Upstream version 10.39.225.0

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

diff --git a/src/third_party/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h b/src/third_party/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h
index 93fcfa8..23a4e6b 100644 (file)
--- a/src/third_party/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h
+++ b/src/third_party/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h
@@ -10,13 +10,72 @@
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_GENERIC_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_GENERIC_H_
 
+#include "webrtc/common_types.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 namespace RtpFormatVideoGeneric {
-  static const uint8_t kKeyFrameBit = 0x01;
-  static const uint8_t kFirstPacketBit = 0x02;
+static const uint8_t kKeyFrameBit = 0x01;
+static const uint8_t kFirstPacketBit = 0x02;
 }  // namespace RtpFormatVideoGeneric
-}  // namespace webrtc
 
+class RtpPacketizerGeneric : public RtpPacketizer {
+ public:
+  // Initialize with payload from encoder.
+  // The payload_data must be exactly one encoded generic frame.
+  RtpPacketizerGeneric(FrameType frametype, size_t max_payload_len);
+
+  virtual ~RtpPacketizerGeneric();
+
+  virtual void SetPayloadData(
+      const uint8_t* payload_data,
+      size_t payload_size,
+      const RTPFragmentationHeader* fragmentation) OVERRIDE;
+
+  // Get the next payload with generic payload header.
+  // buffer is a pointer to where the output will be written.
+  // bytes_to_send is an output variable that will contain number of bytes
+  // written to buffer. The parameter last_packet is true for the last packet of
+  // the frame, false otherwise (i.e., call the function again to get the
+  // next packet).
+  // Returns true on success or false if there was no payload to packetize.
+  virtual bool NextPacket(uint8_t* buffer,
+                          size_t* bytes_to_send,
+                          bool* last_packet) OVERRIDE;
+
+  virtual ProtectionType GetProtectionType() OVERRIDE;
+
+  virtual StorageType GetStorageType(uint32_t retransmission_settings) OVERRIDE;
+
+  virtual std::string ToString() OVERRIDE;
+
+ private:
+  const uint8_t* payload_data_;
+  size_t payload_size_;
+  const size_t max_payload_len_;
+  FrameType frame_type_;
+  uint32_t payload_length_;
+  uint8_t generic_header_;
+
+  DISALLOW_COPY_AND_ASSIGN(RtpPacketizerGeneric);
+};
+
+// Depacketizer for generic codec.
+class RtpDepacketizerGeneric : public RtpDepacketizer {
+ public:
+  explicit RtpDepacketizerGeneric(RtpData* const callback);
+
+  virtual ~RtpDepacketizerGeneric() {}
+
+  virtual bool Parse(WebRtcRTPHeader* rtp_header,
+                     const uint8_t* payload_data,
+                     size_t payload_data_length) OVERRIDE;
+
+ private:
+  RtpData* const callback_;
+
+  DISALLOW_COPY_AND_ASSIGN(RtpDepacketizerGeneric);
+};
+}  // namespace webrtc
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_GENERIC_H_