view rtpdec_asf.c @ 5966:9a86b49201bb libavformat

Revert svn rev 21857, readd first_rtcp_ntp_time in RTPDemuxContext In order to sync RTP streams that get their initial RTCP timestamp at different times, propagate the NTP timestamp of the first RTCP packet to all other streams. This makes the timestamps of returned packets start at (near) zero instead of at any random offset.

/*
 * 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 libavformat/rtpdec_asf.c
 * @brief Microsoft RTP/ASF support
 * @author Ronald S. Bultje <[e-mail]>
 */

#include <libavutil/base64.h>
#include <libavutil/avstring.h>
#include <libavutil/intreadwrite.h>
#include "rtp.h"
#include "rtpdec_asf.h"
#include "rtsp.h"
#include "asf.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 ByteIOContext,
 * 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(ByteIOContext *pb, uint8_t *buf, int len)
{
    init_put_byte(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;
}

void ff_wms_parse_sdp_a_line(AVFormatContext *s, const char *p)
{
    if (av_strstart(p, "pgmpu:data:application/vnd.ms.wms-hdr.asfv1;base64,", &p)) {
        ByteIOContext 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;
        }
        av_open_input_stream(&rt->asf_ctx, &pb, "", &asf_demuxer, NULL);
        rt->asf_pb_pos = url_ftell(&pb);
        av_free(buf);
        rt->asf_ctx->pb = NULL;
    }
}

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 {
    ByteIOContext *pktbuf, pb;
    char *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)
{
    ByteIOContext *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;

        if (len < 4)
            return -1;

        init_put_byte(pb, buf, len, 0, NULL, NULL, NULL, NULL);
        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);

        av_freep(&asf->buf);
        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);
            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).
             */
            if (len_off != len) {
                av_log_missing_feature(s,
                    "RTSP-MS packet splitting", 1);
                return -1;
            }
            asf->buf = av_malloc(len - off);
            out_len  = len - off;
            memcpy(asf->buf, buf + off, len - off);
        }

        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);