/*
* RTSP/SDP client
* 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 "avformat.h"
#include <sys/time.h>
#include <unistd.h> /* for select() prototype */
#include "network.h"
#include "rtp_internal.h"
//#define DEBUG
//#define DEBUG_RTP_TCP
enum RTSPClientState {
RTSP_STATE_IDLE,
RTSP_STATE_PLAYING,
RTSP_STATE_PAUSED,
};
typedef struct RTSPState {
URLContext *rtsp_hd; /* RTSP TCP connexion handle */
int nb_rtsp_streams;
struct RTSPStream **rtsp_streams;
enum RTSPClientState state;
int64_t seek_timestamp;
/* XXX: currently we use unbuffered input */
// ByteIOContext rtsp_gb;
int seq; /* RTSP command sequence number */
char session_id[512];
enum RTSPProtocol protocol;
char last_reply[2048]; /* XXX: allocate ? */
RTPDemuxContext *cur_rtp;
} RTSPState;
typedef struct RTSPStream {
URLContext *rtp_handle; /* RTP stream handle */
RTPDemuxContext *rtp_ctx; /* RTP parse context */
int stream_index; /* corresponding stream index, if any. -1 if none (MPEG2TS case) */
int interleaved_min, interleaved_max; /* interleave ids, if TCP transport */
char control_url[1024]; /* url for this stream (from SDP) */
int sdp_port; /* port (from SDP content - not used in RTSP) */
struct in_addr sdp_ip; /* IP address (from SDP content - not used in RTSP) */
int sdp_ttl; /* IP TTL (from SDP content - not used in RTSP) */
int sdp_payload_type; /* payload type - only used in SDP */
rtp_payload_data_t rtp_payload_data; /* rtp payload parsing infos from SDP */
RTPDynamicProtocolHandler *dynamic_handler; ///< Only valid if it's a dynamic protocol. (This is the handler structure)
void *dynamic_protocol_context; ///< Only valid if it's a dynamic protocol. (This is any private data associated with the dynamic protocol)
} RTSPStream;
static int rtsp_read_play(AVFormatContext *s);
/* XXX: currently, the only way to change the protocols consists in
changing this variable */
int rtsp_default_protocols = (1 << RTSP_PROTOCOL_RTP_UDP);
FFRTSPCallback *ff_rtsp_callback = NULL;
static int rtsp_probe(AVProbeData *p)
{
if (strstart(p->filename, "rtsp:", NULL))
return AVPROBE_SCORE_MAX;
return 0;
}
static int redir_isspace(int c)
{
return (c == ' ' || c == '\t' || c == '\n' || c == '\r');
}
static void skip_spaces(const char **pp)
{
const char *p;
p = *pp;
while (redir_isspace(*p))
p++;
*pp = p;
}
static void get_word_sep(char *buf, int buf_size, const char *sep,
const char **pp)
{
const char *p;
char *q;
p = *pp;
if (*p == '/')
p++;
skip_spaces(&p);
q = buf;
while (!strchr(sep, *p) && *p != '\0') {
if ((q - buf) < buf_size - 1)
*q++ = *p;
p++;
}
if (buf_size > 0)
*q = '\0';
*pp = p;
}
static void get_word(char *buf, int buf_size, const char **pp)
{
const char *p;
char *q;
p = *pp;
skip_spaces(&p);
q = buf;
while (!redir_isspace(*p) && *p != '\0') {
if ((q - buf) < buf_size - 1)
*q++ = *p;
p++;
}
if (buf_size > 0)
*q = '\0';
*pp = p;
}
/* parse the rtpmap description: <codec_name>/<clock_rate>[/<other
params>] */
static int sdp_parse_rtpmap(AVCodecContext *codec, RTSPStream *rtsp_st, int payload_type, const char *p)
{
char buf[256];
int i;
AVCodec *c;
const char *c_name;
/* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
see if we can handle this kind of payload */
get_word_sep(buf, sizeof(buf), "/", &p);
if (payload_type >= RTP_PT_PRIVATE) {
RTPDynamicProtocolHandler *handler= RTPFirstDynamicPayloadHandler;
while(handler) {
if (!strcmp(buf, handler->enc_name) && (codec->codec_type == handler->codec_type)) {
codec->codec_id = handler->codec_id;
rtsp_st->dynamic_handler= handler;
if(handler->open) {
rtsp_st->dynamic_protocol_context= handler->open();
}
break;
}
handler= handler->next;
}
} else {
/* We are in a standard case ( from http://www.iana.org/assignments/rtp-parameters) */
/* search into AVRtpPayloadTypes[] */
for (i = 0; AVRtpPayloadTypes[i].pt >= 0; ++i)
if (!strcmp(buf, AVRtpPayloadTypes[i].enc_name) && (codec->codec_type == AVRtpPayloadTypes[i].codec_type)){
codec->codec_id = AVRtpPayloadTypes[i].codec_id;
break;
}
}
c = avcodec_find_decoder(codec->codec_id);
if (c && c->name)
c_name = c->name;
else
c_name = (char *)NULL;
if (c_name) {
get_word_sep(buf, sizeof(buf), "/", &p);
i = atoi(buf);
switch (codec->codec_type) {
case CODEC_TYPE_AUDIO:
av_log(codec, AV_LOG_DEBUG, " audio codec set to : %s\n", c_name);
codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
if (i > 0) {
codec->sample_rate = i;
get_word_sep(buf, sizeof(buf), "/", &p);
i = atoi(buf);
if (i > 0)
codec->channels = i;
// TODO: there is a bug here; if it is a mono stream, and less than 22000Hz, faad upconverts to stereo and twice the
// frequency. No problem, but the sample rate is being set here by the sdp line. Upcoming patch forthcoming. (rdm)
}
av_log(codec, AV_LOG_DEBUG, " audio samplerate set to : %i\n", codec->sample_rate);
av_log(codec, AV_LOG_DEBUG, " audio channels set to : %i\n", codec->channels);
break;
case CODEC_TYPE_VIDEO:
av_log(codec, AV_LOG_DEBUG, " video codec set to : %s\n", c_name);
break;
default:
break;
}
return 0;
}
return -1;
}
/* return the length and optionnaly the data */
static int hex_to_data(uint8_t *data, const char *p)
{
int c, len, v;
len = 0;
v = 1;
for(;;) {
skip_spaces(&p);
if (p == '\0')
break;
c = toupper((unsigned char)*p++);
if (c >= '0' && c <= '9')
c = c - '0';
else if (c >= 'A' && c <= 'F')
c = c - 'A' + 10;
else
break;
v = (v << 4) | c;
if (v & 0x100) {
if (data)
data[len] = v;
len++;
v = 1;
}
}
return len;
}
static void sdp_parse_fmtp_config(AVCodecContext *codec, char *attr, char *value)
{
switch (codec->codec_id) {
case CODEC_ID_MPEG4:
case CODEC_ID_AAC:
if (!strcmp(attr, "config")) {
/* decode the