ietf.h

一个用于智能手机的多媒体库适合S60 WinCE的跨平台开发库

	char *a_sdplang, *a_lang;

	/*all attributes not defined in RFC 2327 for the presentation*/
	GF_List *Attributes;
	
	/*list of media in the SDP*/
	GF_List *media_desc;
} GF_SDPInfo;


/*
  Memory Consideration: the destructors free all non-NULL string. You should therefore 
  be carefull while (de-)assigning the strings. The function gf_sdp_info_parse() performs a complete 
  reset of the GF_SDPInfo
*/
/*constructor*/
GF_SDPInfo *gf_sdp_info_new();
/*destructor*/
void gf_sdp_info_del(GF_SDPInfo *sdp);
/*reset all structures (destroys substructure too)*/
void gf_sdp_info_reset(GF_SDPInfo *sdp);
/*Parses a memory SDP buffer*/
GF_Err gf_sdp_info_parse(GF_SDPInfo *sdp, char *sdp_text, u32 text_size);
/*check the consistency of the GF_SDPInfo*/
GF_Err gf_sdp_info_check(GF_SDPInfo *sdp);
/*write the SDP to a new buffer and returns it. Automatically checks the SDP before calling*/
GF_Err gf_sdp_info_write(GF_SDPInfo *sdp, char **out_str_buf);


/*
	Const/dest for GF_SDPMedia
*/
GF_SDPMedia *gf_sdp_media_new();
void gf_sdp_media_del(GF_SDPMedia *media);

/*
	Const/dest for GF_SDPConnection
*/
GF_SDPConnection *gf_sdp_conn_new();
void gf_sdp_conn_del(GF_SDPConnection *conn);

/*
	Const/dest for SDP FMTP
*/
GF_SDP_FMTP *gf_sdp_fmtp_new();
void gf_sdp_fmtp_del(GF_SDP_FMTP *fmtp);



/*
	RTP packetizer
*/


/*RTP<->SL mapping*/
typedef struct 
{
	/*1 - required options*/

	/*mode, or "" if no mode ("generic" should be used instead)*/
	char mode[30];
	
	/*config of the stream if carried in SDP*/
	char *config;
	u32 configSize;
	/* Stream Type*/
	u8 StreamType;
	/* stream profile and level indication - for AVC/H264, 0xPPCCLL, with PP:profile, CC:compatibility, LL:level*/
	u32 PL_ID;


	/*2 - optional options*/
	
	/*size of AUs if constant*/
	u32 ConstantSize;
	/*duration of AUs if constant, in RTP timescale*/
	u32 ConstantDuration;

	/* Object Type Indication */
	u8 ObjectTypeIndication;
	/*audio max displacement when interleaving (eg, de-interleaving window buffer max length) in RTP timescale*/
	u32 maxDisplacement;
	/*de-interleaveBufferSize if not recomputable from maxDisplacement*/
	u32 deinterleaveBufferSize;
	
	/*The number of bits on which the AU-size field is encoded in the AU-header*/
	u32 SizeLength;
	/*The number of bits on which the AU-Index is encoded in the first AU-header*/
	u32 IndexLength;
	/*The number of bits on which the AU-Index-delta field is encoded in any non-first AU-header*/
	u32 IndexDeltaLength;

	/*The number of bits on which the DTS-delta field is encoded in the AU-header*/
	u32 DTSDeltaLength;
	/*The number of bits on which the CTS-delta field is encoded in the AU-header*/
	u32 CTSDeltaLength;
	/*random access point flag present*/
	Bool RandomAccessIndication;
	
	/*The number of bits on which the Stream-state field is encoded in the AU-header (systems only)*/
	u32 StreamStateIndication;
	/*The number of bits that is used to encode the auxiliary-data-size field 
	(no normative usage of this section)*/
	u32 AuxiliaryDataSizeLength;

	/*ISMACryp stuff*/
	u8 IV_length, IV_delta_length;
	u8 KI_length;

	/*internal stuff*/
	/*len of first AU header in an RTP payload*/
	u32 auh_first_min_len;
	u32 auh_min_len;
} GP_RTPSLMap;
	

/*packetizer config flags - some flags are dynamically re-assigned when detecting multiSL / B-Frames / ...*/
enum
{
	/*forces MPEG-4 generic transport if MPEG-4 systems mapping is available*/
	GP_RTP_PCK_FORCE_MPEG4 =	(1),
	/*Enables AUs concatenation in an RTP packet (if payload supports it) - this forces GP_RTP_PCK_SIGNAL_SIZE for MPEG-4*/
	GP_RTP_PCK_USE_MULTI	=	(1<<1),
	/*if set, audio interleaving is used if payload supports it (forces GP_RTP_PCK_USE_MULTI flag)
		THIS IS CURRENTLY NOT IMPLEMENTED*/
	GP_RTP_PCK_USE_INTERLEAVING =	(1<<2),
	/*uses static RTP payloadID if any defined*/
	GP_RTP_PCK_USE_STATIC_ID =	(1<<3),

	/*MPEG-4 generic transport option*/
	/*if flag set, RAP flag is signaled in RTP payload*/
	GP_RTP_PCK_SIGNAL_RAP	=	(1<<4),
	/*if flag set, AU indexes are signaled in RTP payload*/
	GP_RTP_PCK_SIGNAL_AU_IDX	=	(1<<5),
	/*if flag set, AU size is signaled in RTP payload*/
	GP_RTP_PCK_SIGNAL_SIZE	=	(1<<6),
	/*if flag set, CTS is signaled in RTP payload - DTS is automatically set if needed*/
	GP_RTP_PCK_SIGNAL_TS	=	(1<<7),

	/*setup payload for carouseling of systems streams*/
	GP_RTP_PCK_AUTO_CAROUSEL = (1<<8),

	/*use LATM payload for AAC-LC*/
	GP_RTP_PCK_USE_LATM_AAC	=	(1<<9),

	/*ISMACryp options*/
	/*signals that input data is selectively encrypted (eg not all input frames are encrypted) 
	- this is usually automatically set by hinter*/
	GP_RTP_PCK_SELECTIVE_ENCRYPTION =	(1<<10),
	/*signals that each sample will have its own key indicator - ignored in non-multi modes
	if not set and key indicator changes, a new RTP packet will be forced*/
	GP_RTP_PCK_KEY_IDX_PER_AU =	(1<<11),
};



/*
		Generic packetization tools - used by track hinters and future live tools
*/

/*currently supported payload types*/
enum 
{
	/*not defined*/
	GF_RTP_PAYT_UNKNOWN,
	/*use generic MPEG-4 transport - RFC 3016 and RFC 3640*/
	GF_RTP_PAYT_MPEG4,
	/*use generic MPEG-1/2 video transport - RFC 2250*/
	GF_RTP_PAYT_MPEG12_VIDEO,
	/*use generic MPEG-1/2 audio transport - RFC 2250*/
	GF_RTP_PAYT_MPEG12_AUDIO,
	/*use H263 transport - RFC 2429*/
	GF_RTP_PAYT_H263,
	/*use AMR transport - RFC 3267*/
	GF_RTP_PAYT_AMR,
	/*use AMR-WB transport - RFC 3267*/
	GF_RTP_PAYT_AMR_WB,
	/*use QCELP transport - RFC 2658*/
	GF_RTP_PAYT_QCELP,
	/*use EVRC/SMV transport - RFC 3558*/
	GF_RTP_PAYT_EVRC_SMV,
	/*use 3GPP Text transport - no RFC yet, only draft*/
	GF_RTP_PAYT_3GPP_TEXT,
	/*use H264 transport - no RFC yet, only draft*/
	GF_RTP_PAYT_H264_AVC,
	/*use LATM for AAC-LC*/
	GF_RTP_PAYT_LATM,
};



/*
	RTP packetizer
*/



/*
		RTP -> SL packetization tool
	You should ONLY modify the GF_SLHeader while packetizing, all the rest is private
	to the tool.
	Also note that AU start/end is automatically updated, therefore you should only
	set CTS-DTS-OCR-sequenceNumber (which is automatically incremented when spliting a payload)
	-padding-idle infos
	SL flags are computed on the fly, but you may wish to modify them in case of 
	packet drop/... at the encoder side

*/
struct __tag_rtp_packetizer
{
	/*input packet sl header cfg. modify only if needed*/
	GF_SLHeader sl_header;

	/*
	
		PRIVATE _ DO NOT TOUCH
	*/
	
	/*RTP payload type (RFC type, NOT the RTP hdr payT)*/
	u32 rtp_payt;
	/*packetization flags*/
	u32 flags;
	/*Path MTU size without 12-bytes RTP header*/
	u32 Path_MTU;
	/*max packet duration in RTP TS*/
	u32 max_ptime;

	/*payload type of RTP packets - only one payload type can be used in GPAC*/
	u8 PayloadType;

	/*RTP header of current packet*/
	GF_RTPHeader rtp_header;

	/*RTP packet handling callbacks*/
	void (*OnNewPacket)(void *cbk_obj, GF_RTPHeader *header);
	void (*OnPacketDone)(void *cbk_obj, GF_RTPHeader *header);
	void (*OnDataReference)(void *cbk_obj, u32 payload_size, u32 offset_from_orig);
	void (*OnData)(void *cbk_obj, char *data, u32 data_size, Bool is_header);
	void *cbk_obj;
	
	/*rest of struct is for MPEG-4 Generic hinting */
	/*SL to RTP map*/
	GP_RTPSLMap slMap;
	/*SL conf and state*/
	GF_SLConfig sl_config;

	/*set to 1 if firstSL in RTP packet*/
	Bool first_sl_in_rtp;
	Bool has_AU_header;
	/*current info writers*/
	GF_BitStream *pck_hdr, *payload;

	/*AU SN of last au*/
	u32 last_au_sn;

	/*info for the current packet*/
	u32 auh_size, bytesInPacket;

	/*ISMACryp info*/
	Bool force_flush, is_encrypted;
	u64 IV, first_AU_IV;
	char *key_indicator;

	/*AVC non-IDR flag: set if all NAL in current packet are non-IDR (disposable)*/
	Bool avc_non_idr;
};

/*generic rtp builder (packetizer)*/
typedef struct __tag_rtp_packetizer GP_RTPPacketizer;

/*creates a new builder
	@hintType: hint media type, one of the above
	@flags: hint flags (cf above)
	@slc: user-given SL config to use. If none specified, default RFC config is used
	@cbk_obj: callback object passed back in functions
	@OnNewPacket: callback function starting new RTP packet
		@header: rtp header for new packet - note that RTP header flags are not used until PacketDone is called
	@OnPacketDone: callback function closing current RTP packet
		@header: final rtp header for packet
	@OnDataReference: optional, to call each time data from input buffer is added to current RTP packet. 
		If not set, data must be added through OnData
		@payload_size: size of reference data
		@offset_from_orig: start offset in input buffer
	@OnData: to call each time data is added to current RTP packet (either extra data from payload or
		data from input when not using referencing)
		@is_head: signal the data added MUST be inserted at the begining of the payload. Otherwise data
		is concatenated as recieved
*/
GP_RTPPacketizer *gf_rtp_builder_new(u32 hintType, 
						GF_SLConfig *slc, 
						u32 flags,
						void *cbk_obj, 
						void (*OnNewPacket)(void *cbk, GF_RTPHeader *header),
						void (*OnPacketDone)(void *cbk, GF_RTPHeader *header),
						void (*OnDataReference)(void *cbk, u32 payload_size, u32 offset_from_orig),
						void (*OnData)(void *cbk, char *data, u32 data_size, Bool is_head)
					);

/*destroy builder*/
void gf_rtp_builder_del(GP_RTPPacketizer *builder);

/*
		init the builder
	@PathMTU: the network transmission unit size (UDP packet size for IP, ..)
	@max_ptime: maximum packet duration IN RTP TIMESCALE
	@StreamType: MPEG-4 system stream type - MUST always be provided for payloads format specifying 
		audio or video streams
	@OTI : MPEG-4 system objectTypeIndication - may be 0 if stream is not mpeg4 systems

			*** all other params are for MultiSL draft ***
	
	  @avgSize: average size of an AU. This is not always known (real-time encoding). 
In this case you should specify a rough compute indicating how many packets could be 
stored per RTP packet. for ex AAC stereo at 44100 k / 64kbps , one AU ~= 380 bytes
so 3 AUs for 1500 MTU is ok - BE CAREFULL: MultiSL adds some SL info on top of the 12
byte RTP header so you should specify a smaller size
The packetizer will ALWAYS make sure there's no pb storing the packets so specifying 
more will result in a slight overhead in the SL mapping but the gain to singleSL 
will still be worth it.
	-Nota: at init, the packetizer can decide to switch to SingleSL if the average size 
specified is too close to the PathMTU

	@maxSize: max size of an AU. If unknown (real-time) set to 0
	@avgTS: average CTS progression (1000/FPS for video)
	@maxDTS: maximum DTS offset in case of bidirectional coding. 
	@IV_length: size (in bytes) of IV when ISMACrypted
	@KI_length: size (in bytes) of key indicator when ISMACrypted
	@pref_mode: MPEG-4 generic only, specifies the payload mode - can be NULL (mode generic)
*/

void gf_rtp_builder_init(GP_RTPPacketizer *builder, u8 PayloadType, u32 PathMTU, u32 max_ptime,
					   u32 StreamType, u32 OTI, u32 PL_ID,
					   u32 avgSize, u32 maxSize, 
					   u32 avgTS, u32 maxDTS,
					   u32 IV_length, u32 KI_length,
					   char *pref_mode);

/*set frame crypto info*/
void gp_rtp_builder_set_cryp_info(GP_RTPPacketizer *builder, u64 IV, char *key_indicator, Bool is_encrypted);
/*packetize input buffer
@data, @data_size: input buffer
@IsAUEnd: set to one if this buffer is the last of the AU
@FullAUSize: complete access unit size if known, 0 otherwise
@duration: sample duration in rtp timescale (only needed for 3GPP text streams)
@descIndex: sample description index (only needed for 3GPP text streams)
*/
GF_Err gf_rtp_builder_process(GP_RTPPacketizer *builder, char *data, u32 data_size, u8 IsAUEnd, u32 FullAUSize, u32 duration, u8 descIndex);

/*format the "fmtp: " attribute for the MPEG-4 generic packetizer. sdpline shall be at least 2000 char*/
GF_Err gf_rtp_builder_format_sdp(GP_RTPPacketizer *builder, char *payload_name, char *sdpLine, char *dsi, u32 dsi_size);
/*formats SDP payload name and media name - both MUST be at least 20 bytes*/
Bool gf_rtp_builder_get_payload_name(GP_RTPPacketizer *builder, char *szPayloadName, char *szMediaName);





/*rtp payload flags*/
enum
{
	/*AU end was detected (eg next packet is AU start)*/
	GF_RTP_NEW_AU = (1),
	/*AMR config*/
	GF_RTP_AMR_ALIGN = (1<<1),
	/*for RFC3016, signals bitstream inspection for RAP discovery*/
	GF_RTP_M4V_CHECK_RAP = (1<<2),
	/*AWFULL hack at rtp level to cope with ffmpeg h264 crashes when jumping in stream without IDR*/
	GF_RTP_AVC_WAIT_RAP = (1<<3),
	/*ISMACryp stuff*/
	GF_RTP_HAS_ISMACRYP = (1<<4),
	GF_RTP_ISMA_SEL_ENC = (1<<5),
	GF_RTP_ISMA_HAS_KEY_IDX = (1<<6)
};

/*
		SL -> RTP packetization tool

*/
struct __tag_rtp_depacketizer
{
	/*depacketize routine*/
	void (*depacketize)(struct __tag_rtp_depacketizer *rtp, GF_RTPHeader *hdr, char *payload, u32 size);

	/*output packet sl header cfg*/
	GF_SLHeader sl_hdr;

	/*RTP payload type (RFC type, NOT the RTP hdr payT)*/
	u32 payt;
	/*depacketization flags*/
	u32 flags;

	/*callback routine*/
	void (*on_sl_packet)(void *udta, char *payload, u32 size, GF_SLHeader *hdr, GF_Err e);
	void *udta;

	/*SL <-> RTP map*/
	GP_RTPSLMap sl_map;
	u32 clock_rate;

	/*inter-packet reconstruction bitstream (for 3GP text and H264)*/
	GF_BitStream *inter_bs;

	/*H264/AVC config*/
	u32 h264_pck_mode;
	
	/*3GP text reassembler state*/
	u8 nb_txt_frag, cur_txt_frag, sidx, txt_len, nb_mod_frag;

	/*ISMACryp*/
	u32 isma_scheme;
	char *key;
};

/*generic rtp builder (packetizer)*/
typedef struct __tag_rtp_depacketizer GF_RTPDepacketizer;

GF_RTPDepacketizer *gf_rtp_depacketizer_new(GF_SDPMedia *media, void (*sl_packet_cbk)(void *udta, char *payload, u32 size, GF_SLHeader *hdr, GF_Err e), void *udta);
void gf_rtp_depacketizer_del(GF_RTPDepacketizer *rtp);
void gf_rtp_depacketizer_reset(GF_RTPDepacketizer *rtp, Bool full_reset);
void gf_rtp_depacketizer_process(GF_RTPDepacketizer *rtp, GF_RTPHeader *hdr, char *payload, u32 size);

void gf_rtp_depacketizer_get_slconfig(GF_RTPDepacketizer *rtp, GF_SLConfig *sl);


#ifdef __cplusplus
}
#endif

#endif		/*_GF_IETF_H_*/