1 files changed, 213 insertions, 0 deletions
diff --git a/third_party/libwebrtc/webrtc/api/rtpparameters.cc b/third_party/libwebrtc/webrtc/api/rtpparameters.cc
new file mode 100644
index 0000000000..550b2f917a
--- /dev/null
+++ b/third_party/libwebrtc/webrtc/api/rtpparameters.cc
@@ -0,0 +1,213 @@
+/*
+ *  Copyright (c) 2017 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 "api/rtpparameters.h"
+
+#include <algorithm>
+#include <sstream>
+#include <string>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+RtcpFeedback::RtcpFeedback() {}
+RtcpFeedback::RtcpFeedback(RtcpFeedbackType type) : type(type) {}
+RtcpFeedback::RtcpFeedback(RtcpFeedbackType type,
+                           RtcpFeedbackMessageType message_type)
+    : type(type), message_type(message_type) {}
+RtcpFeedback::~RtcpFeedback() {}
+
+RtpCodecCapability::RtpCodecCapability() {}
+RtpCodecCapability::~RtpCodecCapability() {}
+
+RtpHeaderExtensionCapability::RtpHeaderExtensionCapability() {}
+RtpHeaderExtensionCapability::RtpHeaderExtensionCapability(
+    const std::string& uri)
+    : uri(uri) {}
+RtpHeaderExtensionCapability::RtpHeaderExtensionCapability(
+    const std::string& uri,
+    int preferred_id)
+    : uri(uri), preferred_id(preferred_id) {}
+RtpHeaderExtensionCapability::~RtpHeaderExtensionCapability() {}
+
+RtpExtension::RtpExtension() {}
+RtpExtension::RtpExtension(const std::string& uri, int id) : uri(uri), id(id) {}
+RtpExtension::RtpExtension(const std::string& uri, int id, bool encrypt)
+    : uri(uri), id(id), encrypt(encrypt) {}
+RtpExtension::~RtpExtension() {}
+
+RtpFecParameters::RtpFecParameters() {}
+RtpFecParameters::RtpFecParameters(FecMechanism mechanism)
+    : mechanism(mechanism) {}
+RtpFecParameters::RtpFecParameters(FecMechanism mechanism, uint32_t ssrc)
+    : ssrc(ssrc), mechanism(mechanism) {}
+RtpFecParameters::~RtpFecParameters() {}
+
+RtpRtxParameters::RtpRtxParameters() {}
+RtpRtxParameters::RtpRtxParameters(uint32_t ssrc) : ssrc(ssrc) {}
+RtpRtxParameters::~RtpRtxParameters() {}
+
+RtpEncodingParameters::RtpEncodingParameters() {}
+RtpEncodingParameters::~RtpEncodingParameters() {}
+
+RtpCodecParameters::RtpCodecParameters() {}
+RtpCodecParameters::~RtpCodecParameters() {}
+
+RtpCapabilities::RtpCapabilities() {}
+RtpCapabilities::~RtpCapabilities() {}
+
+RtpParameters::RtpParameters() {}
+RtpParameters::~RtpParameters() {}
+
+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";
+const int RtpExtension::kTimestampOffsetDefaultId = 2;
+
+const char RtpExtension::kAbsSendTimeUri[] =
+    "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time";
+const int RtpExtension::kAbsSendTimeDefaultId = 3;
+
+const char RtpExtension::kVideoRotationUri[] = "urn:3gpp:video-orientation";
+const int RtpExtension::kVideoRotationDefaultId = 4;
+
+const char RtpExtension::kTransportSequenceNumberUri[] =
+    "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01";
+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::kVideoContentTypeUri[] =
+    "http://www.webrtc.org/experiments/rtp-hdrext/video-content-type";
+const int RtpExtension::kVideoContentTypeDefaultId = 7;
+
+const char RtpExtension::kVideoTimingUri[] =
+    "http://www.webrtc.org/experiments/rtp-hdrext/video-timing";
+const int RtpExtension::kVideoTimingDefaultId = 8;
+
+const char RtpExtension::kEncryptHeaderExtensionsUri[] =
+    "urn:ietf:params:rtp-hdrext:encrypt";
+
+const char* RtpExtension::kRtpStreamIdUri =
+    "urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id";
+const int RtpExtension::kRtpStreamIdDefaultId = 9;
+
+const char* RtpExtension::kRepairedRtpStreamIdUri =
+    "urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id";
+const int RtpExtension::kRepairedRtpStreamIdDefaultId = 10;
+
+const char* RtpExtension::kMIdUri =
+    "urn:ietf:params:rtp-hdrext:sdes:mid";
+const int RtpExtension::kMIdDefaultId = 11;
+
+const char* RtpExtension::kCsrcAudioLevelUri =
+    "urn:ietf:params:rtp-hdrext:csrc-audio-level";
+const int RtpExtension::kCsrcAudioLevelDefaultId = 12;
+
+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 ||
+         uri == webrtc::RtpExtension::kRtpStreamIdUri ||
+         uri == webrtc::RtpExtension::kRepairedRtpStreamIdUri ||
+         uri == webrtc::RtpExtension::kMIdUri ||
+         uri == webrtc::RtpExtension::kCsrcAudioLevelUri;
+}
+
+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 ||
+         uri == webrtc::RtpExtension::kVideoContentTypeUri ||
+         uri == webrtc::RtpExtension::kVideoTimingUri ||
+         uri == webrtc::RtpExtension::kRtpStreamIdUri ||
+         uri == webrtc::RtpExtension::kRepairedRtpStreamIdUri ||
+         uri == webrtc::RtpExtension::kMIdUri;
+}
+
+bool RtpExtension::IsEncryptionSupported(const std::string& uri) {
+  return uri == webrtc::RtpExtension::kAudioLevelUri ||
+         uri == webrtc::RtpExtension::kTimestampOffsetUri ||
+#if !defined(ENABLE_EXTERNAL_AUTH)
+         // TODO(jbauch): Figure out a way to always allow "kAbsSendTimeUri"
+         // here and filter out later if external auth is really used in
+         // srtpfilter. External auth is used by Chromium and replaces the
+         // extension header value of "kAbsSendTimeUri", so it must not be
+         // encrypted (which can't be done by Chromium).
+         uri == webrtc::RtpExtension::kAbsSendTimeUri ||
+#endif
+         uri == webrtc::RtpExtension::kVideoRotationUri ||
+         uri == webrtc::RtpExtension::kTransportSequenceNumberUri ||
+         uri == webrtc::RtpExtension::kPlayoutDelayUri ||
+         uri == webrtc::RtpExtension::kVideoContentTypeUri;
+}
+
+const RtpExtension* RtpExtension::FindHeaderExtensionByUri(
+    const std::vector<RtpExtension>& extensions,
+    const std::string& uri) {
+  for (const auto& extension : extensions) {
+    if (extension.uri == uri) {
+      return &extension;
+    }
+  }
+  return nullptr;
+}
+
+std::vector<RtpExtension> RtpExtension::FilterDuplicateNonEncrypted(
+    const std::vector<RtpExtension>& extensions) {
+  std::vector<RtpExtension> filtered;
+  for (auto extension = extensions.begin(); extension != extensions.end();
+       ++extension) {
+    if (extension->encrypt) {
+      filtered.push_back(*extension);
+      continue;
+    }
+
+    // Only add non-encrypted extension if no encrypted with the same URI
+    // is also present...
+    if (std::find_if(extension + 1, extensions.end(),
+                     [extension](const RtpExtension& check) {
+                       return extension->uri == check.uri;
+                     }) != extensions.end()) {
+      continue;
+    }
+
+    // ...and has not been added before.
+    if (!FindHeaderExtensionByUri(filtered, extension->uri)) {
+      filtered.push_back(*extension);
+    }
+  }
+  return filtered;
+}
+}  // namespace webrtc