Experimental VP9 support was added to the muxer recently.
Reviewed-by: Ronald S. Bultje <rsbultje@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
This merges commits 8e2ea69135 and
096a8effa3 by Anton Khirnov, with the
following change:
- extract_extradata_check() is added to know if the codec is supported
by the bsf before trying to initialize it. This behaviour is similar to
the old AVCodecParser.split checks.
The FATE reference changes are due to the filtered out NAL units that
the old AVCodecParser.split implementation left alone.
Decoding is unchanged as the functions that parse extradata simply
ignored said unnecessary NAL units.
Signed-off-by: James Almer <jamrial@gmail.com>
This fixes a proble where ffmpeg would cause crash to do a seek when the network disconnect.
The log like this:
01-01 10:53:03.441 6580 6580 F DEBUG : backtrace:
01-01 10:53:03.441 6580 6580 F DEBUG : #00 pc 0002942e /system/lib/libavformat.so (ffurl_write+9)
Signed-off-by: tiejun.peng <tiejun.peng@foxmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This adds partial support for the RFC 4175 (raw video over RTP). The
only supported formats are the YCbCr-4:2:2 8 bit because it's natively
supported by FFmpeg with pixel format UYVY, and 10 bit which requires
the vrawdepay codec to convert the payload in a format handled by
FFmpeg.
Signed-off-by: Damien Riegel <damien.riegel@savoirfairelinux.com>
Reviewed-by: Rostislav Pehlivanov <atomnuker@gmail.com>
* commit '537b5b773b317af79d3a5b576ee9683e15ed84f6':
rtmpdh: Do global initialization before running the test
Merged-by: James Almer <jamrial@gmail.com>
This avoids an integer overflow
the solution matches oggparsevorbis.c and 45581ed15d
Fixes: 700242
Found-by: Thomas Guilbert <tguilbert@google.com>
Reviewed-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* commit '3cc3463f306f425f76bd962755df1132eeac6dfa':
avisynth: Support pix_fmts added to AviSynth+
This commit is mostly a noop, see
92916e8542.
Cosmetics and a small fix are merged.
Merged-by: Clément Bœsch <u@pkh.me>
Adding an MOV format option to turn on/off the editlist supporting code, introduced in ca6cae73db
Signed-off-by: Sasi Inguva <isasi@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This is more robust in case some change or corner case causes them to be
dereferenced before being set
Fixes CID1396274, CID1396275
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
refer to SPEC:
Annex E. The FLV File Format said:
E.3 TheFLVFileBody have a table:
Field Type Comment
PreviousTagSize0 UI32 Always 0
Reviewed-by: Bela Bodecs <bodecsb@vivanet.hu>
Reviewed-by: Hendrik Leppkes <h.leppkes@gmail.com>
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
* commit 'c541a44e029e8a4f21db028c34fee3ad1c10a409':
Revert "rtmpproto: Don't include a client version in the unencrypted C1 handshake"
Merged-by: Clément Bœsch <u@pkh.me>
The old "API" that signaled rotation as a metadata value has been
replaced by DISPLAYMATRIX side data quite a while ago.
There is no reason to make muxers/demuxers/API users support both. In
addition, the metadata API is dangerous, as user tags could "leak" into
it, creating unintended features or bugs.
ffmpeg CLI has to be updated to use the new API. In particular, we must
not allow to leak the "rotate" tag into the muxer. Some muxers will
catch this properly (like mov), but others (like mkv) can add it as
generic tag. Note applications, which use libavformat and assume the
old rotate API, will interpret such "rotate" user tags as rotate
metadata (which it is not), and incorrectly rotate the video.
The ffmpeg/ffplay tools drop the use of the old API for muxing and
demuxing, as all muxers/demuxers support the new API. This will mean
that the tools will not mistakenly interpret per-track "rotate" user
tags as rotate metadata. It will _not_ be treated as regression.
Unfortunately, hacks have been added, that allow the user to override
rotation by setting metadata explicitly, e.g. via
-metadata:s:v:0 rotate=0
See references to trac #4560. fate-filter-meta-4560-rotate0 tests this.
It's easier to adjust the hack for supporting it than arguing for its
removal, so ffmpeg CLI now explicitly catches this case, and essentially
replaces the "rotate" value with a display matrix side data. (It would
be easier for both user and implementation to create an explicit option
for rotation.)
When the code under FF_API_OLD_ROTATE_API is disabled, one FATE
reference file has to be updated (because "rotate" is not exported
anymore).
Tested-by: Michael Niedermayer <michael@niedermayer.cc>
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
* commit 'ee050797664c7c74cae262ffab05006b55d47a11':
openssl: Support version 1.1.0.
This commit is mostly a noop, see 798c6ecce5
Included the simplifications by Martin Storsjö and fixed the
GET_BIO_DATA() macro to prevent a warning after the simplifications.
Merged-by: Clément Bœsch <u@pkh.me>
* commit '4b07ebf1eb13561492f7e3c30a67f34415016b3e':
mov: Update colr values
Mostly noop, see a3cab3d433
Only the use of av_color_{primaries,transfer,space}_name() is merged.
Merged-by: Clément Bœsch <u@pkh.me>
This reverts commit 1c193ac1f9, reversing
changes made to 7ebc9f8df4.
Several FATE tests started failing after this merge, so it's reverted
until it can be properly fixed.
* commit '8e2ea691351c5079cdab245ff7bfa5c0f3e3bfe4':
lavf: use the new bitstream filter for extracting extradata
Merged-by: James Almer <jamrial@gmail.com>
* commit 'c359d624d3efc3fd1d83210d78c4152bd329b765':
hevcdec: move decoder-independent declarations into a separate header
Merged-by: James Almer <jamrial@gmail.com>
* commit '7d8d726be7dc46343ab1c98c339c1ed44bcb07c1':
rtmpproto: Don't include a client version in the unencrypted C1 handshake
Merged-by: Clément Bœsch <u@pkh.me>
* commit '9f23f77a532ca9c2b7dc4b5328bc413e4f6f5b56':
rtmpproto: Don't include the libavformat version as "clientid"
Merged-by: Clément Bœsch <u@pkh.me>
* commit 'bad4aad4037f59ba0ad656164be9ab8f7a0fa2d4':
avidec: Do not special case palette on big-endian
This commit is a noop, see 64cafe340b
Merged-by: Clément Bœsch <u@pkh.me>
* commit '8ea35af7620e4f73f9e8c072e1c0fac9a04ec161':
avio: add a new flag for marking streams seekable by timestamp
Merged-by: James Almer <jamrial@gmail.com>
This patch deprecates anything that has to do with merging/splitting
side data. Automatic side data merging (and splitting), as well as all
API symbols involved in it, are removed completely.
Two FF_API_ defines are dedicated to deprecating API symbols related to
this: FF_API_MERGE_SD_API removes av_packet_split/merge_side_data in
libavcodec, and FF_API_LAVF_KEEPSIDE_FLAG deprecates
AVFMT_FLAG_KEEP_SIDE_DATA in libavformat.
Since it was claimed that changing the default from merging side data to
not doing it is an ABI change, there are two additional FF_API_ defines,
which stop using the side data merging/splitting by default (and remove
any code in avformat/avcodec doing this): FF_API_MERGE_SD in libavcodec,
and FF_API_LAVF_MERGE_SD in libavformat.
It is very much intended that FF_API_MERGE_SD and FF_API_LAVF_MERGE_SD
are quickly defined to 0 in the next ABI bump, while the API symbols are
retained for a longer time for the sake of compatibility.
AVFMT_FLAG_KEEP_SIDE_DATA will (very much intentionally) do nothing for
most of the time it will still be defined. Keep in mind that no code
exists that actually tries to unset this flag for any reason, nor does
such code need to exist. Code setting this flag explicitly will work as
before. Thus it's ok for AVFMT_FLAG_KEEP_SIDE_DATA to do nothing once
side data merging has been removed from libavformat.
In order to avoid that anyone in the future does this incorrectly, here
is a small guide how to update the internal code on bumps:
- next ABI bump (probably soon):
- define FF_API_LAVF_MERGE_SD to 0, and remove all code covered by it
- define FF_API_MERGE_SD to 0, and remove all code covered by it
- next API bump (typically two years in the future or so):
- define FF_API_LAVF_KEEPSIDE_FLAG to 0, and remove all code covered
by it
- define FF_API_MERGE_SD_API to 0, and remove all code covered by it
This forces anyone who actually wants packet side data to temporarily
use deprecated API to get it all. If you ask me, this is batshit fucked
up crazy, but it's how we roll. Making AVFMT_FLAG_KEEP_SIDE_DATA to be
set by default was rejected as an ABI change, so I'm going all the way
to get rid of this once and for all.
Reviewed-by: James Almer <jamrial@gmail.com>
Reviewed-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
These values are defined to be 32bit in the specification,
so it makes more sense to store them as fixed width.
Based on a patch by Micahel Niedermayer <michael@niedermayer.cc>.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
* commit '0638b99cdba52554691fc668d9e477bc184c7a33':
aiff: Skip padding byte for odd-sized chunks
Also removes to odd-size checks from get_aiff_header and get_meta to use
the generic path introduced by the original commit.
Merged-by: Matthieu Bouron <matthieu.bouron@gmail.com>
Preparation for potentially disabling merged side data by default in the
libs. Do this in particular because it affects fate tests.
The changed tests either reflect added packet side data, or the changed
packet size due to merged side data removal reducing the packet size.
The current form of the messages indicating matches in the white
or black lists seems to be a bit too much relying on context.
Make the messages more explicit.
Signed-off-by: Alexander Strasser <eclipse7@gmx.net>
the SECOND_LEVEL* flags process and name is too long
extract all of them output to funtions, make code clear
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
Provides a way to change bandwidth parameter inside DASH manifest after a non-CBR H.264 encoding.
Caller now is able to compute the bitrate by itself, after all packets have been written, and then set that value in AVFormatContext->streams->codecpar->bit_rate before calling av_write_trailer. As a result that value will be set in DASH manifest.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Ever since the codecpar changes, this has been always printed when
opening a flv file. This is because the codecpar changes made all
streams to be added lazily as read_packet is called.
Public fields were added after the private fields (negating the entire
point of this). New private fields go into AVStreamInternal anyway.
The new marker was set by guessing which fields are supposed to be
private and wshich not. recommended_encoder_configuration is accessed by
ffserver_config.c directly, and is supposed to use the public API.
ffmpeg.c accesses AVStream.cur_dts, even though it's a private field,
but that seems to be an older error.
Allow all struct fields to be accessed directly, as long as they're
public.
Before this change, many fields were "public", but could be accessed via
AVOption only. This meant they were effectively not public, but were
present for documentation purposes, which was incredibly confusing at
best.
MSVC doesn't support the %s time format, and instead of returning an
error the invalid parameter handler is invoked which (by default)
terminates the process.
Reviewed-by：Steven Liu <lq@chinaffmpeg.org>
Signed-off-by: Hendrik Leppkes <h.leppkes@gmail.com>
refer to ticket id: #6170
rename file from temp to origin name after complete current segment
Reviewed-by: Aman Gupta <ffmpeg@tmm1.net>
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
Signed-off-by: Nicolas Roy-Renaud <nicolas.roy-renaud.1@ens.etsmtl.ca>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This dts value can end up in the list in the absence of durations and is in that
case semantically identical to AV_NOPTS_VALUE. We can alternatively prevent
storing RELATIVE_TS_BASE if there is no duration.
Fixes Ticket3640
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes Ticket 6018
This fixes a regression, and allows playback of files containing mpeg4video that are otherwise
not supported
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
When the http method is not set, the method will use POST for ts,
PUT for m3u8, it is not unify, now set it unify.
This ticket id: #5315
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Reviewed-by: Moritz Barsnick <barsnick@gmx.net>
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
Skips using temporary files when outputting to a protocol other than
"file", which enables dash to output content over network
protocols. The logic has been copied from the HLS format.
Reviewed-by: Steven Liu <lingjiujianke@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This commit optimizes HTTP performance by reducing forward seeks, instead
favoring a read-ahead and discard on the current connection (referred to
as a short seek) for seeks that are within a TCP window's worth of data.
This improves performance because with TCP flow control, a window's worth
of data will be in the local socket buffer already or in-flight from the
sender once congestion control on the sender is fully utilizing the window.
Note: this approach doesn't attempt to differentiate from a newly opened
connection which may not be fully utilizing the window due to congestion
control vs one that is. The receiver can't get at this information, so we
assume worst case; that full window is in use (we did advertise it after all)
and that data could be in-flight
The previous behavior of closing the connection, then opening a new
with a new HTTP range value results in a massive amounts of discarded
and re-sent data when large TCP windows are used. This has been observed
on MacOS/iOS which starts with an initial window of 256KB and grows up to
1MB depending on the bandwidth-product delay.
When seeking within a window's worth of data and we close the connection,
then open a new one within the same window's worth of data, we discard
from the current offset till the end of the window. Then on the new
connection the server ends up re-sending the previous data from new
offset till the end of old window.
Example (assumes full window utilization):
TCP window size: 64KB
Position: 32KB
Forward seek position: 40KB
* (Next window)
32KB |--------------| 96KB |---------------| 160KB
*
40KB |---------------| 104KB
Re-sent amount: 96KB - 40KB = 56KB
For a real world test example, I have MP4 file of ~25MB, which ffplay
only reads ~16MB and performs 177 seeks. With current ffmpeg, this results
in 177 HTTP GETs and ~73MB worth of TCP data communication. With this
patch, ffmpeg issues 4 HTTP GETs and 3 seeks for a total of ~22MB of TCP data
communication.
To support this feature, the short seek logic in avio_seek() has been
extended to call a function to get the short seek threshold value. This
callback has been plumbed to the URLProtocol structure, which now has
infrastructure in HTTP and TCP to get the underlying receiver window size
via SO_RCVBUF. If the underlying URL and protocol don't support returning
a short seek threshold, the default s->short_seek_threshold is used
This feature has been tested on Windows 7 and MacOS/iOS. Windows support
is slightly complicated by the fact that when TCP window auto-tuning is
enabled, SO_RCVBUF doesn't report the real window size, but it does if
SO_RCVBUF was manually set (disabling auto-tuning). So we can only use
this optimization on Windows in the later case
Signed-off-by: Joel Cunningham <joel.cunningham@me.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Reported-by: SleepProgger <security@gnutp.com>
Reviewed-by: Steven Liu <lingjiujianke@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
hls-encoder currenlty does not provide stream level metadata to mpegts
muxer. This patch fixes track #3848 bug.
Signed-off-by: Bela Bodecs <bodecsb@vivanet.hu>
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
This enables having multiple tracks of the same type which would
be treated as different things by the media server (as opposed to
different bit rate versions of the same track). According to the
smooth streaming specification, just setting the systemLanguage
tag is not enough to note that a track with the same attributes
differs from another one.
Reviewed-by: Martin
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
When user use the hls_wrap, there have many problem:
1. some platform refersh the old but usefull segment
2. CDN(Content Delivery Network) Deliver HLS not friendly
The hls_wrap is used to wrap segments for use little space,
now user can use hls_list_size and hls_flags delete_segments
instead it.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Reviewed-by: Carl Eugen Hoyos <ceffmpeg@gmail.com>
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
This way it's clear the size field accounts for the footer length plus every
tag entry, but not the header.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
The size field in the header/footer accounts for the entire APE tag
structure except the 32 bytes from header, for compatibility with
APEv1.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
According to the spec[1], a value of 0 means the footer is present and a value
of 1 means it's absent, the exact opposite of header presence flag where 1
means present and 0 absent.
The reason for this is compatibility with APEv1 tags, where there's no header,
footer presence was mandatory for all files, and the flags field was a zeroed
reserved field.
[1] http://wiki.hydrogenaud.io/index.php?title=Ape_Tags_Flags
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Core of patch is from paul@paulmehta.com
Reference https://crbug.com/643952 (senc,saiz portions)
Signed-off-by: Matt Wolenetz <wolenetz@chromium.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Core of patch is from paul@paulmehta.com
Reference https://crbug.com/643952 (udta_string portion)
Signed-off-by: Matt Wolenetz <wolenetz@chromium.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Core of patch is from paul@paulmehta.com
Reference https://crbug.com/643951
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Check value reduced as the code does not support values beyond INT_MAX
Also the check is moved to a more common place and before integer truncation
Core of patch is from paul@paulmehta.com
Reference https://crbug.com/643950
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Check value reduced as the code does not support larger lengths
Adds a `-hls_flags +temp_file` which will write segment data to
filename.tmp, and then rename to filename when the segment is complete.
This patch is similar in spirit to one used in Plex's ffmpeg fork, and
allows a transcoding webserver to ensure incomplete segment files are
never served up accidentally.
Reviewed-by: Hendrik Leppkes <h.leppkes@gmail.com>
Reviewed-by: Bodecs Bela <bodecsb@vivanet.hu>
Signed-off-by: Aman Gupta <aman@tmm1.net>
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
Blocks are marked as key frames whenever the "reference" field is
zero. This breaks for non-keyframe Blocks with a reference timestamp
of zero.
The likelihood of reference timestamp being zero is increased by a
longstanding bug in muxing that encodes reference timestamp as the
absolute time of the referenced frame (rather than relative to the
current Block timestamp, as described in MKV spec).
Now using INT64_MIN to denote "no reference".
Reported to chromium at http://crbug.com/497889 (contains sample)
Not starting a new segment if the elapsed microsecs since the start of the day
equals the the elapsed microsecs since the start of the day at the time of the
last cut seems plain wrong to me, Deti do you remember the original reason
behind this check?
Signed-off-by: Marton Balint <cus@passwd.hu>
* commit '7f549b8338ed3775fec4bf10421ff5744e5866dd':
riff: don't overwrite bps from WAVEFORMATEX if EXTENSIBLE doesn't contain that data.
Only cosmetics, the change was already present.
Merged-by: Clément Bœsch <cboesch@gopro.com>
* commit '90bc423212396e96a02edc1118982ab7f7766a63':
mov: Wrap stsc index and count compare in a separate function
The mov_stsc_index_valid() function is replaced with a macro to prevent
signdness issues (index is not always signed, and count is always
unsigned currently).
The comparison is also adjusted to reduce the risk of overflows.
Merged-by: Clément Bœsch <u@pkh.me>
Retain the ranges of frame indexes when applying edit list in
mov_fix_index. The index ranges are then used to keep track of the frame
index of the current sample. In case of a discontinuity in frame indexes
due to edit, update the auxiliary info position accordingly.
Reviewed-by: Sasi Inguva <isasi@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Allows the user to reserve space for the ODML master index. A sufficient
sized master index in the AVI header avoids storing follow-up master
indexes within the 'movi' data later. If the option is omitted or zero
the index size is estimated from output duration and bitrate.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Tobias Rapp <t.rapp@noa-archive.com>
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
The current implementation creates new segments comparing
pkt->pts - first_pts > nb_segs * min_seg_duration
This works fine, but if the keyframe interval is smaller than "min_seg_duration"
segments shorter than the minimum segment duration are created.
Example: keyint=50, min_seg_duration=3000000
segment 1 contains keyframe 1 (duration=2s < total_duration=3s)
and keyframe 2 (duration=4s >= total_duration=3s)
segment 2 contains keyframe 3 (duration=6s >= total_duration=6s)
segment 3 contains keyframe 4 (duration=8s < total_duration=9s)
and keyframe 5 (duration=10s >= total_duration=9s)
...
Segment 2 is only 2s long, shorter than min_seg_duration = 3s.
To fix this, new segments are created based on the actual written duration.
Otherwise the option name "min_seg_duration" is misleading.
Signed-off-by: Peter Große <pegro@friiks.de>
Signed-off-by: Martin Storsjö <martin@martin.st>
If set, adds a UTCTiming tag in the manifest.
This is part of the recommendations listed in the "Guidelines for
Implementations: DASH-IF Interoperability Points" [1][2]
Section 4.7 describes means for the Availability Time Synchronization.
A usable default is "https://time.akamai.com/?iso"
[1] http://dashif.org/guidelines/
[2] http://dashif.org/wp-content/uploads/2016/12/DASH-IF-IOP-v4.0-clean.pdf
(current version as of writing)
Signed-off-by: Peter Große <pegro@friiks.de>
Signed-off-by: Martin Storsjö <martin@martin.st>
Codec 4 (frame size 98) uses joint stereo per spec and examples.
Also removed an incorrect "align" var which wasn't used anyway (it was overwrittern).
Probably all/only .AT3 of frame size 98 are JS, too.
Signed-off-by: bnnm <bananaman255@gmail.com>
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Servers seem to be happy to receive the wrapped-around value as long
as they receive a report, otherwise they timeout.
Initially reported and analyzed by Thomas Bernhard.
When detecting a swapped AC3 marker the data of the frame is swapped. However, in subsequent frames the data swapped is taken from the first frame rather than the current frame.
Signed-off-by: Marijn Meijles <marijn@bitpit.net>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
to avoid rebuffering on the clientside for difficult network conditions.
Signed-off-by: Anton Schubert <ischluff@mailbox.org>
Signed-off-by: Martin Storsjö <martin@martin.st>
Appends Z to timestamp to force ISO8601 datetime parsing as UTC.
Without Z, some browsers (Chrome) interpret the timestamp as
localtime and others (Firefox) interpret it as UTC.
Signed-off-by: Anton Schubert <ischluff@mailbox.org>
Signed-off-by: Martin Storsjö <martin@martin.st>
From e24d95c0e06a878d401ee34fd6742fcaddeeb95f Mon Sep 17 00:00:00 2001
From: Joel Cunningham <joel.cunningham@me.com>
Date: Mon, 9 Jan 2017 13:37:51 -0600
Subject: [PATCH] tcp: set socket buffer sizes before listen/connect/accept
Attempting to set SO_RCVBUF and SO_SNDBUF on TCP sockets after connection
establishment is incorrect and some stacks ignore the set call on the socket at
this point. This has been observed on MacOS/iOS. Windows 7 has some peculiar
behavior where setting SO_RCVBUF after applies only if the buffer is increasing
from the default while decreases are ignored. This is possibly how the incorrect
usage has gone unnoticed
Unix Network Programming Vol. 1: The Sockets Networking API (3rd edition, seciton 7.5):
"When setting the size of the TCP socket receive buffer, the ordering of the
function calls is important. This is because of TCP's window scale option,
which is exchanged with the peer on SYN segments when the connection is
established. For a client, this means the SO_RCVBUF socket option must be
set before calling connect. For a server, this means the socket option must
be set for the listening socket before calling listen. Setting this option
for the connected socket will have no effect whatsoever on the possible window
scale option because accept does not return with the connected socket until
TCP's three-way handshake is complete. This is why the option must be set on
the listening socket. (The sizes of the socket buffers are always inherited from
the listening socket by the newly created connected socket)"
Signed-off-by: Joel Cunningham <joel.cunningham@me.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
When bytes_read overflowed, last_bytes_read did not yet overflow
and no bytes-read report was created leading to a timeout.
Analyzed-by: Thomas Bernhard
Fixes ticket #5836.
If fifo is enabled on tee muxer, ffmpeg exits because of an unknown option passed to fifo muxer.
Option name "format_options" was replaced by "format_opts" on tee muxer.
Signed-off-by: Felipe Astroza <felipe@astroza.cl>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This happens because segment_end() returns an error, so seg_write_packet
never proceeds to segment_start(), and seg->avf->pb is never re-set,
so we crash with a null pb when av_write_trailer flushes the packet
queue.
This doesn't seem to be clearly recoverable, so I'm just failing more
gracefully.
Repro:
ffmpeg -i input.ts -f segment -c copy -segment_list /noaxx.m3u8 test-%05d.ts
(assuming you don't have write access to /)
When use http method to delete the old segments,
there is only io_open, hove not io_close yet,
this patch is used to fix it
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
