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Side by Side Diff: talk/app/webrtc/jsepsessiondescription.cc

Issue 1610243002: Move talk/app/webrtc to webrtc/api (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Removed processing of api.gyp for Chromium builds Created 4 years, 10 months ago
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1 /*
2 * libjingle
3 * Copyright 2012 Google Inc.
4 *
5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met:
7 *
8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice,
11 * this list of conditions and the following disclaimer in the documentation
12 * and/or other materials provided with the distribution.
13 * 3. The name of the author may not be used to endorse or promote products
14 * derived from this software without specific prior written permission.
15 *
16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
26 */
27
28 #include "talk/app/webrtc/jsepsessiondescription.h"
29
30 #include "talk/app/webrtc/webrtcsdp.h"
31 #include "talk/session/media/mediasession.h"
32 #include "webrtc/base/arraysize.h"
33 #include "webrtc/base/stringencode.h"
34
35 using rtc::scoped_ptr;
36 using cricket::SessionDescription;
37
38 namespace webrtc {
39
40 static const char* kSupportedTypes[] = {
41 JsepSessionDescription::kOffer,
42 JsepSessionDescription::kPrAnswer,
43 JsepSessionDescription::kAnswer
44 };
45
46 static bool IsTypeSupported(const std::string& type) {
47 bool type_supported = false;
48 for (size_t i = 0; i < arraysize(kSupportedTypes); ++i) {
49 if (kSupportedTypes[i] == type) {
50 type_supported = true;
51 break;
52 }
53 }
54 return type_supported;
55 }
56
57 const char SessionDescriptionInterface::kOffer[] = "offer";
58 const char SessionDescriptionInterface::kPrAnswer[] = "pranswer";
59 const char SessionDescriptionInterface::kAnswer[] = "answer";
60
61 const int JsepSessionDescription::kDefaultVideoCodecId = 100;
62 // This is effectively a max value of the frame rate. 30 is default from camera.
63 const int JsepSessionDescription::kDefaultVideoCodecFramerate = 60;
64 const char JsepSessionDescription::kDefaultVideoCodecName[] = "VP8";
65 // Used as default max video codec size before we have it in signaling.
66 #if defined(ANDROID) || defined(WEBRTC_IOS)
67 // Limit default max video codec size for Android to avoid
68 // HW VP8 codec initialization failure for resolutions higher
69 // than 1280x720 or 720x1280.
70 // Same patch for iOS to support 720P in portrait mode.
71 const int JsepSessionDescription::kMaxVideoCodecWidth = 1280;
72 const int JsepSessionDescription::kMaxVideoCodecHeight = 1280;
73 #else
74 const int JsepSessionDescription::kMaxVideoCodecWidth = 1920;
75 const int JsepSessionDescription::kMaxVideoCodecHeight = 1080;
76 #endif
77 const int JsepSessionDescription::kDefaultVideoCodecPreference = 1;
78
79 SessionDescriptionInterface* CreateSessionDescription(const std::string& type,
80 const std::string& sdp,
81 SdpParseError* error) {
82 if (!IsTypeSupported(type)) {
83 return NULL;
84 }
85
86 JsepSessionDescription* jsep_desc = new JsepSessionDescription(type);
87 if (!jsep_desc->Initialize(sdp, error)) {
88 delete jsep_desc;
89 return NULL;
90 }
91 return jsep_desc;
92 }
93
94 JsepSessionDescription::JsepSessionDescription(const std::string& type)
95 : type_(type) {
96 }
97
98 JsepSessionDescription::~JsepSessionDescription() {}
99
100 bool JsepSessionDescription::Initialize(
101 cricket::SessionDescription* description,
102 const std::string& session_id,
103 const std::string& session_version) {
104 if (!description)
105 return false;
106
107 session_id_ = session_id;
108 session_version_ = session_version;
109 description_.reset(description);
110 candidate_collection_.resize(number_of_mediasections());
111 return true;
112 }
113
114 bool JsepSessionDescription::Initialize(const std::string& sdp,
115 SdpParseError* error) {
116 return SdpDeserialize(sdp, this, error);
117 }
118
119 bool JsepSessionDescription::AddCandidate(
120 const IceCandidateInterface* candidate) {
121 if (!candidate || candidate->sdp_mline_index() < 0)
122 return false;
123 size_t mediasection_index = 0;
124 if (!GetMediasectionIndex(candidate, &mediasection_index)) {
125 return false;
126 }
127 if (mediasection_index >= number_of_mediasections())
128 return false;
129 const std::string& content_name =
130 description_->contents()[mediasection_index].name;
131 const cricket::TransportInfo* transport_info =
132 description_->GetTransportInfoByName(content_name);
133 if (!transport_info) {
134 return false;
135 }
136
137 cricket::Candidate updated_candidate = candidate->candidate();
138 if (updated_candidate.username().empty()) {
139 updated_candidate.set_username(transport_info->description.ice_ufrag);
140 }
141 if (updated_candidate.password().empty()) {
142 updated_candidate.set_password(transport_info->description.ice_pwd);
143 }
144
145 scoped_ptr<JsepIceCandidate> updated_candidate_wrapper(
146 new JsepIceCandidate(candidate->sdp_mid(),
147 static_cast<int>(mediasection_index),
148 updated_candidate));
149 if (!candidate_collection_[mediasection_index].HasCandidate(
150 updated_candidate_wrapper.get()))
151 candidate_collection_[mediasection_index].add(
152 updated_candidate_wrapper.release());
153
154 return true;
155 }
156
157 size_t JsepSessionDescription::number_of_mediasections() const {
158 if (!description_)
159 return 0;
160 return description_->contents().size();
161 }
162
163 const IceCandidateCollection* JsepSessionDescription::candidates(
164 size_t mediasection_index) const {
165 if (mediasection_index >= candidate_collection_.size())
166 return NULL;
167 return &candidate_collection_[mediasection_index];
168 }
169
170 bool JsepSessionDescription::ToString(std::string* out) const {
171 if (!description_ || !out)
172 return false;
173 *out = SdpSerialize(*this);
174 return !out->empty();
175 }
176
177 bool JsepSessionDescription::GetMediasectionIndex(
178 const IceCandidateInterface* candidate,
179 size_t* index) {
180 if (!candidate || !index) {
181 return false;
182 }
183 *index = static_cast<size_t>(candidate->sdp_mline_index());
184 if (description_ && !candidate->sdp_mid().empty()) {
185 bool found = false;
186 // Try to match the sdp_mid with content name.
187 for (size_t i = 0; i < description_->contents().size(); ++i) {
188 if (candidate->sdp_mid() == description_->contents().at(i).name) {
189 *index = i;
190 found = true;
191 break;
192 }
193 }
194 if (!found) {
195 // If the sdp_mid is presented but we can't find a match, we consider
196 // this as an error.
197 return false;
198 }
199 }
200 return true;
201 }
202
203 } // namespace webrtc
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