This is mapped to the faststart flag (which in this case
perhaps should be called "shift and write index at the
start of the file"), which for fragmented files will
write a sidx index at the start.
When segmenting DASH into files, there's usually one sidx
at the start of each segment (although it's not clear to me
whether that actually is necessary). When storing all of it
in one file, the MPD doesn't necessarily need to describe
the individual segments, but the offsets of the fragments can be
fetched from one large sidx atom at the start of the file. This
allows creating files for the DASH ISO BMFF on-demand profile.
Signed-off-by: Martin Storsjö <martin@martin.st>
Previously only tfra entries were added for the first track in each moof.
The frag_info array used for tfra can also be used for writing
other kinds of fragment indexes, where it's more important to
include all tracks.
When the separate_moof option is enabled (as in ismv), we write
a separate moof for each track, so this doesn't make any difference
in that case.
Signed-off-by: Martin Storsjö <martin@martin.st>
This is mostly to serve as a reference example on how to segment
the output from the mp4 muxer, capable of writing the segment
list in four different ways:
- SegmentTemplate with SegmentTimeline
- SegmentTemplate with implicit segments
- SegmentList with individual files
- SegmentList with one single file per track, and byte ranges
The muxer is able to serve live content (with optional windowing)
or create a static segmented MPD.
In advanced cases, users will probably want to do the segmenting
in their own application code.
Signed-off-by: Martin Storsjö <martin@martin.st>
A flag "dash" is added, which enables the necessary flags for
creating DASH compatible fragments.
When this is enabled, one sidx atom is written for each track
before every moof atom.
Signed-off-by: Martin Storsjö <martin@martin.st>
By calling this after writing the moof the first time (for
calculating the moof size), we can avoid intermediate storage
of tfrf_offset in MOVTrack.
Signed-off-by: Martin Storsjö <martin@martin.st>
When writing fragmented streams with an empty initial moov,
we won't have any samples in any tracks when writing the
moov atom, thus trust that any tracks that are added actually
will be present.
Signed-off-by: Martin Storsjö <martin@martin.st>
This is needed because Icecast since version 2.4.1 doesn't default
to audio/mpeg anymore. AVOption default not used here, since a later
check if -content_type is set is performed and would break.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This allows for proper error reporting. Only do
this for non-legacy requests as only Icecast >2.4.0
will reply with a proper status.
Libav seems to accept both, 100 and 200 status codes, but
let's stay close to spec.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This avoids a potential crash if writing a fragmented psp mp4
(which probably is only a hypothetical scenario).
Signed-off-by: Martin Storsjö <martin@martin.st>
Previously we wrote decoding timestamps here, while the specs
say it should be presentation timestamps.
Signed-off-by: Martin Storsjö <martin@martin.st>
When using the new first_trun flag instead of checking the track id,
we don't need to have a special case for the separate_moof flag
any longer.
This simplifies the complicated codepath ever so slightly.
Signed-off-by: Martin Storsjö <martin@martin.st>
In this case, shift tracks to start from zero instead (potentially
stretching the first sample in tracks that start later than the
first one).
Some software does not support edit lists at all, the adobe flash
player seems to be one of these. This results in AV sync errors when
edit lists are used to adjust AV sync.
Some players, such as QuickTime, don't respect the duration for
audio packets, so if an audio track starts later than the video
track and the first audio sample gets a duration longer than the
actual amount of data in it, the result will be out of sync.
Based on patches by Michael Niedermayer.
Signed-off-by: Martin Storsjö <martin@martin.st>
This is the same logic as is invoked on AVFMT_TS_NEGATIVE,
but which can be enabled manually, or can be enabled
in muxers which only need it in certain conditions.
Also allow using the same mechanism to force streams to start
at 0.
Signed-off-by: Martin Storsjö <martin@martin.st>
The only parameters needed by the demuxers are the sample rate and sample
count, which can be trivially extracted manually, without resorting to
an avpriv function.
It will not be set unless the codec context is used as the encoding
context, which is discouraged. For MP2, av_get_audio_frame_duration()
will already set the frame size properly. For MP3, set the frame size
explicitly.
It is never an error if this method failed. If rt->live was
explicitly set to 0 (known to be a recorded file), print it
as a warning, otherwise print it as a debug message.
Based on a patch by Michael Niedermayer.
Signed-off-by: Martin Storsjö <martin@martin.st>
The tfdt atom shouldn't be needed in those cases, we already
write tfxd atoms for ismv anyway, which is roughly equivalent.
This avoids having to declare the iso6 brand for ismv files.
Signed-off-by: Martin Storsjö <martin@martin.st>
ISO/IEC 14496-12:2012/Cor 1:2013 is explicit about how this should be
handled. All zeros doesn't mean that the full file has got a zero
duration, only that the track samples described within the initial moov
have got zero duration. An all ones duration means an indeterminate
duration.
Keep writing a duration consisting of all ones for the ISM mode -
older windows media player versions won't play a file if this is
zero. (Newer windows media player versions play either version fine.)
Signed-off-by: Martin Storsjö <martin@martin.st>
Similarly to the omit_tfhd_offset flag added in e7bf085b, this
avoids writing absolute byte positions to the file, making them
more easily streamable.
This is a new feature from 14496-12:2012, so application support
isn't necessarily too widespread yet (support for it in libav was
added in 20f95f21f in July 2014).
Signed-off-by: Martin Storsjö <martin@martin.st>
The custom IO flag actually never is set for muxers, only for
demuxers, so the check was pointless (unless a user intentionally
would set the flag to signal using custom IO).
Signed-off-by: Martin Storsjö <martin@martin.st>
If one track doesn't have any samples within a moof, no traf/trun
is written for it. When the omit_tfhd_offset flag is set, none
of the tfhd atoms have any base_data_offset set, and the implicit
offset (end of previous track fragment data, or start of the moof
for the first trun) is used.
Signed-off-by: Martin Storsjö <martin@martin.st>
should be the raw amount of pixels (for example 3840x1080 for full HD side by
side) and the DisplayWidth/Height in pixels should be the amount of pixels for
one plane (1920x1080 for that full HD stream)."
So, move the aspect ratio check in the mkv_write_stereo_mode() function
and always write the embl when stereo format and/or aspect ration is set.
Also add a few comments to that function.
CC: libav-stable@libav.org
Found-by: Asan Usipov <asan.usipov@gmail.com>
While a standalone implementation is nice, we already depend on
gmtime and gmtime_r in a number of places.
Signed-off-by: Martin Storsjö <martin@martin.st>
gmtime isn't thread safe in general. In msvcrt (which lacks gmtime_r),
the buffer used by gmtime is thread specific though.
One call to localtime is left in avconv_opt.c, where thread safety
shouldn't matter (instead of making avconv depend on the libavutil
internal header).
Signed-off-by: Martin Storsjö <martin@martin.st>
If the buffer provided to strftime is too small, the buffer contents
are indeterminate - it does not guarantee actually null terminating
the buffer.
Signed-off-by: Martin Storsjö <martin@martin.st>
None of these are likely unless the user is writing a file with two billion
streams or a duration of around two months.
CC: libav-stable@libav.org
Bug-Id: CID 700568 / CID 700569 / CID 700570 /
CID 700571 / CID 700572 / CID 700573
The new function wraps errno so that its value is correctly reported
when other functions overwrite it (eg. in case of logging).
CC: libav-stable@libav.org
Bug-Id: CID 1135748
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
The quality scale field is only supposed to be present if the fourth bit
is set. In practice, lame always sets it, but other tools might not.
CC:libav-stable@libav.org
The ones left using av_gettime are NTP timestamps (for RTCP,
which is specified to send the actual current realtime clock
in RTCP SR packets), and the NUT muxer timestamper, which is
documented as using wallclock time.
Signed-off-by: Martin Storsjö <martin@martin.st>
Prevent possible memory leaks.
Connect to nginx and request a non-existent resource to
trigger the issue.
CC: libav-stable@libav.org
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Uwe L. Korn <uwelk@xhochy.com>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Some RTMP commands need the most recent timestamp as their parameter, so
keep track of it. This must be the most recent one and not e.g. the max
received timestamp as it can decrease again through seeking.
Signed-off-by: Martin Storsjö <martin@martin.st>
In (non-live) streams with no metadata, the duration of a stream can
be retrieved by calling the RTMP function getStreamLength with the
playpath. The server will return a positive duration upon the request if
the duration is known, otherwise either no response or a duration of 0
will be returned.
Signed-off-by: Martin Storsjö <martin@martin.st>
Packets that contain a number as a result to a rtmp function call are
structured the same way (String, Number, Null, Number). This new method
also includes more bounds checks to better handle packets that are not
structured as expected.
Signed-off-by: Martin Storsjö <martin@martin.st>
These allow getting the absolute start timestamp of a fragment
without reading preceding timestamps. This fixes sync between
tracks if starting from fragments in different streams that don't
align exactly.
This also is a prerequisite for producing DASH content.
Signed-off-by: Martin Storsjö <martin@martin.st>
Icecast uses HTTP 1.0 while Libav uses HTTP 1.1 and enables by
default chunked post.
Icecast actually forwards the HTTP chunk headers to the listener
as part of the media stream (without the chunk encoding HTTP headers)
causing the players to lose sync.
Disabling the option is enough to feed icecast properly.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
This is necessary to get the right timestamp offset for content
that starts with dts != 0.
This currently only helps when writing fragmented files with a non-empty
moov atom. When writing an empty moov atom, we don't have any packets
yet, so we don't know the starting dts for the tracks.
Signed-off-by: Martin Storsjö <martin@martin.st>
This makes sure that audio preroll for e.g. AAC is signaled correctly.
Previously we only wrote the edit list correctly if we had negative
dts but started with pts == 0 (e.g. for video with B-frames).
Signed-off-by: Martin Storsjö <martin@martin.st>
In these cases, only drop dts. Because if we drop both we have no
timestamps at all for some files.
This improves playback of HLS streams from GoPro cameras.
Signed-off-by: Martin Storsjö <martin@martin.st>
Trying to write to a stream id larger the the maximum requested is
a programming error, still there is no reason to leave a
reachable abort() in the codebase.
CC: libav-stable@libav.org
This makes the field consistent with AVInputFormat.mime_type and the
argument type of av_match_name.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
By using ff_avc_write_annexb_extradata instead of the h264_mp4toannexb
BSF, the code for doing the conversion itself is kept much shorter,
there's less state to restore at the end, we don't risk leaving the
AVCodecContext in an inconsistent state if returning early due to
errors, etc.
Also add a missing free if the base64 encoding fails.
Signed-off-by: Martin Storsjö <martin@martin.st>
The -hls_allow_cache parameter enables explicitly setting the
EXT-X-ALLOW-CACHE tag in the manifest file. That tag indicates
whether the client MAY or MUST NOT cache downloaded media
segments for later replay.
Valid values are 1 (=YES) or 0 (=NO) and the EXT-X-ALLOW-CACHE
will not show in the manifest for other values (or if
-hls_allow_cache is not used.
Signed-off-by: Martin Storsjö <martin@martin.st>
When AVFMT_FLAG_NOBUFFER is set, the packets are not added to the
AVFormatContext packet list, so they need to be freed when they are
no longer needed.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
The RFC spec draft only specifies the "H265" name - there is no
specification saying how to interpret "HEVC" (if such a packet
format is specified it could be an entirely different format).
Since this is a very new standard (still a draft), there is little
need for compatibility with existing, broken implementations. Therefore
remove the extra alias, to avoid the risk of encouraging incorrect
usage.
Intentionally keeping the ff_hevc_dynamic_handler name for the
handler, to use "hevc" consistently as name for the codec instead
of "h265" within the library internals as long as there only is one
single variant in actual use.
Signed-off-by: Martin Storsjö <martin@martin.st>
In practice this hint is ignored - the rtp muxer always overwrites
the stream time base without taking the hint into account. But as
a general practice this is the correct way to pass a time base hint
on to a chained muxer.
This avoids warnings about using the codec time base as hint
being deprecated.
Signed-off-by: Martin Storsjö <martin@martin.st>
The size variable is (correctly) unsigned, but is passed to several functions
which take signed parameters, such as avio_read, sometimes after having
numbers added to it. So ensure that size remains within the bounds that
these functions can handle.
CC: libav-stable@libav.org
Signed-off-by: Diego Biurrun <diego@biurrun.de>
Previously, the returned error codes were intentionally ignored
(see fadd3a6821), to avoid aborting if the directory already
existed. If the mkdir actually failed, this was caught when
opening files within the directory fails anyway.
By handling the error code here (but explicitly ignoring EEXIST),
the error messages and return codes in these cases are more
appropriate and less confusing.
Signed-off-by: Martin Storsjö <martin@martin.st>
Convert the Matroska stereo format to the Stereo3D format, and add a
Stereo3D side data to the stream.
Bump the doctype version supported.
Bug-Id: 728 / https://bugs.debian.org/757185
If the remote end of a connection oriented socket hangs up, generating
an EPIPE error is preferable over an unhandled SIGPIPE signal.
Signed-off-by: Martin Storsjö <martin@martin.st>
