Upstream version 10.39.225.0

[platform/framework/web/crosswalk.git] / src / third_party / webrtc / modules / rtp_rtcp / source / rtp_format.h

diff --git a/src/third_party/webrtc/modules/rtp_rtcp/source/rtp_format.h b/src/third_party/webrtc/modules/rtp_rtcp/source/rtp_format.h
index d7dba48..07dc225 100644 (file)
--- a/src/third_party/webrtc/modules/rtp_rtcp/source/rtp_format.h
+++ b/src/third_party/webrtc/modules/rtp_rtcp/source/rtp_format.h
@@ -11,6 +11,8 @@
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
 
+#include <string>
+
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/interface/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
@@ -19,7 +21,10 @@ namespace webrtc {
 
 class RtpPacketizer {
  public:
-  static RtpPacketizer* Create(RtpVideoCodecTypes type, size_t max_payload_len);
+  static RtpPacketizer* Create(RtpVideoCodecTypes type,
+                               size_t max_payload_len,
+                               const RTPVideoTypeHeader* rtp_type_header,
+                               FrameType frame_type);
 
   virtual ~RtpPacketizer() {}
 
@@ -37,6 +42,12 @@ class RtpPacketizer {
   virtual bool NextPacket(uint8_t* buffer,
                           size_t* bytes_to_send,
                           bool* last_packet) = 0;
+
+  virtual ProtectionType GetProtectionType() = 0;
+
+  virtual StorageType GetStorageType(uint32_t retransmission_settings) = 0;
+
+  virtual std::string ToString() = 0;
 };
 
 class RtpDepacketizer {