Store the file duration in the same timebase it arrives (i.e.
milliseconds) and only convert it to the file duration units (100ns)
when it's actually written, thus simplifying some calculations. Also,
store the duration as unsigned, since it cannot be negative.
CC: libav-stable@libav.org
Bug-ID: CVE-2016-2326
Open decoders first, next encoders. This makes sure that that
subtitle_header is always set properly, without relying on
avformat_find_stream_info() setting it.
Currently, AVStream contains an embedded AVCodecContext instance, which
is used by demuxers to export stream parameters to the caller and by
muxers to receive stream parameters from the caller. It is also used
internally as the codec context that is passed to parsers.
In addition, it is also widely used by the callers as the decoding (when
demuxer) or encoding (when muxing) context, though this has been
officially discouraged since Libav 11.
There are multiple important problems with this approach:
- the fields in AVCodecContext are in general one of
* stream parameters
* codec options
* codec state
However, it's not clear which ones are which. It is consequently
unclear which fields are a demuxer allowed to set or a muxer allowed to
read. This leads to erratic behaviour depending on whether decoding or
encoding is being performed or not (and whether it uses the AVStream
embedded codec context).
- various synchronization issues arising from the fact that the same
context is used by several different APIs (muxers/demuxers,
parsers, bitstream filters and encoders/decoders) simultaneously, with
there being no clear rules for who can modify what and the different
processes being typically delayed with respect to each other.
- avformat_find_stream_info() making it necessary to support opening
and closing a single codec context multiple times, thus
complicating the semantics of freeing various allocated objects in the
codec context.
Those problems are resolved by replacing the AVStream embedded codec
context with a newly added AVCodecParameters instance, which stores only
the stream parameters exported by the demuxers or read by the muxers.
This API is intended to allow passing around codec parameters without
using full AVCodecContext (which also contains codec options and
encoder/decoder state).
Instead of a linked list constructed at av_register_all(), store them
in a constant array of pointers.
Since no registration is necessary now, this removes some global state
from lavf. This will also allow the urlprotocol layer caller to limit
the available protocols in a simple and flexible way in the following
commits.
The timebase change in the zmbv-8bit test is due to the fact that
previously the timebase string was evaluated as floating point, then
converted to a rational. After this commit, the timebase is passed
directly as is.
It makes more sense to not claim anything about the SAR if we don't know
anything. No changes in the FATE tests, since this is what avconv ends
up doing anyway.
lavfi has been considered to be stable for a while now, so it is enabled
in most configurations. Supporting avplay without lavfi requires a lot
of nontrivial ifdef mess for no good reason.
Some (de)muxers open additional files beyond the main IO context.
Currently, they call avio_open() directly, which prevents the caller
from using custom IO for such streams.
This commit adds callbacks to AVFormatContext that default to
avio_open2()/avio_close(), but can be overridden by the caller. All
muxers and demuxers using AVIO are switched to using those callbacks
instead of calling avio_open()/avio_close() directly.
(de)muxers that use the URLProtocol layer directly instead of AVIO
remain unconverted for now. This should be fixed in later commits.
When there is a non-zero decoding delay due to reordering, the first dts
should be lower than the first pts (since the first packet fed to the
decoder does not produce any output).
Use the same scheme used in mpegvideo_enc (which comes from x264
originally) -- wait for first two timestamps and extrapolate linearly to
the past to produce the first dts value.
When B-frames are enabled and the encoder returns success, all currently
pending buffers immediately become valid and can be returned to the
caller. We can only return one packet at a time, so all the other
pending buffers should be transferred to a new 'ready' fifo, from where
they can be returned in subsequent calls (in which the encoder does not
produce any new output). This bug was hidden by the incorrect testing of
the encoder return value (the return value was overwritten before it was
tested).
Before 741b494fa8, when the reference list
modification description was invalid, the code would substitute the
corresponding reference from the initial ("default") reference list.
After that commit, it will just return an error.
Since there are apparently invalid samples in the wild that used to play
fine with the old code, it is a good idea to re-add some sort of error
resilience here. So, when the reference list modification results in a
missing frame, substitute a previous reference frame for it. The
relevant sample again decodes fine with the same output as previously.
The current muxer behaviour is to create streams in read_header() based
on the audio/video presence flags, but fill in the stream parameters
later when we actually get some packets for them. This is rather shady,
since other demuxers set the stream parameters immediately when the
stream is created and do not touch the stream codec context after that.
Change the flv demuxer to behave in the same way as other similar
demuxers -- create the streams only when we get a packet for them.
Almost all the places from which this function is called already check
the header manually and in the two that don't (the mp3 muxer) the check
should not cause any problems.
It will not be set unless the muxing codec context is also the encoding
context, which is discouraged. When the frame size is not known from
av_get_audio_frame_duration(), the fallback should still be good enough.
It will not be set if the stream codec context is not the encoding
context. Use av_get_audio_frame_duration() instead, it should work for
all audio codecs supported by the muxer.