Commit Graph

118 Commits

Author SHA1 Message Date
Martin Storsjö c136a813d7 rtp: Support packetization/depacketization of opus
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-10-09 11:57:11 +03:00
Martin Storsjö 69673138c5 rtpdec: Remove a useless ff_ prefix from a static symbol
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-09-26 19:05:18 +03:00
Dmitry Samonenko b6bf1490da rtpdec: Support depacketizing speex
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-09-26 19:05:10 +03:00
Samuel Pitoiset 3c19815416 rtp: Depacketization of JPEG (RFC 2435)
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-09-09 22:22:21 +03:00
Anton Khirnov 36ef5369ee Replace all CODEC_ID_* with AV_CODEC_ID_* 2012-08-07 16:00:24 +02:00
Anton Khirnov c4ef6a3e4b Add missing libavutil/time.h includes. 2012-07-28 09:02:07 +02:00
Ronald S. Bultje dfb57fc596 rtpdec: Don't explicitly include unistd.h any longer
unistd.h used to be required for gethostname. On windows, gethostname
is provided by winsock2.h. Now network.h includes both unistd.h and
winsock2.h if they exist.

Signed-off-by: Martin Storsjö <martin@martin.st>
2012-06-29 15:38:10 +03:00
Martin Storsjö 89c3960544 rtpdec: Add a depacketizer for iLBC
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-06-18 22:01:04 +03:00
Martin Storsjö 456001486e rtsp: Don't expose the MS-RTSP RTX data stream to the caller
This avoids exposing a dummy AVStream which won't get any data
and which will make avformat_find_stream_info wait for info about
this stream.

Signed-off-by: Martin Storsjö <martin@martin.st>
2012-04-08 12:04:22 +03:00
Justin Ruggles 5602a464c9 avcodec: add a Vorbis parser to get packet duration
This also allows for removing some of the Vorbis-related hacks.
2012-03-03 16:43:11 -05:00
Martin Storsjö 298a587f44 rtp: Factorize the check for distinguishing RTCP packets from RTP
The binary doesn't change after this patch.

Signed-off-by: Martin Storsjö <martin@martin.st>
2012-02-16 17:45:33 +01:00
Martin Storsjö 08bddfcde5 rtpdec: Support H263 in RFC 2190 format
This is different from the "modern" RTP payload formats for H263
as defined by RFC 4629, 2429 and 3555. According to the newer RFCs,
this old one is to be considered deprecated and only be used for
interoperating with legacy systems.

Signed-off-by: Martin Storsjö <martin@martin.st>
2012-02-14 20:05:31 +02:00
Martin Storsjö ad7beb2cac rtpdec: Use our own SSRC in the SDES field when sending RRs
The s->ssrc field is the sender's SSRC, we use ssrc + 1 to get
a collision free "unique" SSRC for ourselves in the RR part.
The SDES block in the RTCP packet should describe ourselves,
not the sender.

This was fixed for the RR part in 952139a322, but wasn't
fixed for the SDES part until now.

This could cause some Axis cameras to send RTCP BYE packets
to us due to the SSRC collision.

Signed-off-by: Martin Storsjö <martin@martin.st>
2012-01-21 22:18:12 +02:00
Miroslav Slugeň 06d7325ab1 rtpdec: Add support for G726 audio
This requires using a separate init function, since there
isn't necessarily any fmtp lines for this codec, so
parse_sdp_a_line won't be called. Incorporating it with the
alloc function wouldn't do either, since it is called before
the full rtpmap line is parsed (where the sample rate is
extracted).

Signed-off-by: Martin Storsjö <martin@martin.st>
2011-11-30 17:39:32 +02:00
John Brooks 525c5b08fb rtpdec: only use RTCP for PTS when synchronizing multiple streams
RTCP timestamps are only necessary to synchronize time between
multiple streams. For a single stream, the RTP packet timestamp
provides more reliable timing. As a result, single-stream RTP
sessions should now have accurate and monotonic PTS.

Signed-off-by: Martin Storsjö <martin@martin.st>
2011-11-18 10:47:28 +02:00
John Brooks 12348ca25e rtpdec: unwrap RTP timestamps for PTS calculation
The timestamp field in RTPDemuxContext was unused before this.

Signed-off-by: Martin Storsjö <martin@martin.st>
2011-11-18 10:31:17 +02:00
John Brooks b8a1b880ee rtpdec: Simplify finalize_packet
Signed-off-by: Martin Storsjö <martin@martin.st>
2011-11-11 14:26:37 +02:00
Reimar Döffinger bb3244dee2 Replace all usage of strcasecmp/strncasecmp
All current usages of it are incompatible with localization.
For example strcasecmp("i", "I") != 0 is possible, but would
break many of the places where it is used.

Instead use our own implementations that always treat the data
as ASCII.

Signed-off-by: Martin Storsjö <martin@martin.st>
2011-11-06 11:52:57 +02:00
Martin Storsjö bfc6db4477 rtpdec: Add ff_ prefix to all nonstatic symbols
Signed-off-by: Martin Storsjö <martin@martin.st>
2011-10-12 14:48:12 +03:00
John Brooks 07b77fe387 rtpdec: Read the packet length for all RTCP packet types
This allows skipping past unsupported RTCP packet types, as
RFC 3550 section 6.1 mandates.

Currently this only has any practical effect if a sender puts
an unrecognized type before RTCP_BYE in a compounded packet, or
(incorrectly) does not put RTCP_SR first.

Signed-off-by: Martin Storsjö <martin@martin.st>
2011-10-12 14:37:42 +03:00
John Brooks 5d6ecf5345 rtpdec: Fix the minimum packet length for RTCP SR packets
We actually read 20 bytes of these packets.

Signed-off-by: Martin Storsjö <martin@martin.st>
2011-10-12 14:37:39 +03:00
Diego Biurrun 4cc843facd rtp: remove disabled code 2011-07-21 23:03:10 +02:00
Mans Rullgard 0ebcdf5cda Do not include mathematics.h in avutil.h
Signed-off-by: Mans Rullgard <mans@mansr.com>
2011-07-03 21:42:06 +01:00
Mans Rullgard 5e1166b31b Mark some variables with av_unused
Most of these variables are only used in av_dlog statements, some
are required but not used by other macros.

Signed-off-by: Mans Rullgard <mans@mansr.com>
2011-06-03 12:59:05 +01:00
Diego Biurrun 046f081b46 configure: Do not unconditionally add -D_POSIX_C_SOURCE to CPPFLAGS.
Adding _POSIX_C_SOURCE to CPPFLAGS globally produces all sorts of problems
since it causes certain system functions to be hidden on some (BSD) systems.
The solution is to only add the flag on systems that really require it, i.e.
glibc-based ones.

This change makes BSD systems compile out-of-the-box without the need for
adding specific flags manually.  It also allows dropping a number of flags
set manually on a file-per-file basis, but were only present to work around
breakage introduced by the presence of _POSIX_C_SOURCE.

Also add _XOPEN_SOURCE to CPPFLAGS for glibc systems.  We use XSI extensions
in several places already, so it is preferable to define it globally instead
of littering source files with individual #defines only needed for glibc.
2011-05-12 11:41:59 +02:00
Anton Khirnov 925e908bc7 avio: make url_write() internal. 2011-04-04 17:45:20 +02:00
Anton Khirnov 403ee835e7 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 be
cleaned up when somebody is up for it.
2011-04-03 22:47:32 +02:00
Anton Khirnov 6dc7d80de7 avio: avio_ prefix for url_close_dyn_buf 2011-04-03 22:47:05 +02:00
Anton Khirnov b92c545282 avio: avio_ prefix for url_open_dyn_buf 2011-04-03 22:46:56 +02:00
Mans Rullgard 2912e87a6c Replace FFmpeg with Libav in licence headers
Signed-off-by: Mans Rullgard <mans@mansr.com>
2011-03-19 13:33:20 +00:00
Anton Khirnov b7f2fdde74 avio: rename put_flush_packet -> avio_flush
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
2011-03-16 22:59:39 -04:00
Anton Khirnov 77eb5504d3 avio: avio: avio_ prefixes for put_* functions
In the name of consistency:
put_byte           -> avio_w8
put_<type>         -> avio_w<type>
put_buffer         -> avio_write

put_nbyte will be made private
put_tag will be merged with avio_put_str

Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
2011-02-21 14:25:15 -05:00
Anton Khirnov ae628ec1fd avio: rename ByteIOContext to AVIOContext.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
2011-02-20 08:37:15 -05:00
Luca Barbato dfd2a005eb Replace dprintf with av_dlog
dprintf clashes with POSIX.1-2008
2011-01-29 23:55:37 +01:00
Diego Elio Pettenò 119cc033fc Make RTPFirstDynamicPayloadHandler static to rtpdec.c
Signed-off-by: Mans Rullgard <mans@mansr.com>
2011-01-25 01:45:34 +00:00
Diego Elio Pettenò 69ad22c7a7 Make ff_realmedia_mp3_dynamic_handler static.
Signed-off-by: Mans Rullgard <mans@mansr.com>
2011-01-25 01:37:32 +00:00
Martin Storsjö 79d482b108 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 earlier
timestamping code never wrote anything into pkt->pts. The rtpdec_asf
depacketizer just sets the dts of the packet, so if the generic RTP
timestamping is used, too, we get inconsistent timestamps.

Therefore, skip the generic RTP timestamp algorithm if the depacketizer
already has set something.

This fixes "Invalid timestamps" warnings, present since SVN rev 26187.

Originally committed as revision 26241 to svn://svn.ffmpeg.org/ffmpeg/trunk
2011-01-06 11:33:06 +00:00
Martin Storsjö 3a1cdcc798 rtpdec: Emit timestamps for packets before the first RTCP packet, too
Emitted timestamps in each stream start from 0, for the first received
RTP packet. Once an RTCP packet is received, that one is used for
sync, emitting timestamps that fit seamlessly into the earlier ones.

Originally committed as revision 26187 to svn://svn.ffmpeg.org/ffmpeg/trunk
2011-01-01 22:27:16 +00:00
Martin Storsjö bbd8f5477d 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 base
with 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.

Originally committed as revision 26022 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-15 21:06:25 +00:00
Luca Barbato a4a3bade0a 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
2010-12-15 17:16:37 +00:00
Martin Storsjö 86b6e387cc 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 stream
differ. Previously, the AVStream time_base was unconditionally set to
the sample rate (which initially was set to one value when parsing the
rtpmap field in the SDP, but later overridden by an a=SampleRate field).

Additionally, this makes the code actually use the stream time base set
in rtpmap for video codecs, instead of hardcoding it to always be 90 kHz.

Originally committed as revision 25908 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-07 13:29:44 +00:00
Martin Storsjö 2eeefe205f rtpdec: Handle MP3 in RealRTSP
This fixes playback of a RealRTSP/MP3 URL from the RTSP samples on
MultimediaWiki.

Originally committed as revision 25906 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-07 13:28:13 +00:00
Martin Storsjö 4838cdab21 rtpdec: Skip padding bytes at the end of packets
Originally committed as revision 25896 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-06 12:43:38 +00:00
Martin Storsjö 1e515c4280 rtpdec: Add functions for finding depacketizers by name or payload id
Originally committed as revision 25891 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-05 19:39:50 +00:00
Martin Storsjö 35014efcc6 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
2010-12-05 19:37:45 +00:00
Martin Storsjö 946df0598b rtpdec: Return AVERROR(EAGAIN) for mpegts parsing errors
This indicates that there was no error that needs to be reported to the
caller, so we can move on to parse the next packet immediately, if
available. The only error code that ff_mpegts_parse_packet can return
indicates that there was no packet to return from the provided data, for
which it returns -1.

Originally committed as revision 25496 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-10-15 21:32:21 +00:00
Martin Storsjö 65cdee9c95 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
2010-10-13 08:47:34 +00:00
Martin Storsjö ddcf841191 rtpdec: Handle wrapping seq numbers in has_next_packet properly
Originally committed as revision 25461 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-10-13 08:15:39 +00:00
Martin Storsjö d678a6fd82 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
2010-10-13 08:14:30 +00:00
Martin Storsjö 91ec7aea20 rtpdec: Return AVERROR(EAGAIN) if out of data for mpegts, pass returned error codes through
Originally committed as revision 25459 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-10-13 08:13:53 +00:00