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1467 lines
43 KiB
1467 lines
43 KiB
/* |
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* RTSP/SDP client |
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* Copyright (c) 2002 Fabrice Bellard. |
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* |
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* This file is part of FFmpeg. |
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* |
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* FFmpeg is free software; you can redistribute it and/or |
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* modify it under the terms of the GNU Lesser General Public |
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* License as published by the Free Software Foundation; either |
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* version 2.1 of the License, or (at your option) any later version. |
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* |
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* FFmpeg is distributed in the hope that it will be useful, |
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* but WITHOUT ANY WARRANTY; without even the implied warranty of |
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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* Lesser General Public License for more details. |
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* |
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* You should have received a copy of the GNU Lesser General Public |
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* License along with FFmpeg; if not, write to the Free Software |
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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*/ |
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#include "avformat.h" |
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|
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#include <sys/time.h> |
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#include <unistd.h> /* for select() prototype */ |
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#include "network.h" |
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|
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#include "rtp_internal.h" |
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|
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//#define DEBUG |
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//#define DEBUG_RTP_TCP |
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|
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enum RTSPClientState { |
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RTSP_STATE_IDLE, |
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RTSP_STATE_PLAYING, |
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RTSP_STATE_PAUSED, |
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}; |
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|
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typedef struct RTSPState { |
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URLContext *rtsp_hd; /* RTSP TCP connexion handle */ |
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int nb_rtsp_streams; |
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struct RTSPStream **rtsp_streams; |
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|
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enum RTSPClientState state; |
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int64_t seek_timestamp; |
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|
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/* XXX: currently we use unbuffered input */ |
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// ByteIOContext rtsp_gb; |
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int seq; /* RTSP command sequence number */ |
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char session_id[512]; |
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enum RTSPProtocol protocol; |
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char last_reply[2048]; /* XXX: allocate ? */ |
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RTPDemuxContext *cur_rtp; |
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} RTSPState; |
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|
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typedef struct RTSPStream { |
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URLContext *rtp_handle; /* RTP stream handle */ |
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RTPDemuxContext *rtp_ctx; /* RTP parse context */ |
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|
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int stream_index; /* corresponding stream index, if any. -1 if none (MPEG2TS case) */ |
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int interleaved_min, interleaved_max; /* interleave ids, if TCP transport */ |
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char control_url[1024]; /* url for this stream (from SDP) */ |
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|
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int sdp_port; /* port (from SDP content - not used in RTSP) */ |
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struct in_addr sdp_ip; /* IP address (from SDP content - not used in RTSP) */ |
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int sdp_ttl; /* IP TTL (from SDP content - not used in RTSP) */ |
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int sdp_payload_type; /* payload type - only used in SDP */ |
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rtp_payload_data_t rtp_payload_data; /* rtp payload parsing infos from SDP */ |
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RTPDynamicProtocolHandler *dynamic_handler; ///< Only valid if it's a dynamic protocol. (This is the handler structure) |
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void *dynamic_protocol_context; ///< Only valid if it's a dynamic protocol. (This is any private data associated with the dynamic protocol) |
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} RTSPStream; |
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|
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static int rtsp_read_play(AVFormatContext *s); |
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|
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/* XXX: currently, the only way to change the protocols consists in |
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changing this variable */ |
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int rtsp_default_protocols = (1 << RTSP_PROTOCOL_RTP_UDP); |
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|
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static int rtsp_probe(AVProbeData *p) |
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{ |
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if (strstart(p->filename, "rtsp:", NULL)) |
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return AVPROBE_SCORE_MAX; |
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return 0; |
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} |
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|
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static int redir_isspace(int c) |
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{ |
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return (c == ' ' || c == '\t' || c == '\n' || c == '\r'); |
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} |
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static void skip_spaces(const char **pp) |
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{ |
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const char *p; |
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p = *pp; |
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while (redir_isspace(*p)) |
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p++; |
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*pp = p; |
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} |
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|
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static void get_word_sep(char *buf, int buf_size, const char *sep, |
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const char **pp) |
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{ |
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const char *p; |
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char *q; |
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p = *pp; |
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if (*p == '/') |
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p++; |
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skip_spaces(&p); |
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q = buf; |
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while (!strchr(sep, *p) && *p != '\0') { |
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if ((q - buf) < buf_size - 1) |
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*q++ = *p; |
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p++; |
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} |
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if (buf_size > 0) |
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*q = '\0'; |
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*pp = p; |
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} |
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|
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static void get_word(char *buf, int buf_size, const char **pp) |
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{ |
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const char *p; |
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char *q; |
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p = *pp; |
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skip_spaces(&p); |
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q = buf; |
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while (!redir_isspace(*p) && *p != '\0') { |
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if ((q - buf) < buf_size - 1) |
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*q++ = *p; |
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p++; |
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} |
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if (buf_size > 0) |
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*q = '\0'; |
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*pp = p; |
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} |
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|
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/* parse the rtpmap description: <codec_name>/<clock_rate>[/<other |
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params>] */ |
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static int sdp_parse_rtpmap(AVCodecContext *codec, RTSPStream *rtsp_st, int payload_type, const char *p) |
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{ |
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char buf[256]; |
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int i; |
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AVCodec *c; |
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const char *c_name; |
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|
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/* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and |
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see if we can handle this kind of payload */ |
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get_word_sep(buf, sizeof(buf), "/", &p); |
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if (payload_type >= RTP_PT_PRIVATE) { |
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RTPDynamicProtocolHandler *handler= RTPFirstDynamicPayloadHandler; |
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while(handler) { |
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if (!strcmp(buf, handler->enc_name) && (codec->codec_type == handler->codec_type)) { |
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codec->codec_id = handler->codec_id; |
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rtsp_st->dynamic_handler= handler; |
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if(handler->open) { |
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rtsp_st->dynamic_protocol_context= handler->open(); |
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} |
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break; |
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} |
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handler= handler->next; |
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} |
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} else { |
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/* We are in a standard case ( from http://www.iana.org/assignments/rtp-parameters) */ |
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/* search into AVRtpPayloadTypes[] */ |
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for (i = 0; AVRtpPayloadTypes[i].pt >= 0; ++i) |
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if (!strcmp(buf, AVRtpPayloadTypes[i].enc_name) && (codec->codec_type == AVRtpPayloadTypes[i].codec_type)){ |
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codec->codec_id = AVRtpPayloadTypes[i].codec_id; |
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break; |
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} |
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} |
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c = avcodec_find_decoder(codec->codec_id); |
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if (c && c->name) |
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c_name = c->name; |
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else |
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c_name = (char *)NULL; |
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|
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if (c_name) { |
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get_word_sep(buf, sizeof(buf), "/", &p); |
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i = atoi(buf); |
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switch (codec->codec_type) { |
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case CODEC_TYPE_AUDIO: |
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av_log(codec, AV_LOG_DEBUG, " audio codec set to : %s\n", c_name); |
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codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE; |
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codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS; |
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if (i > 0) { |
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codec->sample_rate = i; |
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get_word_sep(buf, sizeof(buf), "/", &p); |
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i = atoi(buf); |
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if (i > 0) |
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codec->channels = i; |
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// TODO: there is a bug here; if it is a mono stream, and less than 22000Hz, faad upconverts to stereo and twice the |
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// frequency. No problem, but the sample rate is being set here by the sdp line. Upcoming patch forthcoming. (rdm) |
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} |
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av_log(codec, AV_LOG_DEBUG, " audio samplerate set to : %i\n", codec->sample_rate); |
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av_log(codec, AV_LOG_DEBUG, " audio channels set to : %i\n", codec->channels); |
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break; |
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case CODEC_TYPE_VIDEO: |
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av_log(codec, AV_LOG_DEBUG, " video codec set to : %s\n", c_name); |
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break; |
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default: |
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break; |
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} |
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return 0; |
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} |
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|
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return -1; |
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} |
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|
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/* return the length and optionnaly the data */ |
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static int hex_to_data(uint8_t *data, const char *p) |
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{ |
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int c, len, v; |
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len = 0; |
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v = 1; |
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for(;;) { |
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skip_spaces(&p); |
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if (p == '\0') |
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break; |
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c = toupper((unsigned char)*p++); |
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if (c >= '0' && c <= '9') |
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c = c - '0'; |
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else if (c >= 'A' && c <= 'F') |
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c = c - 'A' + 10; |
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else |
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break; |
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v = (v << 4) | c; |
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if (v & 0x100) { |
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if (data) |
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data[len] = v; |
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len++; |
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v = 1; |
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} |
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} |
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return len; |
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} |
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static void sdp_parse_fmtp_config(AVCodecContext *codec, char *attr, char *value) |
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{ |
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switch (codec->codec_id) { |
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case CODEC_ID_MPEG4: |
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case CODEC_ID_AAC: |
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if (!strcmp(attr, "config")) { |
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/* decode the hexa encoded parameter */ |
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int len = hex_to_data(NULL, value); |
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codec->extradata = av_mallocz(len + FF_INPUT_BUFFER_PADDING_SIZE); |
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if (!codec->extradata) |
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return; |
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codec->extradata_size = len; |
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hex_to_data(codec->extradata, value); |
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} |
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break; |
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default: |
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break; |
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} |
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return; |
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} |
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typedef struct attrname_map |
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{ |
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const char *str; |
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uint16_t type; |
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uint32_t offset; |
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} attrname_map_t; |
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|
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/* All known fmtp parmeters and the corresping RTPAttrTypeEnum */ |
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#define ATTR_NAME_TYPE_INT 0 |
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#define ATTR_NAME_TYPE_STR 1 |
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static attrname_map_t attr_names[]= |
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{ |
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{"SizeLength", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, sizelength)}, |
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{"IndexLength", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, indexlength)}, |
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{"IndexDeltaLength", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, indexdeltalength)}, |
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{"profile-level-id", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, profile_level_id)}, |
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{"StreamType", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, streamtype)}, |
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{"mode", ATTR_NAME_TYPE_STR, offsetof(rtp_payload_data_t, mode)}, |
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{NULL, -1, -1}, |
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}; |
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|
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/** parse the attribute line from the fmtp a line of an sdp resonse. This is broken out as a function |
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* because it is used in rtp_h264.c, which is forthcoming. |
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*/ |
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int rtsp_next_attr_and_value(const char **p, char *attr, int attr_size, char *value, int value_size) |
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{ |
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skip_spaces(p); |
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if(**p) |
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{ |
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get_word_sep(attr, attr_size, "=", p); |
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if (**p == '=') |
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(*p)++; |
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get_word_sep(value, value_size, ";", p); |
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if (**p == ';') |
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(*p)++; |
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return 1; |
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} |
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return 0; |
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} |
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|
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/* parse a SDP line and save stream attributes */ |
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static void sdp_parse_fmtp(AVStream *st, const char *p) |
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{ |
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char attr[256]; |
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char value[4096]; |
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int i; |
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RTSPStream *rtsp_st = st->priv_data; |
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AVCodecContext *codec = st->codec; |
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rtp_payload_data_t *rtp_payload_data = &rtsp_st->rtp_payload_data; |
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|
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/* loop on each attribute */ |
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while(rtsp_next_attr_and_value(&p, attr, sizeof(attr), value, sizeof(value))) |
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{ |
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/* grab the codec extra_data from the config parameter of the fmtp line */ |
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sdp_parse_fmtp_config(codec, attr, value); |
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/* Looking for a known attribute */ |
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for (i = 0; attr_names[i].str; ++i) { |
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if (!strcasecmp(attr, attr_names[i].str)) { |
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if (attr_names[i].type == ATTR_NAME_TYPE_INT) |
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*(int *)((char *)rtp_payload_data + attr_names[i].offset) = atoi(value); |
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else if (attr_names[i].type == ATTR_NAME_TYPE_STR) |
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*(char **)((char *)rtp_payload_data + attr_names[i].offset) = av_strdup(value); |
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} |
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} |
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} |
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} |
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|
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/** Parse a string \p in the form of Range:npt=xx-xx, and determine the start |
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* and end time. |
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* Used for seeking in the rtp stream. |
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*/ |
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static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end) |
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{ |
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char buf[256]; |
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|
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skip_spaces(&p); |
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if (!stristart(p, "npt=", &p)) |
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return; |
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|
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*start = AV_NOPTS_VALUE; |
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*end = AV_NOPTS_VALUE; |
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|
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get_word_sep(buf, sizeof(buf), "-", &p); |
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*start = parse_date(buf, 1); |
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if (*p == '-') { |
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p++; |
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get_word_sep(buf, sizeof(buf), "-", &p); |
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*end = parse_date(buf, 1); |
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} |
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// av_log(NULL, AV_LOG_DEBUG, "Range Start: %lld\n", *start); |
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// av_log(NULL, AV_LOG_DEBUG, "Range End: %lld\n", *end); |
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} |
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typedef struct SDPParseState { |
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/* SDP only */ |
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struct in_addr default_ip; |
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int default_ttl; |
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} SDPParseState; |
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|
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static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1, |
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int letter, const char *buf) |
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{ |
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RTSPState *rt = s->priv_data; |
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char buf1[64], st_type[64]; |
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const char *p; |
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int codec_type, payload_type, i; |
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AVStream *st; |
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RTSPStream *rtsp_st; |
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struct in_addr sdp_ip; |
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int ttl; |
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|
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#ifdef DEBUG |
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printf("sdp: %c='%s'\n", letter, buf); |
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#endif |
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|
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p = buf; |
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switch(letter) { |
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case 'c': |
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get_word(buf1, sizeof(buf1), &p); |
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if (strcmp(buf1, "IN") != 0) |
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return; |
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get_word(buf1, sizeof(buf1), &p); |
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if (strcmp(buf1, "IP4") != 0) |
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return; |
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get_word_sep(buf1, sizeof(buf1), "/", &p); |
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if (inet_aton(buf1, &sdp_ip) == 0) |
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return; |
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ttl = 16; |
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if (*p == '/') { |
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p++; |
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get_word_sep(buf1, sizeof(buf1), "/", &p); |
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ttl = atoi(buf1); |
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} |
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if (s->nb_streams == 0) { |
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s1->default_ip = sdp_ip; |
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s1->default_ttl = ttl; |
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} else { |
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st = s->streams[s->nb_streams - 1]; |
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rtsp_st = st->priv_data; |
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rtsp_st->sdp_ip = sdp_ip; |
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rtsp_st->sdp_ttl = ttl; |
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} |
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break; |
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case 's': |
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pstrcpy(s->title, sizeof(s->title), p); |
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break; |
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case 'i': |
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if (s->nb_streams == 0) { |
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pstrcpy(s->comment, sizeof(s->comment), p); |
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break; |
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} |
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break; |
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case 'm': |
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/* new stream */ |
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get_word(st_type, sizeof(st_type), &p); |
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if (!strcmp(st_type, "audio")) { |
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codec_type = CODEC_TYPE_AUDIO; |
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} else if (!strcmp(st_type, "video")) { |
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codec_type = CODEC_TYPE_VIDEO; |
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} else { |
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return; |
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} |
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rtsp_st = av_mallocz(sizeof(RTSPStream)); |
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if (!rtsp_st) |
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return; |
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rtsp_st->stream_index = -1; |
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dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st); |
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|
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rtsp_st->sdp_ip = s1->default_ip; |
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rtsp_st->sdp_ttl = s1->default_ttl; |
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|
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get_word(buf1, sizeof(buf1), &p); /* port */ |
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rtsp_st->sdp_port = atoi(buf1); |
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|
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get_word(buf1, sizeof(buf1), &p); /* protocol (ignored) */ |
|
|
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/* XXX: handle list of formats */ |
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get_word(buf1, sizeof(buf1), &p); /* format list */ |
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rtsp_st->sdp_payload_type = atoi(buf1); |
|
|
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if (!strcmp(AVRtpPayloadTypes[rtsp_st->sdp_payload_type].enc_name, "MP2T")) { |
|
/* no corresponding stream */ |
|
} else { |
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st = av_new_stream(s, 0); |
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if (!st) |
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return; |
|
st->priv_data = rtsp_st; |
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rtsp_st->stream_index = st->index; |
|
st->codec->codec_type = codec_type; |
|
if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) { |
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/* if standard payload type, we can find the codec right now */ |
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rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type); |
|
} |
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} |
|
/* put a default control url */ |
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pstrcpy(rtsp_st->control_url, sizeof(rtsp_st->control_url), s->filename); |
|
break; |
|
case 'a': |
|
if (strstart(p, "control:", &p) && s->nb_streams > 0) { |
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char proto[32]; |
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/* get the control url */ |
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st = s->streams[s->nb_streams - 1]; |
|
rtsp_st = st->priv_data; |
|
|
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/* XXX: may need to add full url resolution */ |
|
url_split(proto, sizeof(proto), NULL, 0, NULL, 0, NULL, NULL, 0, p); |
|
if (proto[0] == '\0') { |
|
/* relative control URL */ |
|
pstrcat(rtsp_st->control_url, sizeof(rtsp_st->control_url), "/"); |
|
pstrcat(rtsp_st->control_url, sizeof(rtsp_st->control_url), p); |
|
} else { |
|
pstrcpy(rtsp_st->control_url, sizeof(rtsp_st->control_url), p); |
|
} |
|
} else if (strstart(p, "rtpmap:", &p)) { |
|
/* NOTE: rtpmap is only supported AFTER the 'm=' tag */ |
|
get_word(buf1, sizeof(buf1), &p); |
|
payload_type = atoi(buf1); |
|
for(i = 0; i < s->nb_streams;i++) { |
|
st = s->streams[i]; |
|
rtsp_st = st->priv_data; |
|
if (rtsp_st->sdp_payload_type == payload_type) { |
|
sdp_parse_rtpmap(st->codec, rtsp_st, payload_type, p); |
|
} |
|
} |
|
} else if (strstart(p, "fmtp:", &p)) { |
|
/* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */ |
|
get_word(buf1, sizeof(buf1), &p); |
|
payload_type = atoi(buf1); |
|
for(i = 0; i < s->nb_streams;i++) { |
|
st = s->streams[i]; |
|
rtsp_st = st->priv_data; |
|
if (rtsp_st->sdp_payload_type == payload_type) { |
|
if(rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->parse_sdp_a_line) { |
|
if(!rtsp_st->dynamic_handler->parse_sdp_a_line(st, rtsp_st->dynamic_protocol_context, buf)) { |
|
sdp_parse_fmtp(st, p); |
|
} |
|
} else { |
|
sdp_parse_fmtp(st, p); |
|
} |
|
} |
|
} |
|
} else if(strstart(p, "framesize:", &p)) { |
|
// let dynamic protocol handlers have a stab at the line. |
|
get_word(buf1, sizeof(buf1), &p); |
|
payload_type = atoi(buf1); |
|
for(i = 0; i < s->nb_streams;i++) { |
|
st = s->streams[i]; |
|
rtsp_st = st->priv_data; |
|
if (rtsp_st->sdp_payload_type == payload_type) { |
|
if(rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->parse_sdp_a_line) { |
|
rtsp_st->dynamic_handler->parse_sdp_a_line(st, rtsp_st->dynamic_protocol_context, buf); |
|
} |
|
} |
|
} |
|
} else if(strstart(p, "range:", &p)) { |
|
int64_t start, end; |
|
|
|
// this is so that seeking on a streamed file can work. |
|
rtsp_parse_range_npt(p, &start, &end); |
|
s->start_time= start; |
|
s->duration= (end==AV_NOPTS_VALUE)?AV_NOPTS_VALUE:end-start; // AV_NOPTS_VALUE means live broadcast (and can't seek) |
|
} |
|
break; |
|
} |
|
} |
|
|
|
static int sdp_parse(AVFormatContext *s, const char *content) |
|
{ |
|
const char *p; |
|
int letter; |
|
char buf[1024], *q; |
|
SDPParseState sdp_parse_state, *s1 = &sdp_parse_state; |
|
|
|
memset(s1, 0, sizeof(SDPParseState)); |
|
p = content; |
|
for(;;) { |
|
skip_spaces(&p); |
|
letter = *p; |
|
if (letter == '\0') |
|
break; |
|
p++; |
|
if (*p != '=') |
|
goto next_line; |
|
p++; |
|
/* get the content */ |
|
q = buf; |
|
while (*p != '\n' && *p != '\r' && *p != '\0') { |
|
if ((q - buf) < sizeof(buf) - 1) |
|
*q++ = *p; |
|
p++; |
|
} |
|
*q = '\0'; |
|
sdp_parse_line(s, s1, letter, buf); |
|
next_line: |
|
while (*p != '\n' && *p != '\0') |
|
p++; |
|
if (*p == '\n') |
|
p++; |
|
} |
|
return 0; |
|
} |
|
|
|
static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp) |
|
{ |
|
const char *p; |
|
int v; |
|
|
|
p = *pp; |
|
skip_spaces(&p); |
|
v = strtol(p, (char **)&p, 10); |
|
if (*p == '-') { |
|
p++; |
|
*min_ptr = v; |
|
v = strtol(p, (char **)&p, 10); |
|
*max_ptr = v; |
|
} else { |
|
*min_ptr = v; |
|
*max_ptr = v; |
|
} |
|
*pp = p; |
|
} |
|
|
|
/* XXX: only one transport specification is parsed */ |
|
static void rtsp_parse_transport(RTSPHeader *reply, const char *p) |
|
{ |
|
char transport_protocol[16]; |
|
char profile[16]; |
|
char lower_transport[16]; |
|
char parameter[16]; |
|
RTSPTransportField *th; |
|
char buf[256]; |
|
|
|
reply->nb_transports = 0; |
|
|
|
for(;;) { |
|
skip_spaces(&p); |
|
if (*p == '\0') |
|
break; |
|
|
|
th = &reply->transports[reply->nb_transports]; |
|
|
|
get_word_sep(transport_protocol, sizeof(transport_protocol), |
|
"/", &p); |
|
if (*p == '/') |
|
p++; |
|
get_word_sep(profile, sizeof(profile), "/;,", &p); |
|
lower_transport[0] = '\0'; |
|
if (*p == '/') { |
|
p++; |
|
get_word_sep(lower_transport, sizeof(lower_transport), |
|
";,", &p); |
|
} |
|
if (!strcasecmp(lower_transport, "TCP")) |
|
th->protocol = RTSP_PROTOCOL_RTP_TCP; |
|
else |
|
th->protocol = RTSP_PROTOCOL_RTP_UDP; |
|
|
|
if (*p == ';') |
|
p++; |
|
/* get each parameter */ |
|
while (*p != '\0' && *p != ',') { |
|
get_word_sep(parameter, sizeof(parameter), "=;,", &p); |
|
if (!strcmp(parameter, "port")) { |
|
if (*p == '=') { |
|
p++; |
|
rtsp_parse_range(&th->port_min, &th->port_max, &p); |
|
} |
|
} else if (!strcmp(parameter, "client_port")) { |
|
if (*p == '=') { |
|
p++; |
|
rtsp_parse_range(&th->client_port_min, |
|
&th->client_port_max, &p); |
|
} |
|
} else if (!strcmp(parameter, "server_port")) { |
|
if (*p == '=') { |
|
p++; |
|
rtsp_parse_range(&th->server_port_min, |
|
&th->server_port_max, &p); |
|
} |
|
} else if (!strcmp(parameter, "interleaved")) { |
|
if (*p == '=') { |
|
p++; |
|
rtsp_parse_range(&th->interleaved_min, |
|
&th->interleaved_max, &p); |
|
} |
|
} else if (!strcmp(parameter, "multicast")) { |
|
if (th->protocol == RTSP_PROTOCOL_RTP_UDP) |
|
th->protocol = RTSP_PROTOCOL_RTP_UDP_MULTICAST; |
|
} else if (!strcmp(parameter, "ttl")) { |
|
if (*p == '=') { |
|
p++; |
|
th->ttl = strtol(p, (char **)&p, 10); |
|
} |
|
} else if (!strcmp(parameter, "destination")) { |
|
struct in_addr ipaddr; |
|
|
|
if (*p == '=') { |
|
p++; |
|
get_word_sep(buf, sizeof(buf), ";,", &p); |
|
if (inet_aton(buf, &ipaddr)) |
|
th->destination = ntohl(ipaddr.s_addr); |
|
} |
|
} |
|
while (*p != ';' && *p != '\0' && *p != ',') |
|
p++; |
|
if (*p == ';') |
|
p++; |
|
} |
|
if (*p == ',') |
|
p++; |
|
|
|
reply->nb_transports++; |
|
} |
|
} |
|
|
|
void rtsp_parse_line(RTSPHeader *reply, const char *buf) |
|
{ |
|
const char *p; |
|
|
|
/* NOTE: we do case independent match for broken servers */ |
|
p = buf; |
|
if (stristart(p, "Session:", &p)) { |
|
get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p); |
|
} else if (stristart(p, "Content-Length:", &p)) { |
|
reply->content_length = strtol(p, NULL, 10); |
|
} else if (stristart(p, "Transport:", &p)) { |
|
rtsp_parse_transport(reply, p); |
|
} else if (stristart(p, "CSeq:", &p)) { |
|
reply->seq = strtol(p, NULL, 10); |
|
} else if (stristart(p, "Range:", &p)) { |
|
rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end); |
|
} |
|
} |
|
|
|
static int url_readbuf(URLContext *h, unsigned char *buf, int size) |
|
{ |
|
int ret, len; |
|
|
|
len = 0; |
|
while (len < size) { |
|
ret = url_read(h, buf+len, size-len); |
|
if (ret < 1) |
|
return ret; |
|
len += ret; |
|
} |
|
return len; |
|
} |
|
|
|
/* skip a RTP/TCP interleaved packet */ |
|
static void rtsp_skip_packet(AVFormatContext *s) |
|
{ |
|
RTSPState *rt = s->priv_data; |
|
int ret, len, len1; |
|
uint8_t buf[1024]; |
|
|
|
ret = url_readbuf(rt->rtsp_hd, buf, 3); |
|
if (ret != 3) |
|
return; |
|
len = (buf[1] << 8) | buf[2]; |
|
#ifdef DEBUG |
|
printf("skipping RTP packet len=%d\n", len); |
|
#endif |
|
/* skip payload */ |
|
while (len > 0) { |
|
len1 = len; |
|
if (len1 > sizeof(buf)) |
|
len1 = sizeof(buf); |
|
ret = url_readbuf(rt->rtsp_hd, buf, len1); |
|
if (ret != len1) |
|
return; |
|
len -= len1; |
|
} |
|
} |
|
|
|
static void rtsp_send_cmd(AVFormatContext *s, |
|
const char *cmd, RTSPHeader *reply, |
|
unsigned char **content_ptr) |
|
{ |
|
RTSPState *rt = s->priv_data; |
|
char buf[4096], buf1[1024], *q; |
|
unsigned char ch; |
|
const char *p; |
|
int content_length, line_count; |
|
unsigned char *content = NULL; |
|
|
|
memset(reply, 0, sizeof(RTSPHeader)); |
|
|
|
rt->seq++; |
|
pstrcpy(buf, sizeof(buf), cmd); |
|
snprintf(buf1, sizeof(buf1), "CSeq: %d\r\n", rt->seq); |
|
pstrcat(buf, sizeof(buf), buf1); |
|
if (rt->session_id[0] != '\0' && !strstr(cmd, "\nIf-Match:")) { |
|
snprintf(buf1, sizeof(buf1), "Session: %s\r\n", rt->session_id); |
|
pstrcat(buf, sizeof(buf), buf1); |
|
} |
|
pstrcat(buf, sizeof(buf), "\r\n"); |
|
#ifdef DEBUG |
|
printf("Sending:\n%s--\n", buf); |
|
#endif |
|
url_write(rt->rtsp_hd, buf, strlen(buf)); |
|
|
|
/* parse reply (XXX: use buffers) */ |
|
line_count = 0; |
|
rt->last_reply[0] = '\0'; |
|
for(;;) { |
|
q = buf; |
|
for(;;) { |
|
if (url_readbuf(rt->rtsp_hd, &ch, 1) != 1) |
|
break; |
|
if (ch == '\n') |
|
break; |
|
if (ch == '$') { |
|
/* XXX: only parse it if first char on line ? */ |
|
rtsp_skip_packet(s); |
|
} else if (ch != '\r') { |
|
if ((q - buf) < sizeof(buf) - 1) |
|
*q++ = ch; |
|
} |
|
} |
|
*q = '\0'; |
|
#ifdef DEBUG |
|
printf("line='%s'\n", buf); |
|
#endif |
|
/* test if last line */ |
|
if (buf[0] == '\0') |
|
break; |
|
p = buf; |
|
if (line_count == 0) { |
|
/* get reply code */ |
|
get_word(buf1, sizeof(buf1), &p); |
|
get_word(buf1, sizeof(buf1), &p); |
|
reply->status_code = atoi(buf1); |
|
} else { |
|
rtsp_parse_line(reply, p); |
|
pstrcat(rt->last_reply, sizeof(rt->last_reply), p); |
|
pstrcat(rt->last_reply, sizeof(rt->last_reply), "\n"); |
|
} |
|
line_count++; |
|
} |
|
|
|
if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0') |
|
pstrcpy(rt->session_id, sizeof(rt->session_id), reply->session_id); |
|
|
|
content_length = reply->content_length; |
|
if (content_length > 0) { |
|
/* leave some room for a trailing '\0' (useful for simple parsing) */ |
|
content = av_malloc(content_length + 1); |
|
(void)url_readbuf(rt->rtsp_hd, content, content_length); |
|
content[content_length] = '\0'; |
|
} |
|
if (content_ptr) |
|
*content_ptr = content; |
|
} |
|
|
|
|
|
/* close and free RTSP streams */ |
|
static void rtsp_close_streams(RTSPState *rt) |
|
{ |
|
int i; |
|
RTSPStream *rtsp_st; |
|
|
|
for(i=0;i<rt->nb_rtsp_streams;i++) { |
|
rtsp_st = rt->rtsp_streams[i]; |
|
if (rtsp_st) { |
|
if (rtsp_st->rtp_ctx) |
|
rtp_parse_close(rtsp_st->rtp_ctx); |
|
if (rtsp_st->rtp_handle) |
|
url_close(rtsp_st->rtp_handle); |
|
if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context) |
|
rtsp_st->dynamic_handler->close(rtsp_st->dynamic_protocol_context); |
|
} |
|
av_free(rtsp_st); |
|
} |
|
av_free(rt->rtsp_streams); |
|
} |
|
|
|
static int rtsp_read_header(AVFormatContext *s, |
|
AVFormatParameters *ap) |
|
{ |
|
RTSPState *rt = s->priv_data; |
|
char host[1024], path[1024], tcpname[1024], cmd[2048]; |
|
URLContext *rtsp_hd; |
|
int port, i, j, ret, err; |
|
RTSPHeader reply1, *reply = &reply1; |
|
unsigned char *content = NULL; |
|
RTSPStream *rtsp_st; |
|
int protocol_mask; |
|
AVStream *st; |
|
|
|
/* extract hostname and port */ |
|
url_split(NULL, 0, NULL, 0, |
|
host, sizeof(host), &port, path, sizeof(path), s->filename); |
|
if (port < 0) |
|
port = RTSP_DEFAULT_PORT; |
|
|
|
/* open the tcp connexion */ |
|
snprintf(tcpname, sizeof(tcpname), "tcp://%s:%d", host, port); |
|
if (url_open(&rtsp_hd, tcpname, URL_RDWR) < 0) |
|
return AVERROR_IO; |
|
rt->rtsp_hd = rtsp_hd; |
|
rt->seq = 0; |
|
|
|
/* describe the stream */ |
|
snprintf(cmd, sizeof(cmd), |
|
"DESCRIBE %s RTSP/1.0\r\n" |
|
"Accept: application/sdp\r\n", |
|
s->filename); |
|
rtsp_send_cmd(s, cmd, reply, &content); |
|
if (!content) { |
|
err = AVERROR_INVALIDDATA; |
|
goto fail; |
|
} |
|
if (reply->status_code != RTSP_STATUS_OK) { |
|
err = AVERROR_INVALIDDATA; |
|
goto fail; |
|
} |
|
|
|
/* now we got the SDP description, we parse it */ |
|
ret = sdp_parse(s, (const char *)content); |
|
av_freep(&content); |
|
if (ret < 0) { |
|
err = AVERROR_INVALIDDATA; |
|
goto fail; |
|
} |
|
|
|
protocol_mask = rtsp_default_protocols; |
|
|
|
/* for each stream, make the setup request */ |
|
/* XXX: we assume the same server is used for the control of each |
|
RTSP stream */ |
|
|
|
for(j = RTSP_RTP_PORT_MIN, i = 0; i < rt->nb_rtsp_streams; ++i) { |
|
char transport[2048]; |
|
|
|
rtsp_st = rt->rtsp_streams[i]; |
|
|
|
/* compute available transports */ |
|
transport[0] = '\0'; |
|
|
|
/* RTP/UDP */ |
|
if (protocol_mask & (1 << RTSP_PROTOCOL_RTP_UDP)) { |
|
char buf[256]; |
|
|
|
/* first try in specified port range */ |
|
if (RTSP_RTP_PORT_MIN != 0) { |
|
while(j <= RTSP_RTP_PORT_MAX) { |
|
snprintf(buf, sizeof(buf), "rtp://?localport=%d", j); |
|
if (url_open(&rtsp_st->rtp_handle, buf, URL_RDWR) == 0) { |
|
j += 2; /* we will use two port by rtp stream (rtp and rtcp) */ |
|
goto rtp_opened; |
|
} |
|
} |
|
} |
|
|
|
/* then try on any port |
|
** if (url_open(&rtsp_st->rtp_handle, "rtp://", URL_RDONLY) < 0) { |
|
** err = AVERROR_INVALIDDATA; |
|
** goto fail; |
|
** } |
|
*/ |
|
|
|
rtp_opened: |
|
port = rtp_get_local_port(rtsp_st->rtp_handle); |
|
if (transport[0] != '\0') |
|
pstrcat(transport, sizeof(transport), ","); |
|
snprintf(transport + strlen(transport), sizeof(transport) - strlen(transport) - 1, |
|
"RTP/AVP/UDP;unicast;client_port=%d-%d", |
|
port, port + 1); |
|
} |
|
|
|
/* RTP/TCP */ |
|
else if (protocol_mask & (1 << RTSP_PROTOCOL_RTP_TCP)) { |
|
if (transport[0] != '\0') |
|
pstrcat(transport, sizeof(transport), ","); |
|
snprintf(transport + strlen(transport), sizeof(transport) - strlen(transport) - 1, |
|
"RTP/AVP/TCP"); |
|
} |
|
|
|
else if (protocol_mask & (1 << RTSP_PROTOCOL_RTP_UDP_MULTICAST)) { |
|
if (transport[0] != '\0') |
|
pstrcat(transport, sizeof(transport), ","); |
|
snprintf(transport + strlen(transport), |
|
sizeof(transport) - strlen(transport) - 1, |
|
"RTP/AVP/UDP;multicast"); |
|
} |
|
snprintf(cmd, sizeof(cmd), |
|
"SETUP %s RTSP/1.0\r\n" |
|
"Transport: %s\r\n", |
|
rtsp_st->control_url, transport); |
|
rtsp_send_cmd(s, cmd, reply, NULL); |
|
if (reply->status_code != RTSP_STATUS_OK || |
|
reply->nb_transports != 1) { |
|
err = AVERROR_INVALIDDATA; |
|
goto fail; |
|
} |
|
|
|
/* XXX: same protocol for all streams is required */ |
|
if (i > 0) { |
|
if (reply->transports[0].protocol != rt->protocol) { |
|
err = AVERROR_INVALIDDATA; |
|
goto fail; |
|
} |
|
} else { |
|
rt->protocol = reply->transports[0].protocol; |
|
} |
|
|
|
/* close RTP connection if not choosen */ |
|
if (reply->transports[0].protocol != RTSP_PROTOCOL_RTP_UDP && |
|
(protocol_mask & (1 << RTSP_PROTOCOL_RTP_UDP))) { |
|
url_close(rtsp_st->rtp_handle); |
|
rtsp_st->rtp_handle = NULL; |
|
} |
|
|
|
switch(reply->transports[0].protocol) { |
|
case RTSP_PROTOCOL_RTP_TCP: |
|
rtsp_st->interleaved_min = reply->transports[0].interleaved_min; |
|
rtsp_st->interleaved_max = reply->transports[0].interleaved_max; |
|
break; |
|
|
|
case RTSP_PROTOCOL_RTP_UDP: |
|
{ |
|
char url[1024]; |
|
|
|
/* XXX: also use address if specified */ |
|
snprintf(url, sizeof(url), "rtp://%s:%d", |
|
host, reply->transports[0].server_port_min); |
|
if (rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) { |
|
err = AVERROR_INVALIDDATA; |
|
goto fail; |
|
} |
|
} |
|
break; |
|
case RTSP_PROTOCOL_RTP_UDP_MULTICAST: |
|
{ |
|
char url[1024]; |
|
int ttl; |
|
|
|
ttl = reply->transports[0].ttl; |
|
if (!ttl) |
|
ttl = 16; |
|
snprintf(url, sizeof(url), "rtp://%s:%d?multicast=1&ttl=%d", |
|
host, |
|
reply->transports[0].server_port_min, |
|
ttl); |
|
if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) { |
|
err = AVERROR_INVALIDDATA; |
|
goto fail; |
|
} |
|
} |
|
break; |
|
} |
|
/* open the RTP context */ |
|
st = NULL; |
|
if (rtsp_st->stream_index >= 0) |
|
st = s->streams[rtsp_st->stream_index]; |
|
if (!st) |
|
s->ctx_flags |= AVFMTCTX_NOHEADER; |
|
rtsp_st->rtp_ctx = rtp_parse_open(s, st, rtsp_st->rtp_handle, rtsp_st->sdp_payload_type, &rtsp_st->rtp_payload_data); |
|
|
|
if (!rtsp_st->rtp_ctx) { |
|
err = AVERROR_NOMEM; |
|
goto fail; |
|
} else { |
|
if(rtsp_st->dynamic_handler) { |
|
rtsp_st->rtp_ctx->dynamic_protocol_context= rtsp_st->dynamic_protocol_context; |
|
rtsp_st->rtp_ctx->parse_packet= rtsp_st->dynamic_handler->parse_packet; |
|
} |
|
} |
|
} |
|
|
|
rt->state = RTSP_STATE_IDLE; |
|
rt->seek_timestamp = 0; /* default is to start stream at position |
|
zero */ |
|
if (ap->initial_pause) { |
|
/* do not start immediately */ |
|
} else { |
|
if (rtsp_read_play(s) < 0) { |
|
err = AVERROR_INVALIDDATA; |
|
goto fail; |
|
} |
|
} |
|
return 0; |
|
fail: |
|
rtsp_close_streams(rt); |
|
av_freep(&content); |
|
url_close(rt->rtsp_hd); |
|
return err; |
|
} |
|
|
|
static int 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; |
|
|
|
#ifdef DEBUG_RTP_TCP |
|
printf("tcp_read_packet:\n"); |
|
#endif |
|
redo: |
|
for(;;) { |
|
ret = url_readbuf(rt->rtsp_hd, buf, 1); |
|
#ifdef DEBUG_RTP_TCP |
|
printf("ret=%d c=%02x [%c]\n", ret, buf[0], buf[0]); |
|
#endif |
|
if (ret != 1) |
|
return -1; |
|
if (buf[0] == '$') |
|
break; |
|
} |
|
ret = url_readbuf(rt->rtsp_hd, buf, 3); |
|
if (ret != 3) |
|
return -1; |
|
id = buf[0]; |
|
len = (buf[1] << 8) | buf[2]; |
|
#ifdef DEBUG_RTP_TCP |
|
printf("id=%d len=%d\n", id, len); |
|
#endif |
|
if (len > buf_size || len < 12) |
|
goto redo; |
|
/* get the data */ |
|
ret = url_readbuf(rt->rtsp_hd, buf, len); |
|
if (ret != len) |
|
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 udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st, |
|
uint8_t *buf, int buf_size) |
|
{ |
|
RTSPState *rt = s->priv_data; |
|
RTSPStream *rtsp_st; |
|
fd_set rfds; |
|
int fd1, fd2, fd_max, n, i, ret; |
|
struct timeval tv; |
|
|
|
for(;;) { |
|
if (url_interrupt_cb()) |
|
return -1; |
|
FD_ZERO(&rfds); |
|
fd_max = -1; |
|
for(i = 0; i < rt->nb_rtsp_streams; i++) { |
|
rtsp_st = rt->rtsp_streams[i]; |
|
/* currently, we cannot probe RTCP handle because of blocking restrictions */ |
|
rtp_get_file_handles(rtsp_st->rtp_handle, &fd1, &fd2); |
|
if (fd1 > fd_max) |
|
fd_max = fd1; |
|
FD_SET(fd1, &rfds); |
|
} |
|
tv.tv_sec = 0; |
|
tv.tv_usec = 100 * 1000; |
|
n = select(fd_max + 1, &rfds, NULL, NULL, &tv); |
|
if (n > 0) { |
|
for(i = 0; i < rt->nb_rtsp_streams; i++) { |
|
rtsp_st = rt->rtsp_streams[i]; |
|
rtp_get_file_handles(rtsp_st->rtp_handle, &fd1, &fd2); |
|
if (FD_ISSET(fd1, &rfds)) { |
|
ret = url_read(rtsp_st->rtp_handle, buf, buf_size); |
|
if (ret > 0) { |
|
*prtsp_st = rtsp_st; |
|
return ret; |
|
} |
|
} |
|
} |
|
} |
|
} |
|
} |
|
|
|
static int rtsp_read_packet(AVFormatContext *s, |
|
AVPacket *pkt) |
|
{ |
|
RTSPState *rt = s->priv_data; |
|
RTSPStream *rtsp_st; |
|
int ret, len; |
|
uint8_t buf[RTP_MAX_PACKET_LENGTH]; |
|
|
|
/* get next frames from the same RTP packet */ |
|
if (rt->cur_rtp) { |
|
ret = rtp_parse_packet(rt->cur_rtp, pkt, NULL, 0); |
|
if (ret == 0) { |
|
rt->cur_rtp = NULL; |
|
return 0; |
|
} else if (ret == 1) { |
|
return 0; |
|
} else { |
|
rt->cur_rtp = NULL; |
|
} |
|
} |
|
|
|
/* read next RTP packet */ |
|
redo: |
|
switch(rt->protocol) { |
|
default: |
|
case RTSP_PROTOCOL_RTP_TCP: |
|
len = tcp_read_packet(s, &rtsp_st, buf, sizeof(buf)); |
|
break; |
|
case RTSP_PROTOCOL_RTP_UDP: |
|
case RTSP_PROTOCOL_RTP_UDP_MULTICAST: |
|
len = udp_read_packet(s, &rtsp_st, buf, sizeof(buf)); |
|
if (rtsp_st->rtp_ctx) |
|
rtp_check_and_send_back_rr(rtsp_st->rtp_ctx, len); |
|
break; |
|
} |
|
if (len < 0) |
|
return AVERROR_IO; |
|
ret = rtp_parse_packet(rtsp_st->rtp_ctx, pkt, buf, len); |
|
if (ret < 0) |
|
goto redo; |
|
if (ret == 1) { |
|
/* more packets may follow, so we save the RTP context */ |
|
rt->cur_rtp = rtsp_st->rtp_ctx; |
|
} |
|
return 0; |
|
} |
|
|
|
static int rtsp_read_play(AVFormatContext *s) |
|
{ |
|
RTSPState *rt = s->priv_data; |
|
RTSPHeader reply1, *reply = &reply1; |
|
char cmd[1024]; |
|
|
|
av_log(s, AV_LOG_DEBUG, "hello state=%d\n", rt->state); |
|
|
|
if (rt->state == RTSP_STATE_PAUSED) { |
|
snprintf(cmd, sizeof(cmd), |
|
"PLAY %s RTSP/1.0\r\n", |
|
s->filename); |
|
} else { |
|
snprintf(cmd, sizeof(cmd), |
|
"PLAY %s RTSP/1.0\r\n" |
|
"Range: npt=%0.3f-\r\n", |
|
s->filename, |
|
(double)rt->seek_timestamp / AV_TIME_BASE); |
|
} |
|
rtsp_send_cmd(s, cmd, reply, NULL); |
|
if (reply->status_code != RTSP_STATUS_OK) { |
|
return -1; |
|
} else { |
|
rt->state = RTSP_STATE_PLAYING; |
|
return 0; |
|
} |
|
} |
|
|
|
/* pause the stream */ |
|
static int rtsp_read_pause(AVFormatContext *s) |
|
{ |
|
RTSPState *rt = s->priv_data; |
|
RTSPHeader reply1, *reply = &reply1; |
|
char cmd[1024]; |
|
|
|
rt = s->priv_data; |
|
|
|
if (rt->state != RTSP_STATE_PLAYING) |
|
return 0; |
|
|
|
snprintf(cmd, sizeof(cmd), |
|
"PAUSE %s RTSP/1.0\r\n", |
|
s->filename); |
|
rtsp_send_cmd(s, cmd, reply, NULL); |
|
if (reply->status_code != RTSP_STATUS_OK) { |
|
return -1; |
|
} else { |
|
rt->state = RTSP_STATE_PAUSED; |
|
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 = timestamp; |
|
switch(rt->state) { |
|
default: |
|
case RTSP_STATE_IDLE: |
|
break; |
|
case RTSP_STATE_PLAYING: |
|
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; |
|
RTSPHeader reply1, *reply = &reply1; |
|
char cmd[1024]; |
|
|
|
#if 0 |
|
/* NOTE: it is valid to flush the buffer here */ |
|
if (rt->protocol == RTSP_PROTOCOL_RTP_TCP) { |
|
url_fclose(&rt->rtsp_gb); |
|
} |
|
#endif |
|
snprintf(cmd, sizeof(cmd), |
|
"TEARDOWN %s RTSP/1.0\r\n", |
|
s->filename); |
|
rtsp_send_cmd(s, cmd, reply, NULL); |
|
|
|
rtsp_close_streams(rt); |
|
url_close(rt->rtsp_hd); |
|
return 0; |
|
} |
|
|
|
#ifdef CONFIG_RTSP_DEMUXER |
|
AVInputFormat rtsp_demuxer = { |
|
"rtsp", |
|
"RTSP input format", |
|
sizeof(RTSPState), |
|
rtsp_probe, |
|
rtsp_read_header, |
|
rtsp_read_packet, |
|
rtsp_read_close, |
|
rtsp_read_seek, |
|
.flags = AVFMT_NOFILE, |
|
.read_play = rtsp_read_play, |
|
.read_pause = rtsp_read_pause, |
|
}; |
|
#endif |
|
|
|
static int sdp_probe(AVProbeData *p1) |
|
{ |
|
const char *p = p1->buf, *p_end = p1->buf + p1->buf_size; |
|
|
|
/* we look for a line beginning "c=IN IP4" */ |
|
while (p < p_end && *p != '\0') { |
|
if (p + sizeof("c=IN IP4") - 1 < p_end && strstart(p, "c=IN IP4", NULL)) |
|
return AVPROBE_SCORE_MAX / 2; |
|
|
|
while(p < p_end - 1 && *p != '\n') p++; |
|
if (++p >= p_end) |
|
break; |
|
if (*p == '\r') |
|
p++; |
|
} |
|
return 0; |
|
} |
|
|
|
#define SDP_MAX_SIZE 8192 |
|
|
|
static int sdp_read_header(AVFormatContext *s, |
|
AVFormatParameters *ap) |
|
{ |
|
RTSPState *rt = s->priv_data; |
|
RTSPStream *rtsp_st; |
|
int size, i, err; |
|
char *content; |
|
char url[1024]; |
|
AVStream *st; |
|
|
|
/* read the whole sdp file */ |
|
/* XXX: better loading */ |
|
content = av_malloc(SDP_MAX_SIZE); |
|
size = get_buffer(&s->pb, content, SDP_MAX_SIZE - 1); |
|
if (size <= 0) { |
|
av_free(content); |
|
return AVERROR_INVALIDDATA; |
|
} |
|
content[size] ='\0'; |
|
|
|
sdp_parse(s, content); |
|
av_free(content); |
|
|
|
/* open each RTP stream */ |
|
for(i=0;i<rt->nb_rtsp_streams;i++) { |
|
rtsp_st = rt->rtsp_streams[i]; |
|
|
|
snprintf(url, sizeof(url), "rtp://%s:%d?multicast=1&ttl=%d", |
|
inet_ntoa(rtsp_st->sdp_ip), |
|
rtsp_st->sdp_port, |
|
rtsp_st->sdp_ttl); |
|
if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) { |
|
err = AVERROR_INVALIDDATA; |
|
goto fail; |
|
} |
|
/* open the RTP context */ |
|
st = NULL; |
|
if (rtsp_st->stream_index >= 0) |
|
st = s->streams[rtsp_st->stream_index]; |
|
if (!st) |
|
s->ctx_flags |= AVFMTCTX_NOHEADER; |
|
rtsp_st->rtp_ctx = rtp_parse_open(s, st, rtsp_st->rtp_handle, rtsp_st->sdp_payload_type, &rtsp_st->rtp_payload_data); |
|
if (!rtsp_st->rtp_ctx) { |
|
err = AVERROR_NOMEM; |
|
goto fail; |
|
} else { |
|
if(rtsp_st->dynamic_handler) { |
|
rtsp_st->rtp_ctx->dynamic_protocol_context= rtsp_st->dynamic_protocol_context; |
|
rtsp_st->rtp_ctx->parse_packet= rtsp_st->dynamic_handler->parse_packet; |
|
} |
|
} |
|
} |
|
return 0; |
|
fail: |
|
rtsp_close_streams(rt); |
|
return err; |
|
} |
|
|
|
static int sdp_read_packet(AVFormatContext *s, |
|
AVPacket *pkt) |
|
{ |
|
return rtsp_read_packet(s, pkt); |
|
} |
|
|
|
static int sdp_read_close(AVFormatContext *s) |
|
{ |
|
RTSPState *rt = s->priv_data; |
|
rtsp_close_streams(rt); |
|
return 0; |
|
} |
|
|
|
#ifdef CONFIG_SDP_DEMUXER |
|
AVInputFormat sdp_demuxer = { |
|
"sdp", |
|
"SDP", |
|
sizeof(RTSPState), |
|
sdp_probe, |
|
sdp_read_header, |
|
sdp_read_packet, |
|
sdp_read_close, |
|
}; |
|
#endif |
|
|
|
#ifdef CONFIG_REDIR_DEMUXER |
|
/* dummy redirector format (used directly in av_open_input_file now) */ |
|
static int redir_probe(AVProbeData *pd) |
|
{ |
|
const char *p; |
|
p = pd->buf; |
|
while (redir_isspace(*p)) |
|
p++; |
|
if (strstart(p, "http://", NULL) || |
|
strstart(p, "rtsp://", NULL)) |
|
return AVPROBE_SCORE_MAX; |
|
return 0; |
|
} |
|
|
|
/* called from utils.c */ |
|
int redir_open(AVFormatContext **ic_ptr, ByteIOContext *f) |
|
{ |
|
char buf[4096], *q; |
|
int c; |
|
AVFormatContext *ic = NULL; |
|
|
|
/* parse each URL and try to open it */ |
|
c = url_fgetc(f); |
|
while (c != URL_EOF) { |
|
/* skip spaces */ |
|
for(;;) { |
|
if (!redir_isspace(c)) |
|
break; |
|
c = url_fgetc(f); |
|
} |
|
if (c == URL_EOF) |
|
break; |
|
/* record url */ |
|
q = buf; |
|
for(;;) { |
|
if (c == URL_EOF || redir_isspace(c)) |
|
break; |
|
if ((q - buf) < sizeof(buf) - 1) |
|
*q++ = c; |
|
c = url_fgetc(f); |
|
} |
|
*q = '\0'; |
|
//printf("URL='%s'\n", buf); |
|
/* try to open the media file */ |
|
if (av_open_input_file(&ic, buf, NULL, 0, NULL) == 0) |
|
break; |
|
} |
|
*ic_ptr = ic; |
|
if (!ic) |
|
return AVERROR_IO; |
|
else |
|
return 0; |
|
} |
|
|
|
AVInputFormat redir_demuxer = { |
|
"redir", |
|
"Redirector format", |
|
0, |
|
redir_probe, |
|
NULL, |
|
NULL, |
|
NULL, |
|
}; |
|
#endif
|
|
|