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RTP/RTSP and MPEG4-AAC audio

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- preliminary support for mpeg4-aac rtp payload (no interleaving support)
  - use udp transport as default (makes more sense with rtp, doesn't it ?)
  - some code factorization, so adding support for new rtp payload will be easier
  (I hope ;-)
patch by (Romain DEGEZ: romain degez, smartjog com)

Originally committed as revision 4306 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Romain Degez
2005-05-26 07:47:51 +00:00
committed by Michael Niedermayer
parent 3072f0cb2e
commit d1ccf0e0a6
4 changed files with 462 additions and 181 deletions
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libavformat/rtp.h
+80 -18
View File
@@ -19,22 +19,6 @@
#ifndef RTP_H
#define RTP_H
enum RTPPayloadType {
RTP_PT_ULAW = 0,
RTP_PT_GSM = 3,
RTP_PT_G723 = 4,
RTP_PT_ALAW = 8,
RTP_PT_S16BE_STEREO = 10,
RTP_PT_S16BE_MONO = 11,
RTP_PT_MPEGAUDIO = 14,
RTP_PT_JPEG = 26,
RTP_PT_H261 = 31,
RTP_PT_MPEGVIDEO = 32,
RTP_PT_MPEG2TS = 33,
RTP_PT_H263 = 34, /* old H263 encapsulation */
RTP_PT_PRIVATE = 96,
};
#define RTP_MIN_PACKET_LENGTH 12
#define RTP_MAX_PACKET_LENGTH 1500 /* XXX: suppress this define */
@@ -43,8 +27,8 @@ int rtp_get_codec_info(AVCodecContext *codec, int payload_type);
int rtp_get_payload_type(AVCodecContext *codec);
typedef struct RTPDemuxContext RTPDemuxContext;
RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_type);
typedef struct rtp_payload_data_s rtp_payload_data_s;
RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_type, rtp_payload_data_s *rtp_payload_data);
int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
const uint8_t *buf, int len);
void rtp_parse_close(RTPDemuxContext *s);
@@ -58,4 +42,82 @@ void rtp_get_file_handles(URLContext *h, int *prtp_fd, int *prtcp_fd);
extern URLProtocol rtp_protocol;
#define RTP_PT_PRIVATE 96
#define RTP_VERSION 2
#define RTP_MAX_SDES 256 /* maximum text length for SDES */
/* RTCP paquets use 0.5 % of the bandwidth */
#define RTCP_TX_RATIO_NUM 5
#define RTCP_TX_RATIO_DEN 1000
/* Structure listing usefull vars to parse RTP packet payload*/
typedef struct rtp_payload_data_s
{
int sizelength;
int indexlength;
int indexdeltalength;
int profile_level_id;
int streamtype;
int objecttype;
char *mode;
/* mpeg 4 AU headers */
struct AUHeaders {
int size;
int index;
int cts_flag;
int cts;
int dts_flag;
int dts;
int rap_flag;
int streamstate;
} *au_headers;
int nb_au_headers;
int au_headers_length_bytes;
int cur_au_index;
} rtp_payload_data_t;
typedef struct AVRtpPayloadType_s
{
int pt;
const char enc_name[50]; /* XXX: why 50 ? */
enum CodecType codec_type;
enum CodecID codec_id;
int clock_rate;
int audio_channels;
} AVRtpPayloadType_t;
typedef struct AVRtpDynamicPayloadType_s /* payload type >= 96 */
{
const char enc_name[50]; /* XXX: still why 50 ? ;-) */
enum CodecType codec_type;
enum CodecID codec_id;
} AVRtpDynamicPayloadType_t;
typedef enum {
RTCP_SR = 200,
RTCP_RR = 201,
RTCP_SDES = 202,
RTCP_BYE = 203,
RTCP_APP = 204
} rtcp_type_t;
typedef enum {
RTCP_SDES_END = 0,
RTCP_SDES_CNAME = 1,
RTCP_SDES_NAME = 2,
RTCP_SDES_EMAIL = 3,
RTCP_SDES_PHONE = 4,
RTCP_SDES_LOC = 5,
RTCP_SDES_TOOL = 6,
RTCP_SDES_NOTE = 7,
RTCP_SDES_PRIV = 8,
RTCP_SDES_IMG = 9,
RTCP_SDES_DOOR = 10,
RTCP_SDES_SOURCE = 11
} rtcp_sdes_type_t;
extern AVRtpPayloadType_t AVRtpPayloadTypes[];
extern AVRtpDynamicPayloadType_t AVRtpDynamicPayloadTypes[];
#endif /* RTP_H */