libssh provides a function for parsing ~/.ssh/config for ssh connection parameters like user, hostname, identity file
and port. This patch makes ffmpeg use this function to take parameters from the config file for everything that's not
explicitely set in the url. It also supports host aliases, i.e. using a shorthand in the url and replacing it with the
hostname / IP address specified for the shorthand in the config file.
Signed-off-by: Florian Jacob <projects+ffmpeg@florianjacob.de>
Signed-off-by: Lukasz Marek <lukasz.m.luki2@gmail.com>
* commit '448c8cfe4c53e9e806effd8505b46d57fa707061':
movenc: Support setting fragment_index before the moov atom is written
Conflicts:
libavformat/movenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '0c5e380c2c266d2e8a13c000cc527529db837f10':
movenc: Don't rely on the fragment index for vc1 info gathering
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'cf402d6fa88acd647cdff993429583bec8a34fdc':
rtpenc_mpegts: Set chain->rtp_ctx only after avformat_write_header succeeded
Reviewed-by: Andreas Cadhalpun <andreas.cadhalpun@googlemail.com>
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'c83dd2d2a458075a58895c384372f57c1ec26276':
rtpenc_mpegts: Free the right ->pb in the error path in the init function
Reviewed-by: Andreas Cadhalpun <andreas.cadhalpun@googlemail.com>
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This way, the caller doesn't need to coordinate setting the option
after the moov atom has been written. The downside is that it is
no longer possible to use the option for checking whether the moov
atom already has been written, but a caller is able to keep track
of that by other means anyway.
Signed-off-by: Martin Storsjö <martin@martin.st>
The previous use of the mov->fragments field, for determining whether
written packets were part of the first fragment or not, didn't
work as intended when using the empty_moov flag.
Signed-off-by: Martin Storsjö <martin@martin.st>
By making sure we at each time only have one pointer set, either a
local variable or one in the context, we avoid potential double frees
in the cleanup routines. If chain->rtp_ctx is set, it is closed by
calling avformat_write_trailer, but that shouldn't be called unless
avformat_write_header succeeded.
This issue was pointed out by Andreas Cadhalpun.
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit 'ad94c6ca0b86c463f476b26606259a2041dcddc9':
siff: Use the correct type for packet size variables
Conflicts:
libavformat/siff.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'bfeb83a8b7d3fcf09a54d8dbc9c521e10bb17530':
rtpdec_hevc: Drop extra sanity check for size of input packet
Merged-by: Michael Niedermayer <michaelni@gmx.at>
If resyncing leads to the same position as previously, it will again
lead to a resync attempt, resulting in an infinite loop.
Thus don't seek back beyond the last syncpoint.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
In this case len is always at least 3, since it is checked against
RTP_HEVC_PAYLOAD_HEADER_SIZE + 1 before entering the switch block.
Bug-Id: CID 1238784
A negative time base can trigger assertions.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '46d4d8575979a24a8d026d9805039b724e0e3e5f':
movenc: Avoid writing separate flags for the first sample if not necessary
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '00d751d4fc20ec88d2cc2c9f39ec8b9e9c8cdeba':
movenc: Set tfhd default sample flags based on actual samples, if possible
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This avoids assuming that e.g. audio samples are marked as
sync samples.
This allows omitting the sample flags from trun, if the default
flags happen to be right for all the samples.
Signed-off-by: Martin Storsjö <martin@martin.st>
If EOF is reached, while skipping bytes, avio_tell(pb) won't change
anymore, resulting in an infinite loop.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
They are used in a switch statement, but it is not guaranteed that the
COMM case (where they are set to 0) is reached before the other cases.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '6448f15af02f2c3cf0df8cb8237957e426041f2d':
mxfdec: Fix the error handling for when strftime fails
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Use correct context, reduce log level, don't assume it is a video stream,
and print the tag of the unknown stream.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
a876585215 had the unintended side effect of returning AVERROR(ENOMEM)
when track->entry is zero, while the code intentionally wants to
continue in that case.
Signed-off-by: Martin Storsjö <martin@martin.st>
The mov muxer already supports picking up extradata that wasn't
present during the avformat_write_header call - we just need to
propagate it. Since the dash muxer uses delay_moov, we have time
up until the first segment is written to get extradata filled in.
Also update the codec description string when the extradata becomes
available.
Signed-off-by: Martin Storsjö <martin@martin.st>
SAMPLE-AES encryption is not commonly used yet, but without this patch
ffmpeg is thinking that the hls segments are not encrypted which
produces broken files.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
It is used in adx_read_packet, which currently depends on the
decoder/parser setting this value between reading the file header and
demuxing the first packet.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
The chunk size is limited to UINT16_MAX (written by avio_wb16), so make
sure that the packet size is not too large.
Such large frames need to be split into slices smaller than 64 kB, but
that is currently supported neither by the rv10/rv20 encoders nor the rm
muxer.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
As this is depricated it should not be on by default, it is only
supported for MOV containers, depends on avpriv_get_gamma_from_trc()
Enable by:
-movflags +write_gama
This will use the color_trc to supply a gamma value, if desired an
explicit value may be supplied using the -mov_gamma option supplying
a suitable floating point value, values <=1e-6 will not be written.
Signed-off-by: Kevin Wheatley <kevin.j.wheatley@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
While the native jpeg2000 decoder can determine pixel format correctly
from the codestream, libopenjpeg wrapper cannot. To make sure that
the output is correct when using libopenjpeg to decode digital
cinema files, we do detection from the metadata included in the MXF
wrapper.
If the container has "JPEG 2000 Coding Parameters" metadata element
with Rsiz value set to one of digital cinema profiles, we can safely
assume that the given input file is DCI compliant, therefore the
pixel format should be XYZ.
Reviewed-by: Tomas Härdin <tomas.hardin@codemill.se>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'b72b212a4c43563f1b9fc3ce9a5ff91f89b857ac':
rdt: Use a separate variable to clarify the different value namespaces
Conflicts:
libavformat/rdt.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'c13ab42a0a3979b2c292b2315487f5f5e14ba26b':
rtpdec_qt: Use a local variable instead of RTP_FLAG_KEY
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The original flags variable contains rtpdec flags, while the
rmflags variable contains RM flag bits which have a completely
different definition.
Signed-off-by: Martin Storsjö <martin@martin.st>
The only case where RTP_FLAG_KEY actually is needed is
in RDT, where such a flag needs to be passed via the
rtpdec parse function's flags parameter.
Signed-off-by: Martin Storsjö <martin@martin.st>
Nothing in the framework nor in the rest of the depacketizer actually
uses this flag - the chained demuxer sets the keyframe flag properly on
demuxed packets already.
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit '25f613f8be3b51e4396b93cda131e4631ba54302':
dca: Move syncword definitions to a separate header
Conflicts:
libavcodec/dca_parser.c
libavformat/dtsdec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '1dc19729e92a96620000e09eba8e58cb458c9486':
rtpdec_asf: Don't free the payload context in the .close function
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This fixes an oversight in 96084251, in a refactoring done on top
of Gilles' original patch.
Pointed out by Gilles Chanteperdrix.
Signed-off-by: Martin Storsjö <martin@martin.st>
Fixes the duration field of the OpenDML master index "indx" chunk to
contain the number of samples instead of the number of packets for
(linear/PCM) audio streams.
This matches the OpenDML V1.02 standard text which states that the
duration field shall contain "time span in stream ticks".
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '0af3b65880573aa9b3375362eaab4f84140c7dde':
asf: do not export XMP metadata by default
Conflicts:
libavformat/asfdec.c
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Similarly to what has been done for MOV, display XMP metadata only when
users explicitly require it.
The Extensible Metadata Platform tag can contain various kind of data
which are not strictly related to the video file, such as history of
edits and saves from the project file.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
Next commit will revert the PTS seeking so this is not needed anymore
This reverts commit 38e641a060.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The chunk size is limited to 0xFFFF (written by avio_wb16), so make
sure that the packet size is not too large.
Such large frames need to be split into slices smaller than 64 kB, but
that is currently supported neither by the rv10/rv20 encoders nor the rm
muxer.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
See Ticket244
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This reverts commit 4abfa387b8.
This commit broke playback of fragmented mp4 files with b-frames.
While investigating this, it turned out that the general framework
isn't ready for a PTS-based index yet. Revert this change until
a better thought out solution is in place.
Signed-off-by: Martin Storsjö <martin@martin.st>
Commit 96084251e5 introduced a change in
the parser implementation which broke it. Restore the original
implementation.
Signed-off-by: Gilles Chanteperdrix <gilles.chanteperdrix@xenomai.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This reverts commit 26524e3581.
If we want the T.140 codec to have the AV_CODEC_ID_TEXT codec id,
its type needs to be AVMEDIA_TYPE_SUBTITLE, so, keep interpreting
the text media type as AVMEDIA_TYPE_SUBTITLE.
Signed-off-by: Gilles Chanteperdrix <gilles.chanteperdrix@xenomai.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This makes more sense than mapping to AV_CODEC_ID_SUBRIP. Nothing
indicates that a T.140 track contains subrip sub-titles.
Signed-off-by: Gilles Chanteperdrix <gilles.chanteperdrix@xenomai.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '4f6cd883f06f7893a2b60a41e7a4f8ae633dac2f':
rtpenc: Don't set max_frames_per_packet based on the packet frame size or frame rate
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'bde2bba45c2f2df27a8534028bda09a6e7f835e2':
rtpenc: Restructure if statements in packetizers to simplify adding more conditions
Conflicts:
libavformat/rtpenc_xiph.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'f8c01257f93ceda3e03bc4e540a51022d1e2bff2':
rtpenc: Always do the default initialization regardless of codecs
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'd16c8d28d4e2fca3af1054ffbf635c8cee755fc8':
rtpenc_aac: Use AV_WB16 instead of manual bitshifts
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '9c9b0218e85fcd969308632f75af48a4ce229541':
rtpenc_aac: Merge a definition with a declaration
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '1fc64e2e07787bbca82a72c146588e850e6d098a':
rtpenc: Write conditional statements on separate lines
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '0662440b991361fdb5e732712d997a73e4692e34':
rtpenc_aac: Set a default value for max_frames_per_packet at init
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '12b3459979f5ea6481660cd2c99a0381e2b5ba37':
rtpenc_amr: Use s->num_frames instead of s->buf_ptr - s->buf
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'b9d2d6843a49f9df1d1ae1afe817d9b48c445919':
tls: Pass AVOptions dictionaries through to the chained protocol
Conflicts:
libavformat/tls.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'e14f98c62fdf8744b07419314095d1b3248cce75':
tcp: Clarify the units for the timeout avoptions
Conflicts:
libavformat/tcp.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Instead check the timestamps while muxing, to avoid buffering a
too long timestamp range into one single packet.
This makes the AMR and AAC packetization slightly less efficient,
since we set a possibly unnecessarily high max_frames_per_packet.
(These packetizers end up doing a memmove of the TOC bytes if
sending a packet before max_frames_per_packet is achieved, and
we end up setting max_frames_per_packet to a value that should
be high enough for most uses.)
All packetizers that use max_frames_per_packet now set it either
to a default value, or to a value calculated based on other
parameters, so none of them rely on the previous default setting.
For iLBC, copy one frame at a time, to allow checking the timestamp
range for each of them - basically doing potentially multiple
loops to simplify the code instead of trying to calculate the
number of frames to buffer while honoring s1->max_delay.
This is in preparation for reducing the coupling between libavformat
and libavcodec, by not having the muxers use the encoder field
frame_size (which may not be available during e.g. stream copy).
Signed-off-by: Martin Storsjö <martin@martin.st>
Factorize out the s->num_frames check at the start of the if statements,
simplifying adding more alternative causes for sending the buffered
frames.
Signed-off-by: Martin Storsjö <martin@martin.st>
This avoids having to jump to the defaultcase in the switch. Manually
override the stream time base back to 90 kHz for the few audio codecs
that don't use the sample rate as time base (mp2, mp3).
Signed-off-by: Martin Storsjö <martin@martin.st>
This doesn't fix any bug, but makes the code simpler for later
patches, and more straightforward to read as is.
Signed-off-by: Martin Storsjö <martin@martin.st>
After sending a fragmented frame, len (s->buf_ptr - s->buf) isn't
zero, while s->num_frames is zero as intended. Using s->num_frames
makes it work as intended, and is less convoluted than keeping track
of (resetting) s->buf_ptr.
This avoids sending stray data after sending a fragmented aac packet.
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
Don't prefix them ffio_url, which is misleading, sounding too
much like the urlprotocol layer (like ffurl_*).
Signed-off-by: Martin Storsjö <martin@martin.st>
Other codecs/channel numbers are not supported by this muxer.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Ohter packet sizes are not supported by this muxer.
This avoids a null pointer dereference of pkt->data.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This avoids a null pointer dereference of pkt->data.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
It is used in adx_read_packet, which currently depends on the decoder/parser setting this value between reading the file header and demuxing the first packet.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This assumes CBR (which is true for all samples i have)
Previous version reviewed by: Oleksij Rempel <linux@rempel-privat.de>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Fixes out of array read
Fixes: asan_heap-oob_ae74b5_3610_cov_1739568095_test.3g2
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '8e32b1f0963d01d4f5d4803eb721f162e0d58d9a':
libavformat: Use ffio_free_dyn_buf where applicable
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '8a273a746061a112e5e35066a8fd8e146d821a62':
avio: Add an internal utility function for freeing dynamic buffers
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '078d43e23a7a3d64aafee8a58b380d3e139b3020':
rtpdec: Free depacketizers if the init function failed
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'bb4a310bb85f43e62240145a656b1e5285b14239':
rtpdec: Don't free the payload context in the .free function
Conflicts:
libavformat/rtpdec_latm.c
libavformat/rtpdec_mpeg4.c
libavformat/rtpdec_mpegts.c
libavformat/rtpdec_xiph.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '78791c086bcaf9eb084c27555b31fea8bbb7624a':
rtpdec: Use .init instead of .alloc to set default values
Conflicts:
libavformat/rdt.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '3567b91e49c6ae101c9a35c90f46b8ad9890ac15':
rtpdec_hevc: Share the implementation of fragmented packets with h264
Conflicts:
libavformat/rtpdec_h264.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'f3449062a8d100ac4f703647336c32b126aa99f1':
rtpdec_hevc: Reduce indentation level by returning early on errors
Conflicts:
libavformat/rtpdec_hevc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '8633fb47db2ec39eb8bd1bd65302af75a94ff5d0':
rtpdec_hevc: Share the implementation of parsing a=framesize with h264
Conflicts:
libavformat/rtpdec_h264.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '5956f489d0452ff6dea6b6b81b4fa8e596fc5684':
rtpdec_hevc: Add asterisks at the start of each long comment line
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '5d8cae45737bed6239bd6b6e0698802dbe1463c8':
rtpdec: Get rid of all trivial .alloc/.free functions
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'e72605f80bf5cbe32053a554ccc137e0a99cf3dd':
rtpdec: Allow allocating and freeing the private data without explicit functions
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'b7a4c319fda22aa91ce29692d728ec6103b514f6':
rtpdec: Allow setting the need_parsing field in RTPDynamicProtocolHandler
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'b651c9139e1ab222d5aab9151dcd7d6e40e49885':
rtpdec_mpa_robust: Move .enc_name to the start of the struct
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '353b492d0f2a21ae8eb829db1ac01b54b2a4d202':
rtpdec: Change enc_name to a pointer instead of a fixed-size buffer
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Many of these functions were named foo_free_context, and since
the functions no longer should free the context itself, only
allocated elements within it, the previous naming was slightly
misleading.
Signed-off-by: Martin Storsjö <martin@martin.st>
This is different from how it is handled in codecs/demuxers/muxers
though (where the close function isn't called if the open function
failed), but since the number of depacketizers that have an .init
function is quite limited, this is easy to change.
The main point is that if the init function failed, we shouldn't
try to use that depacketizer at all - this makes sure that the
parse function doesn't need to check for the things that were
initialized in the init function.
Signed-off-by: Martin Storsjö <martin@martin.st>
This makes it more consistent with depacketizers that don't have any
.free function at all, where the payload context is freed by the
surrounding framework. Always free the context in the surrounding
framework, having the individual depacketizers only free any data
they've specifically allocated themselves.
This is similar to how this works for demuxer/muxers/codecs - a
component shouldn't free the priv_data that the framework has
allocated for it.
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit '74d318f138f2a3f1b2fe81aea826d80d1e60f54c':
rtsp: Fix the indentation of a linewrapped statement
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '26524e358147aade6e9dd18fff42d61b966bbc70':
rtsp: Interpret the text media type as AVMEDIA_TYPE_DATA
See: afb0e5a810
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Using the copy codec ACLR atoms where incorrectly written
During the creation of the ACLR atom we are assuming the vos_data
contains the DNxHD header. This change makes this explicit and
ensures we don't over write the stream with the extra_data.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
They share a great deal of common structure; only a few minor
bits in the headers differ.
This also fixes an off-by-one in sending of the last fragment
of large HEVC nals (where it previously sent len+2 bytes, even
if it should have been len+RTP_HEVC_HEADERS_SIZE aka len+3).
Signed-off-by: Martin Storsjö <martin@martin.st>
This allows getting rid of quite a bit of boilerplate in depacketizers.
The default value (initializing need_parsing to 0, aka
AVSTREAM_PARSE_NONE) is the same as it is initialized to by default
in AVStream.
Signed-off-by: Martin Storsjö <martin@martin.st>
Fixes the following link error:
nutdec.c:(.text+0x2d47): undefined reference to `ff_codec_movvideo_tags'
isom.c:(.text+0x332): undefined reference to `avpriv_mpeg4audio_get_config'
isom.c:(.text+0x39e): undefined reference to `avpriv_mpa_freq_tab'
* commit 'cdcc370293a159c321e41af7f0eef141c62d698d':
rtsp: punch holes again after pause
See: 22bb5bd7a3
Merged-by: Michael Niedermayer <michaelni@gmx.at>
When a client behind a NAT issues a pause command, and stay paused for a
long time, the router may stop the RTP/RTCP port redirection. Resend the
hole punching packets before each PLAY command to cause the router to
restart the port redirection in that case.
Move the existing code for sending the packets from the SETUP phase
to the PLAY phase.
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit 'a388e72d1a6b0888cc1591cb699f61a9c1089cf4':
rtpenc_hevc: Aggregate multiple NAL units into one RTP packet, if possible
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'e2220e734f3d01145ef9aefbd7b6ff29a89ae159':
rtpenc_h264: Aggregate multiple NAL units into one RTP packet, if possible
Merged-by: Michael Niedermayer <michaelni@gmx.at>
When a client behind a NAT issues a pause command, and stay paused for a
long time, the router may stop the RTP/RTCP port redirection. Resend the
hole punching packets after each PLAY command to cause the router to
restart the port redirection in that case.
Signed-off-by: Gilles Chanteperdrix <gilles.chanteperdrix@xenomai.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'ff394ca087d41941d2157e7a4e356e3ad312494e':
rtpdec_h264: Add a missing closing paren in a log message
Merged-by: Michael Niedermayer <michaelni@gmx.at>
When receiving an RTCP packet, the difference between the last RTCP
timestamp and the base timestamp may be negative. As these timestamps
are of the uint32_t type, the result becomes a large integer. Cast
the difference to int32_t to avoid this issue.
The result of this issue is very large start times for RTSP
streams, and difficulty to restart correctly after a pause.
Signed-off-by: Gilles Chanteperdrix <gilles.chanteperdrix@xenomai.org>
Signed-off-by: Martin Storsjö <martin@martin.st>
When receiving an RTCP packet, the difference between the last RTCP
timestamp and the base timestamp may be negative. As these timestamps
are of the uint32_t type, the result becomes a large integer. Cast
the difference to int32_t to avoid this issue.
The result of this issue is very large start times for RTSP
streams, and difficulty to restart correctly after a pause.
Signed-off-by: Gilles Chanteperdrix <gilles.chanteperdrix@xenomai.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '10e2d8b5562d8729e4eefbcec63a11eb8a0c502c':
rtpdec_hevc: Use a shared function for parsing parameter sets
Merged-by: Michael Niedermayer <michaelni@gmx.at>
ffm encoder fails when codec is not found.
It may happen when stream is being copied.
This commit allows to store such stream and provides
backward compatibility with version prior 2.5 release.
fixes#4266
Signed-off-by: Lukasz Marek <lukasz.m.luki2@gmail.com>
