Add read support of RtpStreamId/RepairedRtpStreamId header extensions.

webrtc/common_types.h

 /*
  *  Copyright (c) 2012 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_COMMON_TYPES_H_
 #define WEBRTC_COMMON_TYPES_H_
 
 #include <stddef.h>
 #include <string.h>
 
 #include <ostream>
 #include <string>
 #include <vector>
 
 #include "webrtc/api/video/video_content_type.h"
 #include "webrtc/api/video/video_rotation.h"
+#include "webrtc/base/array_view.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/typedefs.h"
 
 #if defined(_MSC_VER)
 // Disable "new behavior: elements of array will be default initialized"
 // warning. Affects OverUseDetectorOptions.
 #pragma warning(disable : 4351)
 #endif
 
@@ skipping 655 matching lines @@
 // min = x, max = y indicates that the receiver is free to adapt
 // in the range (x, y) based on network jitter.
 //
 // Note: Given that this gets embedded in a union, it is up-to the owner to
 // initialize these values.
 struct PlayoutDelay {
   int min_ms;
   int max_ms;
 };
 
+// Class to represent RtpStreamId which is a string.
+// Unlike std::string, it can be copied with memcpy and cleared with memset.
+// Empty value represent unset RtpStreamId.
+class StreamId {
+ public:
+  // Stream id is limited to 16 bytes because it is the maximum length
+  // that can be encoded with one-byte header extensions.
+  static constexpr size_t kMaxSize = 16;
+
+  StreamId() { value_[0] = 0; }
+  explicit StreamId(rtc::ArrayView<const char> value) {
+    Set(value.data(), value.size());
+  }
+  StreamId(const StreamId&) = default;
+  StreamId& operator=(const StreamId&) = default;
+
+  bool empty() const { return value_[0] == 0; }
+  const char* data() const { return value_; }
+  size_t size() const { return strnlen(value_, kMaxSize); }
+
+  void Set(rtc::ArrayView<const uint8_t> value) {
+    Set(reinterpret_cast<const char*>(value.data()), value.size());
+  }
+  void Set(const char* data, size_t size);
+
+  friend bool operator==(const StreamId& lhs, const StreamId& rhs) {
+    return strncmp(lhs.value_, rhs.value_, kMaxSize) == 0;
+  }
+  friend bool operator!=(const StreamId& lhs, const StreamId& rhs) {
+    return !(lhs == rhs);
+  }
+
+ private:
+  char value_[kMaxSize];
+};
+
 struct RTPHeaderExtension {
   RTPHeaderExtension();
 
   bool hasTransmissionTimeOffset;
   int32_t transmissionTimeOffset;
   bool hasAbsoluteSendTime;
   uint32_t absoluteSendTime;
   bool hasTransportSequenceNumber;
   uint16_t transportSequenceNumber;
 
   // Audio Level includes both level in dBov and voiced/unvoiced bit. See:
   // https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/
   bool hasAudioLevel;
   bool voiceActivity;
   uint8_t audioLevel;
 
   // For Coordination of Video Orientation. See
   // http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/
   // ts_126114v120700p.pdf
   bool hasVideoRotation;
   VideoRotation videoRotation;
 
   // TODO(ilnik): Refactor this and one above to be rtc::Optional() and remove
   // a corresponding bool flag.
   bool hasVideoContentType;
   VideoContentType videoContentType;
 
   PlayoutDelay playout_delay = {-1, -1};
+
+  // For identification of a stream when ssrc is not signaled. See
+  // https://tools.ietf.org/html/draft-ietf-avtext-rid-09
+  // TODO(danilchap): Update url from draft to release version.
+  StreamId stream_id;
+  StreamId repaired_stream_id;
 };
 
 struct RTPHeader {
   RTPHeader();
 
   bool markerBit;
   uint8_t payloadType;
   uint16_t sequenceNumber;
   uint32_t timestamp;
   uint32_t ssrc;
@@ skipping 105 matching lines @@
 enum class RtcpMode { kOff, kCompound, kReducedSize };
 
 enum NetworkState {
   kNetworkUp,
   kNetworkDown,
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_COMMON_TYPES_H_