virtualenv/FFmpeg
1
0
Fork 0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-18 03:19:31 +02:00
Code Issues Releases Activity
d3325d555b
FFmpeg/libavformat/rtpdec.c
Martin Storsjö 86b6e387cc rtsp/rtpdec: Set the AVStream time_base directly in rtsp when it is known 
This fixes cases where the RTP time base and the sample rate of the stream
differ. Previously, the AVStream time_base was unconditionally set to
the sample rate (which initially was set to one value when parsing the
rtpmap field in the SDP, but later overridden by an a=SampleRate field).

Additionally, this makes the code actually use the stream time base set
in rtpmap for video codecs, instead of hardcoding it to always be 90 kHz.

Originally committed as revision 25908 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-07 13:29:44 +00:00

781 lines
25 KiB
C
Raw Blame History

/*
* 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
*/
/* needed for gethostname() */
#define _XOPEN_SOURCE 600
#include "libavcodec/get_bits.h"
#include "avformat.h"
#include "mpegts.h"
#include <unistd.h>
#include <strings.h>
#include "network.h"
#include "rtpdec.h"
#include "rtpdec_formats.h"
//#define DEBUG
/* TODO: - add RTCP statistics reporting (should be optional).
- add support for h263/mpeg4 packetized output : IDEA: send a
buffer to 'rtp_write_packet' contains all the packets for ONE
frame. Each packet should have a four byte header containing
the length in big endian format (same trick as
'url_open_dyn_packet_buf')
*/
RTPDynamicProtocolHandler ff_realmedia_mp3_dynamic_handler = {
.enc_name = "X-MP3-draft-00",
.codec_type = AVMEDIA_TYPE_AUDIO,
.codec_id = CODEC_ID_MP3ADU,
};
/* statistics functions */
RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
{
handler->next= RTPFirstDynamicPayloadHandler;
RTPFirstDynamicPayloadHandler= 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_h264_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(&ff_realmedia_mp3_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);
}
RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
enum AVMediaType codec_type)
{
RTPDynamicProtocolHandler *handler;
for (handler = RTPFirstDynamicPayloadHandler;
handler; handler = handler->next)
if (!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 = RTPFirstDynamicPayloadHandler;
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 >= 2) {
switch (buf[1]) {
case RTCP_SR:
if (len < 16) {
av_log(NULL, AV_LOG_ERROR, "Invalid length for RTCP SR packet\n");
return AVERROR_INVALIDDATA;
}
payload_len = (AV_RB16(buf + 2) + 1) * 4;
s->last_rtcp_ntp_time = AV_RB64(buf + 8);
if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE)
s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
s->last_rtcp_timestamp = AV_RB32(buf + 16);
buf += payload_len;
len -= payload_len;
break;
case RTCP_BYE:
return -RTCP_BYE;
default:
return -1;
}
}
return -1;
}
#define RTP_SEQ_MOD (1<<16)
/**
* called on parse open packet
*/
static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
{
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 antother 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;
}
#if 0
/**
* This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
* difference between the arrival and sent timestamp. As a result, the jitter and transit statistics values
* never change. I left this in in case someone else can see a way. (rdm)
*/
static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
{
uint32_t transit= arrival_timestamp - sent_timestamp;
int d;
s->transit= transit;
d= FFABS(transit - s->transit);
s->jitter += d - ((s->jitter + 8)>>4);
}
#endif
int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
{
ByteIOContext *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 (!s->rtp_ctx || (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 (url_open_dyn_buf(&pb) < 0)
return -1;
// Receiver Report
put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
put_byte(pb, RTCP_RR);
put_be16(pb, 7); /* length in words - 1 */
// our own SSRC: we use the server's SSRC + 1 to avoid conflicts
put_be32(pb, s->ssrc + 1);
put_be32(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;
put_be32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
put_be32(pb, extended_max); /* max sequence received */
put_be32(pb, stats->jitter>>4); /* jitter */
if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
{
put_be32(pb, 0); /* last SR timestamp */
put_be32(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;
put_be32(pb, middle_32_bits); /* last SR timestamp */
put_be32(pb, delay_since_last); /* delay since last SR */
}
// CNAME
put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
put_byte(pb, RTCP_SDES);
len = strlen(s->hostname);
put_be16(pb, (6 + len + 3) / 4); /* length in words - 1 */
put_be32(pb, s->ssrc);
put_byte(pb, 0x01);
put_byte(pb, len);
put_buffer(pb, s->hostname, len);
// padding
for (len = (6 + len) % 4; len % 4; len++) {
put_byte(pb, 0);
}
put_flush_packet(pb);
len = url_close_dyn_buf(pb, &buf);
if ((len > 0) && buf) {
int result;
dprintf(s->ic, "sending %d bytes of RR\n", len);
result= url_write(s->rtp_ctx, buf, len);
dprintf(s->ic, "result from url_write: %d\n", result);
av_free(buf);
}
return 0;
}
void rtp_send_punch_packets(URLContext* rtp_handle)
{
ByteIOContext *pb;
uint8_t *buf;
int len;
/* Send a small RTP packet */
if (url_open_dyn_buf(&pb) < 0)
return;
put_byte(pb, (RTP_VERSION << 6));
put_byte(pb, 0); /* Payload type */
put_be16(pb, 0); /* Seq */
put_be32(pb, 0); /* Timestamp */
put_be32(pb, 0); /* SSRC */
put_flush_packet(pb);
len = url_close_dyn_buf(pb, &buf);
if ((len > 0) && buf)
url_write(rtp_handle, buf, len);
av_free(buf);
/* Send a minimal RTCP RR */
if (url_open_dyn_buf(&pb) < 0)
return;
put_byte(pb, (RTP_VERSION << 6));
put_byte(pb, RTCP_RR); /* receiver report */
put_be16(pb, 1); /* length in words - 1 */
put_be32(pb, 0); /* our own SSRC */
put_flush_packet(pb);
len = url_close_dyn_buf(pb, &buf);
if ((len > 0) && buf)
url_write(rtp_handle, buf, len);
av_free(buf);
}
/**
* open a new RTP parse context for stream 'st'. 'st' can be NULL for
* MPEG2TS streams to indicate that they should be demuxed inside the
* rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
*/
RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, 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 {
switch(st->codec->codec_id) {
case CODEC_ID_MPEG1VIDEO:
case CODEC_ID_MPEG2VIDEO:
case CODEC_ID_MP2:
case CODEC_ID_MP3:
case CODEC_ID_MPEG4:
case CODEC_ID_H263:
case CODEC_ID_H264:
st->need_parsing = AVSTREAM_PARSE_FULL;
break;
case 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
s->rtp_ctx = rtpc;
gethostname(s->hostname, sizeof(s->hostname));
return s;
}
void
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 (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && timestamp != RTP_NOTS_VALUE) {
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 + addend + delta_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 MPEG2TS 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 = len - ret;
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, flags);
} else {
// at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise.
switch(st->codec->codec_id) {
case CODEC_ID_MP2:
case 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 CODEC_ID_MPEG1VIDEO:
case 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)) {
/* mpeg2 */
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, 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 (buf[1] >= RTCP_SR && buf[1] <= RTCP_APP) {
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 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 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(stream, 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;
}
Reference in New Issue View Git Blame Copy Permalink