Fix suggested by Luca Barbato.
This was causing spurious EOFs when using -rtsp_transport udp, as
reported in https://bugzilla.libav.org/show_bug.cgi?id=1103
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Make easier to handle the polling function before we implement
full threading support.
(cherry picked from libav commit ca960161f0)
Signed-off-by: James Almer <jamrial@gmail.com>
Main use-case is proxying avio through a foreign I/O layer and a custom
AVIO context, without losing latency and performance characteristics.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Merged from Libav commit 173b56218f.
Main use-case is proxying avio through a foreign I/O layer and a custom
AVIO context, without losing latency and performance characteristics.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
This was introduced in bc2a32969e.
The whole block that the statement was added to is only
relevant when used as a demuxer, but the other statements
there have had other if statements guarding them. Make
sure to only run this whole block if being used as a
demuxer.
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
Introduced in 00e122bc / bc2a3296
The whole block that the statement was added to is only
relevant when used as a demuxer, but the other statements
there have had other if statements guarding them. Make
sure to only run this whole block if being used as a
demuxer.
Fixes ticket #5844.
Also set a default_whitelist for mmsh and ffrtmphttp.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* commit '3fdffc032e8ea5676bc0c2551b900c0dc887835b':
rtsp: Use avcodec_descriptor_get instead of avcodec_find_decoder
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
When feeding input RTP packets to the depacketizer via custom IO,
it needs to pick the right stream using the payload type for
RTP packets, and using the SSRC for RTCP packets. If the first
packet is an RTCP packet, we don't (currently) know the SSRC
yet and thus can't pick the right RTP depacketizer to handle it.
By parsing the SSRC attribute in the SDP, we can map initial
RTCP packets to the right stream.
Signed-off-by: Martin Storsjö <martin@martin.st>
Since all URLContexts have the same AVOptions, such AVOptions
will be applied on the outermost context only and removed from the
dict, while they probably make sense on all contexts.
This makes sure that rw_timeout gets propagated to the innermost
URLContext (to make sure it gets passed to the tcp protocol, when
opening a http connection for instance).
Alternatively, such matching options would be kept in the dict
and only removed after the ffurl_connect call.
Signed-off-by: Martin Storsjö <martin@martin.st>
We cannot play multiple multicast streams with the same port at the
same time. This is because both rtp and rtcp port are opened in
read-write mode, so they will not bind to the multicast address. Try
to make rtp port as read-only by default to solve this bug.
Signed-off-by: Zhao Zhili <wantlamy@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Currently, AVStream contains an embedded AVCodecContext instance, which
is used by demuxers to export stream parameters to the caller and by
muxers to receive stream parameters from the caller. It is also used
internally as the codec context that is passed to parsers.
In addition, it is also widely used by the callers as the decoding (when
demuxer) or encoding (when muxing) context, though this has been
officially discouraged since Libav 11.
There are multiple important problems with this approach:
- the fields in AVCodecContext are in general one of
* stream parameters
* codec options
* codec state
However, it's not clear which ones are which. It is consequently
unclear which fields are a demuxer allowed to set or a muxer allowed to
read. This leads to erratic behaviour depending on whether decoding or
encoding is being performed or not (and whether it uses the AVStream
embedded codec context).
- various synchronization issues arising from the fact that the same
context is used by several different APIs (muxers/demuxers,
parsers, bitstream filters and encoders/decoders) simultaneously, with
there being no clear rules for who can modify what and the different
processes being typically delayed with respect to each other.
- avformat_find_stream_info() making it necessary to support opening
and closing a single codec context multiple times, thus
complicating the semantics of freeing various allocated objects in the
codec context.
Those problems are resolved by replacing the AVStream embedded codec
context with a newly added AVCodecParameters instance, which stores only
the stream parameters exported by the demuxers or read by the muxers.
* commit 'e02dcdf6bb6835ef4b49986b85a67efcb3495a7f':
rtsp: Allow $ as interleaved packet indicator before a complete response header
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
Some RTSP servers ("HiIpcam/V100R003 VodServer/1.0.0") respond to
our keepalive GET_PARAMETER request by a truncated RTSP header
(lacking the final empty line to indicate a complete response
header). Prior to 764ec70149, this worked just fine since we
reacted to the $ as interleaved packet indicator anywhere.
Since $ is a valid character within the response header lines,
764ec70149 changed it to be ignored there. But to keep
compatibility with such broken servers, we need to at least
allow reacting to it at the start of lines.
Signed-off-by: Martin Storsjö <martin@martin.st>
packets are queued due to packet reordering until the queue reach its
maximal size or max delay is reached.
This commit adds a warning trace when max delay is reached.
Signed-off-by: Eloi BAIL <eloi.bail@savoirfairelinux.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit '764ec70149728be82304c163ccc4e280f1629201':
rtsp: Only interpret $ as interleaved packet indicator at the start of replies
Merged-by: Michael Niedermayer <michael@niedermayer.cc>
Allow $ as character anywhere within normal RTSP replies - both
within the lines, and as the first character of RTSP header lines.
(The existing old comment indicated that an inline packet could
start at any line within a RTSP reply header, but that doesn't
sound valid to me, and I'm not sure if the existing code
handled that correctly either.)
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit 'b90adb0aba073f9c1b4abca852119947393ced4c':
rtsp: Make sure we don't write too many transport entries into a fixed-size array
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This removes the error logging added in 4e54432164.
This avoids warnings about "Invalid interval start specification 'now'"
for live rtsp streams.
We only try to parse some of the many valid values for time ranges
in RTSP - the other ones are fully valid but not interesting for the
use case in rtsp.c, so we shouldn't warn about them.
(Parsing the time ranges is needed to allow seeking, but e.g. setting
the current realtime clock for the start time doesn't make sense.
av_parse_time has got a different mode for parsing absolute times
as well, which can handle the special case "now", but that doesn't
make much sense for this particular use in rtsp.c.)
Signed-off-by: Martin Storsjö <martin@martin.st>
