Chromium Code Reviews| Index: webrtc/config.cc |
| diff --git a/webrtc/config.cc b/webrtc/config.cc |
| index e0c490d1ecd8038d546b7298fc9a885ab02718df..b35325b07f4d4d4062f6d9f0b0b13a5260548a89 100644 |
| --- a/webrtc/config.cc |
| +++ b/webrtc/config.cc |
| @@ -41,6 +41,9 @@ std::string RtpExtension::ToString() const { |
| std::stringstream ss; |
| ss << "{uri: " << uri; |
| ss << ", id: " << id; |
| + if (encrypted) { |
| + ss << ", encrypted"; |
| + } |
| ss << '}'; |
| return ss.str(); |
| } |
| @@ -72,6 +75,9 @@ const char* RtpExtension::kPlayoutDelayUri = |
| "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay"; |
| const int RtpExtension::kPlayoutDelayDefaultId = 6; |
| +const char* RtpExtension::kEncryptHeaderExtensionsUri = |
| + "urn:ietf:params:rtp-hdrext:encrypt"; |
| + |
| const int RtpExtension::kMinId = 1; |
| const int RtpExtension::kMaxId = 14; |
| @@ -88,6 +94,16 @@ bool RtpExtension::IsSupportedForVideo(const std::string& uri) { |
| uri == webrtc::RtpExtension::kPlayoutDelayUri; |
| } |
| +bool RtpExtension::AllowEncrypt(const std::string& uri) { |
|
pthatcher1
2017/03/21 07:07:06
A more consistent name would be "IsEncryptionSuppo
joachim
2017/03/23 00:04:32
Done.
|
| + // TODO(jbauch): Figure out a way to add "kTimestampOffsetUri" here |
| + // and filter out later if external auth is used in srtpfilter. |
| + return uri == webrtc::RtpExtension::kAudioLevelUri || |
| + uri == webrtc::RtpExtension::kTimestampOffsetUri || |
| + uri == webrtc::RtpExtension::kVideoRotationUri || |
| + uri == webrtc::RtpExtension::kTransportSequenceNumberUri || |
| + uri == webrtc::RtpExtension::kPlayoutDelayUri; |
| +} |
| + |
| VideoStream::VideoStream() |
| : width(0), |
| height(0), |