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ffmpeg/libavformat/rtpdec_asf.c

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/*
* Microsoft RTP/ASF support.
* Copyright (c) 2008 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 Microsoft RTP/ASF support
* @author Ronald S. Bultje <rbultje@ronald.bitfreak.net>
*/
#include "libavutil/base64.h"
#include "libavutil/avstring.h"
#include "libavutil/intreadwrite.h"
#include "rtp.h"
#include "rtpdec_formats.h"
#include "rtsp.h"
#include "asf.h"
#include "avio_internal.h"
/**
* From MSDN 2.2.1.4, we learn that ASF data packets over RTP should not
* contain any padding. Unfortunately, the header min/max_pktsize are not
* updated (thus making min_pktsize invalid). Here, we "fix" these faulty
* min_pktsize values in the ASF file header.
* @return 0 on success, <0 on failure (currently -1).
*/
static int rtp_asf_fix_header(uint8_t *buf, int len)
{
uint8_t *p = buf, *end = buf + len;
if (len < sizeof(ff_asf_guid) * 2 + 22 ||
memcmp(p, ff_asf_header, sizeof(ff_asf_guid))) {
return -1;
}
p += sizeof(ff_asf_guid) + 14;
do {
uint64_t chunksize = AV_RL64(p + sizeof(ff_asf_guid));
if (memcmp(p, ff_asf_file_header, sizeof(ff_asf_guid))) {
if (chunksize > end - p)
return -1;
p += chunksize;
continue;
}
/* skip most of the file header, to min_pktsize */
p += 6 * 8 + 3 * 4 + sizeof(ff_asf_guid) * 2;
if (p + 8 <= end && AV_RL32(p) == AV_RL32(p + 4)) {
/* and set that to zero */
AV_WL32(p, 0);
return 0;
}
break;
} while (end - p >= sizeof(ff_asf_guid) + 8);
return -1;
}
/**
* The following code is basically a buffered AVIOContext,
* with the added benefit of returning -EAGAIN (instead of 0)
* on packet boundaries, such that the ASF demuxer can return
* safely and resume business at the next packet.
*/
static int packetizer_read(void *opaque, uint8_t *buf, int buf_size)
{
return AVERROR(EAGAIN);
}
static void init_packetizer(AVIOContext *pb, uint8_t *buf, int len)
{
ffio_init_context(pb, buf, len, 0, NULL, packetizer_read, NULL, NULL);
/* this "fills" the buffer with its current content */
pb->pos = len;
pb->buf_end = buf + len;
}
int ff_wms_parse_sdp_a_line(AVFormatContext *s, const char *p)
{
int ret = 0;
if (av_strstart(p, "pgmpu:data:application/vnd.ms.wms-hdr.asfv1;base64,", &p)) {
AVIOContext pb;
RTSPState *rt = s->priv_data;
int len = strlen(p) * 6 / 8;
char *buf = av_mallocz(len);
av_base64_decode(buf, p, len);
if (rtp_asf_fix_header(buf, len) < 0)
av_log(s, AV_LOG_ERROR,
"Failed to fix invalid RTSP-MS/ASF min_pktsize\n");
init_packetizer(&pb, buf, len);
if (rt->asf_ctx) {
av_close_input_stream(rt->asf_ctx);
rt->asf_ctx = NULL;
}
ret = av_open_input_stream(&rt->asf_ctx, &pb, "", &ff_asf_demuxer, NULL);
if (ret < 0)
return ret;
av_metadata_copy(&s->metadata, rt->asf_ctx->metadata, 0);
rt->asf_pb_pos = avio_tell(&pb);
av_free(buf);
rt->asf_ctx->pb = NULL;
}
return ret;
}
static int asfrtp_parse_sdp_line(AVFormatContext *s, int stream_index,
PayloadContext *asf, const char *line)
{
if (av_strstart(line, "stream:", &line)) {
RTSPState *rt = s->priv_data;
s->streams[stream_index]->id = strtol(line, NULL, 10);
if (rt->asf_ctx) {
int i;
for (i = 0; i < rt->asf_ctx->nb_streams; i++) {
if (s->streams[stream_index]->id == rt->asf_ctx->streams[i]->id) {
*s->streams[stream_index]->codec =
*rt->asf_ctx->streams[i]->codec;
rt->asf_ctx->streams[i]->codec->extradata_size = 0;
rt->asf_ctx->streams[i]->codec->extradata = NULL;
av_set_pts_info(s->streams[stream_index], 32, 1, 1000);
}
}
}
}
return 0;
}
struct PayloadContext {
AVIOContext *pktbuf, pb;
uint8_t *buf;
};
/**
* @return 0 when a packet was written into /p pkt, and no more data is left;
* 1 when a packet was written into /p pkt, and more packets might be left;
* <0 when not enough data was provided to return a full packet, or on error.
*/
static int asfrtp_parse_packet(AVFormatContext *s, PayloadContext *asf,
AVStream *st, AVPacket *pkt,
uint32_t *timestamp,
const uint8_t *buf, int len, int flags)
{
AVIOContext *pb = &asf->pb;
int res, mflags, len_off;
RTSPState *rt = s->priv_data;
if (!rt->asf_ctx)
return -1;
if (len > 0) {
int off, out_len = 0;
if (len < 4)
return -1;
av_freep(&asf->buf);
ffio_init_context(pb, buf, len, 0, NULL, NULL, NULL, NULL);
while (avio_tell(pb) + 4 < len) {
int start_off = avio_tell(pb);
mflags = avio_r8(pb);
if (mflags & 0x80)
flags |= RTP_FLAG_KEY;
len_off = avio_rb24(pb);
if (mflags & 0x20) /**< relative timestamp */
avio_skip(pb, 4);
if (mflags & 0x10) /**< has duration */
avio_skip(pb, 4);
if (mflags & 0x8) /**< has location ID */
avio_skip(pb, 4);
off = avio_tell(pb);
if (!(mflags & 0x40)) {
/**
* If 0x40 is not set, the len_off field specifies an offset
* of this packet's payload data in the complete (reassembled)
* ASF packet. This is used to spread one ASF packet over
* multiple RTP packets.
*/
if (asf->pktbuf && len_off != avio_tell(asf->pktbuf)) {
uint8_t *p;
url_close_dyn_buf(asf->pktbuf, &p);
asf->pktbuf = NULL;
av_free(p);
}
if (!len_off && !asf->pktbuf &&
(res = url_open_dyn_buf(&asf->pktbuf)) < 0)
return res;
if (!asf->pktbuf)
return AVERROR(EIO);
avio_write(asf->pktbuf, buf + off, len - off);
avio_skip(pb, len - off);
if (!(flags & RTP_FLAG_MARKER))
return -1;
out_len = url_close_dyn_buf(asf->pktbuf, &asf->buf);
asf->pktbuf = NULL;
} else {
/**
* If 0x40 is set, the len_off field specifies the length of
* the next ASF packet that can be read from this payload
* data alone. This is commonly the same as the payload size,
* but could be less in case of packet splitting (i.e.
* multiple ASF packets in one RTP packet).
*/
int cur_len = start_off + len_off - off;
int prev_len = out_len;
out_len += cur_len;
asf->buf = av_realloc(asf->buf, out_len);
memcpy(asf->buf + prev_len, buf + off,
FFMIN(cur_len, len - off));
avio_skip(pb, cur_len);
}
}
init_packetizer(pb, asf->buf, out_len);
pb->pos += rt->asf_pb_pos;
pb->eof_reached = 0;
rt->asf_ctx->pb = pb;
}
for (;;) {
int i;
res = av_read_packet(rt->asf_ctx, pkt);
rt->asf_pb_pos = avio_tell(pb);
if (res != 0)
break;
for (i = 0; i < s->nb_streams; i++) {
if (s->streams[i]->id == rt->asf_ctx->streams[pkt->stream_index]->id) {
pkt->stream_index = i;
return 1; // FIXME: return 0 if last packet
}
}
av_free_packet(pkt);
}
return res == 1 ? -1 : res;
}
static PayloadContext *asfrtp_new_context(void)
{
return av_mallocz(sizeof(PayloadContext));
}
static void asfrtp_free_context(PayloadContext *asf)
{
if (asf->pktbuf) {
uint8_t *p = NULL;
url_close_dyn_buf(asf->pktbuf, &p);
asf->pktbuf = NULL;
av_free(p);
}
av_freep(&asf->buf);
av_free(asf);
}
#define RTP_ASF_HANDLER(n, s, t) \
RTPDynamicProtocolHandler ff_ms_rtp_ ## n ## _handler = { \
.enc_name = s, \
.codec_type = t, \
.codec_id = CODEC_ID_NONE, \
.parse_sdp_a_line = asfrtp_parse_sdp_line, \
.open = asfrtp_new_context, \
.close = asfrtp_free_context, \
.parse_packet = asfrtp_parse_packet, \
}
RTP_ASF_HANDLER(asf_pfv, "x-asf-pf", AVMEDIA_TYPE_VIDEO);
RTP_ASF_HANDLER(asf_pfa, "x-asf-pf", AVMEDIA_TYPE_AUDIO);
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