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![Ronald S. Bultje](/assets/img/avatar_default.png)
and put return type ("static void") on the same line as function name. Originally committed as revision 19519 to svn://svn.ffmpeg.org/ffmpeg/trunk
282 lines
8.9 KiB
C
282 lines
8.9 KiB
C
/*
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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 libavformat/rtp_asf.c
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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 "rtp_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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void ff_wms_parse_sdp_a_line(AVFormatContext *s, const char *p)
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{
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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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av_open_input_stream(&rt->asf_ctx, &pb, "", &asf_demuxer, NULL);
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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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}
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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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char *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_packet_buf(&asf->pktbuf, rt->asf_ctx->packet_size)))
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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", CODEC_TYPE_VIDEO);
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RTP_ASF_HANDLER(asf_pfa, "x-asf-pf", CODEC_TYPE_AUDIO);
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