/* * 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 */ #include #include #include #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);