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avio: make url_open_dyn_packet_buf internal.
It doesn't look fit to be a part of the public API.
Adding a temporary hack to ffserver to be able to use it, should becleaned up when somebody is up for it.
avio: avio_ prefix for url_close_dyn_buf
avio: avio_ prefix for url_open_dyn_buf
Replace FFmpeg with Libav in licence headers
Signed-off-by: Mans Rullgard <mans@mansr.com>
avio: rename put_flush_packet -> avio_flush
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
avio: avio: avio_ prefixes for put_* functions
In the name of consistency:put_byte -> avio_w8put_<type> -> avio_w<type>put_buffer -> avio_write
put_nbyte will be made privateput_tag will be merged with avio_put_str
avio: rename ByteIOContext to AVIOContext.
Replace dprintf with av_dlog
dprintf clashes with POSIX.1-2008
Make RTPFirstDynamicPayloadHandler static to rtpdec.c
Make ff_realmedia_mp3_dynamic_handler static.
rtpdec: Don't set RTP timestamps if they already are set by the depacketizer
For MS-RTSP, we don't always get RTCP packets (never?), so the earliertimestamping code never wrote anything into pkt->pts. The rtpdec_asfdepacketizer just sets the dts of the packet, so if the generic RTP...
rtpdec: Emit timestamps for packets before the first RTCP packet, too
Emitted timestamps in each stream start from 0, for the first receivedRTP packet. Once an RTCP packet is received, that one is used forsync, emitting timestamps that fit seamlessly into the earlier ones....
rtsp: Don't set the RTP time base from the sample rate if no sample rate is set
This also reverts SVN rev 26016, which incorrectly overwrote the time basewith 90 kHz for all streams, regardless of what was set by the SDP parsing.
The stream that triggered the fix in 26016 still works after this commit....
Reinstate default time_base for rtp streams
The generic default is 0/0 and that obviously triggers once the value is used.
Originally committed as revision 26016 to svn://svn.ffmpeg.org/ffmpeg/trunk
rtsp/rtpdec: Set the AVStream time_base directly in rtsp when it is known
This fixes cases where the RTP time base and the sample rate of the streamdiffer. Previously, the AVStream time_base was unconditionally set tothe sample rate (which initially was set to one value when parsing the...
rtpdec: Handle MP3 in RealRTSP
This fixes playback of a RealRTSP/MP3 URL from the RTSP samples onMultimediaWiki.
Originally committed as revision 25906 to svn://svn.ffmpeg.org/ffmpeg/trunk
rtpdec: Skip padding bytes at the end of packets
Originally committed as revision 25896 to svn://svn.ffmpeg.org/ffmpeg/trunk
rtpdec: Add functions for finding depacketizers by name or payload id
Originally committed as revision 25891 to svn://svn.ffmpeg.org/ffmpeg/trunk
rtpdec: Add a dynamic payload handler for the x-Purevoice format, RFC 2658
This fixes roundup issue 2390.
Originally committed as revision 25889 to svn://svn.ffmpeg.org/ffmpeg/trunk
rtpdec: Return AVERROR for mpegts parsing errors
This indicates that there was no error that needs to be reported to thecaller, so we can move on to parse the next packet immediately, ifavailable. The only error code that ff_mpegts_parse_packet can return...
rtpdec: Don't use the no reordering codepath if there already is a queue
Originally committed as revision 25462 to svn://svn.ffmpeg.org/ffmpeg/trunk
rtpdec: Handle wrapping seq numbers in has_next_packet properly
Originally committed as revision 25461 to svn://svn.ffmpeg.org/ffmpeg/trunk
rtpdec: Parse the next packet in the sequence if it is available, if the previous packet didn't return any data
Originally committed as revision 25460 to svn://svn.ffmpeg.org/ffmpeg/trunk
rtpdec: Return AVERROR if out of data for mpegts, pass returned error codes through
Originally committed as revision 25459 to svn://svn.ffmpeg.org/ffmpeg/trunk
rtpdec: Don't call the depacketizer to return more data unless it actually said it has more data
It may have returned a negative number for an error (e.g. AVERROR,if more data is required for it to be able to return a complete packet).
Originally committed as revision 25458 to svn://svn.ffmpeg.org/ffmpeg/trunk
rtpdec: Split out storing of the depacketization return value to a separate function
This makes the code less fragile and easier to understand.
Originally committed as revision 25457 to svn://svn.ffmpeg.org/ffmpeg/trunk
rtpdec: Set prev_ret properly when parsing more data from mpegts RTP packets
Originally committed as revision 25404 to svn://svn.ffmpeg.org/ffmpeg/trunk
rtpdec: Store the previous return value for mpegts when it was -1, too
Originally committed as revision 25403 to svn://svn.ffmpeg.org/ffmpeg/trunk
rtpdec: Keep track of the previous return value from rtp_parse_packet_internal for mpegts packets
Patch by Robert Schlabbach, robert_s at gmx dot net
Originally committed as revision 25402 to svn://svn.ffmpeg.org/ffmpeg/trunk
rtpdec: Handle RTP header extension
This fixes roundup issue 2270.
Originally committed as revision 25372 to svn://svn.ffmpeg.org/ffmpeg/trunk
Add RTP depacketization of the X-QT QuickTime format
Originally committed as revision 25371 to svn://svn.ffmpeg.org/ffmpeg/trunk
rtpdec: Reorder received RTP packets according to the seq number
Reordering is enabled only when receiving over UDP.
Originally committed as revision 25294 to svn://svn.ffmpeg.org/ffmpeg/trunk
rtpdec: Split out the part of rtp_parse_packet that does the parsing of new packets
Originally committed as revision 25293 to svn://svn.ffmpeg.org/ffmpeg/trunk
rtsp/rtpdec: Allow rtp_parse_packet to take ownership of the packet buffer
Do the same change for ff_rdt_parse_packet, too, to keep the interfacessimilar.
Originally committed as revision 25289 to svn://svn.ffmpeg.org/ffmpeg/trunk
Handle G.722 in RTP, and all the exceptions mandated in RFC 3551
Originally committed as revision 25125 to svn://svn.ffmpeg.org/ffmpeg/trunk
rtsp: Return AVERROR_EOF when all streams have received an RTCP BYE packet
Patch by Josh Allmann, joshua dot allmann at gmail
Originally committed as revision 24965 to svn://svn.ffmpeg.org/ffmpeg/trunk
Reindent
Originally committed as revision 24964 to svn://svn.ffmpeg.org/ffmpeg/trunk
rtpdec: Read RTCP compound packets
Originally committed as revision 24963 to svn://svn.ffmpeg.org/ffmpeg/trunk
rtp: Replace hardcoded RTCP packet types with defines
Originally committed as revision 24912 to svn://svn.ffmpeg.org/ffmpeg/trunk
Do not use the server SSRC as client SSRC in the RTP demuxer
Originally committed as revision 24879 to svn://svn.ffmpeg.org/ffmpeg/trunk
Add RTP depacketization of VP8
Originally committed as revision 24798 to svn://svn.ffmpeg.org/ffmpeg/trunk
Add RTP depacketization of MP4A-LATM
Originally committed as revision 24790 to svn://svn.ffmpeg.org/ffmpeg/trunk
Remove mostly unnecessary rtpdec_*.h files, store the declarations in one file
Originally committed as revision 24596 to svn://svn.ffmpeg.org/ffmpeg/trunk
Add a depacketizer for QDM2
Patch by Josh Allmann, joshua dot allmann at gmail, original codeby Ronald S Bultje.
Originally committed as revision 24236 to svn://svn.ffmpeg.org/ffmpeg/trunk
rtpdec: Allow depacketizers to specify that pkt->pts should be left as AV_NOPTS_VALUE
Originally committed as revision 24234 to svn://svn.ffmpeg.org/ffmpeg/trunk
Add RTP depacketization of SVQ3
Originally committed as revision 23941 to svn://svn.ffmpeg.org/ffmpeg/trunk
rtpdec: Malloc the fmtp value buffer
This allows very large value strings, needed for xiph extradata.
Originally committed as revision 23859 to svn://svn.ffmpeg.org/ffmpeg/trunk
rtpdec: Add generic function for iterating over FMTP configuration lines
This will be used for cleaning up code that is common among RTP depacketizers.
Originally committed as revision 23847 to svn://svn.ffmpeg.org/ffmpeg/trunk
RTSP, rtpdec: Move RTPPayloadData into rtpdec_mpeg4 and remove all references to rtp_payload_data in rtpdec and rtsp
Originally committed as revision 23772 to svn://svn.ffmpeg.org/ffmpeg/trunk
rtpdec: Move AAC depacketization code in rtpdec to a proper payload handler
Originally committed as revision 23771 to svn://svn.ffmpeg.org/ffmpeg/trunk
RTSP: Decouple MPEG-4 and AAC specific parts from rtsp.c
Originally committed as revision 23769 to svn://svn.ffmpeg.org/ffmpeg/trunk
Reset RTCP timestamps after seeking, add range start offset to the packets timestamps
If these aren't reset, the timestamps make a huge jump when the next RTCPis received.
Originally committed as revision 22918 to svn://svn.ffmpeg.org/ffmpeg/trunk
Revert svn rev 21857, readd first_rtcp_ntp_time in RTPDemuxContext
In order to sync RTP streams that get their initial RTCP timestamp atdifferent times, propagate the NTP timestamp of the first RTCP packetto all other streams.
This makes the timestamps of returned packets start at (near) zero instead...
Originally committed as revision 22805 to svn://svn.ffmpeg.org/ffmpeg/trunk
Fix leaks in the AAC RTP depacketizer
Originally committed as revision 22804 to svn://svn.ffmpeg.org/ffmpeg/trunk
Rename rtpdec_theora.[ch] to rtpdec_xiph.[ch], as a preparation for mergingthe Vorbis / theora depacketizers.
Patch by Josh Allmann <joshua DOT allmann AT gmail DOT com>.
Originally committed as revision 22765 to svn://svn.ffmpeg.org/ffmpeg/trunk
Define AVMediaType enum, and use it instead of enum CodecType, whichis deprecated and will be dropped at the next major bump.
Originally committed as revision 22735 to svn://svn.ffmpeg.org/ffmpeg/trunk
RTP depacketization of Theora
Patch by Josh Allmann (joshua allmann gmail com)
Originally committed as revision 22636 to svn://svn.ffmpeg.org/ffmpeg/trunk
Rename RTP depacketizer files from rtp_* to rtpdec_*
Originally committed as revision 22109 to svn://svn.ffmpeg.org/ffmpeg/trunk
Remove first_rtcp_ntp_time. This is used to prevent overflow of the timestamp,but doesn't actually do that. What's worse, it creates timestamp adjustmentsthat are different per stream within a session, leading to a/v sync issues.
See discussion in thread "[FFmpeg-devel] rtp streaming x264+audio issues (and...
When using RTP-over-UDP, send dummy packets during stream setup, similar towhat e.g. RealPlayer does. This allows proper port forwarding setup in NAT-based environments.
Patch by Martin Storsjö <$firstname at $firstname dot st>.
Originally committed as revision 21856 to svn://svn.ffmpeg.org/ffmpeg/trunk
RTP/AMR depacketizer, by Martin Storsjö <$firstname at $firstname dot st>.
Originally committed as revision 21740 to svn://svn.ffmpeg.org/ffmpeg/trunk
Fix warnings about implicit function declaration when compiling rtpdec.c
Patch by Alexis Ballier, alexis D ballier A gmail
Originally committed as revision 21601 to svn://svn.ffmpeg.org/ffmpeg/trunk
Add RTP/H.263 depacketizer by Martin Storsjö <$firstname () $firstname st>.
Originally committed as revision 21512 to svn://svn.ffmpeg.org/ffmpeg/trunk
Add support for mp3 over RTP in rtpdec.c
Originally committed as revision 20916 to svn://svn.ffmpeg.org/ffmpeg/trunk
Add a Vorbis payload parser. Implemented by Colin McQuillan as a GSoCqualification task, see "RTP/Vorbis payload implementation (GSoC qualtask)" thread on mailinglist.
Originally committed as revision 18509 to svn://svn.ffmpeg.org/ffmpeg/trunk
Rename bitstream.h to get_bits.h.
Originally committed as revision 18494 to svn://svn.ffmpeg.org/ffmpeg/trunk
Assign the x-pf-asf payload string to be decoded by rtp_asf.c, and add aSDP line handler that parses the streamID in the SDP so that ASF streamdata can be matched to their respective streams in the RTSP demuxer. See"[PATCH] RTSP-MS 12/15: ASF payload support" thread on mailinglist....
Don't let finalize_packet() touch pkt->stream_index. Instead, let individualpayload handlers take care of that themselves at their own option. What thispatch really does is "fix" a bug in MS-RTSP protocol where incoming packetsare always coming in over the connection (UDP) or interleave-id (TCP) of...
Reindent after r17764.
Originally committed as revision 17765 to svn://svn.ffmpeg.org/ffmpeg/trunk
In the current implementation of rtp_parse_packet(), finalize_packet() iscalled for all packets with an internal handler function but only fornon-first packets from dynamic payload parse_packet() handlers. This patchfixes that. Bug was noticed by Luca in "[PATCH] rtpdec.c: don't overwrite...
Implement marker bit, which is used for several RTP payloads currentlyunder review. See "[FFmpeg-devel] RTP mark bit not passed to parse_packet" thread on mailinglist.
Originally committed as revision 17616 to svn://svn.ffmpeg.org/ffmpeg/trunk
Split rtp.h in rtp.h, rtpdec.h, and rtpenc.h
Originally committed as revision 17016 to svn://svn.ffmpeg.org/ffmpeg/trunk
Add "AVFormatContext *ctx" (that being the RTSP demuxer's) as first argumentto the parse_packet() function pointer in RTPDynamicProtocolHandlers. Thisallows these functions to peek back and retrieve values from the demuxer'scontext (or RTSPState). The ASF/RTP payload parser will use this to be able...
Merge rtp_internal.h in rtp.h, and remove rtp_internal.h
Originally committed as revision 16817 to svn://svn.ffmpeg.org/ffmpeg/trunk
cosmetics: Remove pointless period after copyright statement non-sentences.
Originally committed as revision 16684 to svn://svn.ffmpeg.org/ffmpeg/trunk
Rename rtp_payload_data_t to avoid clashes with the POSIX namespace
Originally committed as revision 16115 to svn://svn.ffmpeg.org/ffmpeg/trunk
Remove access into RTPDemuxContext in rtsp.c, which allows making it opaque(and thus preparing for the introduction of RDTDemuxContext) in a next patch.See discussion in "RDT/Realmedia patches #2" thread on ML.
Originally committed as revision 15542 to svn://svn.ffmpeg.org/ffmpeg/trunk
Change function prototype of RTPDynamicPayloadHandler.parse_packet() tonot use RTPDemuxContext, but rather take a pointer to the payload contextdirectly. This allows using payload handlers regardless over the transportover which they were sent, and prepares for the introduction of a future...
Uniformly define _XOPEN_SOURCE to 600.The feature_tests.h header from Sun systems (Solaris/OpenSolaris) will abortthe build if _XOPEN_SOURCE is defined to 500, and C99 is requested (as wellas POSIX.1-2001), and will only accept it to be defined to 600....
Give register_dynamic_payload_handler() in rtpdec.c a ff_ prefix and exportit so that I can use it in rdt.c as well. See discussion in "Realmedia patch" thread on ML.
Originally committed as revision 15233 to svn://svn.ffmpeg.org/ffmpeg/trunk
Do not set timestamp information for a non existing AVStream(fix a bug in the RTP demuxer)
Originally committed as revision 14909 to svn://svn.ffmpeg.org/ffmpeg/trunk
ensure we get explicit definition of various _XOPEN_SOURCE functions we use
Originally committed as revision 14766 to svn://svn.ffmpeg.org/ffmpeg/trunk
RTP: use dprintf(), allow compilation with -DDEBUG
Originally committed as revision 14211 to svn://svn.ffmpeg.org/ffmpeg/trunk
Reindent after last commit
Originally committed as revision 14046 to svn://svn.ffmpeg.org/ffmpeg/trunk
Fix A/V synch for RTP streams that do not contain MPEG1 or 2(correctly compute the presentation times based on the RTP timestampsand the RTCP SR packets)
Originally committed as revision 14045 to svn://svn.ffmpeg.org/ffmpeg/trunk
Use full path for #includes from another directory.
Originally committed as revision 13098 to svn://svn.ffmpeg.org/ffmpeg/trunk
typo fixes
Originally committed as revision 12449 to svn://svn.ffmpeg.org/ffmpeg/trunk
Add a flags field to the RTPDynamicPayloadPacketHandlerProc (PKT_FLAG_*).This can be used later by RDT to get the flags from the RTP packet anduse that for the RealMedia packet (such as whether this RTP packetrepresents a keyframe or not). For discussion, see "[PATCH] Realmedia...
Reindent after r11493 (always use parse_packet() vfunc in rtp_parse_packet()),see "[PATCH] Realmedia / RTSP (RDT)" thread on ML.
Originally committed as revision 11494 to svn://svn.ffmpeg.org/ffmpeg/trunk
Make rtp_parse_packet() always call the vfunc of the dynamic payload handlerif there is one. See "[PATCH] Realmedia / RTSP (RDT)" thread on ML.
Originally committed as revision 11493 to svn://svn.ffmpeg.org/ffmpeg/trunk
Split the RTP demuxing functions out of rtp.c, to simplify the RTP muxer's dependencies
Originally committed as revision 11406 to svn://svn.ffmpeg.org/ffmpeg/trunk