rtp.c File Reference

Go to the source code of this file.

Functions

static void register_dynamic_payload_handler (RTPDynamicProtocolHandler *handler)

void av_register_rtp_dynamic_payload_handlers (void)

int rtp_get_codec_info (AVCodecContext *codec, int payload_type)

int rtp_get_payload_type (AVCodecContext *codec)

return < 0 if unknown payload type

const char * ff_rtp_enc_name (int payload_type)

enum CodecID ff_rtp_codec_id (const char *buf, enum CodecType codec_type)

static int rtcp_parse_packet (RTPDemuxContext *s, const unsigned char *buf, int len)

static void rtp_init_statistics (RTPStatistics *s, uint16_t base_sequence)

called on parse open packet

static void rtp_init_sequence (RTPStatistics *s, uint16_t seq)

called whenever there is a large jump in sequence numbers, or when they get out of probation.

static int rtp_valid_packet_in_sequence (RTPStatistics *s, uint16_t seq)

returns 1 if we should handle this packet.

static void rtcp_update_jitter (RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)

This function is currently unused; without a valid local ntp time, I don't see how we could calculate the difference between the arrival and sent timestamp.

int rtp_check_and_send_back_rr (RTPDemuxContext *s, int count)

some rtp servers assume client is dead if they don't hear from them.

RTPDemuxContext * rtp_parse_open (AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, rtp_payload_data_t *rtp_payload_data)

open a new RTP parse context for stream 'st'.

static int rtp_parse_mp4_au (RTPDemuxContext *s, const uint8_t *buf)

static void finalize_packet (RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)

This was the second switch in rtp_parse packet.

int rtp_parse_packet (RTPDemuxContext *s, AVPacket *pkt, const uint8_t *buf, int len)

Parse an RTP or RTCP packet directly sent as a buffer.

void rtp_parse_close (RTPDemuxContext *s)

static int rtp_write_header (AVFormatContext *s1)

static void rtcp_send_sr (AVFormatContext *s1, int64_t ntp_time)

void ff_rtp_send_data (AVFormatContext *s1, const uint8_t *buf1, int len, int m)

static void rtp_send_samples (AVFormatContext *s1, const uint8_t *buf1, int size, int sample_size)

static void rtp_send_mpegaudio (AVFormatContext *s1, const uint8_t *buf1, int size)

static void rtp_send_raw (AVFormatContext *s1, const uint8_t *buf1, int size)

static void rtp_send_mpegts_raw (AVFormatContext *s1, const uint8_t *buf1, int size)

static int rtp_write_packet (AVFormatContext *s1, AVPacket *pkt)

Variables

AVRtpPayloadType_t AVRtpPayloadTypes []

RTPDynamicProtocolHandler * RTPFirstDynamicPayloadHandler = NULL

static RTPDynamicProtocolHandler mp4v_es_handler = {"MP4V-ES", CODEC_TYPE_VIDEO, CODEC_ID_MPEG4}

static RTPDynamicProtocolHandler mpeg4_generic_handler = {"mpeg4-generic", CODEC_TYPE_AUDIO, CODEC_ID_AAC}

AVOutputFormat rtp_muxer

Function Documentation

static void register_dynamic_payload_handler ( RTPDynamicProtocolHandler * handler ) [static]

Definition at line 187 of file rtp.c.

Referenced by av_register_rtp_dynamic_payload_handlers().

void av_register_rtp_dynamic_payload_handlers ( void )

Definition at line 193 of file rtp.c.

Referenced by av_register_all().

int rtp_get_codec_info	(	AVCodecContext *	codec,
		int	payload_type
	)

Definition at line 200 of file rtp.c.

Referenced by sdp_parse_line().

int rtp_get_payload_type ( AVCodecContext * codec )

return < 0 if unknown payload type

Definition at line 219 of file rtp.c.

Referenced by rtp_write_header(), and sdp_write_media().

const char* ff_rtp_enc_name ( int payload_type )

Definition at line 234 of file rtp.c.

Referenced by rtp_parse_close(), rtp_parse_open(), and sdp_parse_line().

enum CodecID ff_rtp_codec_id	(	const char *	buf,
		enum CodecType	codec_type
	)

Definition at line 246 of file rtp.c.

Referenced by sdp_parse_rtpmap().

static int rtcp_parse_packet	(	RTPDemuxContext *	s,
		const unsigned char *	buf,
		int	len
	)			`[static]`

Definition at line 258 of file rtp.c.

Referenced by rtp_parse_packet().

static void rtp_init_statistics	(	RTPStatistics *	s,
		uint16_t	base_sequence
	)			`[static]`

called on parse open packet

Definition at line 274 of file rtp.c.

Referenced by rtp_parse_open().

static void rtp_init_sequence	(	RTPStatistics *	s,
		uint16_t	seq
	)			`[static]`

called whenever there is a large jump in sequence numbers, or when they get out of probation.

..

Definition at line 284 of file rtp.c.

Referenced by rtp_valid_packet_in_sequence().

static int rtp_valid_packet_in_sequence	(	RTPStatistics *	s,
		uint16_t	seq
	)			`[static]`

returns 1 if we should handle this packet.

Definition at line 300 of file rtp.c.

Referenced by rtp_parse_packet().

static void rtcp_update_jitter	(	RTPStatistics *	s,
		uint32_t	sent_timestamp,
		uint32_t	arrival_timestamp
	)			`[static]`

This function is currently unused; without a valid local ntp time, I don't see how we could calculate the difference between the arrival and sent timestamp.

As a result, the jitter and transit statistics values never change. I left this in in case someone else can see a way. (rdm)

Definition at line 351 of file rtp.c.

int rtp_check_and_send_back_rr	(	RTPDemuxContext *	s,
		int	count
	)

some rtp servers assume client is dead if they don't hear from them.

.. so we send a Receiver Report to the provided ByteIO context (we don't have access to the rtcp handle from here)

Definition at line 361 of file rtp.c.

Referenced by rtsp_read_packet().

RTPDemuxContext* rtp_parse_open	(	AVFormatContext *	s1,
		AVStream *	st,
		URLContext *	rtpc,
		int	payload_type,
		rtp_payload_data_t *	rtp_payload_data
	)

open a new RTP parse context for stream 'st'.

'st' can be NULL for MPEG2TS streams to indicate that they should be demuxed inside the rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned) TODO: change this to not take rtp_payload data, and use the new dynamic payload system.

Definition at line 467 of file rtp.c.

Referenced by rtsp_read_header(), and sdp_read_header().

static int rtp_parse_mp4_au	(	RTPDemuxContext *	s,
		const uint8_t *	buf
	)			`[static]`

Definition at line 507 of file rtp.c.

Referenced by rtp_parse_packet().

static void finalize_packet	(	RTPDemuxContext *	s,
		AVPacket *	pkt,
		uint32_t	timestamp
	)			`[static]`

This was the second switch in rtp_parse packet.

Normalizes time, if required, sets stream_index, etc.

Definition at line 558 of file rtp.c.

Referenced by rtp_parse_packet().

int rtp_parse_packet	(	RTPDemuxContext *	s,
		AVPacket *	pkt,
		const uint8_t *	buf,
		int	len
	)

Parse an RTP or RTCP packet directly sent as a buffer.

Parameters:

	s	RTP parse context.
	pkt	returned packet
	buf	input buffer or NULL to read the next packets
	len	buffer len

Returns:: 0 if a packet is returned, 1 if a packet is returned and more can follow (use buf as NULL to read the next). -1 if no packet (error or no more packet).

Definition at line 598 of file rtp.c.

Referenced by rtsp_read_packet().

void rtp_parse_close ( RTPDemuxContext * s )

Definition at line 744 of file rtp.c.

Referenced by rtsp_close_streams().

static int rtp_write_header ( AVFormatContext * s1 ) [static]

Definition at line 755 of file rtp.c.

static void rtcp_send_sr	(	AVFormatContext *	s1,
		int64_t	ntp_time
	)			`[static]`

Definition at line 828 of file rtp.c.

Referenced by rtp_write_packet().

void ff_rtp_send_data	(	AVFormatContext *	s1,
		const uint8_t *	buf1,
		int	len,
		int	m
	)

Definition at line 855 of file rtp.c.

Referenced by ff_rtp_send_aac(), ff_rtp_send_mpegvideo(), rtp_send_mpegaudio(), rtp_send_mpegts_raw(), rtp_send_raw(), and rtp_send_samples().

static void rtp_send_samples	(	AVFormatContext *	s1,
		const uint8_t *	buf1,
		int	size,
		int	sample_size
	)			`[static]`

Definition at line 880 of file rtp.c.

Referenced by rtp_write_packet().

static void rtp_send_mpegaudio	(	AVFormatContext *	s1,
		const uint8_t *	buf1,
		int	size
	)			`[static]`

Definition at line 908 of file rtp.c.

Referenced by rtp_write_packet().

static void rtp_send_raw	(	AVFormatContext *	s1,
		const uint8_t *	buf1,
		int	size
	)			`[static]`

Definition at line 960 of file rtp.c.

Referenced by rtp_write_packet().

static void rtp_send_mpegts_raw	(	AVFormatContext *	s1,
		const uint8_t *	buf1,
		int	size
	)			`[static]`

Definition at line 982 of file rtp.c.

Referenced by rtp_write_packet().

static int rtp_write_packet	(	AVFormatContext *	s1,
		AVPacket *	pkt
	)			`[static]`

Definition at line 1006 of file rtp.c.

Variable Documentation

AVRtpPayloadType_t AVRtpPayloadTypes[]

Definition at line 47 of file rtp.c.

RTPDynamicProtocolHandler* RTPFirstDynamicPayloadHandler = NULL

Definition at line 182 of file rtp.c.

Referenced by sdp_parse_rtpmap().

RTPDynamicProtocolHandler mp4v_es_handler = {"MP4V-ES", CODEC_TYPE_VIDEO, CODEC_ID_MPEG4} [static]

Definition at line 184 of file rtp.c.

RTPDynamicProtocolHandler mpeg4_generic_handler = {"mpeg4-generic", CODEC_TYPE_AUDIO, CODEC_ID_AAC} [static]

Definition at line 185 of file rtp.c.

AVOutputFormat rtp_muxer

Initial value:

 {
    "rtp",
    "RTP output format",
    NULL,
    NULL,
    sizeof(RTPDemuxContext),
    CODEC_ID_PCM_MULAW,
    CODEC_ID_NONE,
    rtp_write_header,
    rtp_write_packet,
}

Definition at line 1064 of file rtp.c.


Functions
static void	register_dynamic_payload_handler (RTPDynamicProtocolHandler *handler)
void	av_register_rtp_dynamic_payload_handlers (void)
int	rtp_get_codec_info (AVCodecContext *codec, int payload_type)
int	rtp_get_payload_type (AVCodecContext *codec)
	return < 0 if unknown payload type
const char *	ff_rtp_enc_name (int payload_type)
enum CodecID	ff_rtp_codec_id (const char *buf, enum CodecType codec_type)
static int	rtcp_parse_packet (RTPDemuxContext s, const unsigned char buf, int len)
static void	rtp_init_statistics (RTPStatistics *s, uint16_t base_sequence)
	called on parse open packet
static void	rtp_init_sequence (RTPStatistics *s, uint16_t seq)
	called whenever there is a large jump in sequence numbers, or when they get out of probation.
static int	rtp_valid_packet_in_sequence (RTPStatistics *s, uint16_t seq)
	returns 1 if we should handle this packet.
static void	rtcp_update_jitter (RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
	This function is currently unused; without a valid local ntp time, I don't see how we could calculate the difference between the arrival and sent timestamp.
int	rtp_check_and_send_back_rr (RTPDemuxContext *s, int count)
	some rtp servers assume client is dead if they don't hear from them.
RTPDemuxContext *	rtp_parse_open (AVFormatContext s1, AVStream st, URLContext rtpc, int payload_type, rtp_payload_data_t rtp_payload_data)
	open a new RTP parse context for stream 'st'.
static int	rtp_parse_mp4_au (RTPDemuxContext s, const uint8_t buf)
static void	finalize_packet (RTPDemuxContext s, AVPacket pkt, uint32_t timestamp)
	This was the second switch in rtp_parse packet.
int	rtp_parse_packet (RTPDemuxContext s, AVPacket pkt, const uint8_t *buf, int len)
	Parse an RTP or RTCP packet directly sent as a buffer.
void	rtp_parse_close (RTPDemuxContext *s)
static int	rtp_write_header (AVFormatContext *s1)
static void	rtcp_send_sr (AVFormatContext *s1, int64_t ntp_time)
void	ff_rtp_send_data (AVFormatContext s1, const uint8_t buf1, int len, int m)
static void	rtp_send_samples (AVFormatContext s1, const uint8_t buf1, int size, int sample_size)
static void	rtp_send_mpegaudio (AVFormatContext s1, const uint8_t buf1, int size)
static void	rtp_send_raw (AVFormatContext s1, const uint8_t buf1, int size)
static void	rtp_send_mpegts_raw (AVFormatContext s1, const uint8_t buf1, int size)
static int	rtp_write_packet (AVFormatContext s1, AVPacket pkt)
Variables
AVRtpPayloadType_t	AVRtpPayloadTypes []
RTPDynamicProtocolHandler *	RTPFirstDynamicPayloadHandler = NULL
static RTPDynamicProtocolHandler	mp4v_es_handler = {"MP4V-ES", CODEC_TYPE_VIDEO, CODEC_ID_MPEG4}
static RTPDynamicProtocolHandler	mpeg4_generic_handler = {"mpeg4-generic", CODEC_TYPE_AUDIO, CODEC_ID_AAC}
AVOutputFormat	rtp_muxer