1
0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-28 20:53:54 +02:00
FFmpeg/libavformat/rtspdec.c
Michael Niedermayer e161b079de Merge remote-tracking branch 'qatar/master'
* qatar/master: (22 commits)
  configure: add check for w32threads to enable it automatically
  rtmp: do not hardcode invoke numbers
  cinepack: return non-generic errors
  fate-lavf-ts: use -mpegts_transport_stream_id option.
  Add an APIchanges entry and a minor bump for avio changes.
  avio: Mark the old interrupt callback mechanism as deprecated
  avplay: Set the new interrupt callback
  avconv: Set new interrupt callbacks for all AVFormatContexts, use avio_open2() everywhere
  cinepak: remove redundant coordinate checks
  cinepak: check strip_size
  cinepak, simplify, use AV_RB24()
  cinepak: simplify, use FFMIN()
  cinepak: Fix division by zero, ask for sample if encoded_buf_size is 0
  applehttp: Fix seeking in streams not starting at DTS=0
  http: Don't use the normal http proxy mechanism for https
  tls: Handle connection via a http proxy
  http: Reorder two code blocks
  http: Add a new protocol for opening connections via http proxies
  http: Split out the non-chunked buffer reading part from http_read
  segafilm: add support for raw videos
  ...

Conflicts:
	avconv.c
	configure
	doc/APIchanges
	libavcodec/cinepak.c
	libavformat/applehttp.c
	libavformat/version.h
	tests/lavf-regression.sh
	tests/ref/lavf/ts

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-19 02:00:06 +01:00

413 lines
13 KiB
C

/*
* RTSP demuxer
* 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/avstring.h"
#include "libavutil/intreadwrite.h"
#include "libavutil/mathematics.h"
#include "avformat.h"
#include "internal.h"
#include "network.h"
#include "os_support.h"
#include "rtsp.h"
#include "rdt.h"
#include "url.h"
static int rtsp_read_play(AVFormatContext *s)
{
RTSPState *rt = s->priv_data;
RTSPMessageHeader reply1, *reply = &reply1;
int i;
char cmd[1024];
av_log(s, AV_LOG_DEBUG, "hello state=%d\n", rt->state);
rt->nb_byes = 0;
if (!(rt->server_type == RTSP_SERVER_REAL && rt->need_subscription)) {
if (rt->transport == RTSP_TRANSPORT_RTP) {
for (i = 0; i < rt->nb_rtsp_streams; i++) {
RTSPStream *rtsp_st = rt->rtsp_streams[i];
RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
if (!rtpctx)
continue;
ff_rtp_reset_packet_queue(rtpctx);
rtpctx->last_rtcp_ntp_time = AV_NOPTS_VALUE;
rtpctx->first_rtcp_ntp_time = AV_NOPTS_VALUE;
rtpctx->base_timestamp = 0;
rtpctx->timestamp = 0;
rtpctx->unwrapped_timestamp = 0;
rtpctx->rtcp_ts_offset = 0;
}
}
if (rt->state == RTSP_STATE_PAUSED) {
cmd[0] = 0;
} else {
snprintf(cmd, sizeof(cmd),
"Range: npt=%"PRId64".%03"PRId64"-\r\n",
rt->seek_timestamp / AV_TIME_BASE,
rt->seek_timestamp / (AV_TIME_BASE / 1000) % 1000);
}
ff_rtsp_send_cmd(s, "PLAY", rt->control_uri, cmd, reply, NULL);
if (reply->status_code != RTSP_STATUS_OK) {
return -1;
}
if (rt->transport == RTSP_TRANSPORT_RTP &&
reply->range_start != AV_NOPTS_VALUE) {
for (i = 0; i < rt->nb_rtsp_streams; i++) {
RTSPStream *rtsp_st = rt->rtsp_streams[i];
RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
AVStream *st = NULL;
if (!rtpctx || rtsp_st->stream_index < 0)
continue;
st = s->streams[rtsp_st->stream_index];
rtpctx->range_start_offset =
av_rescale_q(reply->range_start, AV_TIME_BASE_Q,
st->time_base);
}
}
}
rt->state = RTSP_STATE_STREAMING;
return 0;
}
/* pause the stream */
static int rtsp_read_pause(AVFormatContext *s)
{
RTSPState *rt = s->priv_data;
RTSPMessageHeader reply1, *reply = &reply1;
if (rt->state != RTSP_STATE_STREAMING)
return 0;
else if (!(rt->server_type == RTSP_SERVER_REAL && rt->need_subscription)) {
ff_rtsp_send_cmd(s, "PAUSE", rt->control_uri, NULL, reply, NULL);
if (reply->status_code != RTSP_STATUS_OK) {
return -1;
}
}
rt->state = RTSP_STATE_PAUSED;
return 0;
}
int ff_rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply)
{
RTSPState *rt = s->priv_data;
char cmd[1024];
unsigned char *content = NULL;
int ret;
/* describe the stream */
snprintf(cmd, sizeof(cmd),
"Accept: application/sdp\r\n");
if (rt->server_type == RTSP_SERVER_REAL) {
/**
* The Require: attribute is needed for proper streaming from
* Realmedia servers.
*/
av_strlcat(cmd,
"Require: com.real.retain-entity-for-setup\r\n",
sizeof(cmd));
}
ff_rtsp_send_cmd(s, "DESCRIBE", rt->control_uri, cmd, reply, &content);
if (!content)
return AVERROR_INVALIDDATA;
if (reply->status_code != RTSP_STATUS_OK) {
av_freep(&content);
return AVERROR_INVALIDDATA;
}
av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", content);
/* now we got the SDP description, we parse it */
ret = ff_sdp_parse(s, (const char *)content);
av_freep(&content);
if (ret < 0)
return ret;
return 0;
}
static int rtsp_probe(AVProbeData *p)
{
if (av_strstart(p->filename, "rtsp:", NULL))
return AVPROBE_SCORE_MAX;
return 0;
}
static int rtsp_read_header(AVFormatContext *s,
AVFormatParameters *ap)
{
RTSPState *rt = s->priv_data;
int ret;
ret = ff_rtsp_connect(s);
if (ret)
return ret;
rt->real_setup_cache = av_mallocz(2 * s->nb_streams * sizeof(*rt->real_setup_cache));
if (!rt->real_setup_cache)
return AVERROR(ENOMEM);
rt->real_setup = rt->real_setup_cache + s->nb_streams;
if (rt->initial_pause) {
/* do not start immediately */
} else {
if (rtsp_read_play(s) < 0) {
ff_rtsp_close_streams(s);
ff_rtsp_close_connections(s);
return AVERROR_INVALIDDATA;
}
}
return 0;
}
int ff_rtsp_tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
uint8_t *buf, int buf_size)
{
RTSPState *rt = s->priv_data;
int id, len, i, ret;
RTSPStream *rtsp_st;
av_dlog(s, "tcp_read_packet:\n");
redo:
for (;;) {
RTSPMessageHeader reply;
ret = ff_rtsp_read_reply(s, &reply, NULL, 1, NULL);
if (ret < 0)
return ret;
if (ret == 1) /* received '$' */
break;
/* XXX: parse message */
if (rt->state != RTSP_STATE_STREAMING)
return 0;
}
ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
if (ret != 3)
return -1;
id = buf[0];
len = AV_RB16(buf + 1);
av_dlog(s, "id=%d len=%d\n", id, len);
if (len > buf_size || len < 8)
goto redo;
/* get the data */
ret = ffurl_read_complete(rt->rtsp_hd, buf, len);
if (ret != len)
return -1;
if (rt->transport == RTSP_TRANSPORT_RDT &&
ff_rdt_parse_header(buf, len, &id, NULL, NULL, NULL, NULL) < 0)
return -1;
/* find the matching stream */
for (i = 0; i < rt->nb_rtsp_streams; i++) {
rtsp_st = rt->rtsp_streams[i];
if (id >= rtsp_st->interleaved_min &&
id <= rtsp_st->interleaved_max)
goto found;
}
goto redo;
found:
*prtsp_st = rtsp_st;
return len;
}
static int resetup_tcp(AVFormatContext *s)
{
RTSPState *rt = s->priv_data;
char host[1024];
int port;
av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port, NULL, 0,
s->filename);
ff_rtsp_undo_setup(s);
return ff_rtsp_make_setup_request(s, host, port, RTSP_LOWER_TRANSPORT_TCP,
rt->real_challenge);
}
static int rtsp_read_packet(AVFormatContext *s, AVPacket *pkt)
{
RTSPState *rt = s->priv_data;
int ret;
RTSPMessageHeader reply1, *reply = &reply1;
char cmd[1024];
retry:
if (rt->server_type == RTSP_SERVER_REAL) {
int i;
for (i = 0; i < s->nb_streams; i++)
rt->real_setup[i] = s->streams[i]->discard;
if (!rt->need_subscription) {
if (memcmp (rt->real_setup, rt->real_setup_cache,
sizeof(enum AVDiscard) * s->nb_streams)) {
snprintf(cmd, sizeof(cmd),
"Unsubscribe: %s\r\n",
rt->last_subscription);
ff_rtsp_send_cmd(s, "SET_PARAMETER", rt->control_uri,
cmd, reply, NULL);
if (reply->status_code != RTSP_STATUS_OK)
return AVERROR_INVALIDDATA;
rt->need_subscription = 1;
}
}
if (rt->need_subscription) {
int r, rule_nr, first = 1;
memcpy(rt->real_setup_cache, rt->real_setup,
sizeof(enum AVDiscard) * s->nb_streams);
rt->last_subscription[0] = 0;
snprintf(cmd, sizeof(cmd),
"Subscribe: ");
for (i = 0; i < rt->nb_rtsp_streams; i++) {
rule_nr = 0;
for (r = 0; r < s->nb_streams; r++) {
if (s->streams[r]->id == i) {
if (s->streams[r]->discard != AVDISCARD_ALL) {
if (!first)
av_strlcat(rt->last_subscription, ",",
sizeof(rt->last_subscription));
ff_rdt_subscribe_rule(
rt->last_subscription,
sizeof(rt->last_subscription), i, rule_nr);
first = 0;
}
rule_nr++;
}
}
}
av_strlcatf(cmd, sizeof(cmd), "%s\r\n", rt->last_subscription);
ff_rtsp_send_cmd(s, "SET_PARAMETER", rt->control_uri,
cmd, reply, NULL);
if (reply->status_code != RTSP_STATUS_OK)
return AVERROR_INVALIDDATA;
rt->need_subscription = 0;
if (rt->state == RTSP_STATE_STREAMING)
rtsp_read_play (s);
}
}
ret = ff_rtsp_fetch_packet(s, pkt);
if (ret < 0) {
if (ret == AVERROR(ETIMEDOUT) && !rt->packets) {
if (rt->lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_TCP)) {
RTSPMessageHeader reply1, *reply = &reply1;
av_log(s, AV_LOG_WARNING, "UDP timeout, retrying with TCP\n");
if (rtsp_read_pause(s) != 0)
return -1;
// TEARDOWN is required on Real-RTSP, but might make
// other servers close the connection.
if (rt->server_type == RTSP_SERVER_REAL)
ff_rtsp_send_cmd(s, "TEARDOWN", rt->control_uri, NULL,
reply, NULL);
rt->session_id[0] = '\0';
if (resetup_tcp(s) == 0) {
rt->state = RTSP_STATE_IDLE;
rt->need_subscription = 1;
if (rtsp_read_play(s) != 0)
return -1;
goto retry;
}
}
}
return ret;
}
rt->packets++;
/* send dummy request to keep TCP connection alive */
if ((av_gettime() - rt->last_cmd_time) / 1000000 >= rt->timeout / 2) {
if (rt->server_type == RTSP_SERVER_WMS ||
(rt->server_type != RTSP_SERVER_REAL &&
rt->get_parameter_supported)) {
ff_rtsp_send_cmd_async(s, "GET_PARAMETER", rt->control_uri, NULL);
} else {
ff_rtsp_send_cmd_async(s, "OPTIONS", "*", NULL);
}
}
return 0;
}
static int rtsp_read_seek(AVFormatContext *s, int stream_index,
int64_t timestamp, int flags)
{
RTSPState *rt = s->priv_data;
rt->seek_timestamp = av_rescale_q(timestamp,
s->streams[stream_index]->time_base,
AV_TIME_BASE_Q);
switch(rt->state) {
default:
case RTSP_STATE_IDLE:
break;
case RTSP_STATE_STREAMING:
if (rtsp_read_pause(s) != 0)
return -1;
rt->state = RTSP_STATE_SEEKING;
if (rtsp_read_play(s) != 0)
return -1;
break;
case RTSP_STATE_PAUSED:
rt->state = RTSP_STATE_IDLE;
break;
}
return 0;
}
static int rtsp_read_close(AVFormatContext *s)
{
RTSPState *rt = s->priv_data;
ff_rtsp_send_cmd_async(s, "TEARDOWN", rt->control_uri, NULL);
ff_rtsp_close_streams(s);
ff_rtsp_close_connections(s);
ff_network_close();
rt->real_setup = NULL;
av_freep(&rt->real_setup_cache);
return 0;
}
const AVClass rtsp_demuxer_class = {
.class_name = "RTSP demuxer",
.item_name = av_default_item_name,
.option = ff_rtsp_options,
.version = LIBAVUTIL_VERSION_INT,
};
AVInputFormat ff_rtsp_demuxer = {
.name = "rtsp",
.long_name = NULL_IF_CONFIG_SMALL("RTSP input format"),
.priv_data_size = sizeof(RTSPState),
.read_probe = rtsp_probe,
.read_header = rtsp_read_header,
.read_packet = rtsp_read_packet,
.read_close = rtsp_read_close,
.read_seek = rtsp_read_seek,
.flags = AVFMT_NOFILE,
.read_play = rtsp_read_play,
.read_pause = rtsp_read_pause,
.priv_class = &rtsp_demuxer_class,
};