| /* |
| * 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/avassert.h" |
| #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" |
| #include "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)); |
| int skip = 6 * 8 + 3 * 4 + sizeof(ff_asf_guid) * 2; |
| if (memcmp(p, ff_asf_file_header, sizeof(ff_asf_guid))) { |
| if (chunksize > end - p) |
| return -1; |
| p += chunksize; |
| continue; |
| } |
| |
| if (end - p < 8 + skip) |
| break; |
| /* skip most of the file header, to min_pktsize */ |
| p += skip; |
| if (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 = { 0 }; |
| RTSPState *rt = s->priv_data; |
| AVDictionary *opts = NULL; |
| int len = strlen(p) * 6 / 8; |
| char *buf = av_mallocz(len); |
| ff_const59 AVInputFormat *iformat; |
| |
| if (!buf) |
| return AVERROR(ENOMEM); |
| 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) { |
| avformat_close_input(&rt->asf_ctx); |
| } |
| |
| if (!(iformat = av_find_input_format("asf"))) |
| return AVERROR_DEMUXER_NOT_FOUND; |
| |
| rt->asf_ctx = avformat_alloc_context(); |
| if (!rt->asf_ctx) { |
| av_free(buf); |
| return AVERROR(ENOMEM); |
| } |
| rt->asf_ctx->pb = &pb; |
| av_dict_set(&opts, "no_resync_search", "1", 0); |
| |
| if ((ret = ff_copy_whiteblacklists(rt->asf_ctx, s)) < 0) { |
| av_dict_free(&opts); |
| return ret; |
| } |
| |
| ret = avformat_open_input(&rt->asf_ctx, "", iformat, &opts); |
| av_dict_free(&opts); |
| if (ret < 0) { |
| av_free(pb.buffer); |
| return ret; |
| } |
| av_dict_copy(&s->metadata, rt->asf_ctx->metadata, 0); |
| rt->asf_pb_pos = avio_tell(&pb); |
| av_free(pb.buffer); |
| rt->asf_ctx->pb = NULL; |
| } |
| return ret; |
| } |
| |
| static int asfrtp_parse_sdp_line(AVFormatContext *s, int stream_index, |
| PayloadContext *asf, const char *line) |
| { |
| if (stream_index < 0) |
| return 0; |
| 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) { |
| avcodec_parameters_copy(s->streams[stream_index]->codecpar, |
| rt->asf_ctx->streams[i]->codecpar); |
| s->streams[stream_index]->need_parsing = |
| rt->asf_ctx->streams[i]->need_parsing; |
| avpriv_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, uint16_t seq, |
| 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, (uint8_t *)buf, len, 0, NULL, NULL, NULL, NULL); |
| |
| while (avio_tell(pb) + 4 < len) { |
| int start_off = avio_tell(pb); |
| |
| mflags = avio_r8(pb); |
| len_off = avio_rb24(pb); |
| if (mflags & 0x20) /**< relative timestamp */ |
| avio_skip(pb, 4); |
| if (mflags & 0x10) /**< has duration */ |
| avio_skip(pb, 4); |
| if (mflags & 0x8) /**< has location ID */ |
| avio_skip(pb, 4); |
| off = avio_tell(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 != avio_tell(asf->pktbuf)) { |
| ffio_free_dyn_buf(&asf->pktbuf); |
| } |
| if (!len_off && !asf->pktbuf && |
| (res = avio_open_dyn_buf(&asf->pktbuf)) < 0) |
| return res; |
| if (!asf->pktbuf) |
| return AVERROR(EIO); |
| |
| avio_write(asf->pktbuf, buf + off, len - off); |
| avio_skip(pb, len - off); |
| if (!(flags & RTP_FLAG_MARKER)) |
| return -1; |
| out_len = avio_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; |
| if (FFMIN(cur_len, len - off) < 0) |
| return -1; |
| if ((res = av_reallocp(&asf->buf, out_len)) < 0) |
| return res; |
| memcpy(asf->buf + prev_len, buf + off, |
| FFMIN(cur_len, len - off)); |
| avio_skip(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 = ff_read_packet(rt->asf_ctx, pkt); |
| rt->asf_pb_pos = avio_tell(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_packet_unref(pkt); |
| } |
| |
| return res == 1 ? -1 : res; |
| } |
| |
| static void asfrtp_close_context(PayloadContext *asf) |
| { |
| ffio_free_dyn_buf(&asf->pktbuf); |
| av_freep(&asf->buf); |
| } |
| |
| #define RTP_ASF_HANDLER(n, s, t) \ |
| const RTPDynamicProtocolHandler ff_ms_rtp_ ## n ## _handler = { \ |
| .enc_name = s, \ |
| .codec_type = t, \ |
| .codec_id = AV_CODEC_ID_NONE, \ |
| .priv_data_size = sizeof(PayloadContext), \ |
| .parse_sdp_a_line = asfrtp_parse_sdp_line, \ |
| .close = asfrtp_close_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); |