Prevent possible memory leaks.
Connect to nginx and request a non-existent resource to
trigger the issue.
CC: libav-stable@libav.org
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Uwe L. Korn <uwelk@xhochy.com>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Some RTMP commands need the most recent timestamp as their parameter, so
keep track of it. This must be the most recent one and not e.g. the max
received timestamp as it can decrease again through seeking.
Signed-off-by: Martin Storsjö <martin@martin.st>
This fixes the abnormally high ts overhead in the files produced by the
HLS and segments muxers. See https://trac.ffmpeg.org/ticket/2857 . For
example makes it much more likely that it can produces streams that fit
under the 64kb App store limit.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Some encoders do not use syncsafe sizes in v2.4 id3 tags. Check the next
tag to try to choose between the two.
Fixes ticket #4003
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '324b23dde1bc8638959eb32419c95a93906db272':
rtmpproto: Add function to read a number response
Merged-by: Michael Niedermayer <michaelni@gmx.at>
In (non-live) streams with no metadata, the duration of a stream can
be retrieved by calling the RTMP function getStreamLength with the
playpath. The server will return a positive duration upon the request if
the duration is known, otherwise either no response or a duration of 0
will be returned.
Signed-off-by: Martin Storsjö <martin@martin.st>
Packets that contain a number as a result to a rtmp function call are
structured the same way (String, Number, Null, Number). This new method
also includes more bounds checks to better handle packets that are not
structured as expected.
Signed-off-by: Martin Storsjö <martin@martin.st>
Add the judement after create a new program to avoid segment fault.
Signed-off-by: Di Wu <di1028.wu@samsung.com>
Reviewed-by: Marton Balint <cus@passwd.hu>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'ced7238cd01cc2199acf9225305628641a27c1d7':
rtpdec_hevc: Use av_realloc instead of av_malloc+memcpy
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '74b02377980321934e33969c84733ace7e9f4eeb':
mov: Correctly check the color transfer characteristics range
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'e5cfc8fdad901c9487fe896421972852f38bcf5':
sdp: Provide out of bound parameter sets for HEVC if extradata is set
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '9b7f932ee568cadfc0f556a061fcc00cb63f9780':
rtpdec_hevc: Parse out of band vps/sps/pps/sei from fmtp lines
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '2d6e58497e76836604364b037df9b00ba3d75b69':
lavf: switch to AVCodecContext.framerate for demuxing
Conflicts:
libavformat/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
These allow getting the absolute start timestamp of a fragment
without reading preceding timestamps. This fixes sync between
tracks if starting from fragments in different streams that don't
align exactly.
This also is a prerequisite for producing DASH content.
Signed-off-by: Martin Storsjö <martin@martin.st>
E-AC-3 samples should contain 6 audio blocks, so concatenate syncframes
in order to achieve this.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This introduces a new option to the mov demuxer: -use_mfra_for
(pts|dts). When it's given and moofs and a MFRA are present, the MFRA's
TFRAs are read for fragment start times.
Unfortunately some programs that produce fragmented mp4s use the TFRA
time field for dts and some for pts. There is no realistic way to detect
which is the case, hence the responsibility is punted onto the user.
This also means that no behavioural change is enabled by default - you
must pass either dts or pts for anything to happen.
Without this change, timestamps for some discontinuous fragmented mp4 are
wrong, and cause audio/video desync and are not usable for generating
HLS.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'e44ee1eb8db7393e9d43207c2e1812720e292e6d':
movenc: Simplify code by using an existing local pointer
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'dad12ce452a9d69c0d9d53c375003947d5f1b02e':
movenc: Print a warning for an unhandled case of nonzero start dts with empty_moov
Conflicts:
libavformat/movenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'dbb472cb2f2f799295a12b4922a6a8be2cccfdee':
movenc: Write edit lists for fragmented files as well, if necessary
Conflicts:
libavformat/movenc.c
The default for writing EDTS for fragmented mp4 is left at disabled
this can be overridden via command line with -use_editlist
but EDTS + fragments still does not fully work, which is why it is
left disabled by default
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '95ee4e2ce774e0339632d067161596bf3dadfc72':
movenc: Add some comments explaining subtle details in writing the edit lists
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '72f801619a1ae91969fee9a7d72519422433c998':
movenc: Adjust edit lists to trim out parts of tracks with negative pts
Conflicts:
libavformat/movenc.c
See: 66b45d8f7a
See: 14fd34d73b
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '8bef43388132b53f59a6e90add18900a3bb4cc60':
smoothstreamingenc: Simplify code by removing a redundant variable
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Icecast uses HTTP 1.0 while Libav uses HTTP 1.1 and enables by
default chunked post.
Icecast actually forwards the HTTP chunk headers to the listener
as part of the media stream (without the chunk encoding HTTP headers)
causing the players to lose sync.
Disabling the option is enough to feed icecast properly.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
This is necessary to get the right timestamp offset for content
that starts with dts != 0.
This currently only helps when writing fragmented files with a non-empty
moov atom. When writing an empty moov atom, we don't have any packets
yet, so we don't know the starting dts for the tracks.
Signed-off-by: Martin Storsjö <martin@martin.st>
This makes sure that audio preroll for e.g. AAC is signaled correctly.
Previously we only wrote the edit list correctly if we had negative
dts but started with pts == 0 (e.g. for video with B-frames).
Signed-off-by: Martin Storsjö <martin@martin.st>
In these cases, only drop dts. Because if we drop both we have no
timestamps at all for some files.
This improves playback of HLS streams from GoPro cameras.
Signed-off-by: Martin Storsjö <martin@martin.st>
Support only one independent substream right now, and only syncframes
containing 6 blocks.
Fixes part of ticket #3074
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Trying to write to a stream id larger the the maximum requested is
a programming error, still there is no reason to leave a
reachable abort() in the codebase.
CC: libav-stable@libav.org
Fix basename computation code to allow just file names without any
directories in the path.
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
In dahsmode Matroska is not writing the first Cluster for every
audio stream in the Cues element.
Signed-off-by: Frank Galligan <frankgalligan@gmail.com>
Reviewed-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
syncframes in E-AC-3 can be combined to provide 6 audio blocks per
sample, thus requiring parsing for proper decoding.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
