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1 /* | 1 /* |
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 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 | 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 | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 #include "webrtc/config.h" | 10 #include "webrtc/config.h" |
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34 bool UlpfecConfig::operator==(const UlpfecConfig& other) const { | 34 bool UlpfecConfig::operator==(const UlpfecConfig& other) const { |
35 return ulpfec_payload_type == other.ulpfec_payload_type && | 35 return ulpfec_payload_type == other.ulpfec_payload_type && |
36 red_payload_type == other.red_payload_type && | 36 red_payload_type == other.red_payload_type && |
37 red_rtx_payload_type == other.red_rtx_payload_type; | 37 red_rtx_payload_type == other.red_rtx_payload_type; |
38 } | 38 } |
39 | 39 |
40 std::string RtpExtension::ToString() const { | 40 std::string RtpExtension::ToString() const { |
41 std::stringstream ss; | 41 std::stringstream ss; |
42 ss << "{uri: " << uri; | 42 ss << "{uri: " << uri; |
43 ss << ", id: " << id; | 43 ss << ", id: " << id; |
| 44 if (encrypt) { |
| 45 ss << ", encrypt"; |
| 46 } |
44 ss << '}'; | 47 ss << '}'; |
45 return ss.str(); | 48 return ss.str(); |
46 } | 49 } |
47 | 50 |
48 const char* RtpExtension::kAudioLevelUri = | 51 const char* RtpExtension::kAudioLevelUri = |
49 "urn:ietf:params:rtp-hdrext:ssrc-audio-level"; | 52 "urn:ietf:params:rtp-hdrext:ssrc-audio-level"; |
50 const int RtpExtension::kAudioLevelDefaultId = 1; | 53 const int RtpExtension::kAudioLevelDefaultId = 1; |
51 | 54 |
52 const char* RtpExtension::kTimestampOffsetUri = | 55 const char* RtpExtension::kTimestampOffsetUri = |
53 "urn:ietf:params:rtp-hdrext:toffset"; | 56 "urn:ietf:params:rtp-hdrext:toffset"; |
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65 const int RtpExtension::kTransportSequenceNumberDefaultId = 5; | 68 const int RtpExtension::kTransportSequenceNumberDefaultId = 5; |
66 | 69 |
67 // This extension allows applications to adaptively limit the playout delay | 70 // This extension allows applications to adaptively limit the playout delay |
68 // on frames as per the current needs. For example, a gaming application | 71 // on frames as per the current needs. For example, a gaming application |
69 // has very different needs on end-to-end delay compared to a video-conference | 72 // has very different needs on end-to-end delay compared to a video-conference |
70 // application. | 73 // application. |
71 const char* RtpExtension::kPlayoutDelayUri = | 74 const char* RtpExtension::kPlayoutDelayUri = |
72 "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay"; | 75 "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay"; |
73 const int RtpExtension::kPlayoutDelayDefaultId = 6; | 76 const int RtpExtension::kPlayoutDelayDefaultId = 6; |
74 | 77 |
| 78 const char* RtpExtension::kEncryptHeaderExtensionsUri = |
| 79 "urn:ietf:params:rtp-hdrext:encrypt"; |
| 80 |
75 const int RtpExtension::kMinId = 1; | 81 const int RtpExtension::kMinId = 1; |
76 const int RtpExtension::kMaxId = 14; | 82 const int RtpExtension::kMaxId = 14; |
77 | 83 |
78 bool RtpExtension::IsSupportedForAudio(const std::string& uri) { | 84 bool RtpExtension::IsSupportedForAudio(const std::string& uri) { |
79 return uri == webrtc::RtpExtension::kAudioLevelUri || | 85 return uri == webrtc::RtpExtension::kAudioLevelUri || |
80 uri == webrtc::RtpExtension::kTransportSequenceNumberUri; | 86 uri == webrtc::RtpExtension::kTransportSequenceNumberUri; |
81 } | 87 } |
82 | 88 |
83 bool RtpExtension::IsSupportedForVideo(const std::string& uri) { | 89 bool RtpExtension::IsSupportedForVideo(const std::string& uri) { |
84 return uri == webrtc::RtpExtension::kTimestampOffsetUri || | 90 return uri == webrtc::RtpExtension::kTimestampOffsetUri || |
85 uri == webrtc::RtpExtension::kAbsSendTimeUri || | 91 uri == webrtc::RtpExtension::kAbsSendTimeUri || |
86 uri == webrtc::RtpExtension::kVideoRotationUri || | 92 uri == webrtc::RtpExtension::kVideoRotationUri || |
87 uri == webrtc::RtpExtension::kTransportSequenceNumberUri || | 93 uri == webrtc::RtpExtension::kTransportSequenceNumberUri || |
88 uri == webrtc::RtpExtension::kPlayoutDelayUri; | 94 uri == webrtc::RtpExtension::kPlayoutDelayUri; |
89 } | 95 } |
90 | 96 |
| 97 bool RtpExtension::IsEncryptionSupported(const std::string& uri) { |
| 98 return uri == webrtc::RtpExtension::kAudioLevelUri || |
| 99 uri == webrtc::RtpExtension::kTimestampOffsetUri || |
| 100 #if !defined(ENABLE_EXTERNAL_AUTH) |
| 101 // TODO(jbauch): Figure out a way to always allow "kAbsSendTimeUri" |
| 102 // here and filter out later if external auth is really used in |
| 103 // srtpfilter. |
| 104 uri == webrtc::RtpExtension::kAbsSendTimeUri || |
| 105 #endif |
| 106 uri == webrtc::RtpExtension::kVideoRotationUri || |
| 107 uri == webrtc::RtpExtension::kTransportSequenceNumberUri || |
| 108 uri == webrtc::RtpExtension::kPlayoutDelayUri; |
| 109 } |
| 110 |
| 111 const RtpExtension* RtpExtension::FindHeaderExtensionByUri( |
| 112 const std::vector<RtpExtension>& extensions, |
| 113 const std::string& uri) { |
| 114 for (const auto& extension : extensions) { |
| 115 if (extension.uri == uri) { |
| 116 return &extension; |
| 117 } |
| 118 } |
| 119 return nullptr; |
| 120 } |
| 121 |
| 122 std::vector<RtpExtension> RtpExtension::FilterDuplicateNonEncrypted( |
| 123 const std::vector<RtpExtension>& extensions) { |
| 124 std::vector<RtpExtension> filtered; |
| 125 for (auto it = extensions.begin(); it != extensions.end(); ++it) { |
| 126 if (it->encrypt) { |
| 127 filtered.push_back(*it); |
| 128 } else { |
| 129 // Only add non-encrypted extension if no encrypted with the same URI |
| 130 // is also present... |
| 131 bool found = false; |
| 132 for (auto it2 = it + 1; it2 != extensions.end(); ++it2) { |
| 133 if (it->uri == it2->uri) { |
| 134 found = true; |
| 135 break; |
| 136 } |
| 137 } |
| 138 |
| 139 if (!found) { |
| 140 // ...and has not been added before. |
| 141 for (const RtpExtension& existing : filtered) { |
| 142 if (it->uri == existing.uri) { |
| 143 found = true; |
| 144 break; |
| 145 } |
| 146 } |
| 147 |
| 148 if (!found) { |
| 149 filtered.push_back(*it); |
| 150 } |
| 151 } |
| 152 } |
| 153 } |
| 154 return filtered; |
| 155 } |
| 156 |
91 VideoStream::VideoStream() | 157 VideoStream::VideoStream() |
92 : width(0), | 158 : width(0), |
93 height(0), | 159 height(0), |
94 max_framerate(-1), | 160 max_framerate(-1), |
95 min_bitrate_bps(-1), | 161 min_bitrate_bps(-1), |
96 target_bitrate_bps(-1), | 162 target_bitrate_bps(-1), |
97 max_bitrate_bps(-1), | 163 max_bitrate_bps(-1), |
98 max_qp(-1) {} | 164 max_qp(-1) {} |
99 | 165 |
100 VideoStream::~VideoStream() = default; | 166 VideoStream::~VideoStream() = default; |
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202 VideoEncoderConfig::Vp9EncoderSpecificSettings::Vp9EncoderSpecificSettings( | 268 VideoEncoderConfig::Vp9EncoderSpecificSettings::Vp9EncoderSpecificSettings( |
203 const VideoCodecVP9& specifics) | 269 const VideoCodecVP9& specifics) |
204 : specifics_(specifics) {} | 270 : specifics_(specifics) {} |
205 | 271 |
206 void VideoEncoderConfig::Vp9EncoderSpecificSettings::FillVideoCodecVp9( | 272 void VideoEncoderConfig::Vp9EncoderSpecificSettings::FillVideoCodecVp9( |
207 VideoCodecVP9* vp9_settings) const { | 273 VideoCodecVP9* vp9_settings) const { |
208 *vp9_settings = specifics_; | 274 *vp9_settings = specifics_; |
209 } | 275 } |
210 | 276 |
211 } // namespace webrtc | 277 } // namespace webrtc |
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