout_cached_bytes_ -= desired_frame_size_bytes_;
73 RTC_CHECK_LT(playout_cached_buffer_start_ + playout_cached_bytes_,
74 bytes_per_10_ms_);
75 return;
76 }
77 memcpy(buffer, &playout_cache_buffer_.get()[playout_cached_buffer_start_],
78 playout_cached_bytes_);
79 // Push another n*10ms of audio to |buffer|. n > 1 if
80 // |desired_frame_size_bytes_| is greater than 10ms of audio. Note that we
81 // write the audio after the cached bytes copied earlier.
82 int8_t* unwritten_buffer = &buffer[playout_cached_bytes_];
83 int bytes_left =
84 static_cast<int>(desired_frame_size_bytes_ - playout_cached_bytes_);
85 // Ceiling of integer division: 1 + ((x - 1) / y)
86 size_t number_of_requests = 1 + (bytes_left - 1) / (bytes_per_10_ms_);
87 for (size_t i = 0; i < number_of_requests; ++i) {
88 device_buffer_->RequestPlayoutData(samples_per_10_ms_);
89 int num_out = device_buffer_->GetPlayoutData(unwritten_buffer);
90 if (static_cast<size_t>(num_out) != samples_per_10_ms_) {
91 RTC_CHECK_EQ(num_out, 0);
92 playout_cached_bytes_ = 0;
93 return;
94 }
95 unwritten_buffer += bytes_per_10_ms_;
96 RTC_CHECK_GE(bytes_left, 0);
97 bytes_left -= static_cast<int>(bytes_per_10_ms_);
98 }
99 RTC_CHECK_LE(bytes_left, 0);
100 // Put the samples that were written to |buffer| but are not used in the
101 // cache.
102 size_t cache_location = desired_frame_size_bytes_;
103 int8_t* cache_ptr = &buffer[cache_location];
104 playout_cached_bytes_ = number_of_requests * bytes_per_10_ms_ -
105 (desired_frame_size_bytes_ - playout_cached_bytes_);
106 // If playout_cached_bytes_ is larger than the cache buffer, uninitialized
107 // memory will be read.
108 RTC_CHECK_LE(playout_cached_bytes_, bytes_per_10_ms_);
109 RTC_CHECK_EQ(static_cast<size_t>(-bytes_left), playout_cached_bytes_);
110 playout_cached_buffer_start_ = 0;
111 memcpy(playout_cache_buffer_.get(), cache_ptr, playout_cached_bytes_);
112 }
113
DeliverRecordedData(const int8_t * buffer,size_t size_in_bytes,int playout_delay_ms,int record_delay_ms)114 void FineAudioBuffer::DeliverRecordedData(const int8_t* buffer,
115 size_t size_in_bytes,
116 int playout_delay_ms,
117 int record_delay_ms) {
118 // Check if the temporary buffer can store the incoming buffer. If not,
119 // move the remaining (old) bytes to the beginning of the temporary buffer
120 // and start adding new samples after the old samples.
121 if (record_write_pos_ + size_in_bytes > required_record_buffer_size_bytes_) {
122 if (record_cached_bytes_ > 0) {
123 memmove(record_cache_buffer_.get(),
124 record_cache_buffer_.get() + record_read_pos_,
125 record_cached_bytes_);
126 }
127 record_write_pos_ = record_cached_bytes_;
128 record_read_pos_ = 0;
129 }
130 // Add recorded samples to a temporary buffer.
131 memcpy(record_cache_buffer_.get() + record_write_pos_, buffer, size_in_bytes);
132 record_write_pos_ += size_in_bytes;
133 record_cached_bytes_ += size_in_bytes;
134 // Consume samples in temporary buffer in chunks of 10ms until there is not
135 // enough data left. The number of remaining bytes in the cache is given by
136 // |record_cached_bytes_| after this while loop is done.
137 while (record_cached_bytes_ >= bytes_per_10_ms_) {
138 device_buffer_->SetRecordedBuffer(
139 record_cache_buffer_.get() + record_read_pos_, samples_per_10_ms_);
140 device_buffer_->SetVQEData(playout_delay_ms, record_delay_ms, 0);
141 device_buffer_->DeliverRecordedData();
142 // Read next chunk of 10ms data.
143 record_read_pos_ += bytes_per_10_ms_;
144 // Reduce number of cached bytes with the consumed amount.
145 record_cached_bytes_ -= bytes_per_10_ms_;
146 }
147 }
148
149 } // namespace webrtc
150