1 | /* |
2 | * Copyright (C) 2018 Apple Inc. All rights reserved. |
3 | * |
4 | * Redistribution and use in source and binary forms, with or without |
5 | * modification, are permitted provided that the following conditions |
6 | * are met: |
7 | * 1. Redistributions of source code must retain the above copyright |
8 | * notice, this list of conditions and the following disclaimer. |
9 | * 2. Redistributions in binary form must reproduce the above copyright |
10 | * notice, this list of conditions and the following disclaimer in the |
11 | * documentation and/or other materials provided with the distribution. |
12 | * |
13 | * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS'' AND ANY |
14 | * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED |
15 | * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE |
16 | * DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS BE LIABLE FOR ANY |
17 | * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES |
18 | * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; |
19 | * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON |
20 | * ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT |
21 | * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS |
22 | * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. |
23 | */ |
24 | |
25 | #include "config.h" |
26 | #include "LibWebRTCUtils.h" |
27 | |
28 | #if USE(LIBWEBRTC) |
29 | |
30 | #include "LibWebRTCMacros.h" |
31 | #include "RTCPeerConnection.h" |
32 | #include "RTCRtpSendParameters.h" |
33 | #include <wtf/text/WTFString.h> |
34 | |
35 | ALLOW_UNUSED_PARAMETERS_BEGIN |
36 | |
37 | #include <webrtc/api/rtpparameters.h> |
38 | #include <webrtc/api/rtptransceiverinterface.h> |
39 | |
40 | ALLOW_UNUSED_PARAMETERS_END |
41 | |
42 | namespace WebCore { |
43 | |
44 | static inline RTCRtpEncodingParameters toRTCEncodingParameters(const webrtc::RtpEncodingParameters& rtcParameters) |
45 | { |
46 | RTCRtpEncodingParameters parameters; |
47 | |
48 | if (rtcParameters.ssrc) |
49 | parameters.ssrc = *rtcParameters.ssrc; |
50 | if (rtcParameters.rtx && rtcParameters.rtx->ssrc) |
51 | parameters.rtx.ssrc = *rtcParameters.rtx->ssrc; |
52 | if (rtcParameters.fec && rtcParameters.fec->ssrc) |
53 | parameters.fec.ssrc = *rtcParameters.fec->ssrc; |
54 | if (rtcParameters.dtx) { |
55 | switch (*rtcParameters.dtx) { |
56 | case webrtc::DtxStatus::DISABLED: |
57 | parameters.dtx = RTCDtxStatus::Disabled; |
58 | break; |
59 | case webrtc::DtxStatus::ENABLED: |
60 | parameters.dtx = RTCDtxStatus::Enabled; |
61 | } |
62 | } |
63 | parameters.active = rtcParameters.active; |
64 | if (rtcParameters.max_bitrate_bps) |
65 | parameters.maxBitrate = *rtcParameters.max_bitrate_bps; |
66 | if (rtcParameters.max_framerate) |
67 | parameters.maxFramerate = *rtcParameters.max_framerate; |
68 | parameters.rid = fromStdString(rtcParameters.rid); |
69 | if (rtcParameters.scale_resolution_down_by) |
70 | parameters.scaleResolutionDownBy = *rtcParameters.scale_resolution_down_by; |
71 | |
72 | return parameters; |
73 | } |
74 | |
75 | static inline RTCRtpHeaderExtensionParameters (const webrtc::RtpHeaderExtensionParameters& rtcParameters) |
76 | { |
77 | RTCRtpHeaderExtensionParameters parameters; |
78 | |
79 | parameters.uri = fromStdString(rtcParameters.uri); |
80 | parameters.id = rtcParameters.id; |
81 | |
82 | return parameters; |
83 | } |
84 | |
85 | static inline webrtc::RtpHeaderExtensionParameters (const RTCRtpHeaderExtensionParameters& parameters) |
86 | { |
87 | webrtc::RtpHeaderExtensionParameters rtcParameters; |
88 | |
89 | rtcParameters.uri = parameters.uri.utf8().data(); |
90 | rtcParameters.id = parameters.id; |
91 | |
92 | return rtcParameters; |
93 | } |
94 | |
95 | static inline RTCRtpCodecParameters toRTCCodecParameters(const webrtc::RtpCodecParameters& rtcParameters) |
96 | { |
97 | RTCRtpCodecParameters parameters; |
98 | |
99 | parameters.payloadType = rtcParameters.payload_type; |
100 | parameters.mimeType = fromStdString(rtcParameters.mime_type()); |
101 | if (rtcParameters.clock_rate) |
102 | parameters.clockRate = *rtcParameters.clock_rate; |
103 | if (rtcParameters.num_channels) |
104 | parameters.channels = *rtcParameters.num_channels; |
105 | |
106 | StringBuilder sdpFmtpLineBuilder; |
107 | sdpFmtpLineBuilder.appendLiteral("a=fmtp:" ); |
108 | sdpFmtpLineBuilder.appendNumber(parameters.payloadType); |
109 | sdpFmtpLineBuilder.append(' '); |
110 | |
111 | bool isFirst = true; |
112 | for (auto& keyValue : rtcParameters.parameters) { |
113 | if (!isFirst) |
114 | sdpFmtpLineBuilder.append(';'); |
115 | else |
116 | isFirst = false; |
117 | |
118 | sdpFmtpLineBuilder.append(StringView { keyValue.first.c_str() }); |
119 | sdpFmtpLineBuilder.append('='); |
120 | sdpFmtpLineBuilder.append(StringView { keyValue.second.c_str() }); |
121 | } |
122 | parameters.sdpFmtpLine = sdpFmtpLineBuilder.toString(); |
123 | |
124 | return parameters; |
125 | } |
126 | |
127 | RTCRtpParameters toRTCRtpParameters(const webrtc::RtpParameters& rtcParameters) |
128 | { |
129 | RTCRtpParameters parameters; |
130 | |
131 | for (auto& extension : rtcParameters.header_extensions) |
132 | parameters.headerExtensions.append(toRTCHeaderExtensionParameters(extension)); |
133 | for (auto& codec : rtcParameters.codecs) |
134 | parameters.codecs.append(toRTCCodecParameters(codec)); |
135 | |
136 | return parameters; |
137 | } |
138 | |
139 | RTCRtpSendParameters toRTCRtpSendParameters(const webrtc::RtpParameters& rtcParameters) |
140 | { |
141 | RTCRtpSendParameters parameters { toRTCRtpParameters(rtcParameters) }; |
142 | |
143 | parameters.transactionId = fromStdString(rtcParameters.transaction_id); |
144 | for (auto& rtcEncoding : rtcParameters.encodings) |
145 | parameters.encodings.append(toRTCEncodingParameters(rtcEncoding)); |
146 | |
147 | switch (rtcParameters.degradation_preference) { |
148 | // FIXME: Support DegradationPreference::DISABLED. |
149 | case webrtc::DegradationPreference::DISABLED: |
150 | case webrtc::DegradationPreference::MAINTAIN_FRAMERATE: |
151 | parameters.degradationPreference = RTCDegradationPreference::MaintainFramerate; |
152 | break; |
153 | case webrtc::DegradationPreference::MAINTAIN_RESOLUTION: |
154 | parameters.degradationPreference = RTCDegradationPreference::MaintainResolution; |
155 | break; |
156 | case webrtc::DegradationPreference::BALANCED: |
157 | parameters.degradationPreference = RTCDegradationPreference::Balanced; |
158 | break; |
159 | }; |
160 | return parameters; |
161 | } |
162 | |
163 | void updateRTCRtpSendParameters(const RTCRtpSendParameters& parameters, webrtc::RtpParameters& rtcParameters) |
164 | { |
165 | rtcParameters.transaction_id = parameters.transactionId.utf8().data(); |
166 | |
167 | if (parameters.encodings.size() != rtcParameters.encodings.size()) { |
168 | // If encodings size is different, setting parameters will fail. Let's make it so. |
169 | rtcParameters.encodings.clear(); |
170 | return; |
171 | } |
172 | |
173 | // We copy all current encodings parameters and only update parameters that can actually be usefully updated. |
174 | for (size_t i = 0; i < parameters.encodings.size(); ++i) { |
175 | rtcParameters.encodings[i].active = parameters.encodings[i].active; |
176 | if (parameters.encodings[i].maxBitrate) |
177 | rtcParameters.encodings[i].max_bitrate_bps = parameters.encodings[i].maxBitrate; |
178 | if (parameters.encodings[i].maxFramerate) |
179 | rtcParameters.encodings[i].max_framerate = parameters.encodings[i].maxFramerate; |
180 | } |
181 | |
182 | rtcParameters.header_extensions.clear(); |
183 | for (auto& extension : parameters.headerExtensions) |
184 | rtcParameters.header_extensions.push_back(fromRTCHeaderExtensionParameters(extension)); |
185 | // Codecs parameters are readonly |
186 | |
187 | switch (parameters.degradationPreference) { |
188 | case RTCDegradationPreference::MaintainFramerate: |
189 | rtcParameters.degradation_preference = webrtc::DegradationPreference::MAINTAIN_FRAMERATE; |
190 | break; |
191 | case RTCDegradationPreference::MaintainResolution: |
192 | rtcParameters.degradation_preference = webrtc::DegradationPreference::MAINTAIN_RESOLUTION; |
193 | break; |
194 | case RTCDegradationPreference::Balanced: |
195 | rtcParameters.degradation_preference = webrtc::DegradationPreference::BALANCED; |
196 | break; |
197 | } |
198 | } |
199 | |
200 | RTCRtpTransceiverDirection toRTCRtpTransceiverDirection(webrtc::RtpTransceiverDirection rtcDirection) |
201 | { |
202 | switch (rtcDirection) { |
203 | case webrtc::RtpTransceiverDirection::kSendRecv: |
204 | return RTCRtpTransceiverDirection::Sendrecv; |
205 | case webrtc::RtpTransceiverDirection::kSendOnly: |
206 | return RTCRtpTransceiverDirection::Sendonly; |
207 | case webrtc::RtpTransceiverDirection::kRecvOnly: |
208 | return RTCRtpTransceiverDirection::Recvonly; |
209 | case webrtc::RtpTransceiverDirection::kInactive: |
210 | return RTCRtpTransceiverDirection::Inactive; |
211 | }; |
212 | |
213 | RELEASE_ASSERT_NOT_REACHED(); |
214 | } |
215 | |
216 | webrtc::RtpTransceiverDirection fromRTCRtpTransceiverDirection(RTCRtpTransceiverDirection direction) |
217 | { |
218 | switch (direction) { |
219 | case RTCRtpTransceiverDirection::Sendrecv: |
220 | return webrtc::RtpTransceiverDirection::kSendRecv; |
221 | case RTCRtpTransceiverDirection::Sendonly: |
222 | return webrtc::RtpTransceiverDirection::kSendOnly; |
223 | case RTCRtpTransceiverDirection::Recvonly: |
224 | return webrtc::RtpTransceiverDirection::kRecvOnly; |
225 | case RTCRtpTransceiverDirection::Inactive: |
226 | return webrtc::RtpTransceiverDirection::kInactive; |
227 | }; |
228 | |
229 | RELEASE_ASSERT_NOT_REACHED(); |
230 | } |
231 | |
232 | webrtc::RtpTransceiverInit fromRtpTransceiverInit(const RTCRtpTransceiverInit& init) |
233 | { |
234 | webrtc::RtpTransceiverInit rtcInit; |
235 | rtcInit.direction = fromRTCRtpTransceiverDirection(init.direction); |
236 | for (auto& stream : init.streams) |
237 | rtcInit.stream_ids.push_back(stream->id().utf8().data()); |
238 | return rtcInit; |
239 | } |
240 | |
241 | } // namespace WebCore |
242 | |
243 | #endif // USE(LIBWEBRTC) |
244 | |