rtcp::Bye packet moved to own file and got a Parse function

webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h

Index: webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h b/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h
similarity index 54%
copy from webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h
copy to webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h
index 49de7be1a803011f9d263a82538f3bdb2f783480..1cd560c5caae3348e6c3996dbd1339d581a9a2ff 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h
@@ -6,39 +6,41 @@
  *  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_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_EXTENDED_JITTER_REPORT_H_
-#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_EXTENDED_JITTER_REPORT_H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_BYE_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_BYE_H_
 
+#include <string>
 #include <vector>
 
 #include "webrtc/base/checks.h"

[åsapersson, 2015/11/20 09:44:58]

needed?

[danilchap, 2015/11/20 11:03:21]

Done.

 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
+#include "webrtc/typedefs.h"
 
 namespace webrtc {
 namespace rtcp {
 
-class ExtendedJitterReport : public RtcpPacket {
+class Bye : public RtcpPacket {
  public:
-  static const uint8_t kPacketType = 195;
-
-  ExtendedJitterReport() : RtcpPacket() {}
+  static const uint8_t kPacketType = 203;
 
-  virtual ~ExtendedJitterReport() {}
+  Bye();
+  virtual ~Bye() {}
 
   // Parse assumes header is already parsed and validated.
   bool Parse(const RTCPUtility::RtcpCommonHeader& header,
              const uint8_t* payload);  // Size of the payload is in the header.
 
-  bool WithJitter(uint32_t jitter);
+  void From(uint32_t ssrc) { sender_ssrc_ = ssrc; }
+  bool WithCsrc(uint32_t csrc);
+  bool WithReason(const std::string& reason);
 
-  size_t jitters_count() const { return inter_arrival_jitters_.size(); }
-  uint32_t jitter(size_t index) const {
-    RTC_DCHECK_LT(index, jitters_count());
-    return inter_arrival_jitters_[index];
-  }
+  uint32_t sender_ssrc() const { return sender_ssrc_; }
+  const std::vector<uint32_t>& csrcs() const { return csrcs_; }
+  const std::string& reason() const { return reason_; }
 
  protected:
   bool Create(uint8_t* packet,
@@ -47,17 +49,17 @@ class ExtendedJitterReport : public RtcpPacket {
               RtcpPacket::PacketReadyCallback* callback) const override;
 
  private:
-  static const int kMaxNumberOfJitters = 0x1f;
+  static const int kMaxNumberOfCsrcs = 0x1f - 1;  // First item is sender SSRC.
 
-  size_t BlockLength() const override {
-    return kHeaderLength + 4 * inter_arrival_jitters_.size();
-  }
+  size_t BlockLength() const override;
 
-  std::vector<uint32_t> inter_arrival_jitters_;
+  uint32_t sender_ssrc_;
+  std::vector<uint32_t> csrcs_;
+  std::string reason_;
 
-  RTC_DISALLOW_COPY_AND_ASSIGN(ExtendedJitterReport);
+  RTC_DISALLOW_COPY_AND_ASSIGN(Bye);
 };
 
 }  // namespace rtcp
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_EXTENDED_JITTER_REPORT_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_BYE_H_