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FFmpeg/libavformat/rdt.c
Michael Niedermayer 0bd42ae72c Merge remote-tracking branch 'qatar/master' 
* qatar/master:
  avformat: Avoid a warning about mixed declarations and code
  BMV demuxer and decoder
  matroskaenc: Make sure the seekhead struct is freed even on seek failure
  mpeg12enc: Remove write-only variables.
  mpeg12enc: Don't set up run-level info for level 0.
  msmpeg4: Don't set up run-level info for level 0.
  avformat: Warn about using network functions without calling avformat_network_init
  avformat: Revise wording
  rdt: Set AVFMT_NOFILE on ff_rdt_demuxer
  rdt: Check the return value of avformat_open
  rtsp: Discard the dynamic handler, if it has an alloc function which failed
  dsputil: use cpuflags in x86 versions of vector_clip_int32()

Conflicts:
	libavcodec/avcodec.h
	libavcodec/version.h
	libavformat/Makefile
	libavformat/allformats.c
	libavformat/version.h

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-08 02:03:14 +01:00

576 lines
18 KiB
C
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/*
* 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
* @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 "avio_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]->id == s->streams[0]->id);
s->prev_set_id = -1;
s->prev_stream_id = -1;
s->prev_timestamp = -1;
s->parse_packet = handler ? handler->parse_packet : NULL;
s->dynamic_protocol_context = priv_data;
return s;
}
void
ff_rdt_parse_close(RDTDemuxContext *s)
{
av_free(s);
}
struct PayloadContext {
AVFormatContext *rmctx;
int nb_rmst;
RMStream **rmst;
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, 1);
/* 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)
{
AVIOContext 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;
ffio_init_context(&pb, rdt->mlti_data, rdt->mlti_data_size, 0,
NULL, NULL, NULL, NULL);
tag = avio_rl32(&pb);
if (tag == MKTAG('M', 'L', 'T', 'I')) {
int num, chunk_nr;
/* read index of MDPR chunk numbers */
num = avio_rb16(&pb);
if (rule_nr < 0 || rule_nr >= num)
return -1;
avio_skip(&pb, rule_nr * 2);
chunk_nr = avio_rb16(&pb);
avio_skip(&pb, (num - 1 - rule_nr) * 2);
/* read MDPR chunks */
num = avio_rb16(&pb);
if (chunk_nr >= num)
return -1;
while (chunk_nr--)
avio_skip(&pb, avio_rb32(&pb));
size = avio_rb32(&pb);
} else {
size = rdt->mlti_data_size;
avio_seek(&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;
AVIOContext pb;
if (rdt->audio_pkt_cnt == 0) {
int pos;
ffio_init_context(&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 = avio_tell(&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 = avio_alloc_context (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,
uint8_t **bufptr, int len)
{
uint8_t *buf = bufptr ? *bufptr : NULL;
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, &timestamp, 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, &timestamp);
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, &timestamp, 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, first = -1;
for (n = 0; n < s->nb_streams; n++)
if (s->streams[n]->id == stream->id) {
int count = s->streams[n]->index + 1;
if (first == -1) first = n;
if (rdt->nb_rmst < count) {
RMStream **rmst= av_realloc(rdt->rmst, count*sizeof(*rmst));
if (!rmst)
return AVERROR(ENOMEM);
memset(rmst + rdt->nb_rmst, 0,
(count - rdt->nb_rmst) * sizeof(*rmst));
rdt->rmst = rmst;
rdt->nb_rmst = count;
}
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 = avformat_new_stream(s, NULL)))
return NULL;
st->id = orig_st->id;
st->codec->codec_type = orig_st->codec->codec_type;
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 = 0, 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++;
while (1) {
if (!(end = strchr(p, ';')))
break;
if (!odd && end != p) {
if (n_rules > 0)
st = add_dstream(s, orig_st);
else
st = orig_st;
if (!st)
break;
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));
int ret = avformat_open_input(&rdt->rmctx, "", &ff_rdt_demuxer, NULL);
if (ret < 0) {
av_free(rdt);
return NULL;
}
return rdt;
}
static void
rdt_free_context (PayloadContext *rdt)
{
int i;
for (i = 0; i < rdt->nb_rmst; i++)
if (rdt->rmst[i]) {
ff_rm_free_rmstream(rdt->rmst[i]);
av_freep(&rdt->rmst[i]);
}
if (rdt->rmctx)
av_close_input_file(rdt->rmctx);
av_freep(&rdt->mlti_data);
av_freep(&rdt->rmst);
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, \
.alloc = rdt_new_context, \
.free = rdt_free_context, \
.parse_packet = rdt_parse_packet \
}
RDT_HANDLER(live_video, "x-pn-multirate-realvideo-live", AVMEDIA_TYPE_VIDEO);
RDT_HANDLER(live_audio, "x-pn-multirate-realaudio-live", AVMEDIA_TYPE_AUDIO);
RDT_HANDLER(video, "x-pn-realvideo", AVMEDIA_TYPE_VIDEO);
RDT_HANDLER(audio, "x-pn-realaudio", AVMEDIA_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);
}
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