mirror of
https://git.ffmpeg.org/ffmpeg.git
synced 2024-12-27 01:42:20 +00:00
7f5af80ba4
* commit 'ce70f28a1732c74a9cd7fec2d56178750bd6e457': avpacket: Replace av_free_packet with av_packet_unref Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
317 lines
10 KiB
C
317 lines
10 KiB
C
/*
|
|
* 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/avassert.h"
|
|
#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"
|
|
#include "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));
|
|
int skip = 6 * 8 + 3 * 4 + sizeof(ff_asf_guid) * 2;
|
|
if (memcmp(p, ff_asf_file_header, sizeof(ff_asf_guid))) {
|
|
if (chunksize > end - p)
|
|
return -1;
|
|
p += chunksize;
|
|
continue;
|
|
}
|
|
|
|
if (end - p < 8 + skip)
|
|
break;
|
|
/* skip most of the file header, to min_pktsize */
|
|
p += skip;
|
|
if (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;
|
|
AVDictionary *opts = NULL;
|
|
int len = strlen(p) * 6 / 8;
|
|
char *buf = av_mallocz(len);
|
|
AVInputFormat *iformat;
|
|
|
|
if (!buf)
|
|
return AVERROR(ENOMEM);
|
|
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) {
|
|
avformat_close_input(&rt->asf_ctx);
|
|
}
|
|
|
|
if (!(iformat = av_find_input_format("asf")))
|
|
return AVERROR_DEMUXER_NOT_FOUND;
|
|
|
|
rt->asf_ctx = avformat_alloc_context();
|
|
if (!rt->asf_ctx) {
|
|
av_free(buf);
|
|
return AVERROR(ENOMEM);
|
|
}
|
|
rt->asf_ctx->pb = &pb;
|
|
av_dict_set(&opts, "no_resync_search", "1", 0);
|
|
|
|
if ((ret = ff_copy_whitelists(rt->asf_ctx, s)) < 0) {
|
|
av_dict_free(&opts);
|
|
return ret;
|
|
}
|
|
|
|
ret = avformat_open_input(&rt->asf_ctx, "", iformat, &opts);
|
|
av_dict_free(&opts);
|
|
if (ret < 0) {
|
|
av_free(buf);
|
|
return ret;
|
|
}
|
|
av_dict_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 (stream_index < 0)
|
|
return 0;
|
|
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;
|
|
s->streams[stream_index]->need_parsing =
|
|
rt->asf_ctx->streams[i]->need_parsing;
|
|
rt->asf_ctx->streams[i]->codec->extradata_size = 0;
|
|
rt->asf_ctx->streams[i]->codec->extradata = NULL;
|
|
avpriv_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, uint16_t seq,
|
|
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, (uint8_t *)buf, len, 0, NULL, NULL, NULL, NULL);
|
|
|
|
while (avio_tell(pb) + 4 < len) {
|
|
int start_off = avio_tell(pb);
|
|
|
|
mflags = avio_r8(pb);
|
|
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)) {
|
|
ffio_free_dyn_buf(&asf->pktbuf);
|
|
}
|
|
if (!len_off && !asf->pktbuf &&
|
|
(res = avio_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 = avio_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;
|
|
if (FFMIN(cur_len, len - off) < 0)
|
|
return -1;
|
|
if ((res = av_reallocp(&asf->buf, out_len)) < 0)
|
|
return res;
|
|
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 = ff_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_packet_unref(pkt);
|
|
}
|
|
|
|
return res == 1 ? -1 : res;
|
|
}
|
|
|
|
static void asfrtp_close_context(PayloadContext *asf)
|
|
{
|
|
ffio_free_dyn_buf(&asf->pktbuf);
|
|
av_freep(&asf->buf);
|
|
}
|
|
|
|
#define RTP_ASF_HANDLER(n, s, t) \
|
|
RTPDynamicProtocolHandler ff_ms_rtp_ ## n ## _handler = { \
|
|
.enc_name = s, \
|
|
.codec_type = t, \
|
|
.codec_id = AV_CODEC_ID_NONE, \
|
|
.priv_data_size = sizeof(PayloadContext), \
|
|
.parse_sdp_a_line = asfrtp_parse_sdp_line, \
|
|
.close = asfrtp_close_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);
|