The chunksize internal variable has two different uses - for
reading, it's the amount of data left of the current chunk
(or -1 if the server doesn't send data in chunked mode), where
it's only an internal state variable. For writing, it's used
to decide whether to enable chunked encoding (by default), by
using the value 0, or disable chunked encoding (value -1).
This, while consistent, doesn't make much sense to expose
as an AVOption. This splits the usage of the internal variable
into two variables, chunksize which is used for reading (as
before), and chunked_post which is the user-settable option,
with the values 0 and 1, where 1 is default.
Signed-off-by: Martin Storsjö <martin@martin.st>
It is found in some 8svx files (e.g. ones created by SoX).
Currently the decoder reuses the 8svx functions because we already have
handling of a single large planar packet for the compressed 8svx codecs.
The return value ret isn't an error code that can be passed
to ERR_error_string().
This makes the error messages printed actually contain useful
information.
Signed-off-by: Martin Storsjö <martin@martin.st>
The caller expects the seekhead struct to be freed when calling
matroska_write_seekhead. Currently, the structure is leaked if the
seek fails.
Signed-off-by: Martin Storsjö <martin@martin.st>
This is to make developers aware of the fact that they will
start using the new init function at some point.
Signed-off-by: Martin Storsjö <martin@martin.st>
It might make sense not to make the function completely mandatory
immediately at the next bump, which might be quite soon after
the function was introduced.
Signed-off-by: Martin Storsjö <martin@martin.st>
This makes rdt work again, which has been broken since
603b8bc2a1. This commit made
opening a demuxer without a file (or in this case, with a filename
which can't be opened) fail, unless the demuxer actually declared
AVFMT_NOFILE.
Signed-off-by: Martin Storsjö <martin@martin.st>
Since 1.0.0, this function is deprecated. A new function,
CRYPTO_THREADID_set_callback is available, but if not set at all,
it uses the address of errno as thread id, which should be
sufficient for most systems.
On windows, it never was necessary to use this function even
before 1.0.0, it used the right win32 API function for this
by default.
Signed-off-by: Martin Storsjö <martin@martin.st>
All current usages of it are incompatible with localization.
For example strcasecmp("i", "I") != 0 is possible, but would
break many of the places where it is used.
Instead use our own implementations that always treat the data
as ASCII.
Signed-off-by: Martin Storsjö <martin@martin.st>
Earlier, sc->samples_per_frame was used for setting the frame size,
but all files don't have that set properly. The frame size is a
known constant for these codecs.
If frame_size isn't set, the mov/3gp muxer refuses to mux it.
This fixes stream copy of audio from
https://roundup.libav.org/file1248/Video_With_AMR-NB_Audio.3gp
to another 3gp file (roundup issue 2468).
Signed-off-by: Martin Storsjö <martin@martin.st>
These packets are valid packets, and consist of 1 byte (which
contains the mode bits).
This had been analyzed and reported by Igor Levin, igor d levin comverse com.
Signed-off-by: Martin Storsjö <martin@martin.st>
Note, this protocol doesn't yet check verify the server
certificate against a local database of trusted CA root
certificates.
Signed-off-by: Martin Storsjö <martin@martin.st>
Streams from RTSP or SDP that do not match an allowed type will
be skipped entirely, which allows video-only or audio-only
streaming from servers that provide both.
Signed-off-by: Martin Storsjö <martin@martin.st>
This requires some workarounds in the WAV muxer and demuxer. We need to write
the correct bits_per_coded_sample and block_align in the muxer. In the
demuxer, we cannot rely on the bits_per_coded_sample value, so we use the bit
rate and sample rate to determine the value.
This avoids having the decoder rely on AVCodecContext.bit_rate, which is not
required to be set by the user for decoding according to our API.
bits_per_coded_sample should be 8.
block_align is calculated incorrectly, but it is not needed anyway.
packet pts should be calculated in samples.
packet duration can be set.
This fixes false positives of has_codec_delay_been_guessed() for
streams where not every input picture generates an output picture,
such as interlaced H264.
Manual replacements are done in this commit.
In many cases, the id is some constant made up number (e.g. 0 for video
and 1 for audio), which is then not used in the demuxer for anything.
Those ids are removed.
This allows setting the filter_src option for these demuxers, too,
which wasn't possible at all before (where the option only was set
via URL parameters for RTSP).
Signed-off-by: Martin Storsjö <martin@martin.st>
avconv doesn't map video streams to a muxer without specifying a
manual stream mapping if the default video codec is CODEC_ID_NONE.
Signed-off-by: Martin Storsjö <martin@martin.st>
It was broken in 3b3ea34655
"Remove all uses of deprecated AVOptions API", where any
presence of a payload_type AVOption caused its value to
be returned, even if it wasn't set (and thus had the default
-1 value).
This caused the RTP muxer to be broken.
Signed-off-by: Martin Storsjö <martin@martin.st>
Eventually, the old way of passing options by adding
stuff to the URL can be dropped.
This avoids having to tamper with the user-specified URL to
pass options on the transport mode. This also works better
with redirects, since the options don't need to be parsed out
from the URL.
Signed-off-by: Martin Storsjö <martin@martin.st>
Use defines for shortening common parts, omit the .dbl named
initializer (since it's the first element in the union).
Signed-off-by: Martin Storsjö <martin@martin.st>
This eases adding options that are common for both. The
AV_OPT_FLAG_EN/DECODING_PARAM still indicates whether they belong
to the muxer or demuxer.
Signed-off-by: Martin Storsjö <martin@martin.st>
The cbSize field should be included in all cases, even with PCM where
its value is ignored.
Fixes encoding PCM audio in Matroska for some players which insist on
a full WAVEFORMATEX structure for A_MS/ACM audio.
Since fate uses wav files for the audio test a larger number of tests
has changed checksums or shifted positions due to the 2 byte longer
wave header.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
The demuxer does not read the seektable, a parser is not possible without a
full decode, and no shorten decoder can handle random seeking because it needs
side info from the seektable.
This will allow the caller to enumerate child contexts in a generic way
and since the API is recursive, it also allows for deeper nesting (e.g.
AVFormatContext->AVIOContext->URLContext)
This will also allow the new setting/reading API to transparently apply
to children contexts.
No application rely on this count being correct as far as
I know, but if we write a nonzero count value, it might just
as well be the right one.
Signed-off-by: Martin Storsjö <martin@martin.st>
DSS enables this automatically if streaming VOD over TCP. If
enabled, the server feeds packets faster than realtime, screwing
up RTCP NTP based timestamps.
Also, DSS doesn't indicate that this was indicated, if it was
enabled automatically (although if it was requested to be enabled,
a header saying that it was enabled is added, but this isn't
added if it is enabled automatically), making it even harder
to detect and work around properly without explicitly asking
for it to be disabled(/enabled, if we were able to support it).
Signed-off-by: Martin Storsjö <martin@martin.st>
This allows skipping past unsupported RTCP packet types, as
RFC 3550 section 6.1 mandates.
Currently this only has any practical effect if a sender puts
an unrecognized type before RTCP_BYE in a compounded packet, or
(incorrectly) does not put RTCP_SR first.
Signed-off-by: Martin Storsjö <martin@martin.st>
This avoids writing these entries doubly if transcoding from
flv to flv, since the muxer blindly writes any and all metadata
keys set, in addition to the fixed fields that the muxer
always writes.
Signed-off-by: Martin Storsjö <martin@martin.st>
Keeping byte values read from the file as unsigned is consistent
with how they are subsequently used and avoids an undefined left
shift by 24 when bit 7 is set.
Signed-off-by: Mans Rullgard <mans@mansr.com>
On the first iteration through this code, last_dts is always
INT64_MIN (AV_NOPTS_VALUE) and the subtraction overflows in
an invalid manner. Although the result is only used if the
input values are valid, performing the subtraction is still
not allowed in a strict environment.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Whitespace of the patch cleaned up by Aurel
Some of the issues have been reported by Steve Manzuik / Microsoft Vulnerability Research (MSVR)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 956c901c68)
Further suggestions from Kostya <kostya.shishkov@gmail.com> have been
implemented by Reinhard Tartler <siretart@tauware.de>
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
Specifying the payload type is useful when the type number has
already been negotiated before creating the stream, for example
in SIP protocol.
Signed-off-by: Martin Storsjö <martin@martin.st>
This patch also introduces CODEC_ID_CELT.
Signed-off-by: Nicolas George <nicolas.george@normalesup.org>
Signed-off-by: Diego Biurrun <diego@biurrun.de>
If the demuxer did not set a codec_tag, there is none and
inventing one makes no sense. This change stops the rawvideo
"decoder" over-writing user-supplied pixfmt with one derived
from the codec_tag. The pixfmt-codec_tag-pixfmt round-trip
is lossy since several pixfmts map to the same codec_tag.
This fixes fate-lavf-pixfmt with avfilter disabled.
Signed-off-by: Mans Rullgard <mans@mansr.com>
On allocation, the array length is multiplied by sizeof(int64_t),
this prevents the multiplication from overflowing.
Signed-off-by: Martin Storsjö <martin@martin.st>
Move the identical code in rtp_write_header() and
ff_sdp_write_media() inside ff_rtp_get_payload_type()
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Make the iff demuxer send the whole audio chunk to the decoder as a
single packet and move stereo interleaving from the iff demuxer to the
decoder.
Based on a patch by Stefano Sabatini.
git.videolan.org/ffmpeg.git
commit e280a4da2a
This makes the RTMP writing code able to handle FLV data
fed in arbitrarily small or large chunks, with multiple
consecutive packets in one write call, or having the FLV
packet header split over numerous write calls.
When used in conjunction with the flv muxer, the AVIO buffer
size still needs to be large enough to fit the initial metadata
packet though, since the size of that packet is written with a
seekback.
Signed-off-by: Martin Storsjö <martin@martin.st>
This extends the ID3v2 parser to allow for reading of non-text (i.e.
other than T***) meta tag frames providing a ff_id3v2_read_all()
function. An additional data structure 'ID3v2ExtraMeta' is introduced
for these tags since AVDictionary is string oriented and unsuitable
for binary data.
A parser for tag frames of type GEOB is implemented, which is needed
to extract keyring information from encrypted OMA files. GEOB data
is parsed into 'ID3v2ExtraMetaGEOB' data structures.
The routine to decode characters from different encodings to UTF-8,
formerly part of the read_ttag() function, is moved to its own
function. Because some tag frames contain subparts of unknown length,
the function is now also able to read until a null character is found.
In addition, the function now takes care of allocating a buffer long
enough to hold the decoded characters.
Signed-off-by: David Goldwich <david.goldwich@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
If not enough bytes are available, keep track of them and skip
them on next call.
In practice, if these trailer bytes are written in a separate
call, there is no other data written in this call, making it
fall into the "FLV packet too small" case currently - working,
but not as intended.
This patch makes the code more robust, handling all cases
except for having the FLV packet header split over multiple
write calls.
Signed-off-by: Martin Storsjö <martin@martin.st>
If the FLV packet is larger than the AVIO buffer, a partial
FLV packet will be flushed to the RTMP protocol.
This commit handles the most common cases of FLV packets
being written in more than one call.
Signed-off-by: Martin Storsjö <martin@martin.st>
pixel_format/video_size only apply to 'rawvideo' (==uncompressed) demuxer
and make no sense for the other raw (== containerless) demuxers. Keep
only the framerate option for those.
Also use unique classes for all raw video demuxers
Fixes MSVR-11-0088.
Credit: Jeong Wook Oh of Microsoft and Microsoft Vulnerability Research (MSVR)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Martin Storsjö <martin@martin.st>
The initial request contains "Range: 0-", which servers normally
have responded with "HTTP/1.1 206 Partial Content" reply with
a Content-Range header, which was used as indicator for seekability.
Apache, since 2.2.20, responds with "HTTP/1.1 200 OK" for these
requests, which is more friendly to caches and proxies, but the
seekability still is indicated via the Accept-Ranges: bytes header.
Signed-off-by: Martin Storsjö <martin@martin.st>
Unlike other containers RealMedia stores its audio packets in scrambled form,
with interleaver ID preceeding audio codec ID. Currently deinterleaving
decision is tied to the codec while it's possible to have non-default
deinterleaver with audio codec (like Int0 deinterleaver instead of specific
one for Sipro).
Signed-off-by: Anton Khirnov <anton@khirnov.net>
The new function provides a more generic interface than av_fifo_peek()
for peeking at a FIFO buffer data.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This fixes the situation when there are not enough entries in the index
(e.g. on initial seek there's only one index entry in the index) and index
search returns just the last known entry. That causes seeking function just to
seek there instead of trying harder to get at the requested position.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Currently for multichannel audio position for the last block position is
stored in index (and used for seeking), which is obviously not correct.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
First, container stores only DTS and not PTS as it was believed.
Second, multiple frames in a packet store timestamp instead of position
after the frame length.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Metadata currently is written only at the start of the file in normal
cases, when transcoding from a rtmp source metadata could be
written later and the offset recorded can exceed 32bit.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
ID3v1 are legacy tags with several limitations; furthermore
avconv/ffmpeg writes the tags in UTF-8 which probably has near-0
software support.
Add a -write_id3v1 option to be able to turn it on; disabled by default.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
AVFormatParameters are converted into corresponding private options in
av_open_input_file/stream() compat wrappers, so accessing them from
demuxers is redundant.
This fixes an issue where packets which start being read
while reading the header stick around after a seek.
Signed-off-by: Zohar Kelrich <lumimies@gmail.com>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Do not try to seek when we already know we are not allowed to.
Silences warning that always happens when streaming.
Signed-off-by: Zohar Kelrich <lumimies@gmail.com>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
mpegts_read_header is used by both mpegts and mpegtsraw, so this
erro check is no longer valid.
Signed-off-by: Mike Williams <mike@mikebwilliams.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Moving the search and parsing of the 'fmt ' info the main loop of
wav_read_header() allows tags that precede it to be parsed. Creating
wav_parse_fmt_tag() makes wav_read_header() easier to read.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Specifically crafted samples can reinit ogg->streams[] while
reading samples, and thus we should not cache old pointers since
these may no longer be valid.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
According to MPEG-TS specs, the continuity_counter shall not be
incremented when the adaptation_field_control of the packet
equals '00' or '10'.
Signed-off-by: Jindrich Makovicka <jindrich.makovicka@nangu.tv>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Declaring tools associated with each library in their respective
makefiles allows these tools to easily depend on the correct
prerequisites and link against the libs they need.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Set DV packet durations using fields_per_frame.
This requires turning gxf_stream_info into the demuxer's context for access to the value in gxf_packet().
Since MPEG-2 seems to work fine this done only for DV.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Parse the extension flag bit when reading the MPEG4 AudioSpecificConfig.
This has nothing to do with SBR/PS contradictory to what was noted when it was removed.
This is needed because not all players (e.g. MPlayer) are able to distinguish
two different Bink audio decoders when codec_tag is set.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Before this, almost all module groups have been used for grouping functions
and fields in structures semantically. This causes them to not appear
properly in the file documentation and needlessly clutters up the "Modules"
index.
Additionally, this commit streamlines some spelling and appearances.
av_open_input_stream used to allow this, even though it makes no sense.
Make it just print a warning instead of failing, thus restoring
compatibility.
Note that avformat_open_input() will still reject this combination.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
EBML_STOP leaves matroska->current_id set. Then matroska_read_seek changes
the stream position without resetting current_id. The next
matroska_parse_cluster fails due to calculation of incorrect pos. So clear
current_id when avio_seek happens in matroska_read_seek.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
This fixes this compilation warning, by making endptr a non-const
pointer, as required by strtol:
In function ‘udp_open’:
warning: passing argument 2 of ‘strtol’ from incompatible pointer type
Signed-off-by: Martin Storsjö <martin@martin.st>
ogg_save() copies streams[], but doesn't keep track of free()'ed
struct members. Thus, if in between a call to ogg_save() and
ogg_restore(), streams[].private was free()'ed, this would result
in a double free -> crash, which happened when e.g. playing small
chained ogg fragments.
For a unicast udp stream to localhost:1234, currently
ffplay udp://:1234?localport=1234
works, but
ffplay udp://:1234
doesn't work.
Signed-off-by: Martin Storsjö <martin@martin.st>
This helps catching cases where the format string doesn't
match what is passed in, or injection bugs where user data
is passed in as format string.
Signed-off-by: Martin Storsjö <martin@martin.st>
In this case, the string that was passed couldn't contain
user-defined data and thus there was no risk for injection
bugs, but it's safer this way, if we later change the
content of the options string.
Signed-off-by: Martin Storsjö <martin@martin.st>
This is enabled with an AVOption on the RTP muxer. The SDP
generator looks for a latm flag in the rtpflags field.
Signed-off-by: Martin Storsjö <martin@martin.st>
