音视频入门基础：RTP专题（5）——FFmpeg源码中，解析SDP的实现

一、引言

FFmpeg源码中通过ff_sdp_parse函数解析SDP。该函数定义在libavformat/rtsp.c中：

int ff_sdp_parse(AVFormatContext *s, const char *content)
{const char *p;int letter, i;char buf[SDP_MAX_SIZE], *q;SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;p = content;for (;;) {p += strspn(p, SPACE_CHARS);letter = *p;if (letter == '\0')break;p++;if (*p != '=')goto next_line;p++;/* get the content */q = buf;while (*p != '\n' && *p != '\r' && *p != '\0') {if ((q - buf) < sizeof(buf) - 1)*q++ = *p;p++;}*q = '\0';sdp_parse_line(s, s1, letter, buf);next_line:while (*p != '\n' && *p != '\0')p++;if (*p == '\n')p++;}for (i = 0; i < s1->nb_default_include_source_addrs; i++)av_freep(&s1->default_include_source_addrs[i]);av_freep(&s1->default_include_source_addrs);for (i = 0; i < s1->nb_default_exclude_source_addrs; i++)av_freep(&s1->default_exclude_source_addrs[i]);av_freep(&s1->default_exclude_source_addrs);return 0;
}

而ff_sdp_parse函数中又会通过sdp_parse_line函数解析SDP中的一行数据：

int ff_sdp_parse(AVFormatContext *s, const char *content)
{
//...for (;;) {//...sdp_parse_line(s, s1, letter, buf);//...}//...return 0;
}

二、sdp_parse_line函数的定义

sdp_parse_line函数定义在FFmpeg源码（本文演示用的FFmpeg源码版本为7.0.1）的源文件libavformat/rtsp.c中：

static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,int letter, const char *buf)
{RTSPState *rt = s->priv_data;char buf1[64], st_type[64];const char *p;enum AVMediaType codec_type;int payload_type;AVStream *st;RTSPStream *rtsp_st;RTSPSource *rtsp_src;struct sockaddr_storage sdp_ip;int ttl;av_log(s, AV_LOG_TRACE, "sdp: %c='%s'\n", letter, buf);p = buf;if (s1->skip_media && letter != 'm')return;switch (letter) {case 'c':get_word(buf1, sizeof(buf1), &p);if (strcmp(buf1, "IN") != 0)return;get_word(buf1, sizeof(buf1), &p);if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))return;get_word_sep(buf1, sizeof(buf1), "/", &p);if (get_sockaddr(s, buf1, &sdp_ip))return;ttl = 16;if (*p == '/') {p++;get_word_sep(buf1, sizeof(buf1), "/", &p);ttl = atoi(buf1);}if (s->nb_streams == 0) {s1->default_ip = sdp_ip;s1->default_ttl = ttl;} else {rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];rtsp_st->sdp_ip = sdp_ip;rtsp_st->sdp_ttl = ttl;}break;case 's':av_dict_set(&s->metadata, "title", p, 0);break;case 'i':if (s->nb_streams == 0) {av_dict_set(&s->metadata, "comment", p, 0);break;}break;case 'm':/* new stream */s1->skip_media  = 0;s1->seen_fmtp   = 0;s1->seen_rtpmap = 0;codec_type = AVMEDIA_TYPE_UNKNOWN;get_word(st_type, sizeof(st_type), &p);if (!strcmp(st_type, "audio")) {codec_type = AVMEDIA_TYPE_AUDIO;} else if (!strcmp(st_type, "video")) {codec_type = AVMEDIA_TYPE_VIDEO;} else if (!strcmp(st_type, "application")) {codec_type = AVMEDIA_TYPE_DATA;} else if (!strcmp(st_type, "text")) {codec_type = AVMEDIA_TYPE_SUBTITLE;}if (codec_type == AVMEDIA_TYPE_UNKNOWN ||!(rt->media_type_mask & (1 << codec_type)) ||rt->nb_rtsp_streams >= s->max_streams) {s1->skip_media = 1;return;}rtsp_st = av_mallocz(sizeof(RTSPStream));if (!rtsp_st)return;rtsp_st->stream_index = -1;dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);rtsp_st->sdp_ip = s1->default_ip;rtsp_st->sdp_ttl = s1->default_ttl;copy_default_source_addrs(s1->default_include_source_addrs,s1->nb_default_include_source_addrs,&rtsp_st->include_source_addrs,&rtsp_st->nb_include_source_addrs);copy_default_source_addrs(s1->default_exclude_source_addrs,s1->nb_default_exclude_source_addrs,&rtsp_st->exclude_source_addrs,&rtsp_st->nb_exclude_source_addrs);get_word(buf1, sizeof(buf1), &p); /* port */rtsp_st->sdp_port = atoi(buf1);get_word(buf1, sizeof(buf1), &p); /* protocol */if (!strcmp(buf1, "udp"))rt->transport = RTSP_TRANSPORT_RAW;else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))rtsp_st->feedback = 1;/* XXX: handle list of formats */get_word(buf1, sizeof(buf1), &p); /* format list */rtsp_st->sdp_payload_type = atoi(buf1);if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {/* no corresponding stream */if (rt->transport == RTSP_TRANSPORT_RAW) {if (CONFIG_RTPDEC && !rt->ts)rt->ts = avpriv_mpegts_parse_open(s);} else {const RTPDynamicProtocolHandler *handler;handler = ff_rtp_handler_find_by_id(rtsp_st->sdp_payload_type, AVMEDIA_TYPE_DATA);init_rtp_handler(handler, rtsp_st, NULL);finalize_rtp_handler_init(s, rtsp_st, NULL);}} else if (rt->server_type == RTSP_SERVER_WMS &&codec_type == AVMEDIA_TYPE_DATA) {/* RTX stream, a stream that carries all the other actual* audio/video streams. Don't expose this to the callers. */} else {st = avformat_new_stream(s, NULL);if (!st)return;st->id = rt->nb_rtsp_streams - 1;rtsp_st->stream_index = st->index;st->codecpar->codec_type = codec_type;if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {const RTPDynamicProtocolHandler *handler;/* if standard payload type, we can find the codec right now */ff_rtp_get_codec_info(st->codecpar, rtsp_st->sdp_payload_type);if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO &&st->codecpar->sample_rate > 0)avpriv_set_pts_info(st, 32, 1, st->codecpar->sample_rate);/* Even static payload types may need a custom depacketizer */handler = ff_rtp_handler_find_by_id(rtsp_st->sdp_payload_type, st->codecpar->codec_type);init_rtp_handler(handler, rtsp_st, st);finalize_rtp_handler_init(s, rtsp_st, st);}if (rt->default_lang[0])av_dict_set(&st->metadata, "language", rt->default_lang, 0);}/* put a default control url */av_strlcpy(rtsp_st->control_url, rt->control_uri,sizeof(rtsp_st->control_url));break;case 'a':if (av_strstart(p, "control:", &p)) {if (rt->nb_rtsp_streams == 0) {if (!strncmp(p, "rtsp://", 7))av_strlcpy(rt->control_uri, p,sizeof(rt->control_uri));} else {char proto[32];/* get the control url */rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];/* XXX: may need to add full url resolution */av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,NULL, NULL, 0, p);if (proto[0] == '\0') {/* relative control URL */if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')av_strlcat(rtsp_st->control_url, "/",sizeof(rtsp_st->control_url));av_strlcat(rtsp_st->control_url, p,sizeof(rtsp_st->control_url));} elseav_strlcpy(rtsp_st->control_url, p,sizeof(rtsp_st->control_url));}} else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {/* NOTE: rtpmap is only supported AFTER the 'm=' tag */get_word(buf1, sizeof(buf1), &p);payload_type = atoi(buf1);rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];if (rtsp_st->stream_index >= 0) {st = s->streams[rtsp_st->stream_index];sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);}s1->seen_rtpmap = 1;if (s1->seen_fmtp) {parse_fmtp(s, rt, payload_type, s1->delayed_fmtp);}} else if (av_strstart(p, "fmtp:", &p) ||av_strstart(p, "framesize:", &p)) {// let dynamic protocol handlers have a stab at the line.get_word(buf1, sizeof(buf1), &p);payload_type = atoi(buf1);if (s1->seen_rtpmap) {parse_fmtp(s, rt, payload_type, buf);} else {s1->seen_fmtp = 1;av_strlcpy(s1->delayed_fmtp, buf, sizeof(s1->delayed_fmtp));}} else if (av_strstart(p, "ssrc:", &p) && s->nb_streams > 0) {rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];get_word(buf1, sizeof(buf1), &p);rtsp_st->ssrc = strtoll(buf1, NULL, 10);} else if (av_strstart(p, "range:", &p)) {int64_t start, end;// this is so that seeking on a streamed file can work.rtsp_parse_range_npt(p, &start, &end);s->start_time = start;/* AV_NOPTS_VALUE means live broadcast (and can't seek) */s->duration   = (end == AV_NOPTS_VALUE) ?AV_NOPTS_VALUE : end - start;} else if (av_strstart(p, "lang:", &p)) {if (s->nb_streams > 0) {get_word(buf1, sizeof(buf1), &p);rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];if (rtsp_st->stream_index >= 0) {st = s->streams[rtsp_st->stream_index];av_dict_set(&st->metadata, "language", buf1, 0);}} elseget_word(rt->default_lang, sizeof(rt->default_lang), &p);} else if (av_strstart(p, "IsRealDataType:integer;",&p)) {if (atoi(p) == 1)rt->transport = RTSP_TRANSPORT_RDT;} else if (av_strstart(p, "SampleRate:integer;", &p) &&s->nb_streams > 0) {st = s->streams[s->nb_streams - 1];st->codecpar->sample_rate = atoi(p);} else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {// RFC 4568rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];get_word(buf1, sizeof(buf1), &p); // ignore tagget_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);p += strspn(p, SPACE_CHARS);if (av_strstart(p, "inline:", &p))get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);} else if (av_strstart(p, "source-filter:", &p)) {int exclude = 0;get_word(buf1, sizeof(buf1), &p);if (strcmp(buf1, "incl") && strcmp(buf1, "excl"))return;exclude = !strcmp(buf1, "excl");get_word(buf1, sizeof(buf1), &p);if (strcmp(buf1, "IN") != 0)return;get_word(buf1, sizeof(buf1), &p);if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6") && strcmp(buf1, "*"))return;// not checking that the destination address actually matches or is wildcardget_word(buf1, sizeof(buf1), &p);while (*p != '\0') {rtsp_src = av_mallocz(sizeof(*rtsp_src));if (!rtsp_src)return;get_word(rtsp_src->addr, sizeof(rtsp_src->addr), &p);if (exclude) {if (s->nb_streams == 0) {dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src);} else {rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];dynarray_add(&rtsp_st->exclude_source_addrs, &rtsp_st->nb_exclude_source_addrs, rtsp_src);}} else {if (s->nb_streams == 0) {dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src);} else {rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];dynarray_add(&rtsp_st->include_source_addrs, &rtsp_st->nb_include_source_addrs, rtsp_src);}}}} else {if (rt->server_type == RTSP_SERVER_WMS)ff_wms_parse_sdp_a_line(s, p);if (s->nb_streams > 0) {rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];if (rt->server_type == RTSP_SERVER_REAL)ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);if (rtsp_st->dynamic_handler &&rtsp_st->dynamic_handler->parse_sdp_a_line)rtsp_st->dynamic_handler->parse_sdp_a_line(s,rtsp_st->stream_index,rtsp_st->dynamic_protocol_context, buf);}}break;}
}

该函数的作用就是解析SDP中的一行数据。由《音视频入门基础：RTP专题（3）——SDP简介》可以知道：一个SDP会话描述由若干行文本组成，每一行文本的格式如下：<type>=<value>，其中，<type> 必须恰好是一个区分大小写的字符，而 <value> 是结构化文本，其格式取决于 <type>。

形参s：既是输入型参数也是输出型参数，指向一个AVFormatContext类型变量。s->pb存放整个SDP的文本数据。

形参s1：既是输入型参数也是输出型参数，指向一个SDPParseState类型变量。SDPParseState结构体定义如下，用于记录SDP解析的状态：

typedef struct SDPParseState {/* SDP only */struct sockaddr_storage default_ip;int            default_ttl;int            skip_media;  ///< set if an unknown m= line occursint nb_default_include_source_addrs; /**< Number of source-specific multicast include source IP address (from SDP content) */struct RTSPSource **default_include_source_addrs; /**< Source-specific multicast include source IP address (from SDP content) */int nb_default_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP address (from SDP content) */struct RTSPSource **default_exclude_source_addrs; /**< Source-specific multicast exclude source IP address (from SDP content) */int seen_rtpmap;int seen_fmtp;char delayed_fmtp[2048];
} SDPParseState;

形参letter：输入型参数，为该行的<type>值。

形参buf：输入型参数，指向该行的<value>文本数据。

三、sdp_parse_line函数的内部实现分析

sdp_parse_line函数中会通过witch-case语句，通过判断形参letter的值，即该行的<type>值，执行不同的解析：

static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,int letter, const char *buf)
{RTSPState *rt = s->priv_data;char buf1[64], st_type[64];const char *p;enum AVMediaType codec_type;int payload_type;AVStream *st;RTSPStream *rtsp_st;RTSPSource *rtsp_src;struct sockaddr_storage sdp_ip;int ttl;av_log(s, AV_LOG_TRACE, "sdp: %c='%s'\n", letter, buf);p = buf;if (s1->skip_media && letter != 'm')return;switch (letter) {
//...}
}

（一）情况一：<type>的值为'c'

<type>的值为'c'时，<value>会包含连接数据信息，此时该行SDP格式为：c=<nettype> <addrtype> <connection-address>，sdp_parse_line函数中会执行下面代码块：

    case 'c':get_word(buf1, sizeof(buf1), &p);if (strcmp(buf1, "IN") != 0)return;get_word(buf1, sizeof(buf1), &p);if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))return;get_word_sep(buf1, sizeof(buf1), "/", &p);if (get_sockaddr(s, buf1, &sdp_ip))return;ttl = 16;if (*p == '/') {p++;get_word_sep(buf1, sizeof(buf1), "/", &p);ttl = atoi(buf1);}if (s->nb_streams == 0) {s1->default_ip = sdp_ip;s1->default_ttl = ttl;} else {rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];rtsp_st->sdp_ip = sdp_ip;rtsp_st->sdp_ttl = ttl;}break;

上述代码块中，首先判断<nettype>的值是否为“IN”（表示“Internet”），如果不为“IN”，sdp_parse_line函数直接返回，终止该行解析：

        get_word(buf1, sizeof(buf1), &p);if (strcmp(buf1, "IN") != 0)return;

判断<addrtype>的值是否为IP4或IP6，如果不为IP4或IP6，sdp_parse_line函数直接返回，终止该行解析：

        get_word(buf1, sizeof(buf1), &p);if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))return;

 获取<connection-address>（连接地址），通过get_sockaddr函数得到对应的struct addrinfo结构链表：

        get_word_sep(buf1, sizeof(buf1), "/", &p);if (get_sockaddr(s, buf1, &sdp_ip))return;

将<connection-address>相关的信息赋值给rtsp_st->sdp_ip：

        ttl = 16;if (*p == '/') {p++;get_word_sep(buf1, sizeof(buf1), "/", &p);ttl = atoi(buf1);}if (s->nb_streams == 0) {s1->default_ip = sdp_ip;s1->default_ttl = ttl;} else {rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];rtsp_st->sdp_ip = sdp_ip;rtsp_st->sdp_ttl = ttl;}break;

（二）情况二：<type>的值为's'

<type>的值为's'时，<value>会包含文本会话名称，sdp_parse_line函数中会执行下面代码块将会话名称存入s->metadata的成员变量中：

    case 's':av_dict_set(&s->metadata, "title", p, 0);break;

（三）情况三：<type>的值为'm'

<type>的值为'm'时，<value>会包含媒体描述信息，此时该行SDP格式为：m=<media> <port> <proto> <fmt> ...，sdp_parse_line函数中会执行下面代码块：

case 'm':/* new stream */s1->skip_media  = 0;s1->seen_fmtp   = 0;s1->seen_rtpmap = 0;codec_type = AVMEDIA_TYPE_UNKNOWN;get_word(st_type, sizeof(st_type), &p);if (!strcmp(st_type, "audio")) {codec_type = AVMEDIA_TYPE_AUDIO;} else if (!strcmp(st_type, "video")) {codec_type = AVMEDIA_TYPE_VIDEO;} else if (!strcmp(st_type, "application")) {codec_type = AVMEDIA_TYPE_DATA;} else if (!strcmp(st_type, "text")) {codec_type = AVMEDIA_TYPE_SUBTITLE;}if (codec_type == AVMEDIA_TYPE_UNKNOWN ||!(rt->media_type_mask & (1 << codec_type)) ||rt->nb_rtsp_streams >= s->max_streams) {s1->skip_media = 1;return;}rtsp_st = av_mallocz(sizeof(RTSPStream));if (!rtsp_st)return;rtsp_st->stream_index = -1;dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);rtsp_st->sdp_ip = s1->default_ip;rtsp_st->sdp_ttl = s1->default_ttl;copy_default_source_addrs(s1->default_include_source_addrs,s1->nb_default_include_source_addrs,&rtsp_st->include_source_addrs,&rtsp_st->nb_include_source_addrs);copy_default_source_addrs(s1->default_exclude_source_addrs,s1->nb_default_exclude_source_addrs,&rtsp_st->exclude_source_addrs,&rtsp_st->nb_exclude_source_addrs);get_word(buf1, sizeof(buf1), &p); /* port */rtsp_st->sdp_port = atoi(buf1);get_word(buf1, sizeof(buf1), &p); /* protocol */if (!strcmp(buf1, "udp"))rt->transport = RTSP_TRANSPORT_RAW;else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))rtsp_st->feedback = 1;/* XXX: handle list of formats */get_word(buf1, sizeof(buf1), &p); /* format list */rtsp_st->sdp_payload_type = atoi(buf1);if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {/* no corresponding stream */if (rt->transport == RTSP_TRANSPORT_RAW) {if (CONFIG_RTPDEC && !rt->ts)rt->ts = avpriv_mpegts_parse_open(s);} else {const RTPDynamicProtocolHandler *handler;handler = ff_rtp_handler_find_by_id(rtsp_st->sdp_payload_type, AVMEDIA_TYPE_DATA);init_rtp_handler(handler, rtsp_st, NULL);finalize_rtp_handler_init(s, rtsp_st, NULL);}} else if (rt->server_type == RTSP_SERVER_WMS &&codec_type == AVMEDIA_TYPE_DATA) {/* RTX stream, a stream that carries all the other actual* audio/video streams. Don't expose this to the callers. */} else {st = avformat_new_stream(s, NULL);if (!st)return;st->id = rt->nb_rtsp_streams - 1;rtsp_st->stream_index = st->index;st->codecpar->codec_type = codec_type;if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {const RTPDynamicProtocolHandler *handler;/* if standard payload type, we can find the codec right now */ff_rtp_get_codec_info(st->codecpar, rtsp_st->sdp_payload_type);if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO &&st->codecpar->sample_rate > 0)avpriv_set_pts_info(st, 32, 1, st->codecpar->sample_rate);/* Even static payload types may need a custom depacketizer */handler = ff_rtp_handler_find_by_id(rtsp_st->sdp_payload_type, st->codecpar->codec_type);init_rtp_handler(handler, rtsp_st, st);finalize_rtp_handler_init(s, rtsp_st, st);}if (rt->default_lang[0])av_dict_set(&st->metadata, "language", rt->default_lang, 0);}/* put a default control url */av_strlcpy(rtsp_st->control_url, rt->control_uri,sizeof(rtsp_st->control_url));break;

上述代码块中，首先读取出<media>的值，让变量codec_type赋值为对应的媒体类型：

        /* new stream */s1->skip_media  = 0;s1->seen_fmtp   = 0;s1->seen_rtpmap = 0;codec_type = AVMEDIA_TYPE_UNKNOWN;get_word(st_type, sizeof(st_type), &p);if (!strcmp(st_type, "audio")) {codec_type = AVMEDIA_TYPE_AUDIO;} else if (!strcmp(st_type, "video")) {codec_type = AVMEDIA_TYPE_VIDEO;} else if (!strcmp(st_type, "application")) {codec_type = AVMEDIA_TYPE_DATA;} else if (!strcmp(st_type, "text")) {codec_type = AVMEDIA_TYPE_SUBTITLE;}

分配一个RTSPStream结构，RTSPStream结构体用于存贮RTSP流的信息：

        rtsp_st = av_mallocz(sizeof(RTSPStream));if (!rtsp_st)return;rtsp_st->stream_index = -1;dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);rtsp_st->sdp_ip = s1->default_ip;rtsp_st->sdp_ttl = s1->default_ttl;copy_default_source_addrs(s1->default_include_source_addrs,s1->nb_default_include_source_addrs,&rtsp_st->include_source_addrs,&rtsp_st->nb_include_source_addrs);copy_default_source_addrs(s1->default_exclude_source_addrs,s1->nb_default_exclude_source_addrs,&rtsp_st->exclude_source_addrs,&rtsp_st->nb_exclude_source_addrs);

读取出<port>，即发送媒体流的传输端口，赋值给rtsp_st->sdp_port：

        get_word(buf1, sizeof(buf1), &p); /* port */rtsp_st->sdp_port = atoi(buf1);

读取出<proto>，即传输协议：

        get_word(buf1, sizeof(buf1), &p); /* protocol */if (!strcmp(buf1, "udp"))rt->transport = RTSP_TRANSPORT_RAW;else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))rtsp_st->feedback = 1;

读取出<fmt>，如果 <proto> 子字段为 “RTP/AVP ”或 “RTP/SAVP”，则 <fmt> 子字段包含 RTP 有效载荷类型编号，将其赋值给rtsp_st->sdp_payload_type：

        /* XXX: handle list of formats */get_word(buf1, sizeof(buf1), &p); /* format list */rtsp_st->sdp_payload_type = atoi(buf1);

（四）情况四：<type>的值为'a'

<type>的值为'a'时，<value>会包含附加信息，sdp_parse_line函数中会执行下面代码块：

    case 'a':if (av_strstart(p, "control:", &p)) {if (rt->nb_rtsp_streams == 0) {if (!strncmp(p, "rtsp://", 7))av_strlcpy(rt->control_uri, p,sizeof(rt->control_uri));} else {char proto[32];/* get the control url */rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];/* XXX: may need to add full url resolution */av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,NULL, NULL, 0, p);if (proto[0] == '\0') {/* relative control URL */if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')av_strlcat(rtsp_st->control_url, "/",sizeof(rtsp_st->control_url));av_strlcat(rtsp_st->control_url, p,sizeof(rtsp_st->control_url));} elseav_strlcpy(rtsp_st->control_url, p,sizeof(rtsp_st->control_url));}} else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {/* NOTE: rtpmap is only supported AFTER the 'm=' tag */get_word(buf1, sizeof(buf1), &p);payload_type = atoi(buf1);rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];if (rtsp_st->stream_index >= 0) {st = s->streams[rtsp_st->stream_index];sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);}s1->seen_rtpmap = 1;if (s1->seen_fmtp) {parse_fmtp(s, rt, payload_type, s1->delayed_fmtp);}} else if (av_strstart(p, "fmtp:", &p) ||av_strstart(p, "framesize:", &p)) {// let dynamic protocol handlers have a stab at the line.get_word(buf1, sizeof(buf1), &p);payload_type = atoi(buf1);if (s1->seen_rtpmap) {parse_fmtp(s, rt, payload_type, buf);} else {s1->seen_fmtp = 1;av_strlcpy(s1->delayed_fmtp, buf, sizeof(s1->delayed_fmtp));}} else if (av_strstart(p, "ssrc:", &p) && s->nb_streams > 0) {rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];get_word(buf1, sizeof(buf1), &p);rtsp_st->ssrc = strtoll(buf1, NULL, 10);} else if (av_strstart(p, "range:", &p)) {int64_t start, end;// this is so that seeking on a streamed file can work.rtsp_parse_range_npt(p, &start, &end);s->start_time = start;/* AV_NOPTS_VALUE means live broadcast (and can't seek) */s->duration   = (end == AV_NOPTS_VALUE) ?AV_NOPTS_VALUE : end - start;} else if (av_strstart(p, "lang:", &p)) {if (s->nb_streams > 0) {get_word(buf1, sizeof(buf1), &p);rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];if (rtsp_st->stream_index >= 0) {st = s->streams[rtsp_st->stream_index];av_dict_set(&st->metadata, "language", buf1, 0);}} elseget_word(rt->default_lang, sizeof(rt->default_lang), &p);} else if (av_strstart(p, "IsRealDataType:integer;",&p)) {if (atoi(p) == 1)rt->transport = RTSP_TRANSPORT_RDT;} else if (av_strstart(p, "SampleRate:integer;", &p) &&s->nb_streams > 0) {st = s->streams[s->nb_streams - 1];st->codecpar->sample_rate = atoi(p);} else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {// RFC 4568rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];get_word(buf1, sizeof(buf1), &p); // ignore tagget_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);p += strspn(p, SPACE_CHARS);if (av_strstart(p, "inline:", &p))get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);} else if (av_strstart(p, "source-filter:", &p)) {int exclude = 0;get_word(buf1, sizeof(buf1), &p);if (strcmp(buf1, "incl") && strcmp(buf1, "excl"))return;exclude = !strcmp(buf1, "excl");get_word(buf1, sizeof(buf1), &p);if (strcmp(buf1, "IN") != 0)return;get_word(buf1, sizeof(buf1), &p);if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6") && strcmp(buf1, "*"))return;// not checking that the destination address actually matches or is wildcardget_word(buf1, sizeof(buf1), &p);while (*p != '\0') {rtsp_src = av_mallocz(sizeof(*rtsp_src));if (!rtsp_src)return;get_word(rtsp_src->addr, sizeof(rtsp_src->addr), &p);if (exclude) {if (s->nb_streams == 0) {dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src);} else {rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];dynarray_add(&rtsp_st->exclude_source_addrs, &rtsp_st->nb_exclude_source_addrs, rtsp_src);}} else {if (s->nb_streams == 0) {dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src);} else {rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];dynarray_add(&rtsp_st->include_source_addrs, &rtsp_st->nb_include_source_addrs, rtsp_src);}}}} else {if (rt->server_type == RTSP_SERVER_WMS)ff_wms_parse_sdp_a_line(s, p);if (s->nb_streams > 0) {rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];if (rt->server_type == RTSP_SERVER_REAL)ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);if (rtsp_st->dynamic_handler &&rtsp_st->dynamic_handler->parse_sdp_a_line)rtsp_st->dynamic_handler->parse_sdp_a_line(s,rtsp_st->stream_index,rtsp_st->dynamic_protocol_context, buf);}}break;

1.a=rtpmap

a=rtpmap时，SDP的该行格式为：

a=rtpmap:<payload type> <encoding name>/<clock rate> [/<encoding parameters>]，sdp_parse_line函数中会执行下面代码块把音视频压缩编码格式赋值给st->codecpar->codec_id，

else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {/* NOTE: rtpmap is only supported AFTER the 'm=' tag */get_word(buf1, sizeof(buf1), &p);payload_type = atoi(buf1);rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];if (rtsp_st->stream_index >= 0) {st = s->streams[rtsp_st->stream_index];sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);}s1->seen_rtpmap = 1;if (s1->seen_fmtp) {parse_fmtp(s, rt, payload_type, s1->delayed_fmtp);}} 

2.a=fmtp

a=fmtp时，SDP的该行信息的格式为：a=fmtp:<format> <format specific parameters>，sdp_parse_line函数中会执行下面代码块进行解析：

else if (av_strstart(p, "fmtp:", &p) ||av_strstart(p, "framesize:", &p)) {// let dynamic protocol handlers have a stab at the line.get_word(buf1, sizeof(buf1), &p);payload_type = atoi(buf1);if (s1->seen_rtpmap) {parse_fmtp(s, rt, payload_type, buf);} else {s1->seen_fmtp = 1;av_strlcpy(s1->delayed_fmtp, buf, sizeof(s1->delayed_fmtp));}} 

对于H.264视频，该行格式一般为：a=fmtp:XX packetization-mode=X; sprop-parameter-sets=XXX,XXX; profile-level-id=XXX。其解析流程可以参考：《音视频入门基础：RTP专题（6）——FFmpeg源码中，解析SDP中的packetization-mode、profile-level-id和sprop-parameter-sets实现》。