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3f95f0dda5
ffmpeg/libavformat/rtpdec.c
Martin Storsjö 3f95f0dda5 rtpdec: Move the URLContext used for RTCP RR out from the context, to a parameter 
Signed-off-by: Martin Storsjö <martin@martin.st>
2013-01-03 15:14:34 +02:00

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/*
* RTP input format
* Copyright (c) 2002 Fabrice Bellard
*
* This file is part of Libav.
*
* Libav 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.
*
* Libav 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 Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/mathematics.h"
#include "libavutil/avstring.h"
#include "libavutil/time.h"
#include "libavcodec/get_bits.h"
#include "avformat.h"
#include "mpegts.h"
#include "network.h"
#include "url.h"
#include "rtpdec.h"
#include "rtpdec_formats.h"
static RTPDynamicProtocolHandler realmedia_mp3_dynamic_handler = {
.enc_name = "X-MP3-draft-00",
.codec_type = AVMEDIA_TYPE_AUDIO,
.codec_id = AV_CODEC_ID_MP3ADU,
};
static RTPDynamicProtocolHandler speex_dynamic_handler = {
.enc_name = "speex",
.codec_type = AVMEDIA_TYPE_AUDIO,
.codec_id = AV_CODEC_ID_SPEEX,
};
static RTPDynamicProtocolHandler opus_dynamic_handler = {
.enc_name = "opus",
.codec_type = AVMEDIA_TYPE_AUDIO,
.codec_id = AV_CODEC_ID_OPUS,
};
/* statistics functions */
static RTPDynamicProtocolHandler *rtp_first_dynamic_payload_handler = NULL;
void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
{
handler->next = rtp_first_dynamic_payload_handler;
rtp_first_dynamic_payload_handler = handler;
}
void av_register_rtp_dynamic_payload_handlers(void)
{
ff_register_dynamic_payload_handler(&ff_mp4v_es_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_mpeg4_generic_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_amr_nb_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_amr_wb_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_h263_1998_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_h263_2000_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_h263_rfc2190_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_ilbc_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_jpeg_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_theora_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_qdm2_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_svq3_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_mp4a_latm_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_vp8_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_qcelp_dynamic_handler);
ff_register_dynamic_payload_handler(&realmedia_mp3_dynamic_handler);
ff_register_dynamic_payload_handler(&speex_dynamic_handler);
ff_register_dynamic_payload_handler(&opus_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);
ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);
ff_register_dynamic_payload_handler(&ff_qt_rtp_aud_handler);
ff_register_dynamic_payload_handler(&ff_qt_rtp_vid_handler);
ff_register_dynamic_payload_handler(&ff_quicktime_rtp_aud_handler);
ff_register_dynamic_payload_handler(&ff_quicktime_rtp_vid_handler);
ff_register_dynamic_payload_handler(&ff_g726_16_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_g726_24_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_g726_32_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_g726_40_dynamic_handler);
}
RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
enum AVMediaType codec_type)
{
RTPDynamicProtocolHandler *handler;
for (handler = rtp_first_dynamic_payload_handler;
handler; handler = handler->next)
if (!av_strcasecmp(name, handler->enc_name) &&
codec_type == handler->codec_type)
return handler;
return NULL;
}
RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id,
enum AVMediaType codec_type)
{
RTPDynamicProtocolHandler *handler;
for (handler = rtp_first_dynamic_payload_handler;
handler; handler = handler->next)
if (handler->static_payload_id && handler->static_payload_id == id &&
codec_type == handler->codec_type)
return handler;
return NULL;
}
static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf,
int len)
{
int payload_len;
while (len >= 4) {
payload_len = FFMIN(len, (AV_RB16(buf + 2) + 1) * 4);
switch (buf[1]) {
case RTCP_SR:
if (payload_len < 20) {
av_log(NULL, AV_LOG_ERROR,
"Invalid length for RTCP SR packet\n");
return AVERROR_INVALIDDATA;
}
s->last_rtcp_ntp_time = AV_RB64(buf + 8);
s->last_rtcp_timestamp = AV_RB32(buf + 16);
if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
if (!s->base_timestamp)
s->base_timestamp = s->last_rtcp_timestamp;
s->rtcp_ts_offset = s->last_rtcp_timestamp - s->base_timestamp;
}
break;
case RTCP_BYE:
return -RTCP_BYE;
}
buf += payload_len;
len -= payload_len;
}
return -1;
}
#define RTP_SEQ_MOD (1 << 16)
static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence)
{
memset(s, 0, sizeof(RTPStatistics));
s->max_seq = base_sequence;
s->probation = 1;
}
/*
* Called whenever there is a large jump in sequence numbers,
* or when they get out of probation...
*/
static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
{
s->max_seq = seq;
s->cycles = 0;
s->base_seq = seq - 1;
s->bad_seq = RTP_SEQ_MOD + 1;
s->received = 0;
s->expected_prior = 0;
s->received_prior = 0;
s->jitter = 0;
s->transit = 0;
}
/* Returns 1 if we should handle this packet. */
static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
{
uint16_t udelta = seq - s->max_seq;
const int MAX_DROPOUT = 3000;
const int MAX_MISORDER = 100;
const int MIN_SEQUENTIAL = 2;
/* source not valid until MIN_SEQUENTIAL packets with sequence
* seq. numbers have been received */
if (s->probation) {
if (seq == s->max_seq + 1) {
s->probation--;
s->max_seq = seq;
if (s->probation == 0) {
rtp_init_sequence(s, seq);
s->received++;
return 1;
}
} else {
s->probation = MIN_SEQUENTIAL - 1;
s->max_seq = seq;
}
} else if (udelta < MAX_DROPOUT) {
// in order, with permissible gap
if (seq < s->max_seq) {
// sequence number wrapped; count another 64k cycles
s->cycles += RTP_SEQ_MOD;
}
s->max_seq = seq;
} else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
// sequence made a large jump...
if (seq == s->bad_seq) {
/* two sequential packets -- assume that the other side
* restarted without telling us; just resync. */
rtp_init_sequence(s, seq);
} else {
s->bad_seq = (seq + 1) & (RTP_SEQ_MOD - 1);
return 0;
}
} else {
// duplicate or reordered packet...
}
s->received++;
return 1;
}
int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd, int count)
{
AVIOContext *pb;
uint8_t *buf;
int len;
int rtcp_bytes;
RTPStatistics *stats = &s->statistics;
uint32_t lost;
uint32_t extended_max;
uint32_t expected_interval;
uint32_t received_interval;
uint32_t lost_interval;
uint32_t expected;
uint32_t fraction;
uint64_t ntp_time = s->last_rtcp_ntp_time; // TODO: Get local ntp time?
if (!fd || (count < 1))
return -1;
/* TODO: I think this is way too often; RFC 1889 has algorithm for this */
/* 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 (avio_open_dyn_buf(&pb) < 0)
return -1;
// Receiver Report
avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
avio_w8(pb, RTCP_RR);
avio_wb16(pb, 7); /* length in words - 1 */
// our own SSRC: we use the server's SSRC + 1 to avoid conflicts
avio_wb32(pb, s->ssrc + 1);
avio_wb32(pb, s->ssrc); // server SSRC
// some placeholders we should really fill...
// RFC 1889/p64
extended_max = stats->cycles + stats->max_seq;
expected = extended_max - stats->base_seq + 1;
lost = expected - stats->received;
lost = FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
expected_interval = expected - stats->expected_prior;
stats->expected_prior = expected;
received_interval = stats->received - stats->received_prior;
stats->received_prior = stats->received;
lost_interval = expected_interval - received_interval;
if (expected_interval == 0 || lost_interval <= 0)
fraction = 0;
else
fraction = (lost_interval << 8) / expected_interval;
fraction = (fraction << 24) | lost;
avio_wb32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
avio_wb32(pb, extended_max); /* max sequence received */
avio_wb32(pb, stats->jitter >> 4); /* jitter */
if (s->last_rtcp_ntp_time == AV_NOPTS_VALUE) {
avio_wb32(pb, 0); /* last SR timestamp */
avio_wb32(pb, 0); /* delay since last SR */
} else {
uint32_t middle_32_bits = s->last_rtcp_ntp_time >> 16; // this is valid, right? do we need to handle 64 bit values special?
uint32_t delay_since_last = ntp_time - s->last_rtcp_ntp_time;
avio_wb32(pb, middle_32_bits); /* last SR timestamp */
avio_wb32(pb, delay_since_last); /* delay since last SR */
}
// CNAME
avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
avio_w8(pb, RTCP_SDES);
len = strlen(s->hostname);
avio_wb16(pb, (6 + len + 3) / 4); /* length in words - 1 */
avio_wb32(pb, s->ssrc + 1);
avio_w8(pb, 0x01);
avio_w8(pb, len);
avio_write(pb, s->hostname, len);
// padding
for (len = (6 + len) % 4; len % 4; len++)
avio_w8(pb, 0);
avio_flush(pb);
len = avio_close_dyn_buf(pb, &buf);
if ((len > 0) && buf) {
int av_unused result;
av_dlog(s->ic, "sending %d bytes of RR\n", len);
result = ffurl_write(fd, buf, len);
av_dlog(s->ic, "result from ffurl_write: %d\n", result);
av_free(buf);
}
return 0;
}
void ff_rtp_send_punch_packets(URLContext *rtp_handle)
{
AVIOContext *pb;
uint8_t *buf;
int len;
/* Send a small RTP packet */
if (avio_open_dyn_buf(&pb) < 0)
return;
avio_w8(pb, (RTP_VERSION << 6));
avio_w8(pb, 0); /* Payload type */
avio_wb16(pb, 0); /* Seq */
avio_wb32(pb, 0); /* Timestamp */
avio_wb32(pb, 0); /* SSRC */
avio_flush(pb);
len = avio_close_dyn_buf(pb, &buf);
if ((len > 0) && buf)
ffurl_write(rtp_handle, buf, len);
av_free(buf);
/* Send a minimal RTCP RR */
if (avio_open_dyn_buf(&pb) < 0)
return;
avio_w8(pb, (RTP_VERSION << 6));
avio_w8(pb, RTCP_RR); /* receiver report */
avio_wb16(pb, 1); /* length in words - 1 */
avio_wb32(pb, 0); /* our own SSRC */
avio_flush(pb);
len = avio_close_dyn_buf(pb, &buf);
if ((len > 0) && buf)
ffurl_write(rtp_handle, buf, len);
av_free(buf);
}
/**
* open a new RTP parse context for stream 'st'. 'st' can be NULL for
* MPEG2-TS streams to indicate that they should be demuxed inside the
* rtp demux (otherwise AV_CODEC_ID_MPEG2TS packets are returned)
*/
RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st,
int payload_type, int queue_size)
{
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->queue_size = queue_size;
rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
s->ts = ff_mpegts_parse_open(s->ic);
if (s->ts == NULL) {
av_free(s);
return NULL;
}
} else if (st) {
switch (st->codec->codec_id) {
case AV_CODEC_ID_MPEG1VIDEO:
case AV_CODEC_ID_MPEG2VIDEO:
case AV_CODEC_ID_MP2:
case AV_CODEC_ID_MP3:
case AV_CODEC_ID_MPEG4:
case AV_CODEC_ID_H263:
case AV_CODEC_ID_H264:
st->need_parsing = AVSTREAM_PARSE_FULL;
break;
case AV_CODEC_ID_VORBIS:
st->need_parsing = AVSTREAM_PARSE_HEADERS;
break;
case AV_CODEC_ID_ADPCM_G722:
/* According to RFC 3551, the stream clock rate is 8000
* even if the sample rate is 16000. */
if (st->codec->sample_rate == 8000)
st->codec->sample_rate = 16000;
break;
default:
break;
}
}
// needed to send back RTCP RR in RTSP sessions
gethostname(s->hostname, sizeof(s->hostname));
return s;
}
void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
RTPDynamicProtocolHandler *handler)
{
s->dynamic_protocol_context = ctx;
s->parse_packet = handler->parse_packet;
}
/**
* This was the second switch in rtp_parse packet.
* Normalizes time, if required, sets stream_index, etc.
*/
static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
{
if (pkt->pts != AV_NOPTS_VALUE || pkt->dts != AV_NOPTS_VALUE)
return; /* Timestamp already set by depacketizer */
if (timestamp == RTP_NOTS_VALUE)
return;
if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && s->ic->nb_streams > 1) {
int64_t addend;
int delta_timestamp;
/* compute pts from timestamp with received ntp_time */
delta_timestamp = timestamp - s->last_rtcp_timestamp;
/* convert to the PTS timebase */
addend = av_rescale(s->last_rtcp_ntp_time - s->first_rtcp_ntp_time,
s->st->time_base.den,
(uint64_t) s->st->time_base.num << 32);
pkt->pts = s->range_start_offset + s->rtcp_ts_offset + addend +
delta_timestamp;
return;
}
if (!s->base_timestamp)
s->base_timestamp = timestamp;
/* assume that the difference is INT32_MIN < x < INT32_MAX,
* but allow the first timestamp to exceed INT32_MAX */
if (!s->timestamp)
s->unwrapped_timestamp += timestamp;
else
s->unwrapped_timestamp += (int32_t)(timestamp - s->timestamp);
s->timestamp = timestamp;
pkt->pts = s->unwrapped_timestamp + s->range_start_offset -
s->base_timestamp;
}
static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
const uint8_t *buf, int len)
{
unsigned int ssrc, h;
int payload_type, seq, ret, flags = 0;
int ext;
AVStream *st;
uint32_t timestamp;
int rv = 0;
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(st ? st->codec : NULL, 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;
/* 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 extension
len -= ext;
buf += ext;
}
if (!st) {
/* specific MPEG2-TS demux support */
ret = ff_mpegts_parse_packet(s->ts, pkt, buf, len);
/* The only error that can be returned from ff_mpegts_parse_packet
* is "no more data to return from the provided buffer", so return
* AVERROR(EAGAIN) for all errors */
if (ret < 0)
return AVERROR(EAGAIN);
if (ret < len) {
s->read_buf_size = FFMIN(len - ret, sizeof(s->buf));
memcpy(s->buf, buf + ret, s->read_buf_size);
s->read_buf_index = 0;
return 1;
}
return 0;
} else if (s->parse_packet) {
rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
s->st, pkt, &timestamp, buf, len, seq, flags);
} else {
/* At this point, the RTP header has been stripped;
* This is ASSUMING that there is only 1 CSRC, which isn't wise. */
switch (st->codec->codec_id) {
case AV_CODEC_ID_MP2:
case AV_CODEC_ID_MP3:
/* better than nothing: skip MPEG audio RTP header */
if (len <= 4)
return -1;
h = AV_RB32(buf);
len -= 4;
buf += 4;
av_new_packet(pkt, len);
memcpy(pkt->data, buf, len);
break;
case AV_CODEC_ID_MPEG1VIDEO:
case AV_CODEC_ID_MPEG2VIDEO:
/* better than nothing: skip MPEG video RTP header */
if (len <= 4)
return -1;
h = AV_RB32(buf);
buf += 4;
len -= 4;
if (h & (1 << 26)) {
/* MPEG-2 */
if (len <= 4)
return -1;
buf += 4;
len -= 4;
}
av_new_packet(pkt, len);
memcpy(pkt->data, buf, len);
break;
default:
av_new_packet(pkt, len);
memcpy(pkt->data, buf, len);
break;
}
pkt->stream_index = st->index;
}
// now perform timestamp things....
finalize_packet(s, pkt, timestamp);
return rv;
}
void ff_rtp_reset_packet_queue(RTPDemuxContext *s)
{
while (s->queue) {
RTPPacket *next = s->queue->next;
av_free(s->queue->buf);
av_free(s->queue);
s->queue = next;
}
s->seq = 0;
s->queue_len = 0;
s->prev_ret = 0;
}
static void enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len)
{
uint16_t seq = AV_RB16(buf + 2);
RTPPacket *cur = s->queue, *prev = NULL, *packet;
/* Find the correct place in the queue to insert the packet */
while (cur) {
int16_t diff = seq - cur->seq;
if (diff < 0)
break;
prev = cur;
cur = cur->next;
}
packet = av_mallocz(sizeof(*packet));
if (!packet)
return;
packet->recvtime = av_gettime();
packet->seq = seq;
packet->len = len;
packet->buf = buf;
packet->next = cur;
if (prev)
prev->next = packet;
else
s->queue = packet;
s->queue_len++;
}
static int has_next_packet(RTPDemuxContext *s)
{
return s->queue && s->queue->seq == (uint16_t) (s->seq + 1);
}
int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
{
return s->queue ? s->queue->recvtime : 0;
}
static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt)
{
int rv;
RTPPacket *next;
if (s->queue_len <= 0)
return -1;
if (!has_next_packet(s))
av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
"RTP: missed %d packets\n", s->queue->seq - s->seq - 1);
/* Parse the first packet in the queue, and dequeue it */
rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len);
next = s->queue->next;
av_free(s->queue->buf);
av_free(s->queue);
s->queue = next;
s->queue_len--;
return rv;
}
static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
uint8_t **bufptr, int len)
{
uint8_t *buf = bufptr ? *bufptr : NULL;
int ret, flags = 0;
uint32_t timestamp;
int rv = 0;
if (!buf) {
/* If parsing of the previous packet actually returned 0 or an error,
* there's nothing more to be parsed from that packet, but we may have
* indicated that we can return the next enqueued packet. */
if (s->prev_ret <= 0)
return rtp_parse_queued_packet(s, pkt);
/* return the next packets, if any */
if (s->st && s->parse_packet) {
/* timestamp should be overwritten by parse_packet, if not,
* the packet is left with pts == AV_NOPTS_VALUE */
timestamp = RTP_NOTS_VALUE;
rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
s->st, pkt, &timestamp, NULL, 0, 0,
flags);
finalize_packet(s, pkt, timestamp);
return rv;
} else {
// TODO: Move to a dynamic packet handler (like above)
if (s->read_buf_index >= s->read_buf_size)
return AVERROR(EAGAIN);
ret = ff_mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
s->read_buf_size - s->read_buf_index);
if (ret < 0)
return AVERROR(EAGAIN);
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 (RTP_PT_IS_RTCP(buf[1])) {
return rtcp_parse_packet(s, buf, len);
}
if ((s->seq == 0 && !s->queue) || s->queue_size <= 1) {
/* First packet, or no reordering */
return rtp_parse_packet_internal(s, pkt, buf, len);
} else {
uint16_t seq = AV_RB16(buf + 2);
int16_t diff = seq - s->seq;
if (diff < 0) {
/* Packet older than the previously emitted one, drop */
av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
"RTP: dropping old packet received too late\n");
return -1;
} else if (diff <= 1) {
/* Correct packet */
rv = rtp_parse_packet_internal(s, pkt, buf, len);
return rv;
} else {
/* Still missing some packet, enqueue this one. */
enqueue_packet(s, buf, len);
*bufptr = NULL;
/* Return the first enqueued packet if the queue is full,
* even if we're missing something */
if (s->queue_len >= s->queue_size)
return rtp_parse_queued_packet(s, pkt);
return -1;
}
}
}
/**
* Parse an RTP or RTCP packet directly sent as a buffer.
* @param s RTP parse context.
* @param pkt returned packet
* @param bufptr pointer to the 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 ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
uint8_t **bufptr, int len)
{
int rv = rtp_parse_one_packet(s, pkt, bufptr, len);
s->prev_ret = rv;
while (rv == AVERROR(EAGAIN) && has_next_packet(s))
rv = rtp_parse_queued_packet(s, pkt);
return rv ? rv : has_next_packet(s);
}
void ff_rtp_parse_close(RTPDemuxContext *s)
{
ff_rtp_reset_packet_queue(s);
if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
ff_mpegts_parse_close(s->ts);
}
av_free(s);
}
int ff_parse_fmtp(AVStream *stream, PayloadContext *data, const char *p,
int (*parse_fmtp)(AVStream *stream,
PayloadContext *data,
char *attr, char *value))
{
char attr[256];
char *value;
int res;
int value_size = strlen(p) + 1;
if (!(value = av_malloc(value_size))) {
av_log(NULL, AV_LOG_ERROR, "Failed to allocate data for FMTP.");
return AVERROR(ENOMEM);
}
// remove protocol identifier
while (*p && *p == ' ')
p++; // strip spaces
while (*p && *p != ' ')
p++; // eat protocol identifier
while (*p && *p == ' ')
p++; // strip trailing spaces
while (ff_rtsp_next_attr_and_value(&p,
attr, sizeof(attr),
value, value_size)) {
res = parse_fmtp(stream, data, attr, value);
if (res < 0 && res != AVERROR_PATCHWELCOME) {
av_free(value);
return res;
}
}
av_free(value);
return 0;
}
int ff_rtp_finalize_packet(AVPacket *pkt, AVIOContext **dyn_buf, int stream_idx)
{
av_init_packet(pkt);
pkt->size = avio_close_dyn_buf(*dyn_buf, &pkt->data);
pkt->stream_index = stream_idx;
pkt->destruct = av_destruct_packet;
*dyn_buf = NULL;
return pkt->size;
}
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