Fix circular dependencies in webrtc_common.

One reason for the circular deps is that common_types.h is a
historical dumping ground for various structs and defines that
are believed to be generally useful. I tried moving things out
that did not appear to be used downstream (StreamCounters,
RtpCounters etc) and moved the things that seemed used
(RtpHeader + supporting structs) to a new file api/rtp_headers.h.
This makes their place in the api more clear while moving out
the things that don't belong in the API in the first place.

I had to extract out typedefs.h from webrtc_common to resolve
another circular dependency. I believe checks includes typedefs,
but common depends on checks.

Bug: webrtc:7745
Change-Id: I725d49616b1ec0cdc8b74be7c078f7a4d46f084b
Reviewed-on: https://webrtc-review.googlesource.com/33001
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21295}

api/rtp_headers.h

@@ -0,0 +1,178 @@
+/*
+ *  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.
+ */
+
+#ifndef API_RTP_HEADERS_H_
+#define API_RTP_HEADERS_H_
+
+#include <stddef.h>
+#include <string.h>
+#include <ostream>
+#include <string>
+#include <vector>
+
+#include "api/array_view.h"
+#include "api/optional.h"
+#include "api/video/video_content_type.h"
+#include "api/video/video_rotation.h"
+#include "api/video/video_timing.h"
+
+#include "rtc_base/checks.h"
+#include "rtc_base/deprecation.h"
+#include "common_types.h"  // NOLINT(build/include)
+#include "typedefs.h"  // NOLINT(build/include)
+
+namespace webrtc {
+
+// Class to represent the value of RTP header extensions that are
+// variable-length strings (e.g., RtpStreamId and RtpMid).
+// Unlike std::string, it can be copied with memcpy and cleared with memset.
+//
+// Empty value represents unset header extension (use empty() to query).
+class StringRtpHeaderExtension {
+ public:
+  // String RTP header extensions are 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;
+
+  static bool IsLegalName(rtc::ArrayView<const char> name);
+
+  StringRtpHeaderExtension() { value_[0] = 0; }
+  explicit StringRtpHeaderExtension(rtc::ArrayView<const char> value) {
+    Set(value.data(), value.size());
+  }
+  StringRtpHeaderExtension(const StringRtpHeaderExtension&) = default;
+  StringRtpHeaderExtension& operator=(const StringRtpHeaderExtension&) =
+      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 StringRtpHeaderExtension& lhs,
+                         const StringRtpHeaderExtension& rhs) {
+    return strncmp(lhs.value_, rhs.value_, kMaxSize) == 0;
+  }
+  friend bool operator!=(const StringRtpHeaderExtension& lhs,
+                         const StringRtpHeaderExtension& rhs) {
+    return !(lhs == rhs);
+  }
+
+ private:
+  char value_[kMaxSize];
+};
+
+// StreamId represents RtpStreamId which is a string.
+typedef StringRtpHeaderExtension StreamId;
+
+// Mid represents RtpMid which is a string.
+typedef StringRtpHeaderExtension Mid;
+
+struct RTPHeaderExtension {
+  RTPHeaderExtension();
+  RTPHeaderExtension(const RTPHeaderExtension& other);
+  RTPHeaderExtension& operator=(const RTPHeaderExtension& other);
+
+  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;
+
+  bool has_video_timing;
+  VideoSendTiming video_timing;
+
+  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;
+
+  // For identifying the media section used to interpret this RTP packet. See
+  // https://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-negotiation-38
+  Mid mid;
+};
+
+struct RTPHeader {
+  RTPHeader();
+  RTPHeader(const RTPHeader& other);
+  RTPHeader& operator=(const RTPHeader& other);
+
+  bool markerBit;
+  uint8_t payloadType;
+  uint16_t sequenceNumber;
+  uint32_t timestamp;
+  uint32_t ssrc;
+  uint8_t numCSRCs;
+  uint32_t arrOfCSRCs[kRtpCsrcSize];
+  size_t paddingLength;
+  size_t headerLength;
+  int payload_type_frequency;
+  RTPHeaderExtension extension;
+};
+
+// RTCP mode to use. Compound mode is described by RFC 4585 and reduced-size
+// RTCP mode is described by RFC 5506.
+enum class RtcpMode { kOff, kCompound, kReducedSize };
+
+enum NetworkState {
+  kNetworkUp,
+  kNetworkDown,
+};
+
+struct RtpKeepAliveConfig final {
+  // If no packet has been sent for |timeout_interval_ms|, send a keep-alive
+  // packet. The keep-alive packet is an empty (no payload) RTP packet with a
+  // payload type of 20 as long as the other end has not negotiated the use of
+  // this value. If this value has already been negotiated, then some other
+  // unused static payload type from table 5 of RFC 3551 shall be used and set
+  // in |payload_type|.
+  int64_t timeout_interval_ms = -1;
+  uint8_t payload_type = 20;
+
+  bool operator==(const RtpKeepAliveConfig& o) const {
+    return timeout_interval_ms == o.timeout_interval_ms &&
+           payload_type == o.payload_type;
+  }
+  bool operator!=(const RtpKeepAliveConfig& o) const { return !(*this == o); }
+};
+
+// Currently only VP8/VP9 specific.
+struct RtpPayloadState {
+  int16_t picture_id = -1;
+};
+
+}  // namespace webrtc
+
+#endif  // API_RTP_HEADERS_H_