Move RtpExtension to api/ directory and config.h/.cc to call/.
BUG=webrtc:5876
R=deadbeef@webrtc.org, solenberg@webrtc.org
Review-Url: https://codereview.webrtc.org/3004723002 .
Cr-Commit-Position: refs/heads/master@{#19639}
diff --git a/webrtc/api/rtpparameters.cc b/webrtc/api/rtpparameters.cc
new file mode 100644
index 0000000..29b14fb
--- /dev/null
+++ b/webrtc/api/rtpparameters.cc
@@ -0,0 +1,190 @@
+/*
+ * 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 "webrtc/api/rtpparameters.h"
+
+#include <algorithm>
+#include <sstream>
+#include <string>
+
+#include "webrtc/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 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 ||
+ uri == webrtc::RtpExtension::kVideoContentTypeUri ||
+ uri == webrtc::RtpExtension::kVideoTimingUri;
+}
+
+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