f884ef00de
* qatar/master: (31 commits) tiffenc: initialize forgotten avctx. avplay: free the active audio packet at exit. avplay: free rdft data used for spectrogram analysis. log.h: make AVClass a named struct fix ac3 encoder documentation vc1: more prettyprinting cosmetics vc1: prettyprint some tables vc1: K&R formatting cosmetics AVOptions: bump minor and add APIchanges entry. cmdutils/avtools: simplify show_help() by using av_opt_child_class_next() AVOptions: rename FF_OPT_TYPE_* => AV_OPT_TYPE_* Remove all uses of deprecated AVOptions API. AVOptions: add av_opt_next, deprecate av_next_option. AVOptions: add functions for evaluating option strings. AVOptions: split get_number(). AVOptions: add av_opt_get*, deprecate av_get*. AVOptions: add av_opt_set*(). AVOptions: add new API for enumerating children. rv34: move inverse transform functions to DSP context flvenc: Write the right metadata entry count ... Conflicts: avconv.c cmdutils.c doc/APIchanges ffplay.c ffprobe.c libavcodec/ac3dec.c libavcodec/h264.c libavcodec/libvpxenc.c libavcodec/libx264.c libavcodec/mpeg12enc.c libavcodec/options.c libavdevice/libdc1394.c libavdevice/v4l2.c libavfilter/vf_drawtext.c libavformat/flvdec.c libavformat/mpegtsenc.c libavformat/options.c libavutil/avutil.h libavutil/opt.c libswscale/utils.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
417 lines
13 KiB
C
417 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 "libavutil/opt.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->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;
|
|
}
|
|
|
|
static const AVOption options[] = {
|
|
{ "initial_pause", "Don't start playing the stream immediately", offsetof(RTSPState, initial_pause), AV_OPT_TYPE_INT, {.dbl = 0}, 0, 1, AV_OPT_FLAG_DECODING_PARAM },
|
|
{ NULL },
|
|
};
|
|
|
|
const AVClass rtsp_demuxer_class = {
|
|
.class_name = "RTSP demuxer",
|
|
.item_name = av_default_item_name,
|
|
.option = 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,
|
|
};
|