* commit 'b8d2630c5327d2818d05c8a48be0417905d8e0fd':
dashenc: Reduce the segment duration if cutting out parts with edit lists
Merged-by: Michael Niedermayer <michaelni@gmx.at>
id should be an integer, not a string. It is also optional, so use
contentType instead which is the proper attribute for these values.
This addresses ticket #4545, fixing an MPD validation error.
Signed-off-by: James Almer <jamrial@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '254f3daba4271c1918d9a7ad155b1442ef93ed29':
nut: Make sure to clean up on read_header failure
Conflicts:
libavformat/nutdec.c
See: 361702660d
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This makes sure that the time + duration of the first segment
matches the start time of the next segment for e.g. AAC audio
with encoder delay.
Signed-off-by: Martin Storsjö <martin@martin.st>
id should be an integer, not a string. It is also optional, so use
contentType instead which is the proper attribute for these values.
This fixes an MPD validation error.
Signed-off-by: James Almer <jamrial@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
This currently works for most users because
avformat_open_input sets it, but this patch fixes any
applications not using that function.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
APIC tags always have a description. Tag writers obviously leave it
empty if there is no description. In this case, libavformat would export
"" as title. Do not set the title instead.
APIC tags always have a description. Tag writers obviously leave it
empty if there is no description. In this case, libavformat would export
"" as title. Do not set the title instead.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '7b734ee55dbb8476d7ad63c7daf55c534cf82d5d':
lavf: Open PICT images with Quickdraw
Conflicts:
tests/fate/image.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
If max in clean_index is set to a negative ast->sample_size, the
following loop never ends:
while (max < 1024)
max += max;
Thus set ast->sample_size to 0 if it would otherwise be negative.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
If bit_rate is negative, it can trigger an av_assert2 in av_rescale_rnd.
Since av_rescale returns int64_t, but st->codec_bit_rate is int, it can
also overflow into a negative value.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This fixes a segmentation fault.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
index_scale is set to matroska->time_scale of type uint64_t.
When index_scale is int, the assignment can overflow and e.g. result
in index_scale = 0. This causes a floating point exception due to the
division by index_scale.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Fix a duplicate memory allocation. priv_data should be allocated
in line 64 call to avformat_alloc_output_context2 since we pass
the correct AVFormat to it. This removes the duplicate allocation.
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Use dyn_duf to write chunks so that we create the actual chunk
file only after the entire chunk data is available. This will help
not confuse other software looking at the chunk file (e.g. a web
server) by seeing a zero length file when ffmpeg is writing into
it.
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Fixes Detecting AAC with such descriptor if the parts needed for detection
are later in the stream
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '82de8d71118f4eafd6a43e9ea9169bd411793798':
mpegts: Update the PSI/SI table only if the version change
Conflicts:
libavformat/mpegts.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The existing check has two problems:
1) i + count can overflow, so that the check '< 256' returns true.
2) In the (i == 'N') case occurs a j-- so that the loop runs once more.
This can trigger the assertion 'nut->header_len[0] == 0' or cause
segmentation faults or infinite hangs.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
If a PAT is finished while a PMT section filter is opened but
not yet finished, the PMT section filter is closed and all
the received data is discarded.
This is usually not an issue but some multiplexers (With very
quick PAT/PMT repetition settings) consistently emit a PMT
section start, then a PAT, and then the rest of the PMT,
causing the aforementioned behavior to result in no PMT being
finished.
In the most pathologic situation the stream information are lost
and the probe fallback miscategorizes subtitles as mp3 audio.
Avoid the issue through eliminating redundant PSI/SI table
updates by checking their version field, which is required by
the standard to be incremented on every change no matter how
minor.
CC: libav-stable@libav.org
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
A negative frame rate triggers an av_assert2 in av_rescale_rnd.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Check extended sync word for 16-bit LE and BE core streams to reduce
probability of alias sync detection. Previously sync word extension was
checked only for 14-bit streams.
This follows up the similar change in avcodec/dca_parser.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Remove the direct profile from UTCTiming element. Per DASH spec,
direct profile value should be the time at which the request was
made to the server and not the time at which the manifest was
written. So ffmpeg cannot write this value. This patch removes
the direct profile and write the UTCTiming element with the http
profile only if a URL is passed as a parameter. Update the fate
test to reflect this change.
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Fix a bug where the chunk muxer doesn't write the very first audio
packet (with pts == 0).
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* 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>
If the file size is much larger than what is indicated in the XING
header, the demuxer assumes it's a concatenated file, and throws away
the (presumably) incorrect duration information. Unfortunately, this
also triggers if the id3 tags are very large (embedded pictures and
such). Then the half-baked heuristic not only breaks the duration
display, but also gapless audio.
Fix it by subtracting the size of the headers (the check is off by some
bytes, but that doesn't matter at all). Note that there could be an
arbitrary amount of tags _after_ the mp3 data, but hopefully these are
not too large to trigger the heuristic in practice.
Also add a warning when this happens.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
While I'm not sure why exactly sure why the old code could end up in the
wrong position, using the generic index code is much simpler and is
known to work correctly.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This fixes the calculation of the number of needed blocks to make
sure that ALL pixels are represented by the result.
Signed-off-by: Martin Storsjö <martin@martin.st>
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>
This fixes a NULL pointer dereference if vst->duration is 0.
The problem was introduced in commit 0588acaf.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
It's the most useful mode, because it seeks accurately, and does not
break features like gapless audio.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
"-usetoc 2" now invokes the generic seek and indexing mode. This mode
skips data until the seek target is reached, and this is exact. It also
makes gapless audio actually work if a seek past the start of the file
is involved.
Change the fate-gapless-mp3 test to use the new mode, and move the old
one to fate-gapless-mp3-toc (since the test forces use of the Xing TOC).
The new mode has a different result for the seek - this result is
actually correct.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The mp2 seek test results change. Whether to skip samples if the file
had no LAME gapless tags was inconsistent. When seeking to the start
of the file, 529 samples were skipped, but when playing from start,
nothing was skipped. This commit changes the behavior on seek to skip
nothing.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Some players do not support setting minimumUpdatePeriod to zero.
This patch adds a new parameter that will let the users set any
value to this field. Also updates the test and the documentation.
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
DASH spec requires the presence of either duration of the period
or the minimumUpdatePeriod element. This patch adds the
minimumUpdatePeriod element hardcoded with the value 0 as the
manifest will never be updated for WebM DASH Live streams. Also
updating the fate test reference file.
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Using the PRIu8 format specifier to print an enum value causes a
compiler warning, so use %d instead.
Fixes ticket #4467.
Signed-off-by: Chris Watkins <watk@chromium.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
With this the returned timestamp should match the packet instead of
the requested timestamp, which may lay between packets
Reviewed-by: wm4 <nfxjfg@googlemail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'd34039b171bebe37bf723a1b03e5651267099739':
rmenc: Drop the temporary buffer for ac3 byteswap
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '2cc3936599b6fc63143036659653d1be0624360f':
dashenc: Add a publishTime field in dynamic manifests
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '9286de045968ad456d4e752651eec22de5e89060':
mov: Double-check that alias path is not an absolute path
Conflicts:
libavformat/mov.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
nlvl_to and nlvl_from can be set to 1 if both alias and target files
are in the same directory, so actually check the first character of the
string. We can do this because MacOS filepaths (alis type 2) are always
converted to UNIX filepaths (alis type 18).
Absolute paths can be stored in alis type 2 and 18 according to my research:
the first is the canonical MacOS filepath, with path level separated by
colons, and the volume name within the filepath, while the second should be the
absolute filesystem path from the mount point.
This avoids waiting for a count to increase which will always be 0 and may
reduce the startup delay for affected streams (rare)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Apparently, some live streams can delete segments too early, maybe
because the client is too far behind. In this case, it's better to skip
the segment, instead of returning EOF. (Yes, the HLS demuxer actually
returns AVERROR_EOF if opening the segment returns a 404 HTTP error.)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '247aa7af7d8197247c181e3fbfe8d93d75e41b29':
avisynth: Simplify shared library name construction
Conflicts:
libavformat/avisynth.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Removing a bunch of questionable hacks makes it work. These hacks
apparently try to make concatenated mp3s with Lame headers seekable,
which doesn't make too much sense anyway. The main change is that we
trust the Xing header file size field now (the same field is used for
seeking with Xing TOC). Note that a mp3 might contain an unknown number
of unsupported additional tags, so we can't reliably compute this size
manually.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Return appropriate error codes and propagate the error codes from
helper functions to the outer calls. Also fix a potential leak in
call to av_realloc.
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
VP80 fourcc are writed for all contexts (without ctx->codec_tag)
how to reproduce the issue:
1) Get any vp9 video (for example http://base-n.de/webm/out9.webm)
2) ffmpeg -i out9.webm -vcodec copy out9.ivf
3) out9.ivf have VP80 fourcc at ivf header
The proposed fix solves this issue
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Replace an unchecked av_malloc call with stack allocation as the size
is always a constant.
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
In order to safely exit when the user tries to use AviSynth 2.5, the
continue_on_fail value for 2.6's functions need to be set to 1.
Otherwise, the library loader fails before the 'upgrade to 2.6'
log message appears.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
txoffer (e.g. http://tori.aoi-chan.com/ ) redirects to the same URI on your
first request, and serves the actual file on the second. It's stupid, but AFAIK
technically compliant. We'd previously see the server not handing back a Range
header and return an error; now, instead, we see that there's a redirect and
keep track of the offset we want while trying again at the new URL.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
With this patch http can be used to listen for POST data to be used as an input stream.
Signed-off-by: Stephan Holljes <klaxa1337@googlemail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This fixes the calculation of the number of needed blocks to make
sure that ALL pixels are represented by the result.
Reviewed-by: Thomas Volkert <silvo@gmx.net>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'be089af38f65dc8b1fe3564f98020fc815577edb':
mov: Rely on box type rather than file type for colr atom
Conflicts:
libavformat/mov.c
See: 0276b95242
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Although it's not allowed to use only allows 'nclc' in ISOM files, there
are samples that do not always respect this rule. This change prevents
atom overread and a spurious color range initialization.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
MicroDVD has a "hack" for specifying the video framerate the subtitle
was authored against. The demuxer reads this hint correctly, but didn't
skip it correctly.
This was not noticed, because the exported packet has its duration set
to 0, making it invisible (depending on the API user's rendering logic).
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Generally, libavformat exports cover art pictures as video streams with
1 packet and AV_DISPOSITION_ATTACHED_PIC set. Only matroskadec exported
it as attachment with codec_id set to AV_CODEC_ID_MJPEG.
Obviously, this should be consistent, so change the Matroska demuxer to
export a AV_DISPOSITION_ATTACHED_PIC pseudo video stream.
Matroska muxing is probably incorrect too. I know that it can create
broken files with an audio track and just 1 video frame when e.g.
remuxing mp3 with APIC to mkv. But for now this commit does not change
anything about muxing, and also continues to write attachments with
AV_CODEC_ID_MJPEG should the muxer application have special knowledge
that the Matroska is broken in this way.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
MSVC does not support the %F and %T format specifiers in strftime.
Replace that with the expanded version. This fixes the broken fate
tests in MSVC (webm-dash-manifest-*).
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This patch adds support for WebM Live Muxing by adding a new WebM
Chunk muxer. It writes out live WebM Chunks which can be used for
playback using Live DASH Clients.
Please see muxers.texi for sample usage.
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This patch adds support for creating DASH manifests for WebM Live
Streams. It also updates the documentation and adds a fate test to
verify the behavior of the new muxer flag.
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This patch adds support for parsing live files (produced by
-f webm_chunk) which contains only the headers but no packets. This
is only used when using -f webm_dash_manifest. There will be a
follow up patch which adds live support to WebM DASH Manifest
muxer.
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Generally, libavformat exports cover art pictures as video streams with
1 packet and AV_DISPOSITION_ATTACHED_PIC set. Only matroskadec exported
it as attachment with codec_id set to AV_CODEC_ID_MJPEG.
Obviously, this should be consistent, so change the Matroska demuxer to
export a AV_DISPOSITION_ATTACHED_PIC pseudo video stream.
Matroska muxing is probably incorrect too. I know that it can create
broken files with an audio track and just 1 video frame when e.g.
remuxing mp3 with APIC to mkv. But for now this commit does not change
anything about muxing, and also continues to write attachments with
AV_CODEC_ID_MJPEG should the muxer application have special knowledge
that the Matroska is broken in this way.
Fixes trac #4423.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '7d097a0fc57f0fa8385962a539c657c2f40b5ed0':
mpegtsenc: Take max_delay into account when buffering multiple audio packets into one PES packet
Conflicts:
libavformat/mpegtsenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
I found another MXF File containing ProRes with the following
codec_uls: 060E2B34040101010E04020102110500
Therefor I relaxed the pattern.
Related to issue #4349
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Make sure we don't buffer up more than max_delay worth of data
before writing a PES packet, even if pes_payload_size is set to
a larger value.
Signed-off-by: Martin Storsjö <martin@martin.st>
The AVSC_API changes in the new headers mean that the 2.6 alphas
are just as incompatible as 2.5 is.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
In order to safely exit when the user tries to use AviSynth 2.5,
the continue_on_fail value for 2.6's functions need to be set to
1. Otherwise, the library loader fails before the 'upgrade to
2.6' log message appears.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Additionally, update some documentation with support for APNG
Signed-off-by: Donny Yang <work@kota.moe>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '8c9c5479c4ba729b4ba868ab541a90b2061a7c2f':
rtp: Add an option to set the send/receive buffer size
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '3c47e7c4350f73fc77d8e76f0dd6d2946b13c5cc':
rtp: Map the urloptions to AVOptions
Conflicts:
libavformat/rtpproto.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '4978850ca2cb1ec6908f5bc79cc592ca454d11e8':
build: Split JPEG-related tables off into a separate component
Conflicts:
configure
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Previously unset, and some software mishandles files if it is absent
Signed-off-by: Tim Nicholson <tim.nicholson@bbc.co.uk>
Reviewed-by: tomas.hardin@codemill.se
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This prevents sub-muxers from trying to seek back to the beginning of the
whole stream, only to find themselves overwriting some video data in the
current (often last) segment.
We only do this when not writing individual header/trailers.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This permits some interesting segmenting techniques with formats like Matroska,
where you can concatenate the header and segments [N, nb_segments) and get
a working file that starts at segment N's start time.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
API allows protocol implementations to provide API that
allows to list directory content.
API is similar to POSIX opendir/readdir/closedir.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
If the user attempts to use AviSynth 2.5, an error message will
now tell them they need to upgrade.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Fixes ticket #4387.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Note, Vittorio Giovara had submitted a very similar fix to me privately
a few hours before this, iam applying Jochens because it comes with a
commit message too and i had not yet applied Vittorios, but For sake
of credit, Vittorio independently solved this first
* commit 'a8c99205ca8703bd849efae13fcf844315c7147d':
avisynth: Fix compilation against current 2.6 header(s).
Merged-by: Michael Niedermayer <michaelni@gmx.at>
AviSynth 2.6 (and by extension, AviSynth+) moves these functions
into AVSC_API. This requires both adjusting their normal use,
and for AvxSynth, adjusting the position/use of the USING_AVISYNTH
ifdefs.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
* commit 'e4fe535d12f4f30df2dd672e30304af112a5a827':
mov: Write the display matrix in order
Conflicts:
libavformat/mov.c
libavutil/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This will allow to copy the matrix as is and it is just cleaner to keep
the matrix in the same order specified by the mov standard (which is
also explicitly described in the documentation).
In order to preserve compatibility, flip the angle sign in the display API
av_display_rotation_set() and av_display_rotation_get(), and improve the
documentation mentioning the rotation direction.
* commit 'e0046bc9c96150fa06146ace9093f06857dd7b23':
movenc: Write the make and model metadata keys for mov style files
Conflicts:
libavformat/movenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
These are essential allowing QuickTime to keep detecting content
as slow-motion - this allows preserving them on stream copy.
Signed-off-by: Martin Storsjö <martin@martin.st>
This commit has no known use case ATM as there are no unsupported video codecs in flv and could
theoretically be use to generate broken files allthough that would be not entirely easy as
tags/codecs still get sanity checked
This reverts commit 76f4b11780.
This allows stream copying video codecs before they are explicitly
supported. The same feature was in the past useful for audio codecs
in flv
This partly reverts the changes from 735ab7c5e0
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '67a2912307c3c08f2725ccae162cfe3426b80184':
movenc: Don't assume that fragment durations in pts is equal to duration in dts
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '5c337353a2546416631a87de4881850d99141c39':
movenc: Move sidx edit list timestamp adjustment into a block
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'eee13d653953083553cceadbbedf6222ef78a006':
movenc: Only adjust the cts offset at the start of fragments if necessary
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'b81b0cc22b22413760423e239ea644c9afdbfa2d':
movenc: Set the last packet duration based on the next packet when autoflushing
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '3041183677bda0a431b36e96a2c76617abaa8183':
dashenc: Heuristically fill in the duration of packets that need it
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '2889c5e16711770437f380f1bead5f72c6a0b17a':
movenc: Heuristically set the duration of the last sample in a fragment if not set
Merged-by: Michael Niedermayer <michaelni@gmx.at>
For strict CFR, they should be pretty much equal, but if the stream
is VFR, there can be a sometimes significant difference.
Calculate the pts duration separately, used in sidx atoms and for
tfrf/tfxd boxes in smooth streaming ismv files.
Also make sure to reduce the duration of sidx entries according to
edit lists.
Signed-off-by: Martin Storsjö <martin@martin.st>
Adjusting it is only necessary when a sidx/tfrf/tfxd atom already has
been written for the previous fragment (since the sidx/tfrf/tfxd atoms
include the duration between the first pts of the previous fragment, to
the first pts of the new fragment).
Signed-off-by: Martin Storsjö <martin@martin.st>
When automatically flushing fragments based on set conditions
(fragmentation on keyframes, after some interval or byte size),
we already have the next packet for one stream - use this for setting
the duration of the last packet in the flushed fragment correctly.
This avoids having to adjust the timestamp of the first packet in
the new fragment since the last duration was unknown.
Unfortunately, this only works for automatic flushing (not for
caller-triggered flushing, like in the dash muxer), and only for the
one single track that triggered the flushing. The duration of the
last sample in all other tracks still is dependent on AVPacket
duration (or heuristics).
Signed-off-by: Martin Storsjö <martin@martin.st>
This avoids that the mp4 muxer does a similar heuristic, adjusting
the timestamps in a way that the dash muxer doesn't know the actual
timestamps written to the file in the end. By making sure that the
mp4 muxer internal heuristic isn't applied, we know the exact
timestamps written to file, so that the timestamps in manifest match
the files.
Signed-off-by: Martin Storsjö <martin@martin.st>
Even if this is a guess, it is way better than writing a zero duration
of the last sample in a fragment (because if the duration is zero,
the first sample of the next fragment will have the same timestamp
as the last sample in the previous one).
Since we normally don't require libavformat muxer users to set
the duration field in AVPacket, we probably can't strictly require
it here either, so don't log this as a strict warning, only as info.
Signed-off-by: Martin Storsjö <martin@martin.st>
Add a missing AVClass member, check whether localaddr is null.
(Previously, localaddr was always a local stack buffer, while it
now also can be an avoption string which can be null.)
This fixes crashes when not passing any localaddr parameter, since
66028b7ba.
Signed-off-by: Martin Storsjö <martin@martin.st>
This fixes a crash, when trying to mux h264 into mxf_opatom.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Previous version reviewed-by: tomas.hardin@codemill.se
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* cehoyos/master:
lavf/mkv: Ignore ff_isom_write_hvcc() return value as the mov muxer does.
lavc/dts: Do not set bitrate for DTS-HD Master and High Resolution.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The current behavior may produce a different sequence of packets
after seeking, compared to demuxing linearly from the beginning.
This is because the MOV demuxer seeks in each stream individually,
based on timestamp, which may set each stream at a slightly different
position than if the file would have been read sequentially.
This makes implementing certain operations, such as segmenting,
quite hard, and slower than need be.
Therefore, add an option which retains the same packet sequence
after seeking, as when a file is demuxed linearly.
The current behavior may produce a different sequence of packets
after seeking, compared to demuxing linearly from the beginning.
This is because the MOV demuxer seeks in each stream individually,
based on timestamp, which may set each stream at a slightly different
position than if the file would have been read sequentially.
This makes implementing certain operations, such as segmenting,
quite hard, and slower than need be.
Therefore, add an option which retains the same packet sequence
after seeking, as when a file is demuxed linearly.
* commit '4326bc364b58e97fc3d89417054a6b7610179a00':
lavf: Do not list mov-only codecs in riff tags
Conflicts:
libavformat/Makefile
libavformat/avidec.c
See: 2e0b5f5c90
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Reviewed-by: tomas.hardin@codemill.se
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Set this field to TRUE if the audio component is to operate on
little-endian data, and FALSE otherwise.
However TRUE and FALSE are not defined. Since this flag is just a boolean,
interpret all values except for 0 as little endian.
Sample-Id: 64bit_FLOAT_Little_Endian.mov
Instead check for all mov code-points when demuxing avi
and print a warning if a video codec is found like this.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
This is incompatible with the omit_tfhd_offset flag (writing
position independent fragments with interleaving requires the
default_base_moof flag).
This makes the moof atoms slightly bigger, but can be better for
playback (improving locality of sample data in the mdat).
Signed-off-by: Martin Storsjö <martin@martin.st>
This is needed if all the data for one track isn't continuous
within the mdat. Normally we make sure all the data for one
track is continuous, but in new cases we will need to have
the samples interleaved.
Signed-off-by: Martin Storsjö <martin@martin.st>
as this kind of allows to circumvent it to some extend.
We also could add a separate parameter or value to choose this
Found-by: ramiro
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Instead check for all mov code-points when demuxing avi
and print a warning if a video codec is found like this.
Fixes a regression similar to the one described in ticket #4307.
* commit 'bacc92b59bfa5d6a1f631e63e46fc1d2fb934e51':
rtpdec_vp9: Drop extra sanity check for size of input packet
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '9272c965d9559a90ee64d46aebd99c117e07f7a3':
matroskaenc: Fix type used for chapter timestamps
Conflicts:
libavformat/matroskaenc.c
See: a4cd057bc7
Merged-by: Michael Niedermayer <michaelni@gmx.at>
In this case len is always at least 1, since it is checked against
RTP_VP9_DESC_REQUIRED_SIZE + 1 and then it is reduced by
RTP_VP9_DESC_REQUIRED_SIZE before entering the has_pic_id check.
Bug-Id: CID 1270811
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>
