mirror of https://github.com/FFmpeg/FFmpeg.git
You can not select more than 25 topics
Topics must start with a letter or number, can include dashes ('-') and can be up to 35 characters long.
561 lines
18 KiB
561 lines
18 KiB
/* |
|
* Realmedia RTSP protocol (RDT) support. |
|
* Copyright (c) 2007 Ronald S. Bultje |
|
* |
|
* 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 |
|
*/ |
|
|
|
/** |
|
* @file libavformat/rdt.c |
|
* @brief Realmedia RTSP protocol (RDT) support |
|
* @author Ronald S. Bultje <rbultje@ronald.bitfreak.net> |
|
*/ |
|
|
|
#include "avformat.h" |
|
#include "libavutil/avstring.h" |
|
#include "rtpdec.h" |
|
#include "rdt.h" |
|
#include "libavutil/base64.h" |
|
#include "libavutil/md5.h" |
|
#include "rm.h" |
|
#include "internal.h" |
|
#include "libavcodec/get_bits.h" |
|
|
|
struct RDTDemuxContext { |
|
AVFormatContext *ic; /**< the containing (RTSP) demux context */ |
|
/** Each RDT stream-set (represented by one RTSPStream) can contain |
|
* multiple streams (of the same content, but with possibly different |
|
* codecs/bitrates). Each such stream is represented by one AVStream |
|
* in the AVFormatContext, and this variable points to the offset in |
|
* that array such that the first is the first stream of this set. */ |
|
AVStream **streams; |
|
int n_streams; /**< streams with identifical content in this set */ |
|
void *dynamic_protocol_context; |
|
DynamicPayloadPacketHandlerProc parse_packet; |
|
uint32_t prev_timestamp; |
|
int prev_set_id, prev_stream_id; |
|
}; |
|
|
|
RDTDemuxContext * |
|
ff_rdt_parse_open(AVFormatContext *ic, int first_stream_of_set_idx, |
|
void *priv_data, RTPDynamicProtocolHandler *handler) |
|
{ |
|
RDTDemuxContext *s = av_mallocz(sizeof(RDTDemuxContext)); |
|
if (!s) |
|
return NULL; |
|
|
|
s->ic = ic; |
|
s->streams = &ic->streams[first_stream_of_set_idx]; |
|
do { |
|
s->n_streams++; |
|
} while (first_stream_of_set_idx + s->n_streams < ic->nb_streams && |
|
s->streams[s->n_streams]->priv_data == s->streams[0]->priv_data); |
|
s->prev_set_id = -1; |
|
s->prev_stream_id = -1; |
|
s->prev_timestamp = -1; |
|
s->parse_packet = handler->parse_packet; |
|
s->dynamic_protocol_context = priv_data; |
|
|
|
return s; |
|
} |
|
|
|
void |
|
ff_rdt_parse_close(RDTDemuxContext *s) |
|
{ |
|
int i; |
|
|
|
for (i = 1; i < s->n_streams; i++) |
|
s->streams[i]->priv_data = NULL; |
|
|
|
av_free(s); |
|
} |
|
|
|
struct PayloadContext { |
|
AVFormatContext *rmctx; |
|
RMStream *rmst[MAX_STREAMS]; |
|
uint8_t *mlti_data; |
|
unsigned int mlti_data_size; |
|
char buffer[RTP_MAX_PACKET_LENGTH + FF_INPUT_BUFFER_PADDING_SIZE]; |
|
int audio_pkt_cnt; /**< remaining audio packets in rmdec */ |
|
}; |
|
|
|
void |
|
ff_rdt_calc_response_and_checksum(char response[41], char chksum[9], |
|
const char *challenge) |
|
{ |
|
int ch_len = strlen (challenge), i; |
|
unsigned char zres[16], |
|
buf[64] = { 0xa1, 0xe9, 0x14, 0x9d, 0x0e, 0x6b, 0x3b, 0x59 }; |
|
#define XOR_TABLE_SIZE 37 |
|
const unsigned char xor_table[XOR_TABLE_SIZE] = { |
|
0x05, 0x18, 0x74, 0xd0, 0x0d, 0x09, 0x02, 0x53, |
|
0xc0, 0x01, 0x05, 0x05, 0x67, 0x03, 0x19, 0x70, |
|
0x08, 0x27, 0x66, 0x10, 0x10, 0x72, 0x08, 0x09, |
|
0x63, 0x11, 0x03, 0x71, 0x08, 0x08, 0x70, 0x02, |
|
0x10, 0x57, 0x05, 0x18, 0x54 }; |
|
|
|
/* some (length) checks */ |
|
if (ch_len == 40) /* what a hack... */ |
|
ch_len = 32; |
|
else if (ch_len > 56) |
|
ch_len = 56; |
|
memcpy(buf + 8, challenge, ch_len); |
|
|
|
/* xor challenge bytewise with xor_table */ |
|
for (i = 0; i < XOR_TABLE_SIZE; i++) |
|
buf[8 + i] ^= xor_table[i]; |
|
|
|
av_md5_sum(zres, buf, 64); |
|
ff_data_to_hex(response, zres, 16); |
|
for (i=0;i<32;i++) response[i] = tolower(response[i]); |
|
|
|
/* add tail */ |
|
strcpy (response + 32, "01d0a8e3"); |
|
|
|
/* calculate checksum */ |
|
for (i = 0; i < 8; i++) |
|
chksum[i] = response[i * 4]; |
|
chksum[8] = 0; |
|
} |
|
|
|
static int |
|
rdt_load_mdpr (PayloadContext *rdt, AVStream *st, int rule_nr) |
|
{ |
|
ByteIOContext pb; |
|
int size; |
|
uint32_t tag; |
|
|
|
/** |
|
* Layout of the MLTI chunk: |
|
* 4:MLTI |
|
* 2:<number of streams> |
|
* Then for each stream ([number_of_streams] times): |
|
* 2:<mdpr index> |
|
* 2:<number of mdpr chunks> |
|
* Then for each mdpr chunk ([number_of_mdpr_chunks] times): |
|
* 4:<size> |
|
* [size]:<data> |
|
* we skip MDPR chunks until we reach the one of the stream |
|
* we're interested in, and forward that ([size]+[data]) to |
|
* the RM demuxer to parse the stream-specific header data. |
|
*/ |
|
if (!rdt->mlti_data) |
|
return -1; |
|
init_put_byte(&pb, rdt->mlti_data, rdt->mlti_data_size, 0, |
|
NULL, NULL, NULL, NULL); |
|
tag = get_le32(&pb); |
|
if (tag == MKTAG('M', 'L', 'T', 'I')) { |
|
int num, chunk_nr; |
|
|
|
/* read index of MDPR chunk numbers */ |
|
num = get_be16(&pb); |
|
if (rule_nr < 0 || rule_nr >= num) |
|
return -1; |
|
url_fskip(&pb, rule_nr * 2); |
|
chunk_nr = get_be16(&pb); |
|
url_fskip(&pb, (num - 1 - rule_nr) * 2); |
|
|
|
/* read MDPR chunks */ |
|
num = get_be16(&pb); |
|
if (chunk_nr >= num) |
|
return -1; |
|
while (chunk_nr--) |
|
url_fskip(&pb, get_be32(&pb)); |
|
size = get_be32(&pb); |
|
} else { |
|
size = rdt->mlti_data_size; |
|
url_fseek(&pb, 0, SEEK_SET); |
|
} |
|
if (ff_rm_read_mdpr_codecdata(rdt->rmctx, &pb, st, rdt->rmst[st->index], size) < 0) |
|
return -1; |
|
|
|
return 0; |
|
} |
|
|
|
/** |
|
* Actual data handling. |
|
*/ |
|
|
|
int |
|
ff_rdt_parse_header(const uint8_t *buf, int len, |
|
int *pset_id, int *pseq_no, int *pstream_id, |
|
int *pis_keyframe, uint32_t *ptimestamp) |
|
{ |
|
GetBitContext gb; |
|
int consumed = 0, set_id, seq_no, stream_id, is_keyframe, |
|
len_included, need_reliable; |
|
uint32_t timestamp; |
|
|
|
/* skip status packets */ |
|
while (len >= 5 && buf[1] == 0xFF /* status packet */) { |
|
int pkt_len; |
|
|
|
if (!(buf[0] & 0x80)) |
|
return -1; /* not followed by a data packet */ |
|
|
|
pkt_len = AV_RB16(buf+3); |
|
buf += pkt_len; |
|
len -= pkt_len; |
|
consumed += pkt_len; |
|
} |
|
if (len < 16) |
|
return -1; |
|
/** |
|
* Layout of the header (in bits): |
|
* 1: len_included |
|
* Flag indicating whether this header includes a length field; |
|
* this can be used to concatenate multiple RDT packets in a |
|
* single UDP/TCP data frame and is used to precede RDT data |
|
* by stream status packets |
|
* 1: need_reliable |
|
* Flag indicating whether this header includes a "reliable |
|
* sequence number"; these are apparently sequence numbers of |
|
* data packets alone. For data packets, this flag is always |
|
* set, according to the Real documentation [1] |
|
* 5: set_id |
|
* ID of a set of streams of identical content, possibly with |
|
* different codecs or bitrates |
|
* 1: is_reliable |
|
* Flag set for certain streams deemed less tolerable for packet |
|
* loss |
|
* 16: seq_no |
|
* Packet sequence number; if >=0xFF00, this is a non-data packet |
|
* containing stream status info, the second byte indicates the |
|
* type of status packet (see wireshark docs / source code [2]) |
|
* if (len_included) { |
|
* 16: packet_len |
|
* } else { |
|
* packet_len = remainder of UDP/TCP frame |
|
* } |
|
* 1: is_back_to_back |
|
* Back-to-Back flag; used for timing, set for one in every 10 |
|
* packets, according to the Real documentation [1] |
|
* 1: is_slow_data |
|
* Slow-data flag; currently unused, according to Real docs [1] |
|
* 5: stream_id |
|
* ID of the stream within this particular set of streams |
|
* 1: is_no_keyframe |
|
* Non-keyframe flag (unset if packet belongs to a keyframe) |
|
* 32: timestamp (PTS) |
|
* if (set_id == 0x1F) { |
|
* 16: set_id (extended set-of-streams ID; see set_id) |
|
* } |
|
* if (need_reliable) { |
|
* 16: reliable_seq_no |
|
* Reliable sequence number (see need_reliable) |
|
* } |
|
* if (stream_id == 0x3F) { |
|
* 16: stream_id (extended stream ID; see stream_id) |
|
* } |
|
* [1] https://protocol.helixcommunity.org/files/2005/devdocs/RDT_Feature_Level_20.txt |
|
* [2] http://www.wireshark.org/docs/dfref/r/rdt.html and |
|
* http://anonsvn.wireshark.org/viewvc/trunk/epan/dissectors/packet-rdt.c |
|
*/ |
|
init_get_bits(&gb, buf, len << 3); |
|
len_included = get_bits1(&gb); |
|
need_reliable = get_bits1(&gb); |
|
set_id = get_bits(&gb, 5); |
|
skip_bits(&gb, 1); |
|
seq_no = get_bits(&gb, 16); |
|
if (len_included) |
|
skip_bits(&gb, 16); |
|
skip_bits(&gb, 2); |
|
stream_id = get_bits(&gb, 5); |
|
is_keyframe = !get_bits1(&gb); |
|
timestamp = get_bits_long(&gb, 32); |
|
if (set_id == 0x1f) |
|
set_id = get_bits(&gb, 16); |
|
if (need_reliable) |
|
skip_bits(&gb, 16); |
|
if (stream_id == 0x1f) |
|
stream_id = get_bits(&gb, 16); |
|
|
|
if (pset_id) *pset_id = set_id; |
|
if (pseq_no) *pseq_no = seq_no; |
|
if (pstream_id) *pstream_id = stream_id; |
|
if (pis_keyframe) *pis_keyframe = is_keyframe; |
|
if (ptimestamp) *ptimestamp = timestamp; |
|
|
|
return consumed + (get_bits_count(&gb) >> 3); |
|
} |
|
|
|
/**< return 0 on packet, no more left, 1 on packet, 1 on partial packet... */ |
|
static int |
|
rdt_parse_packet (AVFormatContext *ctx, PayloadContext *rdt, AVStream *st, |
|
AVPacket *pkt, uint32_t *timestamp, |
|
const uint8_t *buf, int len, int flags) |
|
{ |
|
int seq = 1, res; |
|
ByteIOContext pb; |
|
|
|
if (rdt->audio_pkt_cnt == 0) { |
|
int pos; |
|
|
|
init_put_byte(&pb, buf, len, 0, NULL, NULL, NULL, NULL); |
|
flags = (flags & RTP_FLAG_KEY) ? 2 : 0; |
|
res = ff_rm_parse_packet (rdt->rmctx, &pb, st, rdt->rmst[st->index], len, pkt, |
|
&seq, flags, *timestamp); |
|
pos = url_ftell(&pb); |
|
if (res < 0) |
|
return res; |
|
if (res > 0) { |
|
if (st->codec->codec_id == CODEC_ID_AAC) { |
|
memcpy (rdt->buffer, buf + pos, len - pos); |
|
rdt->rmctx->pb = av_alloc_put_byte (rdt->buffer, len - pos, 0, |
|
NULL, NULL, NULL, NULL); |
|
} |
|
goto get_cache; |
|
} |
|
} else { |
|
get_cache: |
|
rdt->audio_pkt_cnt = |
|
ff_rm_retrieve_cache (rdt->rmctx, rdt->rmctx->pb, |
|
st, rdt->rmst[st->index], pkt); |
|
if (rdt->audio_pkt_cnt == 0 && |
|
st->codec->codec_id == CODEC_ID_AAC) |
|
av_freep(&rdt->rmctx->pb); |
|
} |
|
pkt->stream_index = st->index; |
|
pkt->pts = *timestamp; |
|
|
|
return rdt->audio_pkt_cnt > 0; |
|
} |
|
|
|
int |
|
ff_rdt_parse_packet(RDTDemuxContext *s, AVPacket *pkt, |
|
const uint8_t *buf, int len) |
|
{ |
|
int seq_no, flags = 0, stream_id, set_id, is_keyframe; |
|
uint32_t timestamp; |
|
int rv= 0; |
|
|
|
if (!s->parse_packet) |
|
return -1; |
|
|
|
if (!buf && s->prev_stream_id != -1) { |
|
/* return the next packets, if any */ |
|
timestamp= 0; ///< Should not be used if buf is NULL, but should be set to the timestamp of the packet returned.... |
|
rv= s->parse_packet(s->ic, s->dynamic_protocol_context, |
|
s->streams[s->prev_stream_id], |
|
pkt, ×tamp, NULL, 0, flags); |
|
return rv; |
|
} |
|
|
|
if (len < 12) |
|
return -1; |
|
rv = ff_rdt_parse_header(buf, len, &set_id, &seq_no, &stream_id, &is_keyframe, ×tamp); |
|
if (rv < 0) |
|
return rv; |
|
if (is_keyframe && |
|
(set_id != s->prev_set_id || timestamp != s->prev_timestamp || |
|
stream_id != s->prev_stream_id)) { |
|
flags |= RTP_FLAG_KEY; |
|
s->prev_set_id = set_id; |
|
s->prev_timestamp = timestamp; |
|
} |
|
s->prev_stream_id = stream_id; |
|
buf += rv; |
|
len -= rv; |
|
|
|
if (s->prev_stream_id >= s->n_streams) { |
|
s->prev_stream_id = -1; |
|
return -1; |
|
} |
|
|
|
rv = s->parse_packet(s->ic, s->dynamic_protocol_context, |
|
s->streams[s->prev_stream_id], |
|
pkt, ×tamp, buf, len, flags); |
|
|
|
return rv; |
|
} |
|
|
|
void |
|
ff_rdt_subscribe_rule (char *cmd, int size, |
|
int stream_nr, int rule_nr) |
|
{ |
|
av_strlcatf(cmd, size, "stream=%d;rule=%d,stream=%d;rule=%d", |
|
stream_nr, rule_nr * 2, stream_nr, rule_nr * 2 + 1); |
|
} |
|
|
|
static unsigned char * |
|
rdt_parse_b64buf (unsigned int *target_len, const char *p) |
|
{ |
|
unsigned char *target; |
|
int len = strlen(p); |
|
if (*p == '\"') { |
|
p++; |
|
len -= 2; /* skip embracing " at start/end */ |
|
} |
|
*target_len = len * 3 / 4; |
|
target = av_mallocz(*target_len + FF_INPUT_BUFFER_PADDING_SIZE); |
|
av_base64_decode(target, p, *target_len); |
|
return target; |
|
} |
|
|
|
static int |
|
rdt_parse_sdp_line (AVFormatContext *s, int st_index, |
|
PayloadContext *rdt, const char *line) |
|
{ |
|
AVStream *stream = s->streams[st_index]; |
|
const char *p = line; |
|
|
|
if (av_strstart(p, "OpaqueData:buffer;", &p)) { |
|
rdt->mlti_data = rdt_parse_b64buf(&rdt->mlti_data_size, p); |
|
} else if (av_strstart(p, "StartTime:integer;", &p)) |
|
stream->first_dts = atoi(p); |
|
else if (av_strstart(p, "ASMRuleBook:string;", &p)) { |
|
int n = st_index, first = -1; |
|
|
|
for (n = 0; n < s->nb_streams; n++) |
|
if (s->streams[n]->priv_data == stream->priv_data) { |
|
if (first == -1) first = n; |
|
rdt->rmst[s->streams[n]->index] = ff_rm_alloc_rmstream(); |
|
rdt_load_mdpr(rdt, s->streams[n], (n - first) * 2); |
|
|
|
if (s->streams[n]->codec->codec_id == CODEC_ID_AAC) |
|
s->streams[n]->codec->frame_size = 1; // FIXME |
|
} |
|
} |
|
|
|
return 0; |
|
} |
|
|
|
static void |
|
real_parse_asm_rule(AVStream *st, const char *p, const char *end) |
|
{ |
|
do { |
|
/* can be either averagebandwidth= or AverageBandwidth= */ |
|
if (sscanf(p, " %*1[Aa]verage%*1[Bb]andwidth=%d", &st->codec->bit_rate) == 1) |
|
break; |
|
if (!(p = strchr(p, ',')) || p > end) |
|
p = end; |
|
p++; |
|
} while (p < end); |
|
} |
|
|
|
static AVStream * |
|
add_dstream(AVFormatContext *s, AVStream *orig_st) |
|
{ |
|
AVStream *st; |
|
|
|
if (!(st = av_new_stream(s, 0))) |
|
return NULL; |
|
st->codec->codec_type = orig_st->codec->codec_type; |
|
st->priv_data = orig_st->priv_data; |
|
st->first_dts = orig_st->first_dts; |
|
|
|
return st; |
|
} |
|
|
|
static void |
|
real_parse_asm_rulebook(AVFormatContext *s, AVStream *orig_st, |
|
const char *p) |
|
{ |
|
const char *end; |
|
int n_rules, odd = 0; |
|
AVStream *st; |
|
|
|
/** |
|
* The ASMRuleBook contains a list of comma-separated strings per rule, |
|
* and each rule is separated by a ;. The last one also has a ; at the |
|
* end so we can use it as delimiter. |
|
* Every rule occurs twice, once for when the RTSP packet header marker |
|
* is set and once for if it isn't. We only read the first because we |
|
* don't care much (that's what the "odd" variable is for). |
|
* Each rule contains a set of one or more statements, optionally |
|
* preceeded by a single condition. If there's a condition, the rule |
|
* starts with a '#'. Multiple conditions are merged between brackets, |
|
* so there are never multiple conditions spread out over separate |
|
* statements. Generally, these conditions are bitrate limits (min/max) |
|
* for multi-bitrate streams. |
|
*/ |
|
if (*p == '\"') p++; |
|
for (n_rules = 0; s->nb_streams < MAX_STREAMS;) { |
|
if (!(end = strchr(p, ';'))) |
|
break; |
|
if (!odd && end != p) { |
|
if (n_rules > 0) |
|
st = add_dstream(s, orig_st); |
|
else |
|
st = orig_st; |
|
real_parse_asm_rule(st, p, end); |
|
n_rules++; |
|
} |
|
p = end + 1; |
|
odd ^= 1; |
|
} |
|
} |
|
|
|
void |
|
ff_real_parse_sdp_a_line (AVFormatContext *s, int stream_index, |
|
const char *line) |
|
{ |
|
const char *p = line; |
|
|
|
if (av_strstart(p, "ASMRuleBook:string;", &p)) |
|
real_parse_asm_rulebook(s, s->streams[stream_index], p); |
|
} |
|
|
|
static PayloadContext * |
|
rdt_new_context (void) |
|
{ |
|
PayloadContext *rdt = av_mallocz(sizeof(PayloadContext)); |
|
|
|
av_open_input_stream(&rdt->rmctx, NULL, "", &rdt_demuxer, NULL); |
|
|
|
return rdt; |
|
} |
|
|
|
static void |
|
rdt_free_context (PayloadContext *rdt) |
|
{ |
|
int i; |
|
|
|
for (i = 0; i < MAX_STREAMS; i++) |
|
if (rdt->rmst[i]) { |
|
ff_rm_free_rmstream(rdt->rmst[i]); |
|
av_freep(&rdt->rmst[i]); |
|
} |
|
if (rdt->rmctx) |
|
av_close_input_stream(rdt->rmctx); |
|
av_freep(&rdt->mlti_data); |
|
av_free(rdt); |
|
} |
|
|
|
#define RDT_HANDLER(n, s, t) \ |
|
static RTPDynamicProtocolHandler ff_rdt_ ## n ## _handler = { \ |
|
.enc_name = s, \ |
|
.codec_type = t, \ |
|
.codec_id = CODEC_ID_NONE, \ |
|
.parse_sdp_a_line = rdt_parse_sdp_line, \ |
|
.open = rdt_new_context, \ |
|
.close = rdt_free_context, \ |
|
.parse_packet = rdt_parse_packet \ |
|
}; |
|
|
|
RDT_HANDLER(live_video, "x-pn-multirate-realvideo-live", CODEC_TYPE_VIDEO); |
|
RDT_HANDLER(live_audio, "x-pn-multirate-realaudio-live", CODEC_TYPE_AUDIO); |
|
RDT_HANDLER(video, "x-pn-realvideo", CODEC_TYPE_VIDEO); |
|
RDT_HANDLER(audio, "x-pn-realaudio", CODEC_TYPE_AUDIO); |
|
|
|
void av_register_rdt_dynamic_payload_handlers(void) |
|
{ |
|
ff_register_dynamic_payload_handler(&ff_rdt_video_handler); |
|
ff_register_dynamic_payload_handler(&ff_rdt_audio_handler); |
|
ff_register_dynamic_payload_handler(&ff_rdt_live_video_handler); |
|
ff_register_dynamic_payload_handler(&ff_rdt_live_audio_handler); |
|
}
|
|
|