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authorRonald S. Bultje <rsbultje@gmail.com>2009-07-27 14:00:10 +0000
committerRonald S. Bultje <rsbultje@gmail.com>2009-07-27 14:00:10 +0000
commitc2f3eec445389d67afc8c699ba23915a20cae51c (patch)
tree56948f82774e53c46fcf1a97938870de8f136297
parente9a832e50833b4265fd8ed93fe58038a40131402 (diff)
Implement RTSP-MS/ASF packet parsing - this completes RTSP-MS support. See
discussion in "[PATCH] RTSP-MS 14/15: ASF packet parsing" thread on mailinglist. Originally committed as revision 19516 to svn://svn.ffmpeg.org/ffmpeg/trunk
-rw-r--r--Changelog1
-rw-r--r--libavformat/rtp_asf.c198
-rw-r--r--libavformat/rtsp.h4
3 files changed, 202 insertions, 1 deletions
diff --git a/Changelog b/Changelog
index 613277a084..fe181a6fde 100644
--- a/Changelog
+++ b/Changelog
@@ -28,6 +28,7 @@ version <next>:
- DivX (XSUB) subtitle encoder
- nonfree libamr support for AMR-NB/WB decoding/encoding removed
- Experimental AAC encoder
+- RTP depacketization of ASF and RTSP from WMS servers
diff --git a/libavformat/rtp_asf.c b/libavformat/rtp_asf.c
index b64f4707f8..33a4a31b40 100644
--- a/libavformat/rtp_asf.c
+++ b/libavformat/rtp_asf.c
@@ -27,11 +27,73 @@
#include <libavutil/base64.h>
#include <libavutil/avstring.h>
+#include <libavutil/intreadwrite.h>
#include "rtp.h"
#include "rtp_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)) {
@@ -41,12 +103,16 @@ void ff_wms_parse_sdp_a_line(AVFormatContext *s, const char *p)
char *buf = av_mallocz(len);
av_base64_decode(buf, p, len);
- init_put_byte(&pb, buf, len, 0, NULL, NULL, NULL, NULL);
+ 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;
}
@@ -79,12 +145,142 @@ asfrtp_parse_sdp_line (AVFormatContext *s, int stream_index,
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_packet_buf(&asf->pktbuf, rt->asf_ctx->packet_size)))
+ 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 = { \
s, \
t, \
CODEC_ID_NONE, \
asfrtp_parse_sdp_line, \
+ asfrtp_new_context, \
+ asfrtp_free_context, \
+ asfrtp_parse_packet, \
};
RTP_ASF_HANDLER(asf_pfv, "x-asf-pf", CODEC_TYPE_VIDEO);
diff --git a/libavformat/rtsp.h b/libavformat/rtsp.h
index 37e7ead38e..0772f74cac 100644
--- a/libavformat/rtsp.h
+++ b/libavformat/rtsp.h
@@ -249,6 +249,10 @@ typedef struct RTSPState {
//@{
/** ASF demuxer context for the embedded ASF stream from WMS servers */
AVFormatContext *asf_ctx;
+
+ /** cache for position of the asf demuxer, since we load a new
+ * data packet in the bytecontext for each incoming RTSP packet. */
+ uint64_t asf_pb_pos;
//@}
} RTSPState;