view rtpdec_asf.c @ 5981:c1cd2ffd5cdc libavformat

Disable LATM AAC in mpegts, this is not supported and produce too many bug reports. Also warn the user about it.

/*
 * 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 <[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);