Index: patched-ffmpeg-mt/libavformat/rtpdec_asf.c |
=================================================================== |
--- patched-ffmpeg-mt/libavformat/rtpdec_asf.c (revision 0) |
+++ patched-ffmpeg-mt/libavformat/rtpdec_asf.c (revision 0) |
@@ -0,0 +1,281 @@ |
+/* |
+ * 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 <rbultje@ronald.bitfreak.net> |
+ */ |
+ |
+#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", CODEC_TYPE_VIDEO); |
+RTP_ASF_HANDLER(asf_pfa, "x-asf-pf", CODEC_TYPE_AUDIO); |