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ffmpeg/libavformat/rtp.c

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/*
* RTP input/output format
* Copyright (c) 2002 Fabrice Bellard.
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "avformat.h"
#include "mpegts.h"
#include "bitstream.h"
#include <unistd.h>
#include <sys/types.h>
#include <sys/socket.h>
#include <netinet/in.h>
#ifndef __BEOS__
# include <arpa/inet.h>
#else
# include "barpainet.h"
#endif
#include <netdb.h>
#include "rtp_internal.h"
//#define RTP_H264
#ifdef RTP_H264
#include "rtp_h264.h"
#endif
//#define DEBUG
/* TODO: - add RTCP statistics reporting (should be optional).
- add support for h263/mpeg4 packetized output : IDEA: send a
buffer to 'rtp_write_packet' contains all the packets for ONE
frame. Each packet should have a four byte header containing
the length in big endian format (same trick as
'url_open_dyn_packet_buf')
*/
/* from http://www.iana.org/assignments/rtp-parameters last updated 05 January 2005 */
AVRtpPayloadType_t AVRtpPayloadTypes[]=
{
{0, "PCMU", CODEC_TYPE_AUDIO, CODEC_ID_PCM_MULAW, 8000, 1},
{1, "Reserved", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{2, "Reserved", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{3, "GSM", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
{4, "G723", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
{5, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
{6, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 16000, 1},
{7, "LPC", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
{8, "PCMA", CODEC_TYPE_AUDIO, CODEC_ID_PCM_ALAW, 8000, 1},
{9, "G722", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
{10, "L16", CODEC_TYPE_AUDIO, CODEC_ID_PCM_S16BE, 44100, 2},
{11, "L16", CODEC_TYPE_AUDIO, CODEC_ID_PCM_S16BE, 44100, 1},
{12, "QCELP", CODEC_TYPE_AUDIO, CODEC_ID_QCELP, 8000, 1},
{13, "CN", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
{14, "MPA", CODEC_TYPE_AUDIO, CODEC_ID_MP2, 90000, -1},
{15, "G728", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
{16, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 11025, 1},
{17, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 22050, 1},
{18, "G729", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
{19, "reserved", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1},
{20, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1},
{21, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1},
{22, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1},
{23, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1},
{24, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1},
{25, "CelB", CODEC_TYPE_VIDEO, CODEC_ID_NONE, 90000, -1},
{26, "JPEG", CODEC_TYPE_VIDEO, CODEC_ID_MJPEG, 90000, -1},
{27, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1},
{28, "nv", CODEC_TYPE_VIDEO, CODEC_ID_NONE, 90000, -1},
{29, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1},
{30, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1},
{31, "H261", CODEC_TYPE_VIDEO, CODEC_ID_H261, 90000, -1},
{32, "MPV", CODEC_TYPE_VIDEO, CODEC_ID_MPEG1VIDEO, 90000, -1},
{33, "MP2T", CODEC_TYPE_DATA, CODEC_ID_MPEG2TS, 90000, -1},
{34, "H263", CODEC_TYPE_VIDEO, CODEC_ID_H263, 90000, -1},
{35, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{36, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{37, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{38, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{39, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{40, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{41, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{42, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{43, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{44, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{45, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{46, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{47, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{48, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{49, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{50, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{51, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{52, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{53, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{54, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{55, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{56, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{57, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{58, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{59, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{60, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{61, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{62, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{63, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{64, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{65, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{66, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{67, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{68, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{69, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{70, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{71, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{72, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{73, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{74, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{75, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{76, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{77, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{78, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{79, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{80, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{81, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{82, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{83, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{84, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{85, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{86, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{87, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{88, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{89, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{90, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{91, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{92, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{93, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{94, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{95, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{96, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{97, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{98, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{99, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{100, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{101, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{102, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{103, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{104, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{105, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{106, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{107, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{108, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{109, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{110, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{111, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{112, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{113, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{114, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{115, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{116, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{117, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{118, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{119, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{120, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{121, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{122, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{123, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{124, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{125, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{126, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{127, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
{-1, "", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}
};
/* statistics functions */
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_MPEG4AAC};
static void register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
{
handler->next= RTPFirstDynamicPayloadHandler;
RTPFirstDynamicPayloadHandler= handler;
}
void av_register_rtp_dynamic_payload_handlers()
{
register_dynamic_payload_handler(&mp4v_es_handler);
register_dynamic_payload_handler(&mpeg4_generic_handler);
#ifdef RTP_H264
register_dynamic_payload_handler(&ff_h264_dynamic_handler);
#endif
}
int rtp_get_codec_info(AVCodecContext *codec, int payload_type)
{
if (AVRtpPayloadTypes[payload_type].codec_id != CODEC_ID_NONE) {
codec->codec_type = AVRtpPayloadTypes[payload_type].codec_type;
codec->codec_id = AVRtpPayloadTypes[payload_type].codec_id;
if (AVRtpPayloadTypes[payload_type].audio_channels > 0)
codec->channels = AVRtpPayloadTypes[payload_type].audio_channels;
if (AVRtpPayloadTypes[payload_type].clock_rate > 0)
codec->sample_rate = AVRtpPayloadTypes[payload_type].clock_rate;
return 0;
}
return -1;
}
/* return < 0 if unknown payload type */
int rtp_get_payload_type(AVCodecContext *codec)
{
int i, payload_type;
/* compute the payload type */
for (payload_type = -1, i = 0; AVRtpPayloadTypes[i].pt >= 0; ++i)
if (AVRtpPayloadTypes[i].codec_id == codec->codec_id) {
if (codec->codec_id == CODEC_ID_PCM_S16BE)
if (codec->channels != AVRtpPayloadTypes[i].audio_channels)
continue;
payload_type = AVRtpPayloadTypes[i].pt;
}
return payload_type;
}
static inline uint32_t decode_be32(const uint8_t *p)
{
return (p[0] << 24) | (p[1] << 16) | (p[2] << 8) | p[3];
}
static inline uint64_t decode_be64(const uint8_t *p)
{
return ((uint64_t)decode_be32(p) << 32) | decode_be32(p + 4);
}
static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
{
if (buf[1] != 200)
return -1;
s->last_rtcp_ntp_time = decode_be64(buf + 8);
if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE)
s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
s->last_rtcp_timestamp = decode_be32(buf + 16);
return 0;
}
/**
* 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)
*/
int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
{
ByteIOContext pb;
uint8_t *buf;
int len;
int rtcp_bytes;
if (!s->rtp_ctx || (count < 1))
return -1;
/* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
s->octet_count += count;
rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
RTCP_TX_RATIO_DEN;
rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
if (rtcp_bytes < 28)
return -1;
s->last_octet_count = s->octet_count;
if (url_open_dyn_buf(&pb) < 0)
return -1;
// Receiver Report
put_byte(&pb, (RTP_VERSION << 6) + 1); /* 1 report block */
put_byte(&pb, 201);
put_be16(&pb, 7); /* length in words - 1 */
put_be32(&pb, s->ssrc); // our own SSRC
put_be32(&pb, s->ssrc); // XXX: should be the server's here!
// some placeholders we should really fill...
put_be32(&pb, ((0 << 24) | (0 & 0x0ffffff))); /* 0% lost, total 0 lost */
put_be32(&pb, (0 << 16) | s->seq);
put_be32(&pb, 0x68); /* jitter */
put_be32(&pb, -1); /* last SR timestamp */
put_be32(&pb, 1); /* delay since last SR */
// CNAME
put_byte(&pb, (RTP_VERSION << 6) + 1); /* 1 report block */
put_byte(&pb, 202);
len = strlen(s->hostname);
put_be16(&pb, (6 + len + 3) / 4); /* length in words - 1 */
put_be32(&pb, s->ssrc);
put_byte(&pb, 0x01);
put_byte(&pb, len);
put_buffer(&pb, s->hostname, len);
// padding
for (len = (6 + len) % 4; len % 4; len++) {
put_byte(&pb, 0);
}
put_flush_packet(&pb);
len = url_close_dyn_buf(&pb, &buf);
if ((len > 0) && buf) {
#if defined(DEBUG)
printf("sending %d bytes of RR\n", len);
#endif
url_write(s->rtp_ctx, buf, len);
av_free(buf);
}
return 0;
}
/**
* 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)
*/
RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, rtp_payload_data_t *rtp_payload_data)
{
RTPDemuxContext *s;
s = av_mallocz(sizeof(RTPDemuxContext));
if (!s)
return NULL;
s->payload_type = payload_type;
s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
s->ic = s1;
s->st = st;
s->rtp_payload_data = rtp_payload_data;
if (!strcmp(AVRtpPayloadTypes[payload_type].enc_name, "MP2T")) {
s->ts = mpegts_parse_open(s->ic);
if (s->ts == NULL) {
av_free(s);
return NULL;
}
} else {
switch(st->codec->codec_id) {
case CODEC_ID_MPEG1VIDEO:
case CODEC_ID_MPEG2VIDEO:
case CODEC_ID_MP2:
case CODEC_ID_MP3:
case CODEC_ID_MPEG4:
#ifdef RTP_H264
case CODEC_ID_H264:
#endif
st->need_parsing = 1;
break;
default:
break;
}
}
// needed to send back RTCP RR in RTSP sessions
s->rtp_ctx = rtpc;
gethostname(s->hostname, sizeof(s->hostname));
return s;
}
static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf)
{
int au_headers_length, au_header_size, i;
GetBitContext getbitcontext;
rtp_payload_data_t *infos;
infos = s->rtp_payload_data;
if (infos == NULL)
return -1;
/* decode the first 2 bytes where are stored the AUHeader sections
length in bits */
au_headers_length = BE_16(buf);
if (au_headers_length > RTP_MAX_PACKET_LENGTH)
return -1;
infos->au_headers_length_bytes = (au_headers_length + 7) / 8;
/* skip AU headers length section (2 bytes) */
buf += 2;
init_get_bits(&getbitcontext, buf, infos->au_headers_length_bytes * 8);
/* XXX: Wrong if optionnal additional sections are present (cts, dts etc...) */
au_header_size = infos->sizelength + infos->indexlength;
if (au_header_size <= 0 || (au_headers_length % au_header_size != 0))
return -1;
infos->nb_au_headers = au_headers_length / au_header_size;
infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers);
/* XXX: We handle multiple AU Section as only one (need to fix this for interleaving)
In my test, the faad decoder doesnt behave correctly when sending each AU one by one
but does when sending the whole as one big packet... */
infos->au_headers[0].size = 0;
infos->au_headers[0].index = 0;
for (i = 0; i < infos->nb_au_headers; ++i) {
infos->au_headers[0].size += get_bits_long(&getbitcontext, infos->sizelength);
infos->au_headers[0].index = get_bits_long(&getbitcontext, infos->indexlength);
}
infos->nb_au_headers = 1;
return 0;
}
/**
* Parse an RTP or RTCP packet directly sent as a buffer.
* @param s RTP parse context.
* @param pkt returned packet
* @param buf input buffer or NULL to read the next packets
* @param len buffer len
* @return 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).
*/
int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
const uint8_t *buf, int len)
{
unsigned int ssrc, h;
int payload_type, seq, delta_timestamp, ret;
AVStream *st;
uint32_t timestamp;
if (!buf) {
/* return the next packets, if any */
if(s->st && s->parse_packet) {
return s->parse_packet(s, pkt, 0, NULL, 0);
} else {
if (s->read_buf_index >= s->read_buf_size)
return -1;
ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
s->read_buf_size - s->read_buf_index);
if (ret < 0)
return -1;
s->read_buf_index += ret;
if (s->read_buf_index < s->read_buf_size)
return 1;
else
return 0;
}
}
if (len < 12)
return -1;
if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
return -1;
if (buf[1] >= 200 && buf[1] <= 204) {
rtcp_parse_packet(s, buf, len);
return -1;
}
payload_type = buf[1] & 0x7f;
seq = (buf[2] << 8) | buf[3];
timestamp = decode_be32(buf + 4);
ssrc = decode_be32(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;
#if defined(DEBUG) || 1
if (seq != ((s->seq + 1) & 0xffff)) {
av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
payload_type, seq, ((s->seq + 1) & 0xffff));
}
#endif
s->seq = seq;
len -= 12;
buf += 12;
if (!st) {
/* specific MPEG2TS demux support */
ret = mpegts_parse_packet(s->ts, pkt, buf, len);
if (ret < 0)
return -1;
if (ret < len) {
s->read_buf_size = len - ret;
memcpy(s->buf, buf + ret, s->read_buf_size);
s->read_buf_index = 0;
return 1;
}
} else {
// at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise.
switch(st->codec->codec_id) {
case CODEC_ID_MP2:
/* better than nothing: skip mpeg audio RTP header */
if (len <= 4)
return -1;
h = decode_be32(buf);
len -= 4;
buf += 4;
av_new_packet(pkt, len);
memcpy(pkt->data, buf, len);
break;
case CODEC_ID_MPEG1VIDEO:
/* better than nothing: skip mpeg video RTP header */
if (len <= 4)
return -1;
h = decode_be32(buf);
buf += 4;
len -= 4;
if (h & (1 << 26)) {
/* mpeg2 */
if (len <= 4)
return -1;
buf += 4;
len -= 4;
}
av_new_packet(pkt, len);
memcpy(pkt->data, buf, len);
break;
// moved from below, verbatim. this is because this section handles packets, and the lower switch handles
// timestamps.
// TODO: Put this into a dynamic packet handler...
case CODEC_ID_MPEG4AAC:
if (rtp_parse_mp4_au(s, buf))
return -1;
{
rtp_payload_data_t *infos = s->rtp_payload_data;
if (infos == NULL)
return -1;
buf += infos->au_headers_length_bytes + 2;
len -= infos->au_headers_length_bytes + 2;
/* XXX: Fixme we only handle the case where rtp_parse_mp4_au define
one au_header */
av_new_packet(pkt, infos->au_headers[0].size);
memcpy(pkt->data, buf, infos->au_headers[0].size);
buf += infos->au_headers[0].size;
len -= infos->au_headers[0].size;
}
s->read_buf_size = len;
s->buf_ptr = buf;
pkt->stream_index = s->st->index;
return 0; ///< Temporary return.
break;
default:
if(s->parse_packet) {
return s->parse_packet(s, pkt, timestamp, buf, len);
} else {
av_new_packet(pkt, len);
memcpy(pkt->data, buf, len);
}
break;
}
switch(st->codec->codec_id) {
case CODEC_ID_MP2:
case CODEC_ID_MPEG1VIDEO:
if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
int64_t addend;
/* XXX: is it really necessary to unify the timestamp base ? */
/* compute pts from timestamp with received ntp_time */
delta_timestamp = timestamp - s->last_rtcp_timestamp;
/* convert to 90 kHz without overflow */
addend = (s->last_rtcp_ntp_time - s->first_rtcp_ntp_time) >> 14;
addend = (addend * 5625) >> 14;
pkt->pts = addend + delta_timestamp;
}
break;
case CODEC_ID_MPEG4AAC:
case CODEC_ID_H264:
case CODEC_ID_MPEG4:
pkt->pts = timestamp;
break;
default:
/* no timestamp info yet */
break;
}
pkt->stream_index = s->st->index;
}
return 0;
}
void rtp_parse_close(RTPDemuxContext *s)
{
// TODO: fold this into the protocol specific data fields.
if (!strcmp(AVRtpPayloadTypes[s->payload_type].enc_name, "MP2T")) {
mpegts_parse_close(s->ts);
}
av_free(s);
}
/* rtp output */
static int rtp_write_header(AVFormatContext *s1)
{
RTPDemuxContext *s = s1->priv_data;
int payload_type, max_packet_size, n;
AVStream *st;
if (s1->nb_streams != 1)
return -1;
st = s1->streams[0];
payload_type = rtp_get_payload_type(st->codec);
if (payload_type < 0)
payload_type = RTP_PT_PRIVATE; /* private payload type */
s->payload_type = payload_type;
// following 2 FIXMies could be set based on the current time, theres normaly no info leak, as rtp will likely be transmitted immedeatly
s->base_timestamp = 0; /* FIXME: was random(), what should this be? */
s->timestamp = s->base_timestamp;
s->ssrc = 0; /* FIXME: was random(), what should this be? */
s->first_packet = 1;
max_packet_size = url_fget_max_packet_size(&s1->pb);
if (max_packet_size <= 12)
return AVERROR_IO;
s->max_payload_size = max_packet_size - 12;
switch(st->codec->codec_id) {
case CODEC_ID_MP2:
case CODEC_ID_MP3:
s->buf_ptr = s->buf + 4;
s->cur_timestamp = 0;
break;
case CODEC_ID_MPEG1VIDEO:
s->cur_timestamp = 0;
break;
case CODEC_ID_MPEG2TS:
n = s->max_payload_size / TS_PACKET_SIZE;
if (n < 1)
n = 1;
s->max_payload_size = n * TS_PACKET_SIZE;
s->buf_ptr = s->buf;
break;
default:
s->buf_ptr = s->buf;
break;
}
return 0;
}
/* send an rtcp sender report packet */
static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
{
RTPDemuxContext *s = s1->priv_data;
#if defined(DEBUG)
printf("RTCP: %02x %Lx %x\n", s->payload_type, ntp_time, s->timestamp);
#endif
put_byte(&s1->pb, (RTP_VERSION << 6));
put_byte(&s1->pb, 200);
put_be16(&s1->pb, 6); /* length in words - 1 */
put_be32(&s1->pb, s->ssrc);
put_be64(&s1->pb, ntp_time);
put_be32(&s1->pb, s->timestamp);
put_be32(&s1->pb, s->packet_count);
put_be32(&s1->pb, s->octet_count);
put_flush_packet(&s1->pb);
}
/* send an rtp packet. sequence number is incremented, but the caller
must update the timestamp itself */
static void rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
{
RTPDemuxContext *s = s1->priv_data;
#ifdef DEBUG
printf("rtp_send_data size=%d\n", len);
#endif
/* build the RTP header */
put_byte(&s1->pb, (RTP_VERSION << 6));
put_byte(&s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
put_be16(&s1->pb, s->seq);
put_be32(&s1->pb, s->timestamp);
put_be32(&s1->pb, s->ssrc);
put_buffer(&s1->pb, buf1, len);
put_flush_packet(&s1->pb);
s->seq++;
s->octet_count += len;
s->packet_count++;
}
/* send an integer number of samples and compute time stamp and fill
the rtp send buffer before sending. */
static void rtp_send_samples(AVFormatContext *s1,
const uint8_t *buf1, int size, int sample_size)
{
RTPDemuxContext *s = s1->priv_data;
int len, max_packet_size, n;
max_packet_size = (s->max_payload_size / sample_size) * sample_size;
/* not needed, but who nows */
if ((size % sample_size) != 0)
av_abort();
while (size > 0) {
len = (max_packet_size - (s->buf_ptr - s->buf));
if (len > size)
len = size;
/* copy data */
memcpy(s->buf_ptr, buf1, len);
s->buf_ptr += len;
buf1 += len;
size -= len;
n = (s->buf_ptr - s->buf);
/* if buffer full, then send it */
if (n >= max_packet_size) {
rtp_send_data(s1, s->buf, n, 0);
s->buf_ptr = s->buf;
/* update timestamp */
s->timestamp += n / sample_size;
}
}
}
/* NOTE: we suppose that exactly one frame is given as argument here */
/* XXX: test it */
static void rtp_send_mpegaudio(AVFormatContext *s1,
const uint8_t *buf1, int size)
{
RTPDemuxContext *s = s1->priv_data;
AVStream *st = s1->streams[0];
int len, count, max_packet_size;
max_packet_size = s->max_payload_size;
/* test if we must flush because not enough space */
len = (s->buf_ptr - s->buf);
if ((len + size) > max_packet_size) {
if (len > 4) {
rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
s->buf_ptr = s->buf + 4;
/* 90 KHz time stamp */
s->timestamp = s->base_timestamp +
(s->cur_timestamp * 90000LL) / st->codec->sample_rate;
}
}
/* add the packet */
if (size > max_packet_size) {
/* big packet: fragment */
count = 0;
while (size > 0) {
len = max_packet_size - 4;
if (len > size)
len = size;
/* build fragmented packet */
s->buf[0] = 0;
s->buf[1] = 0;
s->buf[2] = count >> 8;
s->buf[3] = count;
memcpy(s->buf + 4, buf1, len);
rtp_send_data(s1, s->buf, len + 4, 0);
size -= len;
buf1 += len;
count += len;
}
} else {
if (s->buf_ptr == s->buf + 4) {
/* no fragmentation possible */
s->buf[0] = 0;
s->buf[1] = 0;
s->buf[2] = 0;
s->buf[3] = 0;
}
memcpy(s->buf_ptr, buf1, size);
s->buf_ptr += size;
}
s->cur_timestamp += st->codec->frame_size;
}
/* NOTE: a single frame must be passed with sequence header if
needed. XXX: use slices. */
static void rtp_send_mpegvideo(AVFormatContext *s1,
const uint8_t *buf1, int size)
{
RTPDemuxContext *s = s1->priv_data;
AVStream *st = s1->streams[0];
int len, h, max_packet_size;
uint8_t *q;
max_packet_size = s->max_payload_size;
while (size > 0) {
/* XXX: more correct headers */
h = 0;
if (st->codec->sub_id == 2)
h |= 1 << 26; /* mpeg 2 indicator */
q = s->buf;
*q++ = h >> 24;
*q++ = h >> 16;
*q++ = h >> 8;
*q++ = h;
if (st->codec->sub_id == 2) {
h = 0;
*q++ = h >> 24;
*q++ = h >> 16;
*q++ = h >> 8;
*q++ = h;
}
len = max_packet_size - (q - s->buf);
if (len > size)
len = size;
memcpy(q, buf1, len);
q += len;
/* 90 KHz time stamp */
s->timestamp = s->base_timestamp +
av_rescale((int64_t)s->cur_timestamp * st->codec->time_base.num, 90000, st->codec->time_base.den); //FIXME pass timestamps
rtp_send_data(s1, s->buf, q - s->buf, (len == size));
buf1 += len;
size -= len;
}
s->cur_timestamp++;
}
static void rtp_send_raw(AVFormatContext *s1,
const uint8_t *buf1, int size)
{
RTPDemuxContext *s = s1->priv_data;
AVStream *st = s1->streams[0];
int len, max_packet_size;
max_packet_size = s->max_payload_size;
while (size > 0) {
len = max_packet_size;
if (len > size)
len = size;
/* 90 KHz time stamp */
s->timestamp = s->base_timestamp +
av_rescale((int64_t)s->cur_timestamp * st->codec->time_base.num, 90000, st->codec->time_base.den); //FIXME pass timestamps
rtp_send_data(s1, buf1, len, (len == size));
buf1 += len;
size -= len;
}
s->cur_timestamp++;
}
/* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
static void rtp_send_mpegts_raw(AVFormatContext *s1,
const uint8_t *buf1, int size)
{
RTPDemuxContext *s = s1->priv_data;
int len, out_len;
while (size >= TS_PACKET_SIZE) {
len = s->max_payload_size - (s->buf_ptr - s->buf);
if (len > size)
len = size;
memcpy(s->buf_ptr, buf1, len);
buf1 += len;
size -= len;
s->buf_ptr += len;
out_len = s->buf_ptr - s->buf;
if (out_len >= s->max_payload_size) {
rtp_send_data(s1, s->buf, out_len, 0);
s->buf_ptr = s->buf;
}
}
}
/* write an RTP packet. 'buf1' must contain a single specific frame. */
static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
{
RTPDemuxContext *s = s1->priv_data;
AVStream *st = s1->streams[0];
int rtcp_bytes;
int64_t ntp_time;
int size= pkt->size;
uint8_t *buf1= pkt->data;
#ifdef DEBUG
printf("%d: write len=%d\n", pkt->stream_index, size);
#endif
/* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
RTCP_TX_RATIO_DEN;
if (s->first_packet || rtcp_bytes >= 28) {
/* compute NTP time */
/* XXX: 90 kHz timestamp hardcoded */
ntp_time = (pkt->pts << 28) / 5625;
rtcp_send_sr(s1, ntp_time);
s->last_octet_count = s->octet_count;
s->first_packet = 0;
}
switch(st->codec->codec_id) {
case CODEC_ID_PCM_MULAW:
case CODEC_ID_PCM_ALAW:
case CODEC_ID_PCM_U8:
case CODEC_ID_PCM_S8:
rtp_send_samples(s1, buf1, size, 1 * st->codec->channels);
break;
case CODEC_ID_PCM_U16BE:
case CODEC_ID_PCM_U16LE:
case CODEC_ID_PCM_S16BE:
case CODEC_ID_PCM_S16LE:
rtp_send_samples(s1, buf1, size, 2 * st->codec->channels);
break;
case CODEC_ID_MP2:
case CODEC_ID_MP3:
rtp_send_mpegaudio(s1, buf1, size);
break;
case CODEC_ID_MPEG1VIDEO:
rtp_send_mpegvideo(s1, buf1, size);
break;
case CODEC_ID_MPEG2TS:
rtp_send_mpegts_raw(s1, buf1, size);
break;
default:
/* better than nothing : send the codec raw data */
rtp_send_raw(s1, buf1, size);
break;
}
return 0;
}
static int rtp_write_trailer(AVFormatContext *s1)
{
// RTPDemuxContext *s = s1->priv_data;
return 0;
}
AVOutputFormat rtp_muxer = {
"rtp",
"RTP output format",
NULL,
NULL,
sizeof(RTPDemuxContext),
CODEC_ID_PCM_MULAW,
CODEC_ID_NONE,
rtp_write_header,
rtp_write_packet,
rtp_write_trailer,
};
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