PeerStreamer

/*

 * 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 = url_ftell(&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 (url_ftell(pb) + 4 < len) {

            int start_off = url_ftell(pb);

            mflags = get_byte(pb);

            if (mflags & 0x80)

                flags |= RTP_FLAG_KEY;

            len_off = get_be24(pb);

            if (mflags & 0x20)   /**< relative timestamp */

                url_fskip(pb, 4);

            if (mflags & 0x10)   /**< has duration */

                url_fskip(pb, 4);

            if (mflags & 0x8)    /**< has location ID */

                url_fskip(pb, 4);

            off = url_ftell(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 != url_ftell(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);

                put_buffer(asf->pktbuf, buf + off, len - off);

                url_fskip(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));

                url_fskip(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 = url_ftell(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);