forked from KolibriOS/kolibrios
902 lines
29 KiB
C
902 lines
29 KiB
C
/*
|
|
* 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 gsm_dynamic_handler = {
|
|
.enc_name = "GSM",
|
|
.codec_type = AVMEDIA_TYPE_AUDIO,
|
|
.codec_id = AV_CODEC_ID_GSM,
|
|
};
|
|
|
|
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 t140_dynamic_handler = { /* RFC 4103 */
|
|
.enc_name = "t140",
|
|
.codec_type = AVMEDIA_TYPE_SUBTITLE,
|
|
.codec_id = AV_CODEC_ID_TEXT,
|
|
};
|
|
|
|
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 ff_register_rtp_dynamic_payload_handlers(void)
|
|
{
|
|
ff_register_dynamic_payload_handler(&ff_ac3_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_dv_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_h261_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_hevc_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_audio_robust_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(&ff_vp9_dynamic_handler);
|
|
ff_register_dynamic_payload_handler(&gsm_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);
|
|
ff_register_dynamic_payload_handler(&t140_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 (handler->enc_name &&
|
|
!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_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)
|
|
{
|
|
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_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;
|
|
}
|
|
|
|
/**
|
|
* 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, ×tamp, 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 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_relative();
|
|
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_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, ×tamp, 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->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(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(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(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_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;
|
|
}
|