Support encrypted RTP extensions (RFC 6904)

webrtc/config.cc

 /*
  *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 #include "webrtc/config.h"
@@ skipping 23 matching lines @@
 bool UlpfecConfig::operator==(const UlpfecConfig& other) const {
   return ulpfec_payload_type == other.ulpfec_payload_type &&
          red_payload_type == other.red_payload_type &&
          red_rtx_payload_type == other.red_rtx_payload_type;
 }
 
 std::string RtpExtension::ToString() const {
   std::stringstream ss;
   ss << "{uri: " << uri;
   ss << ", id: " << id;
+  if (encrypt) {
+    ss << ", encrypt";
+  }
   ss << '}';
   return ss.str();
 }
 
 const char* RtpExtension::kAudioLevelUri =
     "urn:ietf:params:rtp-hdrext:ssrc-audio-level";
 const int RtpExtension::kAudioLevelDefaultId = 1;
 
 const char* RtpExtension::kTimestampOffsetUri =
     "urn:ietf:params:rtp-hdrext:toffset";
@@ skipping 11 matching lines @@
 const int RtpExtension::kTransportSequenceNumberDefaultId = 5;
 
 // This extension allows applications to adaptively limit the playout delay
 // on frames as per the current needs. For example, a gaming application
 // has very different needs on end-to-end delay compared to a video-conference
 // application.
 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;
 
 bool RtpExtension::IsSupportedForAudio(const std::string& uri) {
   return uri == webrtc::RtpExtension::kAudioLevelUri ||
          uri == webrtc::RtpExtension::kTransportSequenceNumberUri;
 }
 
 bool RtpExtension::IsSupportedForVideo(const std::string& uri) {
   return uri == webrtc::RtpExtension::kTimestampOffsetUri ||
          uri == webrtc::RtpExtension::kAbsSendTimeUri ||
          uri == webrtc::RtpExtension::kVideoRotationUri ||
          uri == webrtc::RtpExtension::kTransportSequenceNumberUri ||
          uri == webrtc::RtpExtension::kPlayoutDelayUri;
 }
 
+bool RtpExtension::IsEncryptionSupported(const std::string& uri) {
+  // 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;
+}
+
+const RtpExtension* RtpExtension::FindHeaderExtensionByUri(
+    const std::vector<RtpExtension>& extensions,
+    const std::string& uri) {
+  for (const auto& extension : extensions) {
+    if (extension.uri == uri)

[Taylor Brandstetter, 2017/03/23 20:10:56]

nit: Our style is to use {}s even for single-line if statements

[joachim, 2017/03/30 22:43:49]

Done (mine too, looks like I missed this after moving the function from
channel.cc).

+      return &extension;
+  }
+  return nullptr;
+}
+
 VideoStream::VideoStream()
     : width(0),
       height(0),
       max_framerate(-1),
       min_bitrate_bps(-1),
       target_bitrate_bps(-1),
       max_bitrate_bps(-1),
       max_qp(-1) {}
 
 VideoStream::~VideoStream() = default;
@@ skipping 101 matching lines @@
 VideoEncoderConfig::Vp9EncoderSpecificSettings::Vp9EncoderSpecificSettings(
     const VideoCodecVP9& specifics)
     : specifics_(specifics) {}
 
 void VideoEncoderConfig::Vp9EncoderSpecificSettings::FillVideoCodecVp9(
     VideoCodecVP9* vp9_settings) const {
   *vp9_settings = specifics_;
 }
 
 }  // namespace webrtc