Move RtpExtension to api/ directory and config.h/.cc to call/.

webrtc/api/rtpparameters.h

 /*
  *  Copyright 2015 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.
  */
 
 #ifndef WEBRTC_API_RTPPARAMETERS_H_
 #define WEBRTC_API_RTPPARAMETERS_H_
 
 #include <string>
 #include <unordered_map>
 #include <vector>
 
 #include "webrtc/api/mediatypes.h"
-#include "webrtc/config.h"
 #include "webrtc/rtc_base/optional.h"
 
 namespace webrtc {
 
 // These structures are intended to mirror those defined by:
 // http://draft.ortc.org/#rtcrtpdictionaries*
 // Contains everything specified as of 2017 Jan 24.
 //
 // They are used when retrieving or modifying the parameters of an
 // RtpSender/RtpReceiver, or retrieving capabilities.
@@ skipping 48 matching lines @@
 struct RtcpFeedback {
   RtcpFeedbackType type = RtcpFeedbackType::CCM;
 
   // Equivalent to ORTC "parameter" field with slight differences:
   // 1. It's an enum instead of a string.
   // 2. Generic NACK feedback is represented by a GENERIC_NACK message type,
   //    rather than an unset "parameter" value.
   rtc::Optional<RtcpFeedbackMessageType> message_type;
 
   // Constructors for convenience.
-  RtcpFeedback() {}
-  explicit RtcpFeedback(RtcpFeedbackType type) : type(type) {}
-  RtcpFeedback(RtcpFeedbackType type, RtcpFeedbackMessageType message_type)
-      : type(type), message_type(message_type) {}
+  RtcpFeedback();
+  explicit RtcpFeedback(RtcpFeedbackType type);
+  RtcpFeedback(RtcpFeedbackType type, RtcpFeedbackMessageType message_type);
+  ~RtcpFeedback();
 
   bool operator==(const RtcpFeedback& o) const {
     return type == o.type && message_type == o.message_type;
   }
   bool operator!=(const RtcpFeedback& o) const { return !(*this == o); }
 };
 
 // RtpCodecCapability is to RtpCodecParameters as RtpCapabilities is to
 // RtpParameters. This represents the static capabilities of an endpoint's
 // implementation of a codec.
 struct RtpCodecCapability {
+  RtpCodecCapability();
+  ~RtpCodecCapability();
+
   // Build MIME "type/subtype" string from |name| and |kind|.
   std::string mime_type() const { return MediaTypeToString(kind) + "/" + name; }
 
   // Used to identify the codec. Equivalent to MIME subtype.
   std::string name;
 
   // The media type of this codec. Equivalent to MIME top-level type.
   cricket::MediaType kind = cricket::MEDIA_TYPE_AUDIO;
 
   // Clock rate in Hertz. If unset, the codec is applicable to any clock rate.
@@ skipping 76 matching lines @@
   std::string uri;
 
   // Preferred value of ID that goes in the packet.
   rtc::Optional<int> preferred_id;
 
   // If true, it's preferred that the value in the header is encrypted.
   // TODO(deadbeef): Not implemented.
   bool preferred_encrypt = false;
 
   // Constructors for convenience.
-  RtpHeaderExtensionCapability() = default;
-  explicit RtpHeaderExtensionCapability(const std::string& uri) : uri(uri) {}
-  RtpHeaderExtensionCapability(const std::string& uri, int preferred_id)
-      : uri(uri), preferred_id(preferred_id) {}
+  RtpHeaderExtensionCapability();
+  explicit RtpHeaderExtensionCapability(const std::string& uri);
+  RtpHeaderExtensionCapability(const std::string& uri, int preferred_id);
+  ~RtpHeaderExtensionCapability();
 
   bool operator==(const RtpHeaderExtensionCapability& o) const {
     return uri == o.uri && preferred_id == o.preferred_id &&
            preferred_encrypt == o.preferred_encrypt;
   }
   bool operator!=(const RtpHeaderExtensionCapability& o) const {
     return !(*this == o);
   }
 };
 
+// RTP header extension, see RFC 5285.
+struct RtpExtension {
+  RtpExtension();
+  RtpExtension(const std::string& uri, int id);
+  RtpExtension(const std::string& uri, int id, bool encrypt);
+  ~RtpExtension();
+  std::string ToString() const;
+  bool operator==(const RtpExtension& rhs) const {
+    return uri == rhs.uri && id == rhs.id && encrypt == rhs.encrypt;
+  }
+  static bool IsSupportedForAudio(const std::string& uri);
+  static bool IsSupportedForVideo(const std::string& uri);
+  // Return "true" if the given RTP header extension URI may be encrypted.
+  static bool IsEncryptionSupported(const std::string& uri);
+
+  // Returns the named header extension if found among all extensions,
+  // nullptr otherwise.
+  static const RtpExtension* FindHeaderExtensionByUri(
+      const std::vector<RtpExtension>& extensions,
+      const std::string& uri);
+
+  // Return a list of RTP header extensions with the non-encrypted extensions
+  // removed if both the encrypted and non-encrypted extension is present for
+  // the same URI.
+  static std::vector<RtpExtension> FilterDuplicateNonEncrypted(
+      const std::vector<RtpExtension>& extensions);
+
+  // Header extension for audio levels, as defined in:
+  // http://tools.ietf.org/html/draft-ietf-avtext-client-to-mixer-audio-level-03
+  static const char kAudioLevelUri[];
+  static const int kAudioLevelDefaultId;
+
+  // Header extension for RTP timestamp offset, see RFC 5450 for details:
+  // http://tools.ietf.org/html/rfc5450
+  static const char kTimestampOffsetUri[];
+  static const int kTimestampOffsetDefaultId;
+
+  // Header extension for absolute send time, see url for details:
+  // http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
+  static const char kAbsSendTimeUri[];
+  static const int kAbsSendTimeDefaultId;
+
+  // Header extension for coordination of video orientation, see url for
+  // details:
+  // http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/ts_126114v120700p.pdf
+  static const char kVideoRotationUri[];
+  static const int kVideoRotationDefaultId;
+
+  // Header extension for video content type. E.g. default or screenshare.
+  static const char kVideoContentTypeUri[];
+  static const int kVideoContentTypeDefaultId;
+
+  // Header extension for video timing.
+  static const char kVideoTimingUri[];
+  static const int kVideoTimingDefaultId;
+
+  // Header extension for transport sequence number, see url for details:
+  // http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions
+  static const char kTransportSequenceNumberUri[];
+  static const int kTransportSequenceNumberDefaultId;
+
+  static const char kPlayoutDelayUri[];
+  static const int kPlayoutDelayDefaultId;
+
+  // Encryption of Header Extensions, see RFC 6904 for details:
+  // https://tools.ietf.org/html/rfc6904
+  static const char kEncryptHeaderExtensionsUri[];
+
+  // Inclusive min and max IDs for one-byte header extensions, per RFC5285.
+  static const int kMinId;
+  static const int kMaxId;
+
+  std::string uri;
+  int id = 0;
+  bool encrypt = false;
+};
+
 // See webrtc/config.h. Has "uri" and "id" fields.
 // TODO(deadbeef): This is missing the "encrypt" flag, which is unimplemented.
 typedef RtpExtension RtpHeaderExtensionParameters;
 
 struct RtpFecParameters {
   // If unset, a value is chosen by the implementation.
   // Works just like RtpEncodingParameters::ssrc.
   rtc::Optional<uint32_t> ssrc;
 
   FecMechanism mechanism = FecMechanism::RED;
 
   // Constructors for convenience.
-  RtpFecParameters() = default;
-  explicit RtpFecParameters(FecMechanism mechanism) : mechanism(mechanism) {}
-  RtpFecParameters(FecMechanism mechanism, uint32_t ssrc)
-      : ssrc(ssrc), mechanism(mechanism) {}
+  RtpFecParameters();
+  explicit RtpFecParameters(FecMechanism mechanism);
+  RtpFecParameters(FecMechanism mechanism, uint32_t ssrc);
+  ~RtpFecParameters();
 
   bool operator==(const RtpFecParameters& o) const {
     return ssrc == o.ssrc && mechanism == o.mechanism;
   }
   bool operator!=(const RtpFecParameters& o) const { return !(*this == o); }
 };
 
 struct RtpRtxParameters {
   // If unset, a value is chosen by the implementation.
   // Works just like RtpEncodingParameters::ssrc.
   rtc::Optional<uint32_t> ssrc;
 
   // Constructors for convenience.
-  RtpRtxParameters() = default;
-  explicit RtpRtxParameters(uint32_t ssrc) : ssrc(ssrc) {}
+  RtpRtxParameters();
+  explicit RtpRtxParameters(uint32_t ssrc);
+  ~RtpRtxParameters();
 
   bool operator==(const RtpRtxParameters& o) const { return ssrc == o.ssrc; }
   bool operator!=(const RtpRtxParameters& o) const { return !(*this == o); }
 };
 
 struct RtpEncodingParameters {
+  RtpEncodingParameters();
+  ~RtpEncodingParameters();
+
   // If unset, a value is chosen by the implementation.
   //
   // Note that the chosen value is NOT returned by GetParameters, because it
   // may change due to an SSRC conflict, in which case the conflict is handled
   // internally without any event. Another way of looking at this is that an
   // unset SSRC acts as a "wildcard" SSRC.
   rtc::Optional<uint32_t> ssrc;
 
   // Can be used to reference a codec in the |codecs| member of the
   // RtpParameters that contains this RtpEncodingParameters. If unset, the
@@ skipping 71 matching lines @@
            scale_framerate_down_by == o.scale_framerate_down_by &&
            active == o.active && rid == o.rid &&
            dependency_rids == o.dependency_rids;
   }
   bool operator!=(const RtpEncodingParameters& o) const {
     return !(*this == o);
   }
 };
 
 struct RtpCodecParameters {
+  RtpCodecParameters();
+  ~RtpCodecParameters();
+
   // Build MIME "type/subtype" string from |name| and |kind|.
   std::string mime_type() const { return MediaTypeToString(kind) + "/" + name; }
 
   // Used to identify the codec. Equivalent to MIME subtype.
   std::string name;
 
   // The media type of this codec. Equivalent to MIME top-level type.
   cricket::MediaType kind = cricket::MEDIA_TYPE_AUDIO;
 
   // Payload type used to identify this codec in RTP packets.
@@ skipping 42 matching lines @@
            max_ptime == o.max_ptime && ptime == o.ptime &&
            rtcp_feedback == o.rtcp_feedback && parameters == o.parameters;
   }
   bool operator!=(const RtpCodecParameters& o) const { return !(*this == o); }
 };
 
 // RtpCapabilities is used to represent the static capabilities of an
 // endpoint. An application can use these capabilities to construct an
 // RtpParameters.
 struct RtpCapabilities {
+  RtpCapabilities();
+  ~RtpCapabilities();
+
   // Supported codecs.
   std::vector<RtpCodecCapability> codecs;
 
   // Supported RTP header extensions.
   std::vector<RtpHeaderExtensionCapability> header_extensions;
 
   // Supported Forward Error Correction (FEC) mechanisms. Note that the RED,
   // ulpfec and flexfec codecs used by these mechanisms will still appear in
   // |codecs|.
   std::vector<FecMechanism> fec;
 
   bool operator==(const RtpCapabilities& o) const {
     return codecs == o.codecs && header_extensions == o.header_extensions &&
            fec == o.fec;
   }
   bool operator!=(const RtpCapabilities& o) const { return !(*this == o); }
 };
 
 // Note that unlike in ORTC, an RtcpParameters structure is not included in
 // RtpParameters, because our API includes an additional "RtpTransport"
 // abstraction on which RTCP parameters are set.
 struct RtpParameters {
+  RtpParameters();
+  ~RtpParameters();
+
   // Used when calling getParameters/setParameters with a PeerConnection
   // RtpSender, to ensure that outdated parameters are not unintentionally
   // applied successfully.
   // TODO(deadbeef): Not implemented.
   std::string transaction_id;
 
   // Value to use for MID RTP header extension.
   // Called "muxId" in ORTC.
   // TODO(deadbeef): Not implemented.
   std::string mid;
@@ skipping 14 matching lines @@
            header_extensions == o.header_extensions &&
            encodings == o.encodings &&
            degradation_preference == o.degradation_preference;
   }
   bool operator!=(const RtpParameters& o) const { return !(*this == o); }
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_RTPPARAMETERS_H_