音视频入门基础：RTP专题（10）——FFmpeg源码中，解析RTP header的实现

一、引言

由《音视频入门基础：RTP专题（9）——FFmpeg接收RTP流的原理和内部实现》可以知道，FFmpeg接收RTP流时，其源码内部会调用rtp_read函数。而rtp_read函数内部会通过recvfrom函数接收基于UDP的RTP音视频数据。一般情况下，每通过一次recvfrom函数接收到数据算一个RTP packet。然后FFmpeg会通过rtp_parse_packet_internal函数解析该RTP packet。

二、rtp_parse_packet_internal函数的定义

rtp_parse_packet_internal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/rtpdec.c中：

static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,const uint8_t *buf, int len)
{unsigned int ssrc;int payload_type, seq, flags = 0;int ext, csrc;AVStream *st;uint32_t timestamp;int rv = 0;csrc         = buf[0] & 0x0f;ext          = buf[0] & 0x10;payload_type = buf[1] & 0x7f;if (buf[1] & 0x80)flags |= RTP_FLAG_MARKER;seq       = AV_RB16(buf + 2);timestamp = AV_RB32(buf + 4);ssrc      = AV_RB32(buf + 8);/* store the ssrc in the RTPDemuxContext */s->ssrc = ssrc;/* NOTE: we can handle only one payload type */if (s->payload_type != payload_type)return -1;st = s->st;// only do something with this if all the rtp checks pass...if (!rtp_valid_packet_in_sequence(&s->statistics, seq)) {av_log(s->ic, AV_LOG_ERROR,"RTP: PT=%02x: bad cseq %04x expected=%04x\n",payload_type, seq, ((s->seq + 1) & 0xffff));return -1;}if (buf[0] & 0x20) {int padding = buf[len - 1];if (len >= 12 + padding)len -= padding;}s->seq = seq;len   -= 12;buf   += 12;len   -= 4 * csrc;buf   += 4 * csrc;if (len < 0)return AVERROR_INVALIDDATA;/* RFC 3550 Section 5.3.1 RTP Header Extension handling */if (ext) {if (len < 4)return -1;/* calculate the header extension length (stored as number* of 32-bit words) */ext = (AV_RB16(buf + 2) + 1) << 2;if (len < ext)return -1;// skip past RTP header extensionlen -= ext;buf += ext;}if (s->handler && s->handler->parse_packet) {rv = s->handler->parse_packet(s->ic, s->dynamic_protocol_context,s->st, pkt, &timestamp, buf, len, seq,flags);} else if (st) {if ((rv = av_new_packet(pkt, len)) < 0)return rv;memcpy(pkt->data, buf, len);pkt->stream_index = st->index;} else {return AVERROR(EINVAL);}// now perform timestamp things....finalize_packet(s, pkt, timestamp);return rv;
}

该函数的作用是：解析一个RTP packet。该函数的前半部分实现了解析该RTP packet的RTP header的功能。

形参s：既是输入型参数也是输出型参数，指向一个RTPDemuxContext类型变量。RTPDemuxContext结构体声明如下，存贮RTP解复用的上下文信息：

struct RTPDemuxContext {AVFormatContext *ic;AVStream *st;int payload_type;uint32_t ssrc;uint16_t seq;uint32_t timestamp;uint32_t base_timestamp;int64_t  unwrapped_timestamp;int64_t  range_start_offset;int max_payload_size;/* used to send back RTCP RR */char hostname[256];int srtp_enabled;struct SRTPContext srtp;/** Statistics for this stream (used by RTCP receiver reports) */RTPStatistics statistics;/** Fields for packet reordering @{ */int prev_ret;     ///< The return value of the actual parsing of the previous packetRTPPacket* queue; ///< A sorted queue of buffered packets not yet returnedint queue_len;    ///< The number of packets in queueint queue_size;   ///< The size of queue, or 0 if reordering is disabled/*@}*//* rtcp sender statistics receive */uint64_t last_rtcp_ntp_time;int64_t last_rtcp_reception_time;uint64_t first_rtcp_ntp_time;uint32_t last_rtcp_timestamp;int64_t rtcp_ts_offset;/* rtcp sender statistics */unsigned int packet_count;unsigned int octet_count;unsigned int last_octet_count;int64_t last_feedback_time;/* dynamic payload stuff */const RTPDynamicProtocolHandler *handler;PayloadContext *dynamic_protocol_context;
};

形参pkt：输出型参数。执行rtp_parse_packet_internal函数后，pkt会得到从该RTP packet解析出来的信息。

形参buf：输入型参数，存放需要被解析的该RTP packet数据的缓冲区。

形参len：输入型参数，该RTP packet的字节数。

返回值：返回非负数表示成功，返回负数表示失败。

三、rtp_parse_packet_internal函数中，解析RTP header的实现

rtp_parse_packet_internal函数中，首先通过下面语句将RTP header中的CSRC count字段读取出来：

    csrc         = buf[0] & 0x0f;

将RTP header中的extension字段读取出来：

    ext          = buf[0] & 0x10;

 

将RTP header中的payload type（有效载荷类型）字段读取出来：

    payload_type = buf[1] & 0x7f;

判断RTP header中的marker字段值的值是否为1，如果为1，设置局部变量flags：

    if (buf[1] & 0x80)flags |= RTP_FLAG_MARKER;

将RTP header中的sequence number（序列号）字段读取出来。关于AV_RB16宏定义的用法可以参考：《FFmpeg源码：AV_RB32、AV_RB16、AV_RB8宏定义分析》：

    seq       = AV_RB16(buf + 2);
//...s->seq = seq;

将RTP header中的timestamp（时间戳）字段读取出来：

    timestamp = AV_RB32(buf + 4);

将RTP header中的synchronization source (SSRC) identifier字段读取出来：

    ssrc      = AV_RB32(buf + 8);/* store the ssrc in the RTPDemuxContext */s->ssrc = ssrc;

判断RTP header中的padding属性的值是否为1，如果为1，表示RTP packet末端会附加填充字节：

    if (buf[0] & 0x20) {int padding = buf[len - 1];if (len >= 12 + padding)len -= padding;}

RTP header中的RTP Fixed Header（RTP header中的固定长度部分）固定占12字节，RTP Fixed Header之后紧接着的是contributing source（CSRC） identifiers，让指针buf移动到contributing source（CSRC） identifiers之后，使得可以读取之后的数据：

    len   -= 12;buf   += 12;len   -= 4 * csrc;buf   += 4 * csrc;if (len < 0)return AVERROR_INVALIDDATA;

如果RTP header中存在RTP Header Extension，跳过RTP Header Extension的读取：

    /* RFC 3550 Section 5.3.1 RTP Header Extension handling */if (ext) {if (len < 4)return -1;/* calculate the header extension length (stored as number* of 32-bit words) */ext = (AV_RB16(buf + 2) + 1) << 2;if (len < ext)return -1;// skip past RTP header extensionlen -= ext;buf += ext;}