This function is entangled with encoder setup, so it is more encoding
code rather than ffmpeg_hw code. This will allow making more encoder
state private in the future.
This function is entangled with decoder setup, so it is more decoding
code rather than ffmpeg_hw code. This will allow making more decoder
state private in the future.
It tracks whether the decoder for this stream ever produced any frames
and its only use is for checking whether a filter input ever received a
frame - those that did not are prioritized by the scheduler.
This is awkward and unnecessarily complicated - checking whether the
filtergraph input format is valid works just as well and does not
require maintaining an extra variable.
Export the corresponding flag in InputFile instead. This will allow
making the demuxer AVFormatContext private in future commits, similarly
to what was previously done for muxers.
There is no point in having a per-stream wallclock start time, since
they are all computed at the same instant. Keep a per-file start time
instead, initialized when the demuxer thread starts.
That is a more appropriate place for this code and will allow hiding
more of InputStream.
The value of repeat_pict extracted from libavformat internal parser no
longer needs to be trasmitted outside of the demuxing thread.
Move readrate handling to the demuxer thread. This has to be done in the
same commit, since it reads InputStream.dts,nb_packets, which are now
set in the demuxer thread.
This way computing it and using it for streamcopy does not need to
happen in sync. Will be useful in following commits, where updating
InputStream.dts will be moved to the demuxing thread.
When an input stream terminates and no frames were successfully decoded,
filtering code will currently configure the filtergraph using demuxer
stream parameters. Use decoder parameters instead, which should be more
reliable. Also, initialize them immediately when an input stream is
bound to a filtergraph input, so that these parameters are always
available (if at all) and filtering code does not need to reach into the
decoder at some arbitrary later point.
Stop using InputStream.dts for generating missing timestamps for decoded
frames, because it contains pre-decoding timestamps and there may be
arbitrary amount of delay between input packets and output frames (e.g.
dependent on the thread count when frame threading is used). It is also
in AV_TIME_BASE (i.e. microseconds), which may introduce unnecessary
rounding issues.
New code maintains a timebase that is the inverse of the LCM of all the
samplerates seen so far, and thus can accurately represent every audio
sample. This timebase is used to generate missing timestamps after
decoding.
Changes the result of the following FATE tests
* pcm_dvd-16-5.1-96000
* lavf-smjpeg
* adpcm-ima-smjpeg
In all of these the timestamps now better correspond to actual frame
durations.
