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>
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>
At least one FATE sample contains such chunks and happens to work simply
by accident (due to find_stream_info() swallowing the error).
CC: libav-stable@libav.org
Update mxf_set_audio_pts to use the container-provided information.
The UL is marked as "to be changed in the future", but the current
samples in the wild do use it.
Prevent out of array writes.
Similar to what Michael Niedermayer did to address the same issue.
Bug-Id: CVE-2014-2263
CC: libav-stable@libav.org
Signed-off-by: Diego Biurrun <diego@biurrun.de>
It is basically a wrapper around av_get_audio_frame_duration(), with a
fallback to AVCodecContext.frame_size. However, that field is set only
when the stream codec context is actually used for encoding or decoding,
which is discouraged.
For muxing, it is generally the responsibility of the caller to set the
packet duration.
For demuxing, if the duration is not stored at the container level, it
should be set by the parser.
Therefore, removing the frame_size fallback should not break any
important case.
The cur_*auth_type variables were set before the http_connect call
prior to 6a463e7fb - their sole purpose is to record the
authentication type used to do the latest request, since parsing
the http response sets the new type in the auth state.
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
Originally, AVFormatContext and a metadata dict were provided to ff_vorbis_comment(),
but this presented issues if an AVStream was being updated or the metadata on
AVFormatContext wasn't actually being updated. To remedy this, ff_vorbis_stream_comment()
explicitly updates a stream's metadata and sets any necessary flags.
ff_vorbis_comment() does not modify any flags, and any calls to it that update
AVFormatContext's metadata (just a single call) must also update
AVFormatContext.event_flags after detecting any metadata changes to the provided
dictionary, as signaled by a positive return value.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Currently, only onMetaData is used, but some providers (wrongly)
put metadata into onCuePoint events, and it's still nice to be
able to use that data.
onCuePoint events also present metadata slightly differently than
onMetaData events: all metadata is found inside an object called
"parameters". In order to extract this metadata, it's easiest to
recurse through the object tree and pull out anything found in
child objects and put it in the top-level metadata.
Reference: http://help.adobe.com/en_US/FlashPlatform/reference/actionscript/2/help.html?content=00001404.html
Signed-off-by: Anton Khirnov <anton@khirnov.net>
If any option named "metadata" is set inside the context, it is pulled up to
the context and then the option is cleared.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
The only flags, for now, indicate if metadata was updated and are set after each call to
av_read_frame(). This comes with the caveat that, on stream start, it might not be set properly
as packets might be buffered in AVFormatContext.packet_buffer before being given to the user
in av_read_frame().
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Previously this logic was only used if the server didn't
respond with Connection: close, but use it even for that case,
if the server response is non-chunked.
Originally the http code has relied on Connection: close to close
the socket when the file/stream is received - the http protocol
code just kept reading from the socket until the socket was closed.
In f240ed18 we added a check for the file size, because some
http servers didn't respond with Connection: close (and wouldn't
close the socket) even though we requested it, which meant that the
http protocol blocked for a long time at the end of files, waiting
for a socket level timeout.
When reading over tls, trying to read at the end of the connection,
when the peer has closed the connection, can produce spurious (but
harmless) warnings. Therefore always voluntarily stop reading when
the specified file size has been received, if not using a chunked
transfer encoding. (For chunked transfers, we already return 0
as soon as we get the chunk header indicating end of stream.)
Signed-off-by: Martin Storsjö <martin@martin.st>
Split return value handling from the actual opening.
Incidentally fixes the https -> http redirect issue reported by
Compn on behalf of rcombs.
CC: libav-stable@libav.org
AVFormatContext->priv_data is not always a MpegTSContext, it can be
RTSPState when decoding a RTP stream. So it is necessary to pass
MpegTSContext pointer explicitly.
Within libav, the write_section_data function doesn't actually use
the MpegTSContext at all, so this doesn't change anything at the
moment (no memory was corrupted before), but it reduces the risk of
anybody trying to touch the MpegTSContext via AVFormatContext->priv_data
in the future.
Signed-off-by: Martin Storsjö <martin@martin.st>
Its contents are meaningful only if the stream codec context is the one
actually used for encoding, which is often not the case (and is
discouraged).
Use AVCodecContext.field_order instead.
librtmp can keep pointers to this string internally, and may
use them at shutdown as well.
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
This typo has existed since this code was added in c16582579.
Newer versions of clang pointed out that this comparison always
was true (since the result of the negation is either 0 or 1, while
AVDISCARD_ALL has the value 48).
Signed-off-by: Martin Storsjö <martin@martin.st>
default-base-is-moof shall be set to track fragments compatible with DASH
Media Segments. So, this is a fundamental support for ISOBMFF ver. DASH.
This is meaningful only when base-data-offset-present is absent and two or
more track fragments are present in a movie fragment.
Signed-off-by: Martin Storsjö <martin@martin.st>
It makes more sense to print the timebase exactly as it is set. Also,
this avoids a divide by zero when av_dump_format() is called on a format
context before writing the header.
As indicated in the function documentation, the header MUST be
checked prior to calling it because no consistency check is done
there.
CC:libav-stable@libav.org
Previously, AVStream.codec.time_base was used for that purpose, which
was quite confusing for the callers. This change also opens the path for
removing AVStream.codec.
The change in the lavf-mkv test is due to the native timebase (1/1000)
being used instead of the default one (1/90000), so the packets are now
sent to the crc muxer in the same order in which they are demuxed
(previously some of them got reordered because of inexact timestamp
conversion).
It has not been properly maintained for years and there is little hope
of that changing in the future.
It appears simpler to write a new replacement from scratch than
unbreaking it.
On big endian machines, the default value set via the faulty
AVOption ended up as 2^32 times too big.
This fixes the fate-lavf-ogg test which currently is broken on
big endian machines, broken since 3831362. Since that commit,
a final zero-sized packet is written to the ogg muxer in that test,
which caused different flushing behaviour on little and big endian
depending on whether the pref_duration option was handled as it
should or not.
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
The '?xyz' form is used by android devices (and according to apple
mailing list archives, also by older iOS devices). The 'loci' field
(defined in 3GPP 26.244) is used by recent iOS devices.
Even though the loci field can contain an altitude, it was plain
0 in my sample. Just export longitude and latitude, in a string
format matching the one used by the '?xyz' metadata field.
Signed-off-by: Martin Storsjö <martin@martin.st>
This allows the caller to write all buffered data to disk, allowing
the caller to know at what byte position in the file a certain
packet starts (any packet written after the flush will be located
after that byte position).
Signed-off-by: Martin Storsjö <martin@martin.st>
In the presence of no metadata, do not set any stream flag in the FLV
header but let the demuxer handle the detection and creation of streams
as data arrives.
Signed-off-by: Martin Storsjö <martin@martin.st>
If no streams were indicated in the FLV header, do not automatically
allocate by default a video and an audio stream. Instead, in the case
that the header did not indicate the presence of any data, allocate no
stream until data actually arrives for one type.
Signed-off-by: Martin Storsjö <martin@martin.st>
The other format (full flac header blocks) should not be exported by any
demuxers anymore.
This allows to drop an avpriv_ function and also simplify the following
commits.
Only copy it manually in the muxers where it makes sense (rtspenc,
sapenc). Don't touch the original AVStream in movenchint, where
the original AVStream should be kept untouched.
This fixes the normal tracks in RTP hinted files after
abb810db - the hint tracks were ok while the normal media tracks
were broken, noticed by Michael Niedermayer.
This reverts abb810db but achieves the same effect for the other
muxers.
Signed-off-by: Martin Storsjö <martin@martin.st>
While it strictly isn't necessary to copy the time base (since
any use of it is scaled in ff_write_chained), it still is better
to signal the actual time base to the caller, avoiding one
unnecessary rescaling. This also lets the caller know what the
actual internal time base is, in case that is useful info
for some caller.
This reverts commit 397ffde115.
Signed-off-by: Martin Storsjö <martin@martin.st>
This results in DefaultDuration not being written when the framerate is
not known, but as this field is purely informative, this should not
break any sane demuxers.
Add the low overhead pipe mode and the extended broadcast mode.
Export the options as 'syncponts' since it impacts only that.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
This corrects the bug that caused the checksums to change in
9767d7c092.
It caused the EOS flag to be set incorrectly; the ogg spec does not
allow it to be set in the middle of a logical bitstream.
Signed-off-by: Andrew Kelley <superjoe30@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
This avoids all the ABI troubles associated with avpriv_.
Since this function is very small and does not depend on any tables,
making it inline should have no adverse effects.
Before, header information for ogg format files was sent with the
first encoded packet.
This patch makes it so that it is possible for API users to
differentiate between headers and encoded audio. This is useful, for
example, when creating an audio stream where you want to send one set
of headers for every client that connects and then the encoded stream
of audio.
Signed-off-by: Martin Storsjö <martin@martin.st>
