Commit Graph

9324 Commits

Author SHA1 Message Date
Diego Biurrun b5a138652f Give less generic names to global library option arrays 2013-08-02 19:19:02 +02:00
Gavriloaie Eugen-Andrei 0d6fa3977b rtmp: Add seek support
Signed-off-by: Martin Storsjö <martin@martin.st>
2013-08-02 20:11:54 +03:00
Diego Biurrun f900f35ac8 flvdec: Eliminate completely silly goto 2013-08-01 12:43:54 +02:00
Diego Biurrun e4529df944 flvdec: K&R formatting cosmetics 2013-08-01 12:43:53 +02:00
Diego Biurrun 390b4d7088 flvdec: Fix = vs. == typo in sample rate check 2013-08-01 12:43:53 +02:00
Martin Storsjö fd8f91e3f4 rtsp: Simplify code for forming the remote peer url
Signed-off-by: Martin Storsjö <martin@martin.st>
2013-08-01 12:11:17 +03:00
Martin Storsjö ee37d5811c rtpproto: Allow specifying a separate rtcp port in ff_rtp_set_remote_url
A separate rtcp port can already be set when opening the rtp
protocol normally, but when doing port setup as in RTSP (where
we first need to open the local ports and pass them to the peer,
and only then receive the remote peer port numbers), we didn't
check the same url parameter as in the normal open routine.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-08-01 10:26:17 +03:00
Martin Storsjö 1851e1d05d rtpproto: Check the size before reading buf[1]
I doubt that anyone ever would try to send a 1 byte packet
via the RTP protocol, but check just in case - it shouldn't
crash at least.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-08-01 09:41:21 +03:00
Martin Storsjö b7e6da988b rtpproto: Move rtpproto specific function declarations to a separate header
Mixing these with the rtp depacketizer functions in rtpdec.h is
no good.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-07-31 21:12:34 +03:00
Martin Storsjö 892b0be1df rtpproto: Simplify the rtp_read function by looping over the fds
This avoids having duplicate code where only the fd parameter
differs.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-07-31 21:12:34 +03:00
Martin Storsjö 7531588fff rtpproto: Remove a misplaced comment
The fdset is a pollfd array nowadays, and it is already populated
at this point.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-07-31 21:12:34 +03:00
Martin Storsjö 54e03ff6af rtpproto: Support nonblocking reads
Signed-off-by: Martin Storsjö <martin@martin.st>
2013-07-31 21:12:34 +03:00
Martin Storsjö 2e814d0329 rtpenc: Simplify code by introducing a macro for rescaling NTP timestamps
Signed-off-by: Martin Storsjö <martin@martin.st>
2013-07-31 21:12:34 +03:00
Michael Niedermayer 9696740af7 hls: Call avformat_find_stream_info() on the chained demuxers
This allows the chained demuxer (or more precisely, the lavf
utility code) to better fill in timestamps on packets from
these, especially for cases where one stream is a raw ADTS
stream.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-07-30 12:31:08 +03:00
Ed Torbett 1f57d60129 rtsp: Support RFC4570 (source specific multicast) more properly.
Add support for domain names, for multiple source addresses,
for exclusions, and for session level specification of addresses.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-07-29 22:58:56 +03:00
Ed Torbett 7497222090 rtpproto: Support more than one SSM include address, support excludes
Signed-off-by: Martin Storsjö <martin@martin.st>
2013-07-29 22:58:55 +03:00
Ed Torbett 7d99c92945 udp: Keep track of include and exclude sources separately
This allows us to explicitly fail if the caller tried to set
both inclusions and exclusions at the same time.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-07-29 22:58:54 +03:00
Ed Torbett 3357bccc5c udp: Allow specifying multicast include/blocks as host names as well
Previously this only allowed literal IP addresses. When these
are conveyed in a SDP file as in RFC4570, host names are allowed
as well.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-07-29 22:58:53 +03:00
Diego Biurrun 0e8c6f221a avisynth: Fix a "AVISynth" vs. "AviSynth" struct name typo
Also fix some similar typos in comments and documentation.
2013-07-29 20:09:59 +02:00
Martin Storsjö 06205b5efd hls: Free packets when skipping packets when seeking
This fixes memory leaks present since 2b3d041cdc.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-07-29 20:15:50 +03:00
Martin Storsjö a2b7eeeb06 hlsproto: Store all durations in AV_TIME_BASE
Also parse segment durations as floating point, which is allowed
since HLS version 3.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-07-29 20:15:50 +03:00
Martin Storsjö c441910399 hls: Store all durations in AV_TIME_BASE
Also parse segment durations as floating point, which is allowed
since HLS version 3.

This is based on a patch by Zhang Rui.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-07-29 20:15:50 +03:00
Martin Storsjö e1d5b24476 hls: Store first_timestamp in units of AV_TIME_BASE
When first_timestamp was stored as-is, its actual time base
wasn't known later in the seek function.

Additionally, the logic (from 795d9594cf) for scaling it
based on stream_index is flawed - stream_index in the seek
function only specifies which stream the seek timestamp refers
to, but obviously doesn't say anything about which stream
first_timestamp belongs to.

In the cases where stream_index was >= 0 and all streams had the
same time base, this didn't matter in practice.

Seeking taking first_timestamp into account is problematic
when one variant is mpegts (with real timestamps) and one variant
is raw ADTS (with timestamps only being accumulated packet
duration), where the variants start at totally different timestamps.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-07-29 20:15:50 +03:00
LYF c110cbf6b5 hls: Create an AVProgram for each variant
Without the information, an application may choose audio from one
variant and video from another variant, which leads to fetching two
variants from the network. This enables av_find_best_stream() to find
matching audio and video streams, so that only one variant is fetched.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-07-29 09:50:09 +03:00
Michael Niedermayer 9d64f23629 hls: Respect the different stream time bases when comparing dts
Also adjust the streams timestamps according to their start
timestamp when comparing. This helps getting correctly interleaved
packets if one stream lacks timestamps (such as a plain ADTS
stream when the other variants are full mpegts) when the others
have timestamps that don't start from zero.

This probably doesn't work properly if such a stream is
temporarily disabled (via the discard flags) and then reenabled,
and such streams are hard to correctly sync against the other
streams as well - but this works better than before at least.

The segment number restriction makes sure all variants advance
roughly at the same pace as well.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-07-29 09:50:08 +03:00
Martin Storsjö c11e33a3d9 hls: Set stream offset before opening a chained demuxer
This makes sure we don't accidentally check discard flags
for the wrong stream.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-07-29 09:50:06 +03:00
Michael Niedermayer cdd2d73d31 hls: Don't check discard flags until the parent demuxer's streams actually exist
If passing the end of one segment while initializing the
chained demuxer, the parent demuxer's streams aren't set up
yet, so we can't recheck the discard flags.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-07-29 09:50:05 +03:00
Michael Niedermayer 82bf8c8783 hls: Copy the time base from the chained demuxer
When a variant stream isn't mpegts but e.g. raw adts, the
default time base (90k) isn't correct.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-07-29 09:50:03 +03:00
Martin Storsjö eb33ba04e0 hls: Return all packets from a previous variant before moving on to the next one
This serves as a safeguard; normally we want to use the dts
comparison to interleave packets from all active variants. If that
dts comparison for some reason doesn't work as intended, make sure
that all packets in all variants for a certain sequence number have
been returned before moving on to the next one.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-07-29 09:50:02 +03:00
Michael Niedermayer c5b46a064d lavf: Don't interpret just slightly broken timestamps as wraparound
This avoids breaking some slightly incorrect (dts > pts) timestamps
in sample HLS streams from Apple.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-07-29 09:49:58 +03:00
Luca Barbato 2219e27b5b oma: correctly mark and decrypt partial packets
Incomplete crypted files would lead to a read after buffer boundary
otherwise.

Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
2013-07-28 22:54:55 +02:00
Luca Barbato 9d0b45ade8 oma: check geob tag boundary
Prevent read after buffer boundary on corrupted tag.

Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
2013-07-28 22:54:55 +02:00
Luca Barbato 4f03a77e52 oma: refactor seek function
Properly propagate seek errors from avio and the generic pcm seek.
2013-07-28 22:54:55 +02:00
Zhang Rui 0f51c398be http: Support reading gzip/deflate compressed data
Derived from VLC's http module.
Original authors:
  Antoine Cellerier <dionoea@videolan.org>
  Sébastien Escudier <sebastien-devel@celeos.eu>
  Rémi Duraffort <ivoire@videolan.org>
  Rémi Denis-Courmont <remi@remlab.net>
  Francois Cartegnie <fcvlcdev@free.fr>

Normally, http servers shouldn't send this to us since we
don't advertise it with an Accept-Encoding header, but some
servers still do it anyway.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-07-28 18:39:49 +03:00
Diego Biurrun 6ff15cd569 Remove unreachable returns 2013-07-27 16:00:41 +02:00
Luca Barbato c8f0b20b4a avidec: Let the inner dv demuxer take care of discarding
CC: libav-stable@libav.org
2013-07-27 16:00:11 +02:00
Luca Barbato 10aa44aa67 avidec: K&R formatting cosmetics
Signed-off-by: Diego Biurrun <diego@biurrun.de>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
2013-07-27 16:00:11 +02:00
Martin Storsjö 979e9e8f36 wtv: Drop some casts that now are unnecessary
Signed-off-by: Martin Storsjö <martin@martin.st>
2013-07-26 23:05:18 +03:00
Martin Storsjö 86f042dcab wtv: Make WTV_SECTOR_BITS a 64 bit constant
This makes sure that values that are left-shifted by this constant
end up casted to 64 bit before shifting, avoiding overflow if the
value ends up larger than 2 GB.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-07-26 23:05:14 +03:00
Hendrik Schreiber b512360184 avio: Don't set the seekable flag if no seek function is provided
Signed-off-by: Martin Storsjö <martin@martin.st>
2013-07-26 21:46:18 +03:00
Martin Storsjö 1297f7b87f rtpenc: Fix some odd comments
Some weird comments stem from the fact that the rtpdec and rtpenc
code was shared earlier.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-07-26 21:43:29 +03:00
Diego Biurrun 03039f4c8c miscellaneous typo fixes 2013-07-25 19:43:32 +02:00
Diego Biurrun 4a2ef39442 cosmetics: Add '0' to float constants ending in '.'. 2013-07-25 11:33:23 +02:00
Ed Torbett 36fb0d02a1 rtsp: Support multicast source filters (RFC 4570)
This supports inclusion of one single IP address for now,
at the media level. Specifying the filter at the session level
(instead of at the media level), multiple source addresses,
exclusion, or using FQDNs instead of plain IP addresses is not
supported (yet at least).

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-07-19 12:02:13 +03:00
Martin Storsjö 4d97ca040b rtpproto: Check the source IP if one single source has been specified
If another peer is sending unicast packets to the same port that
we are listening on, those packets can end up being received despite
using source specific multicast. For those cases, manually check the
source address of received packets against the intended source address.

This only handles the case when the source list is one single IP
address for now, which probably is the most common case.

Based on a patch by Ed Torbett.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-07-19 12:02:09 +03:00
Ed Torbett 336353deaa rtpproto: Support IGMPv3 source specific multicast inclusion
Blocking/exclusion is not supported yet.

The rtp protocol parameter takes the same form as the existing
sources parameter for the udp protocol.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-07-19 12:00:49 +03:00
Martin Storsjö e740929a07 lavf: Make sure avg_frame_rate can be calculated without integer overflow
If either of the deltas is too large for the multiplications to
succeed, don't use this for setting the avg frame rate.

Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Cc: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
2013-07-15 21:39:24 +03:00
Martin Storsjö 31931520df mov: Do not allow updating the time scale after it has been set
The time scale is set in mdhd, and later validated in the
enclosing trak atom once all of its children have been parsed.

A loose mdhd atom outside of a trak atom could update the time
scale of the last stream without any validation.

Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Cc: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
2013-07-15 21:38:58 +03:00
Martin Storsjö 5b4eb243bc mov: Seek back if overreading an individual atom
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Cc: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
2013-07-15 21:37:44 +03:00
Martin Storsjö 1dd1b2332e rtsp: Include an User-Agent header field in all requests
Some rtsp servers like the IP Cam IcyBox IB-CAM2002 need it.

Based on a patch by Carl Eugen Hoyos.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-07-14 20:53:04 +03:00