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ffmpeg / libavformat / rtpdec_asf.c @ a16ebd40

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1 1a30d541 Ronald S. Bultje
/*
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 * Microsoft RTP/ASF support.
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 * Copyright (c) 2008 Ronald S. Bultje
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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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/**
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 * @file
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 * @brief Microsoft RTP/ASF support
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 * @author Ronald S. Bultje <rbultje@ronald.bitfreak.net>
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 */
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#include "libavutil/base64.h"
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#include "libavutil/avstring.h"
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#include "libavutil/intreadwrite.h"
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#include "rtp.h"
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#include "rtpdec_asf.h"
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#include "rtsp.h"
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#include "asf.h"
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/**
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 * From MSDN 2.2.1.4, we learn that ASF data packets over RTP should not
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 * contain any padding. Unfortunately, the header min/max_pktsize are not
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 * updated (thus making min_pktsize invalid). Here, we "fix" these faulty
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 * min_pktsize values in the ASF file header.
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 * @return 0 on success, <0 on failure (currently -1).
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 */
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static int rtp_asf_fix_header(uint8_t *buf, int len)
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{
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    uint8_t *p = buf, *end = buf + len;
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    if (len < sizeof(ff_asf_guid) * 2 + 22 ||
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        memcmp(p, ff_asf_header, sizeof(ff_asf_guid))) {
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        return -1;
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    }
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    p += sizeof(ff_asf_guid) + 14;
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    do {
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        uint64_t chunksize = AV_RL64(p + sizeof(ff_asf_guid));
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        if (memcmp(p, ff_asf_file_header, sizeof(ff_asf_guid))) {
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            if (chunksize > end - p)
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                return -1;
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            p += chunksize;
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            continue;
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        }
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        /* skip most of the file header, to min_pktsize */
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        p += 6 * 8 + 3 * 4 + sizeof(ff_asf_guid) * 2;
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        if (p + 8 <= end && AV_RL32(p) == AV_RL32(p + 4)) {
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            /* and set that to zero */
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            AV_WL32(p, 0);
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            return 0;
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        }
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        break;
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    } while (end - p >= sizeof(ff_asf_guid) + 8);
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    return -1;
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}
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/**
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 * The following code is basically a buffered ByteIOContext,
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 * with the added benefit of returning -EAGAIN (instead of 0)
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 * on packet boundaries, such that the ASF demuxer can return
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 * safely and resume business at the next packet.
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 */
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static int packetizer_read(void *opaque, uint8_t *buf, int buf_size)
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{
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    return AVERROR(EAGAIN);
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}
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static void init_packetizer(ByteIOContext *pb, uint8_t *buf, int len)
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{
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    init_put_byte(pb, buf, len, 0, NULL, packetizer_read, NULL, NULL);
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    /* this "fills" the buffer with its current content */
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    pb->pos     = len;
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    pb->buf_end = buf + len;
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}
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int ff_wms_parse_sdp_a_line(AVFormatContext *s, const char *p)
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{
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    int ret = 0;
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    if (av_strstart(p, "pgmpu:data:application/vnd.ms.wms-hdr.asfv1;base64,", &p)) {
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        ByteIOContext pb;
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        RTSPState *rt = s->priv_data;
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        int len = strlen(p) * 6 / 8;
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        char *buf = av_mallocz(len);
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        av_base64_decode(buf, p, len);
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        if (rtp_asf_fix_header(buf, len) < 0)
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            av_log(s, AV_LOG_ERROR,
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                   "Failed to fix invalid RTSP-MS/ASF min_pktsize\n");
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        init_packetizer(&pb, buf, len);
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        if (rt->asf_ctx) {
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            av_close_input_stream(rt->asf_ctx);
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            rt->asf_ctx = NULL;
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        }
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        ret = av_open_input_stream(&rt->asf_ctx, &pb, "", &asf_demuxer, NULL);
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        if (ret < 0)
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            return ret;
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        rt->asf_pb_pos = url_ftell(&pb);
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        av_free(buf);
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        rt->asf_ctx->pb = NULL;
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    }
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    return ret;
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}
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static int asfrtp_parse_sdp_line(AVFormatContext *s, int stream_index,
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                                 PayloadContext *asf, const char *line)
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{
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    if (av_strstart(line, "stream:", &line)) {
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        RTSPState *rt = s->priv_data;
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        s->streams[stream_index]->id = strtol(line, NULL, 10);
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        if (rt->asf_ctx) {
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            int i;
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            for (i = 0; i < rt->asf_ctx->nb_streams; i++) {
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                if (s->streams[stream_index]->id == rt->asf_ctx->streams[i]->id) {
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                    *s->streams[stream_index]->codec =
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                        *rt->asf_ctx->streams[i]->codec;
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                    rt->asf_ctx->streams[i]->codec->extradata_size = 0;
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                    rt->asf_ctx->streams[i]->codec->extradata = NULL;
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                    av_set_pts_info(s->streams[stream_index], 32, 1, 1000);
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                }
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           }
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        }
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    }
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    return 0;
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}
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struct PayloadContext {
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    ByteIOContext *pktbuf, pb;
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    uint8_t *buf;
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};
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/**
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 * @return 0 when a packet was written into /p pkt, and no more data is left;
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 *         1 when a packet was written into /p pkt, and more packets might be left;
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 *        <0 when not enough data was provided to return a full packet, or on error.
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 */
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static int asfrtp_parse_packet(AVFormatContext *s, PayloadContext *asf,
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                               AVStream *st, AVPacket *pkt,
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                               uint32_t *timestamp,
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                               const uint8_t *buf, int len, int flags)
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{
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    ByteIOContext *pb = &asf->pb;
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    int res, mflags, len_off;
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    RTSPState *rt = s->priv_data;
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    if (!rt->asf_ctx)
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        return -1;
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    if (len > 0) {
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        int off, out_len;
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        if (len < 4)
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            return -1;
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        init_put_byte(pb, buf, len, 0, NULL, NULL, NULL, NULL);
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        mflags = get_byte(pb);
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        if (mflags & 0x80)
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            flags |= RTP_FLAG_KEY;
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        len_off = get_be24(pb);
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        if (mflags & 0x20)   /**< relative timestamp */
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            url_fskip(pb, 4);
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        if (mflags & 0x10)   /**< has duration */
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            url_fskip(pb, 4);
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        if (mflags & 0x8)    /**< has location ID */
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            url_fskip(pb, 4);
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        off = url_ftell(pb);
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        av_freep(&asf->buf);
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        if (!(mflags & 0x40)) {
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            /**
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             * If 0x40 is not set, the len_off field specifies an offset of this
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             * packet's payload data in the complete (reassembled) ASF packet.
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             * This is used to spread one ASF packet over multiple RTP packets.
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             */
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            if (asf->pktbuf && len_off != url_ftell(asf->pktbuf)) {
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                uint8_t *p;
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                url_close_dyn_buf(asf->pktbuf, &p);
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                asf->pktbuf = NULL;
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                av_free(p);
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            }
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            if (!len_off && !asf->pktbuf &&
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                (res = url_open_dyn_buf(&asf->pktbuf)) < 0)
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                return res;
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            if (!asf->pktbuf)
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                return AVERROR(EIO);
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            put_buffer(asf->pktbuf, buf + off, len - off);
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            if (!(flags & RTP_FLAG_MARKER))
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                return -1;
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            out_len     = url_close_dyn_buf(asf->pktbuf, &asf->buf);
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            asf->pktbuf = NULL;
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        } else {
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            /**
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             * If 0x40 is set, the len_off field specifies the length of the
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             * next ASF packet that can be read from this payload data alone.
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             * This is commonly the same as the payload size, but could be
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             * less in case of packet splitting (i.e. multiple ASF packets in
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             * one RTP packet).
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             */
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            if (len_off != len) {
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                av_log_missing_feature(s,
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                    "RTSP-MS packet splitting", 1);
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                return -1;
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            }
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            asf->buf = av_malloc(len - off);
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            out_len  = len - off;
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            memcpy(asf->buf, buf + off, len - off);
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        }
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        init_packetizer(pb, asf->buf, out_len);
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        pb->pos += rt->asf_pb_pos;
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        pb->eof_reached = 0;
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        rt->asf_ctx->pb = pb;
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    }
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    for (;;) {
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        int i;
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        res = av_read_packet(rt->asf_ctx, pkt);
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        rt->asf_pb_pos = url_ftell(pb);
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        if (res != 0)
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            break;
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        for (i = 0; i < s->nb_streams; i++) {
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            if (s->streams[i]->id == rt->asf_ctx->streams[pkt->stream_index]->id) {
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                pkt->stream_index = i;
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                return 1; // FIXME: return 0 if last packet
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            }
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        }
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        av_free_packet(pkt);
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    }
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    return res == 1 ? -1 : res;
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}
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static PayloadContext *asfrtp_new_context(void)
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{
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    return av_mallocz(sizeof(PayloadContext));
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}
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static void asfrtp_free_context(PayloadContext *asf)
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{
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    if (asf->pktbuf) {
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        uint8_t *p = NULL;
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        url_close_dyn_buf(asf->pktbuf, &p);
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        asf->pktbuf = NULL;
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        av_free(p);
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    }
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    av_freep(&asf->buf);
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    av_free(asf);
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}
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#define RTP_ASF_HANDLER(n, s, t) \
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RTPDynamicProtocolHandler ff_ms_rtp_ ## n ## _handler = { \
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    .enc_name         = s, \
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    .codec_type       = t, \
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    .codec_id         = CODEC_ID_NONE, \
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    .parse_sdp_a_line = asfrtp_parse_sdp_line, \
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    .open             = asfrtp_new_context, \
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    .close            = asfrtp_free_context, \
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    .parse_packet     = asfrtp_parse_packet,   \
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};
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RTP_ASF_HANDLER(asf_pfv, "x-asf-pf",  AVMEDIA_TYPE_VIDEO);
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RTP_ASF_HANDLER(asf_pfa, "x-asf-pf",  AVMEDIA_TYPE_AUDIO);