Based on a suggestion by Martin Panter. This is more descriptive,
since it's the actual timestamp field from the RTMP packet,
which might or might not be a delta depending on context (in
some packets it's a delta, in some packets it's an absolute
timestamp, and in some packets it's 0xffffff to indicate that
the actual delta or absolute timestamp is transmitted separately).
Signed-off-by: Martin Storsjö <martin@martin.st>
Related fix in "rtmpdump":
https://repo.or.cz/w/rtmpdump.git/commitdiff/79459a2
Adobe's RTMP specification (21 Dec 2012), section 5.3.1.3 ("Extended
Timestamp"), says "this field is present in Type 3 chunks". Type 3 chunks are
those with the one-byte header size.
This resolves intermittent hangs and segfaults caused by the read function,
and also includes an untested fix for the write function.
The read function was tested with ABC (Australia) News 24 streams, however
they are probably restricted to only Australian internet addresses. Some of
the packets at the start of these streams seem to contain junk timestamp
fields, often requiring the extended field. Test command:
avplay rtmp://cp81899.live.edgefcs.net/live/news24-med@28772
Signed-off-by: Martin Storsjö <martin@martin.st>
Get the last partition offset and use it when footer partition
offset is missing.
Footer partition may not be present and even if present footer
partition offset may not be set in any partition except last one.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Extrapolate audio timestamps based on the number of samples demuxed.
Deal with some MXF nastiness involving fractional number of
samples per EditUnit when seeking (the specs handwave this away).
Further fixes from Tomas Härdin.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
We cannot easily determine if an mpeg TS's packet size is DVHS, FEC
or so on, for that we need to expose the internal raw_packet_size
field.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Since 2007, the Xiph.org Foundation recommends that .ogg only be used
for Ogg Vorbis audio files.
Source: http://wiki.xiph.org/index.php/MIME_Types_and_File_Extensions
However we only do it if we have libvorbis available because the
built in vorbis encoder is not as good.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
Currently ff_interleave_packet_per_dts() waits until it gets a frame for
each stream before outputting packets in interleaved order.
Sparse streams (i.e. streams with much fewer packets than the other
streams, like subtitles or audio with DTX) tend to add up latency and in
specific cases end up allocating a large amount of memory.
Emit the top packet from the packet_buffer if it has a time delta
larger than a specified threshold.
Original report of the issue and initial proposed solution by
mus.svz@gmail.com.
Bug-id: 31
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This fixes playback of mp3 streams in rtp/asf. This used to work
until c6f1dc8, but mostly by coincidence.
Signed-off-by: Martin Storsjö <martin@martin.st>
The normal differential timestamps can't handle negative
differences, thus send a full packet header with an absolute
timestamp in these cases.
Signed-off-by: Martin Storsjö <martin@martin.st>
If the url ends with .flv, we stripped it but didn't initialize
rt->playpath, doing av_strlcat on an uninitialized buffer.
Signed-off-by: Martin Storsjö <martin@martin.st>
The track duration is often not reliable or is not the duration
represented by the number of frames. In those cases, avg_frame_rate
was reported incorrectly. Removing this code falls back to the
default calculation in avformat_find_stream_info().
This is a partial revert of commit c3aeaa540.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Reviewed-by: Stephen Hutchinson <qyot27@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
It could probably also be considered an error if the pointer isn't
null at this point, but then we might risk rejecting some
slightly broken files that we might have handled so far.
Sample-Id: 00000496-google
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
These arrays are normally freed at the end of mov_read_trak,
but make sure they're freed in case mov_read_trak returned
early (due to errors) or in case the atoms that allocate arrays
are encountered at some other point than within a trak (which
we don't have checks against).
Sample-Id: 00000496-google
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
Such files have IndexTableSegments which when parsed cover EditUnit
ranges like this:
[0,1)
[249,250)
[249,377)
[0,249)
where each interval is
[IndexStartPosition, IndexStartPosition + IndexDuration)
This would be reduced to a sparse index like:
[0,1), [249,250)
instead of the full range:
[0,249), [249,377)
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
The Omnia A/XE encoder writes the explicit extra data incorrectly
and wrongly disables parametric stereo. Truncating the extra data
by setting the size to 2 works around this. The AAC extra data
parser will then only parse the correct parts.
Bug-id: 599
The code cannot handle there being none, but that should not happen for
valid files.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC:libav-stable@libav.org
This avoids a memory leak (or having to worry about freeing the
config string) if the colorspace isn't accepted.
Signed-off-by: Martin Storsjö <martin@martin.st>
Some ACTi cameras fail if "*" is passed as the URI.
Signed-off-by: Ismael Luceno <ismael.luceno@corp.bluecherry.net>
Signed-off-by: Martin Storsjö <martin@martin.st>
Directly loads AviSynth through LoadLibrary instead of relying on
Video for Windows, and supports using AvxSynth (via dlopen) to
open scripts on Linux and OS X.
Error messages from AviSynth/AvxSynth are now reported through
av_log and exit, rather than the traditional behavior of generating
an error video that the user would need to watch to diagnose.
The main rewrite was authored by d s <avxsynth.testing@gmail.com>
from the AvxSynth team, with additional contributions by
Oka Motofumi <chikuzen.mo@gmail.com>
Stephen Hutchinson <qyot27@gmail.com>
Diego Biurrun <diego@biurrun.de>
Anton Khirnov <anton@khirnov.net>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This allows both the main playlist itself as well as the variant
playlists to handle redirects combined with relative URLs.
Signed-off-by: Martin Storsjö <martin@martin.st>
It might be passed to code requiring padding, such as lzo decompression.
Fixes invalid reads.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC:libav-stable@libav.org
This is necessary to avoid target config settings bleeding into the host
compilation process with hardcoded tables and the DV VLC tables no longer
present as static tables in a header file.
Generate extradata with SPS/PPS based on container dimensions.
Authors of this commit are: Reimar and Thomas Mundt
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Fixes audio packet pts values in some files generated by AVID TRMG 3.01.
Signed-off-by: Marton Balint <cus@passwd.hu>
Reviewed-by: Tomas Härdin <tomas.hardin@codemill.se>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Imporoves detection of some files in the wild:
- ID3v2 a.k.a. "ea3" header is optional.
- Version and flags in ID3v2 header are unspecified.
Signed-off-by: David Goldwich <david.goldwich@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This simplifies proper error handling in rtsp.c/rtspdec.c. When
broadcasting over RTSP in TCP mode, the AVIOContext is closed and
recreated for each sent packet, and if the recreation fails, we might
try to close a NULL buffer when freeing things at the end.
Previously, if recreating the buffer in rtspdec.c failed, this would
crash later due to trying to close a NULL buffer.
Signed-off-by: Martin Storsjö <martin@martin.st>
This was added in 9b07a2dc02 as an ABI hack to allow older
code built with lavf 52 to register protocols even if the size
of the URLProtocol struct was increased. Later, registering
protocols from outside of lavf was removed and this workaround
isn't needed any longer since lavf 53.
This removes an unchecked malloc and a memory leak for the cases
when this workaround actually was used - which it hasn't since
lavf 53.
Signed-off-by: Martin Storsjö <martin@martin.st>
Also typedef the private data struct and make its name consistent with
the rest of Libav.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
avconv abuses the API by accessing AVStream.parser (which is private).
Removing AVStream.reference_dts in
2ba68dd044 breaks ABI compatibility for an
old avconv using a newer lavf. Fix this by adding a dummy field until
the next bump.
F4V is Adobe's mp4/iso media variant, with the most significant
addition/change being supporting other flash codecs than just
aac/h264.
Signed-off-by: Martin Storsjö <martin@martin.st>
Use an helper function to seek by sector to avoid possible mistakes
due shifting by WTV_SECTOR_BITS a 32bit integer.
Contrary to common intuition, a 32 bit integer left shifted
by a 64 bit integer does not promote the 32 bit integer to
64 bit before shifting.
This makes sure we don't send the Except: 100-continue header
if no authentication credentials have been provided.
Signed-off-by: Martin Storsjö <martin@martin.st>
Normally, all channel ids are between 0 and 10, while they in
uncommon cases can have values up to 64k.
This avoids allocating two arrays for up to 64k entries (at a total
of over 6 MB in size) each when most of them aren't used at all.
Signed-off-by: Martin Storsjö <martin@martin.st>
Inspired by a patch by Jakob van Bethlehem. But instead of doing
an empty POST first to trigger the WWW-Authenticate header (which
would succeed if no auth actually was required), add an Expect:
100-continue header, which is meant to be used exactly for
cases like this.
The header is added if doing a post, and the user has specified
authentication but we don't know the auth method yet.
Not all common HTTP servers support the Expect: 100-continue header,
though, so we only try to use it when it really is needed. The user
can request it to be added for other POST requests as well via
an option - which would allow the caller to know immediately that
the POST has failed (e.g. if no auth was provided but the server
required it, or if the target URL simply doesn't exist).
This is only done for write mode posts (e.g. posts without pre-set
post_data) - for posts with pre-set data, we can just redo the post
if it failed due to 401.
Signed-off-by: Martin Storsjö <martin@martin.st>
The default is to autodetect the auth method. This does require one
extra request (and also closing and reopening the http connection).
For some cases such as HTTP POST, the autodetection is not handled
properly (yet).
No option is added for digest, since this method requires getting
nonce parameters from the server first and can't be used straight
away like Basic.
Signed-off-by: Martin Storsjö <martin@martin.st>
The plain VP6 format is vertically flipped compared to VP6F/VP6A.
Support for the plain VP6 format was added in 09d8c0ae83 (which
also introduced support for muxing VP6F properly in general).
Signed-off-by: Martin Storsjö <martin@martin.st>
This wasn't an issue prior to 58404738, when the whole RTMP packet
was copied at once and the length of the individual embedded flv
packets only were validated by the flv demuxer.
Prior to this patch, this could lead to reads and writes out of bound.
Signed-off-by: Martin Storsjö <martin@martin.st>
If the embedded flv packets were incomplete and we aborted the
copying loop early, make sure the flv buffer is trimmed to
only contain full packets.
Signed-off-by: Martin Storsjö <martin@martin.st>
update_offset is also called from handle_metadata, where the
packet header sizes is already included in the size.
Previously this lead to flv_data/flv_size including 15 uninitialized
bytes at the end after each call to handle_metadata, making the
flv demuxer lose sync with the stream.
Also remove leftover copying in handle_metadata. This is a leftover
from the refactoring in 5840473. (Previously this final mempcy was
the one that copied all the packets at once, while this is done
within the loop right now.) After making sure flv_size is set to
the right size, this write was out of bounds.
Signed-off-by: Martin Storsjö <martin@martin.st>
This was overlooked in d872fb0f7 since I assumed all the realloc
issues in the rtmp code was fixed already.
Signed-off-by: Martin Storsjö <martin@martin.st>
The current magic numbers passed are values in seconds, while the
parameter itself should be passed over the wire in milliseconds.
This makes (some/all?) live streams from Red5 work correctly, that
previously returned StreamNotFound even with "-rtmp_live live". After
this commit, the default 'any' also works on these streams.
Signed-off-by: Martin Storsjö <martin@martin.st>
On (certain streams/setups at least on) Red5, the description string
actually is present, but empty. Therefore, first try loading the
description, but if not found or empty, load the code string instead.
The code string is quite understandable in most cases anyway (like
"NetStream.Play.StreamNotFound").
Signed-off-by: Martin Storsjö <martin@martin.st>
This makes sure errors in setting stream parameters are passed
on to the caller. This avoids successfully opening files while
some parameters aren't filled in properly.
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
Fixes sync in some samples (e.g. bugs 7581 and 8374 in VLC).
Based on a commit by Matthieu Bouron <matthieu.bouron@gmail.com>
Reported-by: Jean-Baptiste Kempf <jb@videolan.org>
CC: libav-stable@libav.org
This avoids setting a negative number of frames, ending up with a
negative average frame rate.
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
If a zero-length video packet is to be returned, just return
AVERROR(EAGAIN) and switch back to the audio stream.
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
This avoids a division by zero for G726.
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
This avoids a division by zero.
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
Even if the sample rate is valid, an invalid bitrate could
pass the mode combination test below.
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
This avoids divisions by zero later (and possibly assertions in
time base scaling), since an invalid rate_flag combined with an
invalid bitrate below could pass the mode combination test.
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
When av_reallocp fails, the associated variables that keep track of
the number of elements in the array (and in some cases, the
separate number of allocated elements) need to be reset.
Not all of these might technically be needed, but it's better to
reset them if in doubt, to make sure variables don't end up
conflicting.
Signed-off-by: Martin Storsjö <martin@martin.st>
Also add options for specifying a certificate and key, which can
be used both when operating as client and as server.
Partially based on a patch by Peter Ross.
Signed-off-by: Martin Storsjö <martin@martin.st>
When passing a dict to the nested protocol, it will consume
the used options from it, so a separate copy needs to be used
when reopening the connection multiple times.
Signed-off-by: Martin Storsjö <martin@martin.st>
A file containing the trusted CA certificates needs to be
supplied via the ca_file AVOption, unless the TLS library
has got a system default file/database set up.
This doesn't check the hostname of the peer certificate with
openssl, which requires a non-trivial piece of code for
manually matching the desired hostname to the string provided
by the certificate, not provided as a library function.
That is, with openssl, this only validates that the received
certificate is signed with the right CA, but not that it is
the actual server we think we're talking to.
Verification is still disabled by default since we can't count
on a proper CA database existing at all times.
Signed-off-by: Martin Storsjö <martin@martin.st>
Fragmenting blindly to a certain duration isn't a good choice
if one should be able to switch between different qualities,
therefore default to keyframes instead.
Signed-off-by: Martin Storsjö <martin@martin.st>
This makes sure other sanity checks for conflicting options
can work properly, e.g. for the conflict between the faststart
flag when using the ismv mode.
Signed-off-by: Martin Storsjö <martin@martin.st>
Use update_offset() as done for rtmp audio, video and notifications and
read update and write the fields instead of replacing them in the rtmp
packet and then memcpying it to the output buffer.
And fix the AMF_DATA_TYPE_ARRAY parsing while at it.
A MIXEDARRAY type, as the ARRAY, store the number of elements in
an uint32 before the list. The ARRAY is strict and does not have
an OBJECT terminator, MIXEDARRAY behaves like an OBJECT type and
a different than stated number of element can be present.
Also make sure the existing length check can't overflow.
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
This makes sure that it doesn't try to free an AVBuffer belonging
to an earlier packet when we free the local packet at the end.
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
This avoids divisions by zero later.
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
Null buffers are useful for simulating writing to a real buffer
for the sake of measuring how many bytes are written.
Signed-off-by: Martin Storsjö <martin@martin.st>
ASF markers only have a start time, so we lose the chapter end times,
but that is ASF for you
Signed-off-by: Vladimir Pantelic <vladoman@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
this was forgotten when we changed ASF to not output the preroll time
Signed-off-by: Vladimir Pantelic <vladoman@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Abort if it is invalid if strict error checking has been requested.
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
This makes the output fragments independent of their position in
the output stream, making the output work better when streamed.
QuickTime Player doesn't support fragmented mp4 without the base
data offset, though.
Signed-off-by: Martin Storsjö <martin@martin.st>
This is a bit more work, but avoids having to fill in
the data offset field afterwards instead of directly when
the rest of the trun atom is written.
This simplifies future cases where this field needs to be set to
something different.
Signed-off-by: Martin Storsjö <martin@martin.st>
A given packet won't always come in contiguously; sometimes
they may be broken up on chunk boundaries by packets of another
channel.
This support primarily involves tracking information about the
data that's been read, so the reader can pick up where it left
off for a given channel.
As a side effect, we no longer over-report the bytes read if
(toread = MIN(size, chunk_size)) == size
Signed-off-by: Martin Storsjö <martin@martin.st>
Since the number of channels is multiplied by 36 and assigned to
to a uint16_t, make sure this calculation didn't overflow. (In
certain cases the calculation could overflow leaving the
truncated block_align at 0, leading to divisions by zero later.)
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
Some files have the duration set to -1 in the mdhd atom, more
or less legitimately. (We produce such files ourselves, for the
initial duration in fragmented mp4 files.)
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
This more closely corresponds to the usage of the field.
Its usage here is unrelated to the channel ID.
Signed-off-by: Martin Storsjö <martin@martin.st>
Channel 4 is typically used by the Flash player to transmit
audio, channel 6 for video, and various stream-specific invokes
get sent over channel 8, which is designated the source channel.
This more closely matches the behavior of the Flash player,
including the transmission of play requests over channel 8.
Signed-off-by: Martin Storsjö <martin@martin.st>
Sending non-monotonic packets (e.g. when the audio and video
streams are monotonic within themselves but not muxed
monotonically) will lead to negative values the RTMP timestamp
field (where timestamps are transmitted only as deltas for each
channel), and this delta can end up being incorrectly written as
a large unsigned number.
Signed-off-by: Martin Storsjö <martin@martin.st>
Since 596e5d4783, this is not necessary anymore. It also allows to
actually disable the flushing, improving write performance (but
possibly giving worse latency in real-time streaming).
Signed-off-by: Martin Storsjö <martin@martin.st>
This is enabled by default and can be disabled with
"-fflags -flush_packets".
Inspired by a patch from Nicolas George <nicolas.george@normalesup.org>.
Signed-off-by: Martin Storsjö <martin@martin.st>
If we really want to support parameter changes, they need to be
signalled along with the AVPackets as parameter change side data,
not just changing the AVCodecContext parameters when a packet
is demuxed (since there may be other earlier packets yet undecoded).
Something similar was already done for the sample rate in 0883109b2,
but some parameters were left changeable.
This avoids having to recheck the channel count for validity for
each decoded frame in (ad)pcm decoders, unless the decoders
explicitly say that they accept parameter changes.
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
Limit the size to INT_MAX/2 (for simplicity) to be sure that
size + BYTES_PER_FRAME_RECORD won't overflow.
Also factorize other existing error return paths.
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
Limit the size to INT_MAX/2 (for simplicity) to be sure that
size + FF_INPUT_BUFFER_PADDING_SIZE won't overflow.
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
The seektable is required for filling in ape->frames[i].pos
further down.
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
This makes sure the faststart vs fragmentation check works as
intended when fragmentation is enabled due to using the ismv mode.
Signed-off-by: Martin Storsjö <martin@martin.st>
This fixes warnings about making integers from pointers without
a cast, and avoids the theoretical case where the lower 32 bits of
the pointer would all be zero where the implicit cast wouldn't give
the right result.
Signed-off-by: Martin Storsjö <martin@martin.st>
This allows creation of frame accurate chapter marks from sources like
DVD and BD where the precise chapter location is not known until the
chapter mark has been reached during reading.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This should improve write performance quite significantly.
---
Tested with both writing a normal mp4, by using the faststart
feature and writing a fragmented mp4 file; all turn out with the
same md5sum as before.
Signed-off-by: Martin Storsjö <martin@martin.st>
Remove the header decoding for PCM audio from mpeg.c and the
20/24bit parts from pcm.c and merge them into a new decoder in
pcm-dvd.c.
The decoder has added support for samples that span multiple
packets and modified 20/24bit group decoding. Both is needed to
decode samples that have been generated with DVD-Lab Pro 2. The
decoding of 16bit PCM and two channel 24bit is identical to
before. No other samples are known to verify the correctness of
the encoding this software does.
The complete list of tested formats is
48kHz/16bit/2-8 channels
48kHz/24bit/2-5 channels
96kHz/16bit/2-4 channels
96kHz/24bit/2 channels
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
When streaming to limelight, the app name is either a full
"appname/subaccount" or "appname/_definst_". In the latter case,
the app name can be simplified into simply "appname", but the
authentication hashing assumes the /_definst_ still to be present.
Signed-off-by: Martin Storsjö <martin@martin.st>
If a client tries to read the file while it's being updated, the client
would get an incomplete manifest. Instead write to a separate temp file
and atomically rename it to replace the previous one.
Signed-off-by: Martin Storsjö <martin@martin.st>
The element was only being written when the value == 1. But the default
value of this element is 1, so this has no useful effect. This element
needs to be written when the value == 0.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
On failures in the write_trailer function, we could also ignore
the errors and try to finish the file despite these errors (which
would only leave an incomplete chapters track). It's probably better
to signal the error clearly to the caller though (and if this
function failed there's no guarantee that there's enough memory to
finish the trailer either).
Signed-off-by: Martin Storsjö <martin@martin.st>
