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
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
|
/*
* Copyright (c) 2015 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 "test/layer_filtering_transport.h"
#include <string.h>
#include <algorithm>
#include <memory>
#include <utility>
#include "api/rtp_headers.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/create_video_rtp_depacketizer.h"
#include "modules/rtp_rtcp/source/rtp_video_header.h"
#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
#include "modules/video_coding/codecs/interface/common_constants.h"
#include "modules/video_coding/codecs/vp8/include/vp8_globals.h"
#include "modules/video_coding/codecs/vp9/include/vp9_globals.h"
#include "rtc_base/checks.h"
namespace webrtc {
namespace test {
LayerFilteringTransport::LayerFilteringTransport(
TaskQueueBase* task_queue,
std::unique_ptr<SimulatedPacketReceiverInterface> pipe,
Call* send_call,
uint8_t vp8_video_payload_type,
uint8_t vp9_video_payload_type,
int selected_tl,
int selected_sl,
const std::map<uint8_t, MediaType>& payload_type_map,
uint32_t ssrc_to_filter_min,
uint32_t ssrc_to_filter_max,
rtc::ArrayView<const RtpExtension> audio_extensions,
rtc::ArrayView<const RtpExtension> video_extensions)
: DirectTransport(task_queue,
std::move(pipe),
send_call,
payload_type_map,
audio_extensions,
video_extensions),
vp8_video_payload_type_(vp8_video_payload_type),
vp9_video_payload_type_(vp9_video_payload_type),
vp8_depacketizer_(CreateVideoRtpDepacketizer(kVideoCodecVP8)),
vp9_depacketizer_(CreateVideoRtpDepacketizer(kVideoCodecVP9)),
selected_tl_(selected_tl),
selected_sl_(selected_sl),
discarded_last_packet_(false),
ssrc_to_filter_min_(ssrc_to_filter_min),
ssrc_to_filter_max_(ssrc_to_filter_max) {}
LayerFilteringTransport::LayerFilteringTransport(
TaskQueueBase* task_queue,
std::unique_ptr<SimulatedPacketReceiverInterface> pipe,
Call* send_call,
uint8_t vp8_video_payload_type,
uint8_t vp9_video_payload_type,
int selected_tl,
int selected_sl,
const std::map<uint8_t, MediaType>& payload_type_map,
rtc::ArrayView<const RtpExtension> audio_extensions,
rtc::ArrayView<const RtpExtension> video_extensions)
: LayerFilteringTransport(task_queue,
std::move(pipe),
send_call,
vp8_video_payload_type,
vp9_video_payload_type,
selected_tl,
selected_sl,
payload_type_map,
/*ssrc_to_filter_min=*/0,
/*ssrc_to_filter_max=*/0xFFFFFFFF,
audio_extensions,
video_extensions) {}
bool LayerFilteringTransport::DiscardedLastPacket() const {
return discarded_last_packet_;
}
bool LayerFilteringTransport::SendRtp(rtc::ArrayView<const uint8_t> packet,
const PacketOptions& options) {
if (selected_tl_ == -1 && selected_sl_ == -1) {
// Nothing to change, forward the packet immediately.
return test::DirectTransport::SendRtp(packet, options);
}
RtpPacket rtp_packet;
rtp_packet.Parse(packet);
if (rtp_packet.Ssrc() < ssrc_to_filter_min_ ||
rtp_packet.Ssrc() > ssrc_to_filter_max_) {
// Nothing to change, forward the packet immediately.
return test::DirectTransport::SendRtp(packet, options);
}
if (rtp_packet.PayloadType() == vp8_video_payload_type_ ||
rtp_packet.PayloadType() == vp9_video_payload_type_) {
const bool is_vp8 = rtp_packet.PayloadType() == vp8_video_payload_type_;
VideoRtpDepacketizer& depacketizer =
is_vp8 ? *vp8_depacketizer_ : *vp9_depacketizer_;
if (auto parsed_payload = depacketizer.Parse(rtp_packet.PayloadBuffer())) {
int temporal_idx;
int spatial_idx;
bool non_ref_for_inter_layer_pred;
bool end_of_frame;
if (is_vp8) {
temporal_idx = absl::get<RTPVideoHeaderVP8>(
parsed_payload->video_header.video_type_header)
.temporalIdx;
spatial_idx = kNoSpatialIdx;
num_active_spatial_layers_ = 1;
non_ref_for_inter_layer_pred = false;
end_of_frame = true;
} else {
const auto& vp9_header = absl::get<RTPVideoHeaderVP9>(
parsed_payload->video_header.video_type_header);
temporal_idx = vp9_header.temporal_idx;
spatial_idx = vp9_header.spatial_idx;
non_ref_for_inter_layer_pred = vp9_header.non_ref_for_inter_layer_pred;
end_of_frame = vp9_header.end_of_frame;
if (vp9_header.ss_data_available) {
RTC_DCHECK(vp9_header.temporal_idx == kNoTemporalIdx ||
vp9_header.temporal_idx == 0);
num_active_spatial_layers_ = vp9_header.num_spatial_layers;
}
}
if (spatial_idx == kNoSpatialIdx)
num_active_spatial_layers_ = 1;
RTC_CHECK_GT(num_active_spatial_layers_, 0);
if (selected_sl_ >= 0 &&
spatial_idx ==
std::min(num_active_spatial_layers_ - 1, selected_sl_) &&
end_of_frame) {
// This layer is now the last in the superframe.
rtp_packet.SetMarker(true);
} else {
const bool higher_temporal_layer =
(selected_tl_ >= 0 && temporal_idx != kNoTemporalIdx &&
temporal_idx > selected_tl_);
const bool higher_spatial_layer =
(selected_sl_ >= 0 && spatial_idx != kNoSpatialIdx &&
spatial_idx > selected_sl_);
// Filter out non-reference lower spatial layers since they are not
// needed for decoding of target spatial layer.
const bool lower_non_ref_spatial_layer =
(selected_sl_ >= 0 && spatial_idx != kNoSpatialIdx &&
spatial_idx <
std::min(num_active_spatial_layers_ - 1, selected_sl_) &&
non_ref_for_inter_layer_pred);
if (higher_temporal_layer || higher_spatial_layer ||
lower_non_ref_spatial_layer) {
// Truncate packet to a padding packet.
rtp_packet.SetPayloadSize(0);
rtp_packet.SetPadding(1);
rtp_packet.SetMarker(false);
discarded_last_packet_ = true;
}
}
} else {
RTC_DCHECK_NOTREACHED() << "Parse error";
}
}
return test::DirectTransport::SendRtp(rtp_packet, options);
}
} // namespace test
} // namespace webrtc
|
