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FFmpeg/libavformat/rtpdec.c
Andreas Rheinhardt 790f793844 avutil/common: Don't auto-include mem.h 
There are lots of files that don't need it: The number of object
files that actually need it went down from 2011 to 884 here.

Keep it for external users in order to not cause breakages.

Also improve the other headers a bit while just at it.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2024-03-31 00:08:43 +01:00

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/*
* RTP input 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 "libavutil/mathematics.h"
#include "libavutil/avstring.h"
#include "libavutil/intreadwrite.h"
#include "libavutil/mem.h"
#include "libavutil/time.h"
#include "libavcodec/bytestream.h"
#include "avformat.h"
#include "network.h"
#include "srtp.h"
#include "url.h"
#include "rtpdec.h"
#include "rtpdec_formats.h"
#include "internal.h"
#define MIN_FEEDBACK_INTERVAL 200000 /* 200 ms in us */
static const RTPDynamicProtocolHandler l24_dynamic_handler = {
.enc_name = "L24",
.codec_type = AVMEDIA_TYPE_AUDIO,
.codec_id = AV_CODEC_ID_PCM_S24BE,
};
static const RTPDynamicProtocolHandler gsm_dynamic_handler = {
.enc_name = "GSM",
.codec_type = AVMEDIA_TYPE_AUDIO,
.codec_id = AV_CODEC_ID_GSM,
};
static const RTPDynamicProtocolHandler realmedia_mp3_dynamic_handler = {
.enc_name = "X-MP3-draft-00",
.codec_type = AVMEDIA_TYPE_AUDIO,
.codec_id = AV_CODEC_ID_MP3ADU,
};
static const RTPDynamicProtocolHandler speex_dynamic_handler = {
.enc_name = "speex",
.codec_type = AVMEDIA_TYPE_AUDIO,
.codec_id = AV_CODEC_ID_SPEEX,
};
static const RTPDynamicProtocolHandler opus_dynamic_handler = {
.enc_name = "opus",
.codec_type = AVMEDIA_TYPE_AUDIO,
.codec_id = AV_CODEC_ID_OPUS,
};
static const RTPDynamicProtocolHandler t140_dynamic_handler = { /* RFC 4103 */
.enc_name = "t140",
.codec_type = AVMEDIA_TYPE_SUBTITLE,
.codec_id = AV_CODEC_ID_TEXT,
};
extern const RTPDynamicProtocolHandler ff_rdt_video_handler;
extern const RTPDynamicProtocolHandler ff_rdt_audio_handler;
extern const RTPDynamicProtocolHandler ff_rdt_live_video_handler;
extern const RTPDynamicProtocolHandler ff_rdt_live_audio_handler;
static const RTPDynamicProtocolHandler *const rtp_dynamic_protocol_handler_list[] = {
/* rtp */
&ff_ac3_dynamic_handler,
&ff_amr_nb_dynamic_handler,
&ff_amr_wb_dynamic_handler,
&ff_dv_dynamic_handler,
&ff_g726_16_dynamic_handler,
&ff_g726_24_dynamic_handler,
&ff_g726_32_dynamic_handler,
&ff_g726_40_dynamic_handler,
&ff_g726le_16_dynamic_handler,
&ff_g726le_24_dynamic_handler,
&ff_g726le_32_dynamic_handler,
&ff_g726le_40_dynamic_handler,
&ff_h261_dynamic_handler,
&ff_h263_1998_dynamic_handler,
&ff_h263_2000_dynamic_handler,
&ff_h263_rfc2190_dynamic_handler,
&ff_h264_dynamic_handler,
&ff_hevc_dynamic_handler,
&ff_ilbc_dynamic_handler,
&ff_jpeg_dynamic_handler,
&ff_mp4a_latm_dynamic_handler,
&ff_mp4v_es_dynamic_handler,
&ff_mpeg_audio_dynamic_handler,
&ff_mpeg_audio_robust_dynamic_handler,
&ff_mpeg_video_dynamic_handler,
&ff_mpeg4_generic_dynamic_handler,
&ff_mpegts_dynamic_handler,
&ff_ms_rtp_asf_pfa_handler,
&ff_ms_rtp_asf_pfv_handler,
&ff_qcelp_dynamic_handler,
&ff_qdm2_dynamic_handler,
&ff_qt_rtp_aud_handler,
&ff_qt_rtp_vid_handler,
&ff_quicktime_rtp_aud_handler,
&ff_quicktime_rtp_vid_handler,
&ff_rfc4175_rtp_handler,
&ff_svq3_dynamic_handler,
&ff_theora_dynamic_handler,
&ff_vc2hq_dynamic_handler,
&ff_vorbis_dynamic_handler,
&ff_vp8_dynamic_handler,
&ff_vp9_dynamic_handler,
&gsm_dynamic_handler,
&l24_dynamic_handler,
&opus_dynamic_handler,
&realmedia_mp3_dynamic_handler,
&speex_dynamic_handler,
&t140_dynamic_handler,
/* rdt */
&ff_rdt_video_handler,
&ff_rdt_audio_handler,
&ff_rdt_live_video_handler,
&ff_rdt_live_audio_handler,
NULL,
};
/**
* Iterate over all registered rtp dynamic protocol handlers.
*
* @param opaque a pointer where libavformat will store the iteration state.
* Must point to NULL to start the iteration.
*
* @return the next registered rtp dynamic protocol handler
* or NULL when the iteration is finished
*/
static const RTPDynamicProtocolHandler *rtp_handler_iterate(void **opaque)
{
uintptr_t i = (uintptr_t)*opaque;
const RTPDynamicProtocolHandler *r = rtp_dynamic_protocol_handler_list[i];
if (r)
*opaque = (void*)(i + 1);
return r;
}
const RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
enum AVMediaType codec_type)
{
void *i = 0;
const RTPDynamicProtocolHandler *handler;
while (handler = rtp_handler_iterate(&i)) {
if (handler->enc_name &&
!av_strcasecmp(name, handler->enc_name) &&
codec_type == handler->codec_type)
return handler;
}
return NULL;
}
const RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id,
enum AVMediaType codec_type)
{
void *i = 0;
const RTPDynamicProtocolHandler *handler;
while (handler = rtp_handler_iterate(&i)) {
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(s->ic, AV_LOG_ERROR, "Invalid RTCP SR packet length\n");
return AVERROR_INVALIDDATA;
}
s->last_rtcp_reception_time = av_gettime_relative();
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 = (int32_t)(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;
}
static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp,
uint32_t arrival_timestamp)
{
// Most of this is pretty straight from RFC 3550 appendix A.8
uint32_t transit = arrival_timestamp - sent_timestamp;
uint32_t prev_transit = s->transit;
int32_t d = transit - prev_transit;
// Doing the FFABS() call directly on the "transit - prev_transit"
// expression doesn't work, since it's an unsigned expression. Doing the
// transit calculation in unsigned is desired though, since it most
// probably will need to wrap around.
d = FFABS(d);
s->transit = transit;
if (!prev_transit)
return;
s->jitter += d - (int32_t) ((s->jitter + 8) >> 4);
}
int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd,
AVIOContext *avio, 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;
int32_t lost_interval;
uint32_t expected;
uint32_t fraction;
if ((!fd && !avio) || (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 (!fd)
pb = avio;
else 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;
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 = av_rescale(av_gettime_relative() - s->last_rtcp_reception_time,
65536, AV_TIME_BASE);
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, (7 + 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);
avio_w8(pb, 0); /* END */
// padding
for (len = (7 + len) % 4; len % 4; len++)
avio_w8(pb, 0);
avio_flush(pb);
if (!fd)
return 0;
len = avio_close_dyn_buf(pb, &buf);
if ((len > 0) && buf) {
int av_unused result;
av_log(s->ic, AV_LOG_TRACE, "sending %d bytes of RR\n", len);
result = ffurl_write(fd, buf, len);
av_log(s->ic, AV_LOG_TRACE, "result from ffurl_write: %d\n", result);
av_free(buf);
}
return 0;
}
void ff_rtp_send_punch_packets(URLContext *rtp_handle)
{
uint8_t buf[RTP_MIN_PACKET_LENGTH], *ptr = buf;
/* Send a small RTP packet */
bytestream_put_byte(&ptr, (RTP_VERSION << 6));
bytestream_put_byte(&ptr, 0); /* Payload type */
bytestream_put_be16(&ptr, 0); /* Seq */
bytestream_put_be32(&ptr, 0); /* Timestamp */
bytestream_put_be32(&ptr, 0); /* SSRC */
ffurl_write(rtp_handle, buf, ptr - buf);
/* Send a minimal RTCP RR */
ptr = buf;
bytestream_put_byte(&ptr, (RTP_VERSION << 6));
bytestream_put_byte(&ptr, RTCP_RR); /* receiver report */
bytestream_put_be16(&ptr, 1); /* length in words - 1 */
bytestream_put_be32(&ptr, 0); /* our own SSRC */
ffurl_write(rtp_handle, buf, ptr - buf);
}
static int find_missing_packets(RTPDemuxContext *s, uint16_t *first_missing,
uint16_t *missing_mask)
{
int i;
uint16_t next_seq = s->seq + 1;
RTPPacket *pkt = s->queue;
if (!pkt || pkt->seq == next_seq)
return 0;
*missing_mask = 0;
for (i = 1; i <= 16; i++) {
uint16_t missing_seq = next_seq + i;
while (pkt) {
int16_t diff = pkt->seq - missing_seq;
if (diff >= 0)
break;
pkt = pkt->next;
}
if (!pkt)
break;
if (pkt->seq == missing_seq)
continue;
*missing_mask |= 1 << (i - 1);
}
*first_missing = next_seq;
return 1;
}
int ff_rtp_send_rtcp_feedback(RTPDemuxContext *s, URLContext *fd,
AVIOContext *avio)
{
int len, need_keyframe, missing_packets;
AVIOContext *pb;
uint8_t *buf;
int64_t now;
uint16_t first_missing = 0, missing_mask = 0;
if (!fd && !avio)
return -1;
need_keyframe = s->handler && s->handler->need_keyframe &&
s->handler->need_keyframe(s->dynamic_protocol_context);
missing_packets = find_missing_packets(s, &first_missing, &missing_mask);
if (!need_keyframe && !missing_packets)
return 0;
/* Send new feedback if enough time has elapsed since the last
* feedback packet. */
now = av_gettime_relative();
if (s->last_feedback_time &&
(now - s->last_feedback_time) < MIN_FEEDBACK_INTERVAL)
return 0;
s->last_feedback_time = now;
if (!fd)
pb = avio;
else if (avio_open_dyn_buf(&pb) < 0)
return -1;
if (need_keyframe) {
avio_w8(pb, (RTP_VERSION << 6) | 1); /* PLI */
avio_w8(pb, RTCP_PSFB);
avio_wb16(pb, 2); /* 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
}
if (missing_packets) {
avio_w8(pb, (RTP_VERSION << 6) | 1); /* NACK */
avio_w8(pb, RTCP_RTPFB);
avio_wb16(pb, 3); /* length in words - 1 */
avio_wb32(pb, s->ssrc + 1);
avio_wb32(pb, s->ssrc); // server SSRC
avio_wb16(pb, first_missing);
avio_wb16(pb, missing_mask);
}
avio_flush(pb);
if (!fd)
return 0;
len = avio_close_dyn_buf(pb, &buf);
if (len > 0 && buf) {
ffurl_write(fd, buf, len);
av_free(buf);
}
return 0;
}
static int opus_write_extradata(AVCodecParameters *codecpar)
{
uint8_t *bs;
int ret;
/* This function writes an extradata with a channel mapping family of 0.
* This mapping family only supports mono and stereo layouts. And RFC7587
* specifies that the number of channels in the SDP must be 2.
*/
if (codecpar->ch_layout.nb_channels > 2) {
return AVERROR_INVALIDDATA;
}
ret = ff_alloc_extradata(codecpar, 19);
if (ret < 0)
return ret;
bs = (uint8_t *)codecpar->extradata;
/* Opus magic */
bytestream_put_buffer(&bs, "OpusHead", 8);
/* Version */
bytestream_put_byte (&bs, 0x1);
/* Channel count */
bytestream_put_byte (&bs, codecpar->ch_layout.nb_channels);
/* Pre skip */
bytestream_put_le16 (&bs, 0);
/* Input sample rate */
bytestream_put_le32 (&bs, 48000);
/* Output gain */
bytestream_put_le16 (&bs, 0x0);
/* Mapping family */
bytestream_put_byte (&bs, 0x0);
return 0;
}
/**
* open a new RTP parse context for stream 'st'. 'st' can be NULL for
* MPEG-2 TS streams.
*/
RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st,
int payload_type, int queue_size)
{
RTPDemuxContext *s;
int ret;
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;
av_log(s->ic, AV_LOG_VERBOSE, "setting jitter buffer size to %d\n",
s->queue_size);
rtp_init_statistics(&s->statistics, 0);
if (st) {
switch (st->codecpar->codec_id) {
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->codecpar->sample_rate == 8000)
st->codecpar->sample_rate = 16000;
break;
case AV_CODEC_ID_OPUS:
ret = opus_write_extradata(st->codecpar);
if (ret < 0) {
av_log(s1, AV_LOG_ERROR,
"Error creating opus extradata: %s\n",
av_err2str(ret));
av_free(s);
return NULL;
}
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,
const RTPDynamicProtocolHandler *handler)
{
s->dynamic_protocol_context = ctx;
s->handler = handler;
}
void ff_rtp_parse_set_crypto(RTPDemuxContext *s, const char *suite,
const char *params)
{
if (!ff_srtp_set_crypto(&s->srtp, suite, params))
s->srtp_enabled = 1;
}
static int rtp_set_prft(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp) {
int64_t rtcp_time, delta_timestamp, delta_time;
AVProducerReferenceTime *prft =
(AVProducerReferenceTime *) av_packet_new_side_data(
pkt, AV_PKT_DATA_PRFT, sizeof(AVProducerReferenceTime));
if (!prft)
return AVERROR(ENOMEM);
rtcp_time = ff_parse_ntp_time(s->last_rtcp_ntp_time) - NTP_OFFSET_US;
delta_timestamp = (int64_t)timestamp - (int64_t)s->last_rtcp_timestamp;
delta_time = av_rescale_q(delta_timestamp, s->st->time_base, AV_TIME_BASE_Q);
prft->wallclock = rtcp_time + delta_time;
prft->flags = 24;
return 0;
}
/**
* 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) {
if (rtp_set_prft(s, pkt, timestamp) < 0) {
av_log(s->ic, AV_LOG_WARNING, "rtpdec: failed to set prft");
}
}
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;
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 extension
len -= 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;
}
void ff_rtp_reset_packet_queue(RTPDemuxContext *s)
{
while (s->queue) {
RTPPacket *next = s->queue->next;
av_freep(&s->queue->buf);
av_freep(&s->queue);
s->queue = next;
}
s->seq = 0;
s->queue_len = 0;
s->prev_ret = 0;
}
static int enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len)
{
uint16_t seq = AV_RB16(buf + 2);
RTPPacket **cur = &s->queue, *packet;
/* Find the correct place in the queue to insert the packet */
while (*cur) {
int16_t diff = seq - (*cur)->seq;
if (diff < 0)
break;
cur = &(*cur)->next;
}
packet = av_mallocz(sizeof(*packet));
if (!packet)
return AVERROR(ENOMEM);
packet->recvtime = av_gettime_relative();
packet->seq = seq;
packet->len = len;
packet->buf = buf;
packet->next = *cur;
*cur = packet;
s->queue_len++;
return 0;
}
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)) {
int pkt_missed = s->queue->seq - s->seq - 1;
if (pkt_missed < 0)
pkt_missed += UINT16_MAX;
av_log(s->ic, AV_LOG_WARNING,
"RTP: missed %d packets\n", pkt_missed);
}
/* 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_freep(&s->queue->buf);
av_freep(&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 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->handler && s->handler->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->handler->parse_packet(s->ic, s->dynamic_protocol_context,
s->st, pkt, &timestamp, NULL, 0, 0,
flags);
finalize_packet(s, pkt, timestamp);
return rv;
}
}
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->st) {
int64_t received = av_gettime_relative();
uint32_t arrival_ts = av_rescale_q(received, AV_TIME_BASE_Q,
s->st->time_base);
timestamp = AV_RB32(buf + 4);
// Calculate the jitter immediately, before queueing the packet
// into the reordering queue.
rtcp_update_jitter(&s->statistics, timestamp, arrival_ts);
}
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->ic, 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. */
rv = enqueue_packet(s, buf, len);
if (rv < 0)
return rv;
*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) {
av_log(s->ic, AV_LOG_WARNING, "jitter buffer full\n");
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;
if (s->srtp_enabled && bufptr && ff_srtp_decrypt(&s->srtp, *bufptr, &len) < 0)
return -1;
rv = rtp_parse_one_packet(s, pkt, bufptr, len);
s->prev_ret = rv;
while (rv < 0 && 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);
ff_srtp_free(&s->srtp);
av_free(s);
}
int ff_parse_fmtp(AVFormatContext *s,
AVStream *stream, PayloadContext *data, const char *p,
int (*parse_fmtp)(AVFormatContext *s,
AVStream *stream,
PayloadContext *data,
const char *attr, const char *value))
{
char attr[256];
char *value;
int res;
int value_size = strlen(p) + 1;
if (!(value = av_malloc(value_size))) {
av_log(s, AV_LOG_ERROR, "Failed to allocate data for FMTP.\n");
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(s, 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)
{
int ret;
av_packet_unref(pkt);
pkt->size = avio_close_dyn_buf(*dyn_buf, &pkt->data);
pkt->stream_index = stream_idx;
*dyn_buf = NULL;
if ((ret = av_packet_from_data(pkt, pkt->data, pkt->size)) < 0) {
av_freep(&pkt->data);
return ret;
}
return pkt->size;
}
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