Update To 11.40.268.0

[platform/framework/web/crosswalk.git] / src / third_party / webrtc / modules / rtp_rtcp / source / rtp_format_h264.h

diff --git a/src/third_party/webrtc/modules/rtp_rtcp/source/rtp_format_h264.h b/src/third_party/webrtc/modules/rtp_rtcp/source/rtp_format_h264.h
index 830d765..ba293c8 100644 (file)
--- a/src/third_party/webrtc/modules/rtp_rtcp/source/rtp_format_h264.h
+++ b/src/third_party/webrtc/modules/rtp_rtcp/source/rtp_format_h264.h
@@ -12,6 +12,7 @@
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_
 
 #include <queue>
+#include <string>
 
 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
 
@@ -92,18 +93,11 @@ class RtpPacketizerH264 : public RtpPacketizer {
 // Depacketizer for H264.
 class RtpDepacketizerH264 : public RtpDepacketizer {
  public:
-  explicit RtpDepacketizerH264(RtpData* const callback);
-
   virtual ~RtpDepacketizerH264() {}
 
-  virtual bool Parse(WebRtcRTPHeader* rtp_header,
+  virtual bool Parse(ParsedPayload* parsed_payload,
                      const uint8_t* payload_data,
                      size_t payload_data_length) OVERRIDE;
-
- private:
-  RtpData* const callback_;
-
-  DISALLOW_COPY_AND_ASSIGN(RtpDepacketizerH264);
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_