Tell the user that the RTP muxer needs to be used to packetize
the data - using the RTP protocol on its own isn't enough.
Signed-off-by: Martin Storsjö <martin@martin.st>
Having more than 10 consecutive frames decoded as mp3 should be
considered a clear signal that the sample is mp3 and not mpegps.
Reported-By: Florian Iragne <florian@iragne.fr>
CC: libav-stable@libav.org
Fixes decting channel layout for files with uncommon audio, such as
FL and FR in two separate streams. Introduced in 3bab7cd.
CC: libav-devel@libav.org
Sample-Id: ticket1474.mov
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
This allows to load metadata entries longer than 1024 bytes.
Displaying them is still limited to 1024 characters, but applications
can load them fully now.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
This fixes the build on compilers that interpreted the earlier
code as a variable length array (which we intentionally disallow).
Signed-off-by: Martin Storsjö <martin@martin.st>
This allows one to specify templated segment names for init-segments,
media-segments, and for the base-url in the case of single-file.
Signed-off-by: Martin Storsjö <martin@martin.st>
Currently, when streaming to an RTMP server, any time a packet of type
RTMP_PT_NOTIFY is encountered, the packet is prepended with @setDataFrame
before it gets sent to the server. This is incorrect; only packets for
onMetaData and |RtmpSampleAccess should invoke @setDataFrame on the RTMP
server. Specifically, the current bug manifests itself when trying to
stream onTextData or onCuePoint invocations.
This fix addresses that problem and ensures that the @setDataFrame is
only prepended for onMetaData and |RtmpSampleAccess.
Since data is fed to the rtmp_write function in smaller pieces (depending
on the calling IO buffer size), we can't generally assume that the
whole packet (or even the whole command string) is available at once,
therefore we can only check the command string once the full packet
has been transferred to us for sending.
Based on a patch by Jeffrey Wescott.
Signed-off-by: Martin Storsjö <martin@martin.st>
We try to avoid mixing av_malloc with av_realloc, since av_malloc
may be implemented with functions that can't (formally) be mixed
with the functions used in av_realloc.
Signed-off-by: Martin Storsjö <martin@martin.st>
This reverts commit b9d08c77a4.
After taking MoveFileEx into use, we can replace files with renames
on windows as well.
Signed-off-by: Martin Storsjö <martin@martin.st>
This allows getting the normal unix semantics, where a rename
allows replacing an existing file.
Based on a suggestion by Reimar Döffinger.
Signed-off-by: Martin Storsjö <martin@martin.st>
This doesn't add any dependency on library internals, since this
only is a static inline function that gets built into each of the
calling functions - this is only to reduce the code duplication.
Signed-off-by: Martin Storsjö <martin@martin.st>
This allows setting the right fragment number if doing
random-access writing of fragments, and also allows reading the
current sequence number.
Signed-off-by: Martin Storsjö <martin@martin.st>
This allows creating a later mp4 fragment without sequentially
writing the earlier ones before (when called from a segmenter).
Normally when writing a fragmented mp4 file sequentially, the
first timestamps of a fragment are adjusted to match the
end of the previous fragment, to make sure the timestamp is the
same, even if it is calculated as the sum of previous fragment
durations. (And for the first packet in a file, the offset of
the first packet is written using an edit list.)
When writing an individual mp4 fragment discontinuously like this
(with potentially writing the earlier fragments separately later),
there's a risk of getting a gap in the timeline if the duration
field of the last packet in the previous fragment doesn't match up
with the start time of the next fragment.
Using this requires setting -avoid_negative_ts make_non_negative
(or -avoid_negative_ts 0).
Signed-off-by: Martin Storsjö <martin@martin.st>
This makes sure that the internal utf8 path names are handled
properly - the normal file handling functions assume path names
are in the native codepage, which isn't utf8.
This assumes that the tools outside of lavf don't use the mkdir
definition. (The tools don't do the same reading of command line
parameters as wchar either - they probably won't handle all possible
unicode file parameters properly, but at least work more predictably
if no utf8/wchar conversion is involved.)
This is moved further down in os_support.h, since windows.h shouldn't
be included before winsock2.h, while io.h needs to be included before
the manual defines for lseek functions.
Signed-off-by: Martin Storsjö <martin@martin.st>
On windows, rename(2) will fail if the target file exists. On
unix this trick is used to make sure that people reading the file
either will get the full previous file, or the full new version
of the file, but no intermediate version.
Signed-off-by: Martin Storsjö <martin@martin.st>
This makes sure that segments actually start at a keyframe (and
makes sure we don't split segments twice in a row, with one segment
consisting of only a handful of packets), when one stream uses b-frames
while another one doesn't.
Signed-off-by: Martin Storsjö <martin@martin.st>
Don't write any bitrate attribute if it isn't known. As long as one
doesn't want automatic bitrate switching, playback can work just
fine even if it isn't set.
If strict standard compliance is requested, this is still considered
an error, since the attribute is mandatory according to the spec.
Based on a patch by Rodger Combs.
Signed-off-by: Martin Storsjö <martin@martin.st>
The chained flv muxer wants one set of tags - normally this set
could be signaled via the AVOutputFormat codec_tag field (as
smoothstreamingenc and dashenc do). hdsenc doesn't signal it, since
the FLV codec tag arrays aren't exported from flvenc.c. This can
lead to the caller keeping an original codec tag from the originating
container here, which would then be a mismatch for the FLV muxer.
Since we don't really care about what codec tag the caller might
have set, just clear it and let the lavf muxer layer set the right
one for the chained FLV muxer later instead.
Signed-off-by: Martin Storsjö <martin@martin.st>
When given a stream starting at dts=0, it would previously consider
s->offset as uninitialized and set an offset when the second packet
was written, ending up writing two packets with dts=0. By initializing
this field to AV_NOPTS_VALUE, we make sure that we only initialize it
once, on the first packet.
Signed-off-by: Martin Storsjö <martin@martin.st>
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>
