#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"
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() {}
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 {