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}

modules/rtp_rtcp/include/rtp_rtcp_defines.h

@@ -16,6 +16,7 @@
 #include <vector>
 
 #include "api/audio_codecs/audio_format.h"
+#include "api/rtp_headers.h"
 #include "common_types.h"  // NOLINT(build/include)
 #include "modules/include/module_common_types.h"
 #include "rtc_base/deprecation.h"
@@ -496,5 +497,100 @@ class TransportSequenceNumberAllocator {
   virtual uint16_t AllocateSequenceNumber() = 0;
 };
 
+struct RtpPacketCounter {
+  RtpPacketCounter()
+      : header_bytes(0), payload_bytes(0), padding_bytes(0), packets(0) {}
+
+  void Add(const RtpPacketCounter& other) {
+    header_bytes += other.header_bytes;
+    payload_bytes += other.payload_bytes;
+    padding_bytes += other.padding_bytes;
+    packets += other.packets;
+  }
+
+  void Subtract(const RtpPacketCounter& other) {
+    RTC_DCHECK_GE(header_bytes, other.header_bytes);
+    header_bytes -= other.header_bytes;
+    RTC_DCHECK_GE(payload_bytes, other.payload_bytes);
+    payload_bytes -= other.payload_bytes;
+    RTC_DCHECK_GE(padding_bytes, other.padding_bytes);
+    padding_bytes -= other.padding_bytes;
+    RTC_DCHECK_GE(packets, other.packets);
+    packets -= other.packets;
+  }
+
+  void AddPacket(size_t packet_length, const RTPHeader& header) {
+    ++packets;
+    header_bytes += header.headerLength;
+    padding_bytes += header.paddingLength;
+    payload_bytes +=
+        packet_length - (header.headerLength + header.paddingLength);
+  }
+
+  size_t TotalBytes() const {
+    return header_bytes + payload_bytes + padding_bytes;
+  }
+
+  size_t header_bytes;   // Number of bytes used by RTP headers.
+  size_t payload_bytes;  // Payload bytes, excluding RTP headers and padding.
+  size_t padding_bytes;  // Number of padding bytes.
+  uint32_t packets;      // Number of packets.
+};
+
+// Data usage statistics for a (rtp) stream.
+struct StreamDataCounters {
+  StreamDataCounters();
+
+  void Add(const StreamDataCounters& other) {
+    transmitted.Add(other.transmitted);
+    retransmitted.Add(other.retransmitted);
+    fec.Add(other.fec);
+    if (other.first_packet_time_ms != -1 &&
+        (other.first_packet_time_ms < first_packet_time_ms ||
+         first_packet_time_ms == -1)) {
+      // Use oldest time.
+      first_packet_time_ms = other.first_packet_time_ms;
+    }
+  }
+
+  void Subtract(const StreamDataCounters& other) {
+    transmitted.Subtract(other.transmitted);
+    retransmitted.Subtract(other.retransmitted);
+    fec.Subtract(other.fec);
+    if (other.first_packet_time_ms != -1 &&
+        (other.first_packet_time_ms > first_packet_time_ms ||
+         first_packet_time_ms == -1)) {
+      // Use youngest time.
+      first_packet_time_ms = other.first_packet_time_ms;
+    }
+  }
+
+  int64_t TimeSinceFirstPacketInMs(int64_t now_ms) const {
+    return (first_packet_time_ms == -1) ? -1 : (now_ms - first_packet_time_ms);
+  }
+
+  // Returns the number of bytes corresponding to the actual media payload (i.e.
+  // RTP headers, padding, retransmissions and fec packets are excluded).
+  // Note this function does not have meaning for an RTX stream.
+  size_t MediaPayloadBytes() const {
+    return transmitted.payload_bytes - retransmitted.payload_bytes -
+           fec.payload_bytes;
+  }
+
+  int64_t first_packet_time_ms;    // Time when first packet is sent/received.
+  RtpPacketCounter transmitted;    // Number of transmitted packets/bytes.
+  RtpPacketCounter retransmitted;  // Number of retransmitted packets/bytes.
+  RtpPacketCounter fec;            // Number of redundancy packets/bytes.
+};
+
+// Callback, called whenever byte/packet counts have been updated.
+class StreamDataCountersCallback {
+ public:
+  virtual ~StreamDataCountersCallback() {}
+
+  virtual void DataCountersUpdated(const StreamDataCounters& counters,
+                                   uint32_t ssrc) = 0;
+};
+
 }  // namespace webrtc
 #endif  // MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_