/*
* RTP input format
- * Copyright (c) 2002 Fabrice Bellard.
+ * 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
*/
+
+#include "libavutil/mathematics.h"
+#include "libavutil/avstring.h"
+#include "libavcodec/get_bits.h"
#include "avformat.h"
#include "mpegts.h"
-#include "bitstream.h"
+#include "url.h"
#include <unistd.h>
#include "network.h"
-#include "rtp_internal.h"
-#include "rtp_h264.h"
+#include "rtpdec.h"
+#include "rtpdec_formats.h"
//#define DEBUG
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')
*/
-/* statistics functions */
-RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
+static RTPDynamicProtocolHandler ff_realmedia_mp3_dynamic_handler = {
+ .enc_name = "X-MP3-draft-00",
+ .codec_type = AVMEDIA_TYPE_AUDIO,
+ .codec_id = CODEC_ID_MP3ADU,
+};
-static RTPDynamicProtocolHandler mp4v_es_handler= {"MP4V-ES", CODEC_TYPE_VIDEO, CODEC_ID_MPEG4};
-static RTPDynamicProtocolHandler mpeg4_generic_handler= {"mpeg4-generic", CODEC_TYPE_AUDIO, CODEC_ID_AAC};
+/* statistics functions */
+static RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
-static void register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
+void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
{
handler->next= RTPFirstDynamicPayloadHandler;
RTPFirstDynamicPayloadHandler= handler;
void av_register_rtp_dynamic_payload_handlers(void)
{
- register_dynamic_payload_handler(&mp4v_es_handler);
- register_dynamic_payload_handler(&mpeg4_generic_handler);
- register_dynamic_payload_handler(&ff_h264_dynamic_handler);
+ 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 (!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)
{
- if (buf[1] != 200)
- return -1;
- 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);
- return 0;
+ 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_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)
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, 201);
- put_be16(pb, 7); /* length in words - 1 */
- put_be32(pb, s->ssrc); // our own SSRC
- put_be32(pb, s->ssrc); // XXX: should be the server's here!
+ 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;
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, 202);
+ 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);
+ 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;
-#if defined(DEBUG)
- printf("sending %d bytes of RR\n", len);
-#endif
- result= url_write(s->rtp_ctx, buf, len);
-#if defined(DEBUG)
- printf("result from url_write: %d\n", result);
-#endif
+ 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 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);
+}
+
+
/**
* 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)
- * TODO: change this to not take rtp_payload data, and use the new dynamic payload system.
*/
-RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, rtp_payload_data_t *rtp_payload_data)
+RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size)
{
RTPDemuxContext *s;
s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
s->ic = s1;
s->st = st;
- s->rtp_payload_data = rtp_payload_data;
+ 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 = mpegts_parse_open(s->ic);
+ s->ts = ff_mpegts_parse_open(s->ic);
if (s->ts == NULL) {
av_free(s);
return NULL;
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;
}
return s;
}
-static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf)
+void
+ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
+ RTPDynamicProtocolHandler *handler)
{
- int au_headers_length, au_header_size, i;
- GetBitContext getbitcontext;
- rtp_payload_data_t *infos;
-
- infos = s->rtp_payload_data;
-
- if (infos == NULL)
- return -1;
-
- /* decode the first 2 bytes where the AUHeader sections are stored
- length in bits */
- au_headers_length = AV_RB16(buf);
-
- if (au_headers_length > RTP_MAX_PACKET_LENGTH)
- return -1;
-
- infos->au_headers_length_bytes = (au_headers_length + 7) / 8;
-
- /* skip AU headers length section (2 bytes) */
- buf += 2;
-
- init_get_bits(&getbitcontext, buf, infos->au_headers_length_bytes * 8);
-
- /* XXX: Wrong if optionnal additional sections are present (cts, dts etc...) */
- au_header_size = infos->sizelength + infos->indexlength;
- if (au_header_size <= 0 || (au_headers_length % au_header_size != 0))
- return -1;
-
- infos->nb_au_headers = au_headers_length / au_header_size;
- infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers);
-
- /* XXX: We handle multiple AU Section as only one (need to fix this for interleaving)
- In my test, the FAAD decoder does not behave correctly when sending each AU one by one
- but does when sending the whole as one big packet... */
- infos->au_headers[0].size = 0;
- infos->au_headers[0].index = 0;
- for (i = 0; i < infos->nb_au_headers; ++i) {
- infos->au_headers[0].size += get_bits_long(&getbitcontext, infos->sizelength);
- infos->au_headers[0].index = get_bits_long(&getbitcontext, infos->indexlength);
- }
-
- infos->nb_au_headers = 1;
-
- return 0;
+ s->dynamic_protocol_context = ctx;
+ s->parse_packet = handler->parse_packet;
}
/**
*/
static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
{
- switch(s->st->codec->codec_id) {
- case CODEC_ID_MP2:
- case CODEC_ID_MPEG1VIDEO:
- case CODEC_ID_MPEG2VIDEO:
- if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
- int64_t addend;
-
- int delta_timestamp;
- /* XXX: is it really necessary to unify the timestamp base ? */
- /* compute pts from timestamp with received ntp_time */
- delta_timestamp = timestamp - s->last_rtcp_timestamp;
- /* convert to 90 kHz without overflow */
- addend = (s->last_rtcp_ntp_time - s->first_rtcp_ntp_time) >> 14;
- addend = (addend * 5625) >> 14;
- pkt->pts = addend + delta_timestamp;
- }
- break;
- case CODEC_ID_AAC:
- case CODEC_ID_H264:
- case CODEC_ID_MPEG4:
- pkt->pts = timestamp;
- break;
- default:
- /* no timestamp info yet */
- break;
+ 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;
}
- pkt->stream_index = s->st->index;
+
+ 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;
}
-/**
- * Parse an RTP or RTCP packet directly sent as a buffer.
- * @param s RTP parse context.
- * @param pkt returned packet
- * @param buf 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,
- const uint8_t *buf, int len)
+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;
- if (!buf) {
- /* return the next packets, if any */
- if(s->st && s->parse_packet) {
- timestamp= 0; ///< Should not be used if buf is NULL, but should be set to the timestamp of the packet returned....
- rv= s->parse_packet(s, pkt, ×tamp, 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 -1;
- ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
- s->read_buf_size - s->read_buf_index);
- if (ret < 0)
- return -1;
- 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] >= 200 && buf[1] <= 204) {
- rtcp_parse_packet(s, buf, len);
- return -1;
- }
+ 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);
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 = mpegts_parse_packet(s->ts, pkt, buf, len);
+ 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 -1;
+ 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, pkt, ×tamp, buf, len, flags);
+ rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
+ s->st, pkt, ×tamp, 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;
av_new_packet(pkt, len);
memcpy(pkt->data, buf, len);
break;
- // moved from below, verbatim. this is because this section handles packets, and the lower switch handles
- // timestamps.
- // TODO: Put this into a dynamic packet handler...
- case CODEC_ID_AAC:
- if (rtp_parse_mp4_au(s, buf))
- return -1;
- {
- rtp_payload_data_t *infos = s->rtp_payload_data;
- if (infos == NULL)
- return -1;
- buf += infos->au_headers_length_bytes + 2;
- len -= infos->au_headers_length_bytes + 2;
-
- /* XXX: Fixme we only handle the case where rtp_parse_mp4_au define
- one au_header */
- av_new_packet(pkt, infos->au_headers[0].size);
- memcpy(pkt->data, buf, infos->au_headers[0].size);
- buf += infos->au_headers[0].size;
- len -= infos->au_headers[0].size;
- }
- s->read_buf_size = len;
- rv= 0;
- break;
default:
av_new_packet(pkt, len);
memcpy(pkt->data, buf, len);
break;
}
- // now perform timestamp things....
- finalize_packet(s, pkt, timestamp);
+ 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;
}
-void rtp_parse_close(RTPDemuxContext *s)
+static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
+ uint8_t **bufptr, int len)
{
- // TODO: fold this into the protocol specific data fields.
+ 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, ×tamp, 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 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")) {
- mpegts_parse_close(s->ts);
+ 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;
+}
projects / ffmpeg / blobdiff