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
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
|
// Copyright 2020 The Chromium Authors
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.
#include "third_party/blink/renderer/modules/peerconnection/rtc_encoded_audio_frame_delegate.h"
#include <optional>
#include <utility>
#include "base/time/time.h"
#include "third_party/blink/renderer/core/typed_arrays/dom_array_buffer.h"
#include "third_party/blink/renderer/platform/bindings/exception_code.h"
#include "third_party/blink/renderer/platform/bindings/exception_state.h"
#include "third_party/blink/renderer/platform/bindings/v8_binding.h"
#include "third_party/blink/renderer/platform/peerconnection/webrtc_util.h"
#include "third_party/webrtc/api/frame_transformer_factory.h"
namespace blink {
static constexpr char kRTCEncodedAudioFrameDetachKey[] = "RTCEncodedAudioFrame";
const void* RTCEncodedAudioFramesAttachment::kAttachmentKey;
RTCEncodedAudioFrameDelegate::RTCEncodedAudioFrameDelegate(
std::unique_ptr<webrtc::TransformableAudioFrameInterface> webrtc_frame,
webrtc::ArrayView<const unsigned int> contributing_sources,
std::optional<uint16_t> sequence_number)
: webrtc_frame_(std::move(webrtc_frame)),
contributing_sources_(contributing_sources),
sequence_number_(sequence_number) {}
uint32_t RTCEncodedAudioFrameDelegate::RtpTimestamp() const {
base::AutoLock lock(lock_);
return webrtc_frame_ ? webrtc_frame_->GetTimestamp() : 0;
}
DOMArrayBuffer* RTCEncodedAudioFrameDelegate::CreateDataBuffer(
v8::Isolate* isolate) const {
ArrayBufferContents contents;
{
base::AutoLock lock(lock_);
if (!webrtc_frame_) {
// WebRTC frame already passed, return a detached ArrayBuffer.
DOMArrayBuffer* buffer = DOMArrayBuffer::Create(
/*num_elements=*/static_cast<size_t>(0), /*element_byte_size=*/1);
ArrayBufferContents contents_to_drop;
NonThrowableExceptionState exception_state;
buffer->Transfer(isolate,
V8AtomicString(isolate, kRTCEncodedAudioFrameDetachKey),
contents_to_drop, exception_state);
return buffer;
}
auto data = webrtc_frame_->GetData();
contents = ArrayBufferContents(
data.size(), 1, ArrayBufferContents::kNotShared,
ArrayBufferContents::kDontInitialize,
ArrayBufferContents::AllocationFailureBehavior::kCrash);
CHECK(contents.IsValid());
contents.ByteSpan().copy_from(data);
}
return DOMArrayBuffer::Create(std::move(contents));
}
void RTCEncodedAudioFrameDelegate::SetData(const DOMArrayBuffer* data) {
base::AutoLock lock(lock_);
if (webrtc_frame_ && data) {
webrtc_frame_->SetData(webrtc::ArrayView<const uint8_t>(
static_cast<const uint8_t*>(data->Data()), data->ByteLength()));
}
}
base::expected<void, String> RTCEncodedAudioFrameDelegate::SetRtpTimestamp(
uint32_t timestamp) {
base::AutoLock lock(lock_);
if (!webrtc_frame_) {
return base::unexpected("Underlying webrtc frame doesn't exist.");
}
webrtc_frame_->SetRTPTimestamp(timestamp);
return base::ok();
}
std::optional<uint32_t> RTCEncodedAudioFrameDelegate::Ssrc() const {
base::AutoLock lock(lock_);
return webrtc_frame_ ? std::make_optional(webrtc_frame_->GetSsrc())
: std::nullopt;
}
std::optional<uint8_t> RTCEncodedAudioFrameDelegate::PayloadType() const {
base::AutoLock lock(lock_);
return webrtc_frame_ ? std::make_optional(webrtc_frame_->GetPayloadType())
: std::nullopt;
}
std::optional<std::string> RTCEncodedAudioFrameDelegate::MimeType() const {
base::AutoLock lock(lock_);
return webrtc_frame_ ? std::make_optional(webrtc_frame_->GetMimeType())
: std::nullopt;
}
std::optional<uint16_t> RTCEncodedAudioFrameDelegate::SequenceNumber() const {
return sequence_number_;
}
Vector<uint32_t> RTCEncodedAudioFrameDelegate::ContributingSources() const {
return contributing_sources_;
}
std::optional<base::TimeTicks> RTCEncodedAudioFrameDelegate::ReceiveTime()
const {
base::AutoLock lock(lock_);
if (!webrtc_frame_) {
return std::nullopt;
}
return ConvertToOptionalTimeTicks(webrtc_frame_->ReceiveTime());
}
std::optional<base::TimeTicks> RTCEncodedAudioFrameDelegate::CaptureTime()
const {
base::AutoLock lock(lock_);
if (!webrtc_frame_ ||
webrtc_frame_->GetDirection() !=
webrtc::TransformableFrameInterface::Direction::kReceiver) {
return std::nullopt;
}
return ConvertToOptionalTimeTicks(webrtc_frame_->CaptureTime(),
WebRTCFrameNtpEpoch());
}
std::optional<base::TimeDelta>
RTCEncodedAudioFrameDelegate::SenderCaptureTimeOffset() const {
base::AutoLock lock(lock_);
if (!webrtc_frame_) {
return std::nullopt;
}
return ConvertToOptionalTimeDelta(webrtc_frame_->SenderCaptureTimeOffset());
}
std::unique_ptr<webrtc::TransformableAudioFrameInterface>
RTCEncodedAudioFrameDelegate::PassWebRtcFrame() {
base::AutoLock lock(lock_);
return std::move(webrtc_frame_);
}
std::unique_ptr<webrtc::TransformableAudioFrameInterface>
RTCEncodedAudioFrameDelegate::CloneWebRtcFrame() {
base::AutoLock lock(lock_);
if (!webrtc_frame_) {
return nullptr;
}
return webrtc::CloneAudioFrame(webrtc_frame_.get());
}
} // namespace blink
|
