* RTP input format
* Copyright (c) 2002 Fabrice Bellard
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
- * FFmpeg is distributed in the hope that it will be useful,
+ * Libav is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
+ * License along with Libav; 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 "libavutil/mathematics.h"
+#include "libavutil/avstring.h"
+#include "libavutil/time.h"
#include "libavcodec/get_bits.h"
#include "avformat.h"
#include "mpegts.h"
+#include "url.h"
-#include <unistd.h>
#include "network.h"
#include "rtpdec.h"
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')
+ 'ffio_open_dyn_packet_buf')
*/
+static RTPDynamicProtocolHandler realmedia_mp3_dynamic_handler = {
+ .enc_name = "X-MP3-draft-00",
+ .codec_type = AVMEDIA_TYPE_AUDIO,
+ .codec_id = AV_CODEC_ID_MP3ADU,
+};
+
+static RTPDynamicProtocolHandler speex_dynamic_handler = {
+ .enc_name = "speex",
+ .codec_type = AVMEDIA_TYPE_AUDIO,
+ .codec_id = AV_CODEC_ID_SPEEX,
+};
+
+static RTPDynamicProtocolHandler opus_dynamic_handler = {
+ .enc_name = "opus",
+ .codec_type = AVMEDIA_TYPE_AUDIO,
+ .codec_id = AV_CODEC_ID_OPUS,
+};
+
/* statistics functions */
-RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
+static RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
{
ff_register_dynamic_payload_handler(&ff_amr_wb_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_h263_1998_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_h263_2000_dynamic_handler);
+ ff_register_dynamic_payload_handler(&ff_h263_rfc2190_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
+ ff_register_dynamic_payload_handler(&ff_ilbc_dynamic_handler);
+ ff_register_dynamic_payload_handler(&ff_jpeg_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_theora_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_qdm2_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_svq3_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_mp4a_latm_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_vp8_dynamic_handler);
+ ff_register_dynamic_payload_handler(&ff_qcelp_dynamic_handler);
+ ff_register_dynamic_payload_handler(&realmedia_mp3_dynamic_handler);
+ ff_register_dynamic_payload_handler(&speex_dynamic_handler);
+ ff_register_dynamic_payload_handler(&opus_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);
ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);
ff_register_dynamic_payload_handler(&ff_qt_rtp_vid_handler);
ff_register_dynamic_payload_handler(&ff_quicktime_rtp_aud_handler);
ff_register_dynamic_payload_handler(&ff_quicktime_rtp_vid_handler);
+
+ ff_register_dynamic_payload_handler(&ff_g726_16_dynamic_handler);
+ ff_register_dynamic_payload_handler(&ff_g726_24_dynamic_handler);
+ ff_register_dynamic_payload_handler(&ff_g726_32_dynamic_handler);
+ ff_register_dynamic_payload_handler(&ff_g726_40_dynamic_handler);
+}
+
+RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
+ enum AVMediaType codec_type)
+{
+ RTPDynamicProtocolHandler *handler;
+ for (handler = RTPFirstDynamicPayloadHandler;
+ 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 = 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) {
+ while (len >= 4) {
+ payload_len = FFMIN(len, (AV_RB16(buf + 2) + 1) * 4);
+
switch (buf[1]) {
case RTCP_SR:
- if (len < 16) {
+ if (payload_len < 20) {
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);
+ 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;
+ }
- buf += payload_len;
- len -= payload_len;
break;
case RTCP_BYE:
return -RTCP_BYE;
- default:
- return -1;
}
+
+ buf += payload_len;
+ len -= payload_len;
}
return -1;
}
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)
+int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
{
- ByteIOContext *pb;
+ AVIOContext *pb;
uint8_t *buf;
int len;
int rtcp_bytes;
return -1;
s->last_octet_count = s->octet_count;
- if (url_open_dyn_buf(&pb) < 0)
+ if (avio_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 */
+ 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
- put_be32(pb, s->ssrc + 1);
- put_be32(pb, s->ssrc); // server SSRC
+ 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;
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 */
+ 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)
{
- put_be32(pb, 0); /* last SR timestamp */
- put_be32(pb, 0); /* delay since last SR */
+ avio_wb32(pb, 0); /* last SR timestamp */
+ avio_wb32(pb, 0); /* delay since last SR */
} else {
uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
- put_be32(pb, middle_32_bits); /* last SR timestamp */
- put_be32(pb, delay_since_last); /* delay since last SR */
+ avio_wb32(pb, middle_32_bits); /* last SR timestamp */
+ avio_wb32(pb, delay_since_last); /* delay since last SR */
}
// CNAME
- put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
- put_byte(pb, RTCP_SDES);
+ avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
+ avio_w8(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);
+ avio_wb16(pb, (6 + len + 3) / 4); /* length in words - 1 */
+ avio_wb32(pb, s->ssrc + 1);
+ avio_w8(pb, 0x01);
+ avio_w8(pb, len);
+ avio_write(pb, s->hostname, len);
// padding
for (len = (6 + len) % 4; len % 4; len++) {
- put_byte(pb, 0);
+ avio_w8(pb, 0);
}
- put_flush_packet(pb);
- len = url_close_dyn_buf(pb, &buf);
+ avio_flush(pb);
+ len = avio_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);
+ int av_unused result;
+ av_dlog(s->ic, "sending %d bytes of RR\n", len);
+ result= ffurl_write(s->rtp_ctx, buf, len);
+ av_dlog(s->ic, "result from ffurl_write: %d\n", result);
av_free(buf);
}
return 0;
}
-void rtp_send_punch_packets(URLContext* rtp_handle)
+void ff_rtp_send_punch_packets(URLContext* rtp_handle)
{
- ByteIOContext *pb;
+ AVIOContext *pb;
uint8_t *buf;
int len;
/* Send a small RTP packet */
- if (url_open_dyn_buf(&pb) < 0)
+ if (avio_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 */
+ 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 */
- put_flush_packet(pb);
- len = url_close_dyn_buf(pb, &buf);
+ avio_flush(pb);
+ len = avio_close_dyn_buf(pb, &buf);
if ((len > 0) && buf)
- url_write(rtp_handle, buf, len);
+ ffurl_write(rtp_handle, buf, len);
av_free(buf);
/* Send a minimal RTCP RR */
- if (url_open_dyn_buf(&pb) < 0)
+ if (avio_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 */
+ 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 */
- put_flush_packet(pb);
- len = url_close_dyn_buf(pb, &buf);
+ avio_flush(pb);
+ len = avio_close_dyn_buf(pb, &buf);
if ((len > 0) && buf)
- url_write(rtp_handle, buf, len);
+ ffurl_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)
+ * rtp demux (otherwise AV_CODEC_ID_MPEG2TS packets are returned)
*/
-RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size)
+RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size)
{
RTPDemuxContext *s;
av_free(s);
return NULL;
}
- } else {
- av_set_pts_info(st, 32, 1, 90000);
+ } else if (st) {
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:
+ case AV_CODEC_ID_MPEG1VIDEO:
+ case AV_CODEC_ID_MPEG2VIDEO:
+ case AV_CODEC_ID_MP2:
+ case AV_CODEC_ID_MP3:
+ case AV_CODEC_ID_MPEG4:
+ case AV_CODEC_ID_H263:
+ case AV_CODEC_ID_H264:
st->need_parsing = AVSTREAM_PARSE_FULL;
break;
- case CODEC_ID_ADPCM_G722:
- av_set_pts_info(st, 32, 1, st->codec->sample_rate);
+ case AV_CODEC_ID_VORBIS:
+ st->need_parsing = AVSTREAM_PARSE_HEADERS;
+ break;
+ case AV_CODEC_ID_ADPCM_G722:
/* According to RFC 3551, the stream clock rate is 8000
* even if the sample rate is 16000. */
if (st->codec->sample_rate == 8000)
st->codec->sample_rate = 16000;
break;
default:
- if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
- av_set_pts_info(st, 32, 1, st->codec->sample_rate);
- }
break;
}
}
}
void
-rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
- RTPDynamicProtocolHandler *handler)
+ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
+ RTPDynamicProtocolHandler *handler)
{
s->dynamic_protocol_context = ctx;
s->parse_packet = handler->parse_packet;
*/
static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
{
- if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && timestamp != RTP_NOTS_VALUE) {
+ 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;
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;
+ 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,
return -1;
}
+ if (buf[0] & 0x20) {
+ int padding = buf[len - 1];
+ if (len >= 12 + padding)
+ len -= padding;
+ }
+
s->seq = seq;
len -= 12;
buf += 12;
if (!st) {
/* specific MPEG2TS demux support */
ret = ff_mpegts_parse_packet(s->ts, pkt, buf, len);
- if (ret < 0) {
- s->prev_ret = -1;
- return -1;
- }
+ /* 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;
- s->prev_ret = 1;
return 1;
}
- s->prev_ret = 0;
return 0;
} else if (s->parse_packet) {
rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
} 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:
+ case AV_CODEC_ID_MP2:
+ case AV_CODEC_ID_MP3:
/* better than nothing: skip mpeg audio RTP header */
if (len <= 4)
return -1;
av_new_packet(pkt, len);
memcpy(pkt->data, buf, len);
break;
- case CODEC_ID_MPEG1VIDEO:
- case CODEC_ID_MPEG2VIDEO:
+ case AV_CODEC_ID_MPEG1VIDEO:
+ case AV_CODEC_ID_MPEG2VIDEO:
/* better than nothing: skip mpeg video RTP header */
if (len <= 4)
return -1;
// now perform timestamp things....
finalize_packet(s, pkt, timestamp);
- s->prev_ret = rv;
return rv;
}
static int has_next_packet(RTPDemuxContext *s)
{
- return s->queue && s->queue->seq == s->seq + 1;
+ return s->queue && s->queue->seq == (uint16_t) (s->seq + 1);
}
int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
av_free(s->queue);
s->queue = next;
s->queue_len--;
- return rv ? rv : has_next_packet(s);
+ return rv;
}
-/**
- * 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,
+static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
uint8_t **bufptr, int len)
{
uint8_t* buf = bufptr ? *bufptr : NULL;
int rv= 0;
if (!buf) {
- /* If parsing of the previous packet actually returned 0, there's
- * nothing more to be parsed from that packet, but we may have
+ /* 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)
+ if (s->prev_ret <= 0)
return rtp_parse_queued_packet(s, pkt);
/* return the next packets, if any */
if(s->st && s->parse_packet) {
rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
s->st, pkt, ×tamp, NULL, 0, flags);
finalize_packet(s, pkt, timestamp);
- s->prev_ret = rv;
- return rv ? rv : has_next_packet(s);
+ return rv;
} else {
// TODO: Move to a dynamic packet handler (like above)
- if (s->read_buf_index >= s->read_buf_size) {
- s->prev_ret = -1;
- return -1;
- }
+ 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) {
- s->prev_ret = -1;
- return -1;
- }
+ if (ret < 0)
+ return AVERROR(EAGAIN);
s->read_buf_index += ret;
if (s->read_buf_index < s->read_buf_size)
return 1;
- else {
- s->prev_ret = 0;
- return has_next_packet(s);
- }
+ else
+ return 0;
}
}
if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
return -1;
- if (buf[1] >= RTCP_SR && buf[1] <= RTCP_APP) {
+ if (RTP_PT_IS_RTCP(buf[1])) {
return rtcp_parse_packet(s, buf, len);
}
- if (s->seq == 0 || s->queue_size <= 1) {
+ 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 {
} else if (diff <= 1) {
/* Correct packet */
rv = rtp_parse_packet_internal(s, pkt, buf, len);
- return rv ? rv : has_next_packet(s);
+ return rv;
} else {
/* Still missing some packet, enqueue this one. */
enqueue_packet(s, buf, len);
}
}
-void rtp_parse_close(RTPDemuxContext *s)
+/**
+ * Parse an RTP or RTCP packet directly sent as a buffer.
+ * @param s RTP parse context.
+ * @param pkt returned packet
+ * @param bufptr pointer to the input buffer or NULL to read the next packets
+ * @param len buffer len
+ * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
+ * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
+ */
+int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
+ uint8_t **bufptr, int len)
+{
+ int rv = rtp_parse_one_packet(s, pkt, bufptr, len);
+ s->prev_ret = rv;
+ while (rv == AVERROR(EAGAIN) && has_next_packet(s))
+ rv = rtp_parse_queued_packet(s, pkt);
+ return rv ? rv : has_next_packet(s);
+}
+
+void ff_rtp_parse_close(RTPDemuxContext *s)
{
ff_rtp_reset_packet_queue(s);
if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
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