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
|
/*
* Copyright (c) 2020 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 "audio/channel_send_frame_transformer_delegate.h"
#include <utility>
namespace webrtc {
namespace {
class TransformableOutgoingAudioFrame : public TransformableFrameInterface {
public:
TransformableOutgoingAudioFrame(AudioFrameType frame_type,
uint8_t payload_type,
uint32_t rtp_timestamp,
uint32_t rtp_start_timestamp,
const uint8_t* payload_data,
size_t payload_size,
int64_t absolute_capture_timestamp_ms,
uint32_t ssrc)
: frame_type_(frame_type),
payload_type_(payload_type),
rtp_timestamp_(rtp_timestamp),
rtp_start_timestamp_(rtp_start_timestamp),
payload_(payload_data, payload_size),
absolute_capture_timestamp_ms_(absolute_capture_timestamp_ms),
ssrc_(ssrc) {}
~TransformableOutgoingAudioFrame() override = default;
rtc::ArrayView<const uint8_t> GetData() const override { return payload_; }
void SetData(rtc::ArrayView<const uint8_t> data) override {
payload_.SetData(data.data(), data.size());
}
uint32_t GetTimestamp() const override {
return rtp_timestamp_ + rtp_start_timestamp_;
}
uint32_t GetStartTimestamp() const { return rtp_start_timestamp_; }
uint32_t GetSsrc() const override { return ssrc_; }
AudioFrameType GetFrameType() const { return frame_type_; }
uint8_t GetPayloadType() const override { return payload_type_; }
int64_t GetAbsoluteCaptureTimestampMs() const {
return absolute_capture_timestamp_ms_;
}
Direction GetDirection() const override { return Direction::kSender; }
private:
AudioFrameType frame_type_;
uint8_t payload_type_;
uint32_t rtp_timestamp_;
uint32_t rtp_start_timestamp_;
rtc::Buffer payload_;
int64_t absolute_capture_timestamp_ms_;
uint32_t ssrc_;
};
} // namespace
ChannelSendFrameTransformerDelegate::ChannelSendFrameTransformerDelegate(
SendFrameCallback send_frame_callback,
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
rtc::TaskQueue* encoder_queue)
: send_frame_callback_(send_frame_callback),
frame_transformer_(std::move(frame_transformer)),
encoder_queue_(encoder_queue) {}
void ChannelSendFrameTransformerDelegate::Init() {
frame_transformer_->RegisterTransformedFrameCallback(
rtc::scoped_refptr<TransformedFrameCallback>(this));
}
void ChannelSendFrameTransformerDelegate::Reset() {
frame_transformer_->UnregisterTransformedFrameCallback();
frame_transformer_ = nullptr;
MutexLock lock(&send_lock_);
send_frame_callback_ = SendFrameCallback();
}
void ChannelSendFrameTransformerDelegate::Transform(
AudioFrameType frame_type,
uint8_t payload_type,
uint32_t rtp_timestamp,
uint32_t rtp_start_timestamp,
const uint8_t* payload_data,
size_t payload_size,
int64_t absolute_capture_timestamp_ms,
uint32_t ssrc) {
frame_transformer_->Transform(
std::make_unique<TransformableOutgoingAudioFrame>(
frame_type, payload_type, rtp_timestamp, rtp_start_timestamp,
payload_data, payload_size, absolute_capture_timestamp_ms, ssrc));
}
void ChannelSendFrameTransformerDelegate::OnTransformedFrame(
std::unique_ptr<TransformableFrameInterface> frame) {
MutexLock lock(&send_lock_);
if (!send_frame_callback_)
return;
rtc::scoped_refptr<ChannelSendFrameTransformerDelegate> delegate(this);
encoder_queue_->PostTask(
[delegate = std::move(delegate), frame = std::move(frame)]() mutable {
delegate->SendFrame(std::move(frame));
});
}
void ChannelSendFrameTransformerDelegate::SendFrame(
std::unique_ptr<TransformableFrameInterface> frame) const {
MutexLock lock(&send_lock_);
RTC_DCHECK_RUN_ON(encoder_queue_);
RTC_CHECK_EQ(frame->GetDirection(),
TransformableFrameInterface::Direction::kSender);
if (!send_frame_callback_)
return;
auto* transformed_frame =
static_cast<TransformableOutgoingAudioFrame*>(frame.get());
send_frame_callback_(transformed_frame->GetFrameType(),
transformed_frame->GetPayloadType(),
transformed_frame->GetTimestamp() -
transformed_frame->GetStartTimestamp(),
transformed_frame->GetData(),
transformed_frame->GetAbsoluteCaptureTimestampMs());
}
} // namespace webrtc
|
