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kolibrios-fun/contrib/sdk/sources/ffmpeg/libavformat/rtpdec.c

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upload sdk git-svn-id: svn://kolibrios.org@4349 a494cfbc-eb01-0410-851d-a64ba20cac60
2013-12-15 09:09:20 +01:00
/*
* 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/time.h"
#include "libavcodec/get_bits.h"
#include "avformat.h"
#include "network.h"
#include "srtp.h"
#include "url.h"
#include "rtpdec.h"
#include "rtpdec_formats.h"
#define MIN_FEEDBACK_INTERVAL 200000 /* 200 ms in us */
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,
};
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_amr_nb_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_amr_wb_dynamic_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);
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_mp4a_latm_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_mp4v_es_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_mpeg_audio_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_mpeg_video_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_mpeg4_generic_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_mpegts_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);
ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);
ff_register_dynamic_payload_handler(&ff_qcelp_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_qdm2_dynamic_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_svq3_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_theora_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_vp8_dynamic_handler);
ff_register_dynamic_payload_handler(&opus_dynamic_handler);
ff_register_dynamic_payload_handler(&realmedia_mp3_dynamic_handler);
ff_register_dynamic_payload_handler(&speex_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_reception_time = av_gettime();
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;
}
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() - 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_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);
}
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();
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;
}
/**
* open a new RTP parse context for stream 'st'. 'st' can be NULL for
* MPEG2-TS streams.
*/
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);
if (st) {
switch (st->codec->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->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->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;
}
/**
* 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;
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(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;
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_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, *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;
packet->recvtime = av_gettime();
packet->seq = seq;
packet->len = len;
packet->buf = buf;
packet->next = *cur;
*cur = 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 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();
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->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;
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 == 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);
ff_srtp_free(&s->srtp);
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.\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(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_init_packet(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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