1 /*
2 * Copyright (c) 2004 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 "media/base/media_engine.h"
12
13 #include <stddef.h>
14
15 #include <cstdint>
16 #include <string>
17 #include <utility>
18
19 #include "absl/algorithm/container.h"
20 #include "api/video/video_bitrate_allocation.h"
21 #include "rtc_base/checks.h"
22 #include "rtc_base/string_encode.h"
23
24 namespace cricket {
25
26 RtpCapabilities::RtpCapabilities() = default;
27 RtpCapabilities::~RtpCapabilities() = default;
28
CreateRtpParametersWithOneEncoding()29 webrtc::RtpParameters CreateRtpParametersWithOneEncoding() {
30 webrtc::RtpParameters parameters;
31 webrtc::RtpEncodingParameters encoding;
32 parameters.encodings.push_back(encoding);
33 return parameters;
34 }
35
CreateRtpParametersWithEncodings(StreamParams sp)36 webrtc::RtpParameters CreateRtpParametersWithEncodings(StreamParams sp) {
37 std::vector<uint32_t> primary_ssrcs;
38 sp.GetPrimarySsrcs(&primary_ssrcs);
39 size_t encoding_count = primary_ssrcs.size();
40
41 std::vector<webrtc::RtpEncodingParameters> encodings(encoding_count);
42 for (size_t i = 0; i < encodings.size(); ++i) {
43 encodings[i].ssrc = primary_ssrcs[i];
44 }
45
46 const std::vector<RidDescription>& rids = sp.rids();
47 RTC_DCHECK(rids.size() == 0 || rids.size() == encoding_count);
48 for (size_t i = 0; i < rids.size(); ++i) {
49 encodings[i].rid = rids[i].rid;
50 }
51
52 webrtc::RtpParameters parameters;
53 parameters.encodings = encodings;
54 parameters.rtcp.cname = sp.cname;
55 return parameters;
56 }
57
GetDefaultEnabledRtpHeaderExtensions(const RtpHeaderExtensionQueryInterface & query_interface)58 std::vector<webrtc::RtpExtension> GetDefaultEnabledRtpHeaderExtensions(
59 const RtpHeaderExtensionQueryInterface& query_interface) {
60 std::vector<webrtc::RtpExtension> extensions;
61 for (const auto& entry : query_interface.GetRtpHeaderExtensions()) {
62 if (entry.direction != webrtc::RtpTransceiverDirection::kStopped)
63 extensions.emplace_back(entry.uri, *entry.preferred_id);
64 }
65 return extensions;
66 }
67
CheckRtpParametersValues(const webrtc::RtpParameters & rtp_parameters)68 webrtc::RTCError CheckRtpParametersValues(
69 const webrtc::RtpParameters& rtp_parameters) {
70 using webrtc::RTCErrorType;
71
72 for (size_t i = 0; i < rtp_parameters.encodings.size(); ++i) {
73 if (rtp_parameters.encodings[i].bitrate_priority <= 0) {
74 LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_RANGE,
75 "Attempted to set RtpParameters bitrate_priority to "
76 "an invalid number. bitrate_priority must be > 0.");
77 }
78 if (rtp_parameters.encodings[i].scale_resolution_down_by &&
79 *rtp_parameters.encodings[i].scale_resolution_down_by < 1.0) {
80 LOG_AND_RETURN_ERROR(
81 RTCErrorType::INVALID_RANGE,
82 "Attempted to set RtpParameters scale_resolution_down_by to an "
83 "invalid value. scale_resolution_down_by must be >= 1.0");
84 }
85 if (rtp_parameters.encodings[i].max_framerate &&
86 *rtp_parameters.encodings[i].max_framerate < 0.0) {
87 LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_RANGE,
88 "Attempted to set RtpParameters max_framerate to an "
89 "invalid value. max_framerate must be >= 0.0");
90 }
91 if (rtp_parameters.encodings[i].min_bitrate_bps &&
92 rtp_parameters.encodings[i].max_bitrate_bps) {
93 if (*rtp_parameters.encodings[i].max_bitrate_bps <
94 *rtp_parameters.encodings[i].min_bitrate_bps) {
95 LOG_AND_RETURN_ERROR(webrtc::RTCErrorType::INVALID_RANGE,
96 "Attempted to set RtpParameters min bitrate "
97 "larger than max bitrate.");
98 }
99 }
100 if (rtp_parameters.encodings[i].num_temporal_layers) {
101 if (*rtp_parameters.encodings[i].num_temporal_layers < 1 ||
102 *rtp_parameters.encodings[i].num_temporal_layers >
103 webrtc::kMaxTemporalStreams) {
104 LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_RANGE,
105 "Attempted to set RtpParameters "
106 "num_temporal_layers to an invalid number.");
107 }
108 }
109 if (i > 0 && (rtp_parameters.encodings[i].num_temporal_layers !=
110 rtp_parameters.encodings[i - 1].num_temporal_layers)) {
111 LOG_AND_RETURN_ERROR(
112 RTCErrorType::INVALID_MODIFICATION,
113 "Attempted to set RtpParameters num_temporal_layers "
114 "at encoding layer i: " +
115 rtc::ToString(i) +
116 " to a different value than other encoding layers.");
117 }
118 }
119
120 return webrtc::RTCError::OK();
121 }
122
CheckRtpParametersInvalidModificationAndValues(const webrtc::RtpParameters & old_rtp_parameters,const webrtc::RtpParameters & rtp_parameters)123 webrtc::RTCError CheckRtpParametersInvalidModificationAndValues(
124 const webrtc::RtpParameters& old_rtp_parameters,
125 const webrtc::RtpParameters& rtp_parameters) {
126 using webrtc::RTCErrorType;
127 if (rtp_parameters.encodings.size() != old_rtp_parameters.encodings.size()) {
128 LOG_AND_RETURN_ERROR(
129 RTCErrorType::INVALID_MODIFICATION,
130 "Attempted to set RtpParameters with different encoding count");
131 }
132 if (rtp_parameters.rtcp != old_rtp_parameters.rtcp) {
133 LOG_AND_RETURN_ERROR(
134 RTCErrorType::INVALID_MODIFICATION,
135 "Attempted to set RtpParameters with modified RTCP parameters");
136 }
137 if (rtp_parameters.header_extensions !=
138 old_rtp_parameters.header_extensions) {
139 LOG_AND_RETURN_ERROR(
140 RTCErrorType::INVALID_MODIFICATION,
141 "Attempted to set RtpParameters with modified header extensions");
142 }
143 if (!absl::c_equal(old_rtp_parameters.encodings, rtp_parameters.encodings,
144 [](const webrtc::RtpEncodingParameters& encoding1,
145 const webrtc::RtpEncodingParameters& encoding2) {
146 return encoding1.rid == encoding2.rid;
147 })) {
148 LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
149 "Attempted to change RID values in the encodings.");
150 }
151 if (!absl::c_equal(old_rtp_parameters.encodings, rtp_parameters.encodings,
152 [](const webrtc::RtpEncodingParameters& encoding1,
153 const webrtc::RtpEncodingParameters& encoding2) {
154 return encoding1.ssrc == encoding2.ssrc;
155 })) {
156 LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
157 "Attempted to set RtpParameters with modified SSRC");
158 }
159
160 return CheckRtpParametersValues(rtp_parameters);
161 }
162
CompositeMediaEngine(std::unique_ptr<VoiceEngineInterface> voice_engine,std::unique_ptr<VideoEngineInterface> video_engine)163 CompositeMediaEngine::CompositeMediaEngine(
164 std::unique_ptr<VoiceEngineInterface> voice_engine,
165 std::unique_ptr<VideoEngineInterface> video_engine)
166 : voice_engine_(std::move(voice_engine)),
167 video_engine_(std::move(video_engine)) {}
168
169 CompositeMediaEngine::~CompositeMediaEngine() = default;
170
Init()171 bool CompositeMediaEngine::Init() {
172 voice().Init();
173 return true;
174 }
175
voice()176 VoiceEngineInterface& CompositeMediaEngine::voice() {
177 return *voice_engine_.get();
178 }
179
video()180 VideoEngineInterface& CompositeMediaEngine::video() {
181 return *video_engine_.get();
182 }
183
voice() const184 const VoiceEngineInterface& CompositeMediaEngine::voice() const {
185 return *voice_engine_.get();
186 }
187
video() const188 const VideoEngineInterface& CompositeMediaEngine::video() const {
189 return *video_engine_.get();
190 }
191
192 } // namespace cricket
193