1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
|
/*
* 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.
*/
#include "modules/audio_device/fine_audio_buffer.h"
#include <cstdint>
#include <cstring>
#include "api/array_view.h"
#include "modules/audio_device/audio_device_buffer.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
namespace webrtc {
FineAudioBuffer::FineAudioBuffer(AudioDeviceBuffer* audio_device_buffer)
: audio_device_buffer_(audio_device_buffer),
playout_samples_per_channel_10ms_(rtc::dchecked_cast<size_t>(
audio_device_buffer->PlayoutSampleRate() * 10 / 1000)),
record_samples_per_channel_10ms_(rtc::dchecked_cast<size_t>(
audio_device_buffer->RecordingSampleRate() * 10 / 1000)),
playout_channels_(audio_device_buffer->PlayoutChannels()),
record_channels_(audio_device_buffer->RecordingChannels()) {
RTC_DCHECK(audio_device_buffer_);
RTC_DLOG(LS_INFO) << __FUNCTION__;
if (IsReadyForPlayout()) {
RTC_DLOG(LS_INFO) << "playout_samples_per_channel_10ms: "
<< playout_samples_per_channel_10ms_;
RTC_DLOG(LS_INFO) << "playout_channels: " << playout_channels_;
}
if (IsReadyForRecord()) {
RTC_DLOG(LS_INFO) << "record_samples_per_channel_10ms: "
<< record_samples_per_channel_10ms_;
RTC_DLOG(LS_INFO) << "record_channels: " << record_channels_;
}
}
FineAudioBuffer::~FineAudioBuffer() {
RTC_DLOG(LS_INFO) << __FUNCTION__;
}
void FineAudioBuffer::ResetPlayout() {
playout_buffer_.Clear();
}
void FineAudioBuffer::ResetRecord() {
record_buffer_.Clear();
}
bool FineAudioBuffer::IsReadyForPlayout() const {
return playout_samples_per_channel_10ms_ > 0 && playout_channels_ > 0;
}
bool FineAudioBuffer::IsReadyForRecord() const {
return record_samples_per_channel_10ms_ > 0 && record_channels_ > 0;
}
void FineAudioBuffer::GetPlayoutData(rtc::ArrayView<int16_t> audio_buffer,
int playout_delay_ms) {
RTC_DCHECK(IsReadyForPlayout());
// Ask WebRTC for new data in chunks of 10ms until we have enough to
// fulfill the request. It is possible that the buffer already contains
// enough samples from the last round.
while (playout_buffer_.size() < audio_buffer.size()) {
// Get 10ms decoded audio from WebRTC. The ADB knows about number of
// channels; hence we can ask for number of samples per channel here.
if (audio_device_buffer_->RequestPlayoutData(
playout_samples_per_channel_10ms_) ==
static_cast<int32_t>(playout_samples_per_channel_10ms_)) {
// Append 10ms to the end of the local buffer taking number of channels
// into account.
const size_t num_elements_10ms =
playout_channels_ * playout_samples_per_channel_10ms_;
const size_t written_elements = playout_buffer_.AppendData(
num_elements_10ms, [&](rtc::ArrayView<int16_t> buf) {
const size_t samples_per_channel_10ms =
audio_device_buffer_->GetPlayoutData(buf.data());
return playout_channels_ * samples_per_channel_10ms;
});
RTC_DCHECK_EQ(num_elements_10ms, written_elements);
} else {
// Provide silence if AudioDeviceBuffer::RequestPlayoutData() fails.
// Can e.g. happen when an AudioTransport has not been registered.
const size_t num_bytes = audio_buffer.size() * sizeof(int16_t);
std::memset(audio_buffer.data(), 0, num_bytes);
return;
}
}
// Provide the requested number of bytes to the consumer.
const size_t num_bytes = audio_buffer.size() * sizeof(int16_t);
memcpy(audio_buffer.data(), playout_buffer_.data(), num_bytes);
// Move remaining samples to start of buffer to prepare for next round.
memmove(playout_buffer_.data(), playout_buffer_.data() + audio_buffer.size(),
(playout_buffer_.size() - audio_buffer.size()) * sizeof(int16_t));
playout_buffer_.SetSize(playout_buffer_.size() - audio_buffer.size());
// Cache playout latency for usage in DeliverRecordedData();
playout_delay_ms_ = playout_delay_ms;
}
void FineAudioBuffer::DeliverRecordedData(
rtc::ArrayView<const int16_t> audio_buffer,
int record_delay_ms,
absl::optional<int64_t> capture_time_ns) {
RTC_DCHECK(IsReadyForRecord());
// Always append new data and grow the buffer when needed.
record_buffer_.AppendData(audio_buffer.data(), audio_buffer.size());
// Consume samples from buffer in chunks of 10ms until there is not
// enough data left. The number of remaining samples in the cache is given by
// the new size of the internal `record_buffer_`.
const size_t num_elements_10ms =
record_channels_ * record_samples_per_channel_10ms_;
while (record_buffer_.size() >= num_elements_10ms) {
audio_device_buffer_->SetRecordedBuffer(record_buffer_.data(),
record_samples_per_channel_10ms_,
capture_time_ns);
audio_device_buffer_->SetVQEData(playout_delay_ms_, record_delay_ms);
audio_device_buffer_->DeliverRecordedData();
memmove(record_buffer_.data(), record_buffer_.data() + num_elements_10ms,
(record_buffer_.size() - num_elements_10ms) * sizeof(int16_t));
record_buffer_.SetSize(record_buffer_.size() - num_elements_10ms);
}
}
} // namespace webrtc
|
