This implements reading the tag in the demuxer and adds support for writing it
in the muxer. Some example channel layout tables for muxing are included for
ac3, aac, and alac, but they are not utilized yet.
Use Sound Sample Description Version 2 for all MOV files.
Updated FATE references accordingly.
Note that ADPCM is treated as compressed audio in version 2.
704af3e29c broke publishing
of rtmp streams, at least publishing to Wowza servers.
This changes all invoke commands to use nb_invokes.
Signed-off-by: Martin Storsjö <martin@martin.st>
malloc() is allowed to return NULL when zero is the argument. This
causes us to think malloc has failed and return AVERROR(ENOMEM). In
addition OS X malloc() returns an unfreeable non-NULL pointer for size
zero when alignment is greater than 16.
Pass the correct size in bits to mpeg4audio_get_config and add a flag
to disable parsing of the sync extension when the size is not known.
Latm with AudioMuxVersion 0 does not specify the size of the audio
specific config. Data after the audio specific config can be
misinterpreted as sync extension resulting in random and wrong configs.
Commit 035af99 made avconv always call an encoder when using the
null muxer. While useful for 2-pass encodes, it inadvertently
caused an extra memcpy of raw frames when decoding only.
This hack restores the old behaviour when only decoding while
allowing use of the null muxer with encoded streams as well.
Signed-off-by: Mans Rullgard <mans@mansr.com>
The media_type_mask is initialized via AVOptions for the
rtsp and sdp demuxers, but it isn't available as an option
for the rtp guessing demuxer (since it doesn't really make
sense there). Therefore, it must be manually initialized
instead, since a zero value means no media types at all
are accepted.
Signed-off-by: Martin Storsjö <martin@martin.st>
For the standardized 8 kHz sample rate, this works exactly the same.
For nonstandard sample rates, the different predefined G726
names (G726-16, G726-24, G726-32, G726-40) are interpreted as an
indication of the bits per coded sample, even though their
actual bitrates aren't what the name specifies.
This feels more sane than using free-form names for nonstandard
sample rate/bitrate combinations, e.g like G726-22, G726-33
for 11025 Hz.
Signed-off-by: Martin Storsjö <martin@martin.st>
This avoids overflow if frame_size is over 2147, since both
frame_size and AV_TIME_BASE are plain integers.
Signed-off-by: Martin Storsjö <martin@martin.st>
Not yet complete, for demuxing AAC the AAC header must be generated
manually.
Possibly the decoder could accept the header as extradata to simplify
this.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This simplifies the open functions by avoiding one function
call that needs error checking, reducing the amount of
extra bulk code.
Signed-off-by: Martin Storsjö <martin@martin.st>
This string will be passed to ff_http_auth_create_response
even if no proxy is used, resulting in reading uninitialized
memory. The other auth string is always initialized by
av_url_split.
Signed-off-by: Martin Storsjö <martin@martin.st>
This requires using a separate init function, since there
isn't necessarily any fmtp lines for this codec, so
parse_sdp_a_line won't be called. Incorporating it with the
alloc function wouldn't do either, since it is called before
the full rtpmap line is parsed (where the sample rate is
extracted).
Signed-off-by: Martin Storsjö <martin@martin.st>
Using an unsigned variable avoids problems with overflows.
There is further no need for a 64-bit intermediate here.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This simplifies the decoder so it doesn't have to process an in-packet header
or handle arbitrary-sized packets. It also fixes decoding of files with large
headers.
AVFMT_NOTIMESTAMPS for crc, as it ignores the timestamps.
AVFMT_VARIABLE_FPS for framecrc, as it prints dts.
Many FATE changes, because avconv is no longer duplicating frames in
those tests.
Also added -vsync 0 for some tests to prevent avconv from dropping
frames until it can be fixed more properly.
AVFMT_NOTIMESTAMPS for md5, as it ignores the timestamps.
AVFMT_VARIABLE_FPS for framemd5, as it prints dts.
-vsync 0 for the vp8 test is needed because with vsync 2 the timestamp
guessing code gets confused by an altref frame that is never displayed
and drops a frame later.
Adding the thread count in frame level multithreading to has_b_frames
as an additional delay causes more problems than it solves.
For example inconsistent behaviour during timestamp calculation in
libavformat.
Thread count and frame level multithreading are both set by the user.
If the additional delay caused by frame level multithreading needs
to be considered in the calling code it has all information to take
it into account.
Should it become necessary to calculate a maximum delay inside
libavcodec it should be exported as its own field and not reusing
an existing field.
Based on a patch by Michael Niedermayer.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
Note: FCPublish/FCUnpublish are adobe server specific and not described
in the rtmp specification. Some servers might not cope with them at
all.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
The Apple HTTP Live Streaming demuxer's implementation of
seeking searches for the MPEG TS segment which contains the
requested timestamp. In its current implementation it assumes
that the first segment will start from 0.
But, MPEG TS streams do not necessarily start with timestamp
(near) 0, causing seeking to fail for those streams.
This also occurs when using live streaming of HTTP Live Streams.
In this case sliding playlists may be used, which means that in
that case only the last x encoded segments are stored, the earlier
segments get deleted from disk and removed from the playlist.
Because of this, when starting playback of a stream in the middle
of such a broadcast, the initial segment fetched after parsing
the m3u8 playlist will not start from timestamp (near) 0, causing
(the admittedly limited live) seeking to fail.
This patch changes this demuxers seeking implementation to use
the initial DTS as an offset for searching the segments containing
the requested timestamp.
Signed-off-by: Martin Storsjö <martin@martin.st>
The tls protocol handles connections via proxies internally.
With TLS/SSL, the peer verification requires that the client
speaks directly with the server, since the proxy doesn't have
the remote server's private key.
Signed-off-by: Martin Storsjö <martin@martin.st>
This opens a plain TCP connection through the proxy via the
CONNECT HTTP method. Normally, this is allowed for connections
on port 443, but can in general be used to allow connections
to any port (depending on proxy configuration), and could thus
be used to tunnel any TCP connection via a HTTP proxy.
Signed-off-by: Martin Storsjö <martin@martin.st>
RTCP timestamps are only necessary to synchronize time between
multiple streams. For a single stream, the RTP packet timestamp
provides more reliable timing. As a result, single-stream RTP
sessions should now have accurate and monotonic PTS.
Signed-off-by: Martin Storsjö <martin@martin.st>
TLSv1 is compatible with SSLv3, so this doesn't change much
in terms of compatibility. By explicitly using TLSv1, OpenSSL
sends the server name indication (SNI) header, which we
already set using SSL_set_tlsext_host_name (earlier, this
didn't have any effect).
SNI allows servers to serve SSL content for different host
names with separate certificates on one single port (vhosts).
Signed-off-by: Martin Storsjö <martin@martin.st>
This makes the function accept the format of creation_time
as output by demuxers (e.g. the mov demuxer), making the
creation timestamp stay intact if transcoding.
Signed-off-by: Martin Storsjö <martin@martin.st>
This function is used in muxers for parsing the 'creation_time'
metadata key, for converting it to a time value.
This makes it match the behaviour of the exported 'creation_time'
metadata from demuxers, where it is in UTC, too.
Signed-off-by: Martin Storsjö <martin@martin.st>
Converting to double before the multiplication rather than after
avoids an integer overflow in some cases.
Signed-off-by: Mans Rullgard <mans@mansr.com>
The interrupt callback has to be passed in during opening (setting it
after opening isn't enough), since a blocking open couldn't be
interrupted otherwise.
Options are passed down to procotols and also need to be available
during open() in most cases.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This is a better io interrupt callback function, which has an
opaque parameter, which is given to the interrupt callback.
This allows callers to precisely cancel IO for one single
AVFormatContext, without interrupt other ones in the same
process.
Note, it's not needed in AVIOContext, at the moment.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Add a decoder for the VBLE Lossless Codec, which
still has a cult following. Used to be popular
several years ago on doom9.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
This is needed because the twinvq decoder cannot rely on bit_rate to be set.
The API documentation says that bit_rate is set by libavcodec, not by the
user.
Tested with both Basic and Digest authentication, and tested with
both proxy authentication and authentication for the requested
resource at the same time.
Signed-off-by: Martin Storsjö <martin@martin.st>
The error was hidden before, to avoid showing an error on the
first request where no auth has been provided, when the server
indicates which authentication method to use.
Now the error is printed if an authentication method was used,
but failed.
Signed-off-by: Martin Storsjö <martin@martin.st>
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>
