/*
* RTP input format
- * Copyright (c) 2002 Fabrice Bellard.
+ * Copyright (c) 2002 Fabrice Bellard
*
* This file is part of FFmpeg.
*
*/
/* needed for gethostname() */
-#define _XOPEN_SOURCE 500
+#define _XOPEN_SOURCE 600
-#include "libavcodec/bitstream.h"
+#include "libavcodec/get_bits.h"
#include "avformat.h"
#include "mpegts.h"
#include <unistd.h>
#include "network.h"
-#include "rtp_internal.h"
-#include "rtp_h264.h"
+#include "rtpdec.h"
+#include "rtpdec_formats.h"
//#define DEBUG
/* statistics functions */
RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
-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};
-
void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
{
handler->next= RTPFirstDynamicPayloadHandler;
void av_register_rtp_dynamic_payload_handlers(void)
{
- ff_register_dynamic_payload_handler(&mp4v_es_handler);
- ff_register_dynamic_payload_handler(&mpeg4_generic_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_ms_rtp_asf_pfv_handler);
+ ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);
}
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 >= 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)
// Receiver Report
put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
- put_byte(pb, 201);
+ put_byte(pb, RTCP_RR);
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!
+ // 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;
// CNAME
put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
- put_byte(pb, 202);
+ 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);
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)
- * 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 *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type)
{
RTPDemuxContext *s;
s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
s->ic = s1;
s->st = st;
- s->rtp_payload_data = rtp_payload_data;
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:
+ av_set_pts_info(st, 32, 1, st->codec->sample_rate);
+ /* 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 == CODEC_TYPE_AUDIO) {
+ if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
av_set_pts_info(st, 32, 1, st->codec->sample_rate);
}
break;
return s;
}
-static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf)
+void
+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)
{
- if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
+ if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && timestamp != RTP_NOTS_VALUE) {
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 = addend + delta_timestamp;
+ pkt->pts = s->range_start_offset + addend + delta_timestamp;
}
- pkt->stream_index = s->st->index;
}
/**
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);
+ /* 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 -1;
- ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
+ 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 -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;
+ if (buf[1] >= RTCP_SR && buf[1] <= RTCP_APP) {
+ return rtcp_parse_packet(s, buf, len);
}
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);
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);
if (ret < 0)
return -1;
if (ret < len) {
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 rtp_parse_close(RTPDemuxContext *s)
{
- // TODO: fold this into the protocol specific data fields.
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