| Commit message (Collapse) | Author | Age |
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These fields are already deallocated by uninit_decoder(). Also remove
the wrong/useless log message.
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Now it's used for initialization only for audio and video.
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It doesn't provide this function. The code is not really designed to
work without it, so it will probably mess up big time, but at least
make it compile again.
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See --lavfi-complex option.
This is still quite rough. There's no support for dynamic configuration
of any kind. There are probably corner cases where playback might freeze
or burn 100% CPU (due to dataflow problems when interaction with
libavfilter).
Future possible plans might include:
- freely switch tracks by providing some sort of default track graph
label
- automatically enabling audio visualization
- automatically mix audio or stack video when multiple tracks are
selected at once (similar to how multiple sub tracks can be selected)
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The complex filter support that will be added makes much more complex
use of libavfilter, and I'm not going to bother with adding hacks to
keep libavfilter optional.
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Preparation.
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track can't be NLUL at this point, so the if is redundant. Remove it and
unindent the block. Also, make the function check whether the track is
selected at all, which makes it safer and idempotent.
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The main reason for changing the fullscreen default is that not doing it
would change the vo_rpi default behavior with the previous commit.
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This makes it possible to set video size and position using the
--geometry and/or --autofit options. It's also possible to switch
between fullscreen/non-fullscreen playback during runtime.
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It can be "dangerous". In particular, the decoder might have failed to
initialize, and is now in a broken state. avcodec_flush_buffers() is not
expected to be called in this state, and could trigger undefined
behavior.
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Apparently useful for window embedding.
Fixes #2750.
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For bitmap subs, implement it properly. For libass, you need newest git
master.
Fixes #2791.
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Might possible improve A/V sync, although this is at best approximate.
(AVI is just fucked.)
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I can't explain this, but it seems to be a similar case to the ALSA HDMI
one. I find it hard to tell because of the slightly different names and
conventions in use in libavcodec, WAVEEXT channel masks, decoders, codec
specifications, HDMI, and platform audio APIs.
The fix is the same as the one for ao_alsa (see commit be49da72). This
should fix at least playing 7.1 sources on OSX with 7.1(rear) selected
in Audio MIDI Setup. The ao_alsa commit mentions XBMC, but I couldn't
find out where it does that or if it also does that for CoreAudio. It's
woth noting that PHT (essentially an old XBMC fork) also exhibited the
incorrect behavior (i.e. side and back speakers were swapped).
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Remove flc-frc <-> sl<->sr. This was just plain wrong, and a mistaken
change to make 7.1 work properly on CoreAudio with 7.1(rear) layout.
Also see the following commit.
Add br-br <-> sl<->sr, because we decided that it makes sense.
Note that this "fudging" is applied only if the channel pairs are
replaced, i.e. they would get dropped and be replaced with silence. This
is done to compensate for libswresample's default rematrixing (which
takes care of some more common cases).
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And some other simplifications.
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Avoids "problems". In particular, it makes MMAL output a NOPTS timestamp
if the input timestamp was NOPTS.
Don't do it for other decoders. Ideally, we will at some point in the
future switch to integer fractions for timestamps at least up until the
filter layer. But this would be a larger change, and for now I'd prefer
keeping the not-rounded demuxer timestamps (if we have them).
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Also remove the unused function argument.
Fixes #2784.
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Basically, just make it append " (original)" if the original aspect
ratio is selected.
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This is the "unicode" version of it. It appears Firefox uses it now?
I'm not sure if we still need to support the old variant, but hopefully
not.
Fixes #2782.
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When the directory contains files named such as xx-14.ext and
xx-14.5.ext, remove the extension to sort the table to load the
14 before the 14.5.
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Fixes #2710
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Don't mind me.
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Slightly better.
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Will be helpful for the coming filter support. I planned on merging
audio/video decoding, but this will have to wait a bit longer, so only
remove the duplicate status codes.
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Now I'm not sure if this is really necessary to explicitly mention. But
then there have been issues and pull requests without being properly
looked at for a week or longer.
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Not needed anymore, because the decoder was merged with FFmpeg.
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Since there can be multiple backends for a single API (vaapi can use GLX
or EGL), not logging the exact backend name is annoying. So add it. At
the same time, there is no need to duplicate the name as used by the
--hwdec options, so replace it with using the numeric hwdec API ID.
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Let's fix broken samples with questionable heuristic without real
reasoning. Until this gets fixed properly, this is a good compromise,
though. A proper fix would properly resync audio and video without
brutally resetting the decoders, but on the other hand not doing the
brutal reset would cause issues in other obscure corner cases such
resyncing might cause.
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This code is tricky because it has to wakeup the mainloop to make
progressing during syncing audio, but also has to avoid waking it up
when it's not needed. Failure to do so either burns CPU by not ever
going to sleep, or causes apparent "freezes" by going to sleep (and it
will continue if the mainloop is woken up e.g. due to user input).
In this case, simply starting A/V playback with --start=5 and removing
an unrelated wakeup in osd.c can trigger such a "freeze". The unrelated
wakeup did hide this bug, nonetheless it's a bug.
(Can't wait to rewrite this shitty audio resync code. And it's all my
fault.)
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Explains libdvdnav vs libdvdread situation (as much as I know),
workarounds for bugs, and limitations of image-based subtitles.
(try #4)
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Signed-off-by: wm4 <wm4@nowhere>
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Commit b53cb8de added a delay queue for decoded frames. This is supposed
to be used with copy-back decoders like dxva2-copy and vaapi-copy.
Surfaces returned by them can't be referenced after uninitializing the
decoders, so they have to be released before destroying the decoder.
Move the flush_all() call above decoder uninit accordingly. Also move
the destruction of the AVFrame used for decoding (just for being
defensive - normally it doesn't hold any reference).
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I hope.
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We just need to provide an entrypoint for it, and move the main init
code to a separate function. This gets rid of the messy video chain full
reinit in command.c, which completely destroyed and recreated the video
state for the purpose of mid-stream hw/sw switching.
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Don't give the "software_fallback_decoder" field special meaning. Alwass
set it, and rename it to "decoder". Whether hw decoding is used is
determined by the "hwdec" field already.
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These changes don't make too much sense without context, but are
preparation for later. Then the audio_src/video_src fields will be
actually be NULL under circumstances.
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Before this commit, reinit_audio_chain() did 2 things: create all the
management data structures and initialize the decoder, and handling lazy
filter/output init (as well as dealing with format changes). For the
second purpose, it could be called multiple times (even though it wasn't
really idempotent). This was pretty weird, so make them separate
functions. The new function is actually idempotent too.
It also turns out the reinit functions don't have to call themselves
recursively for the spdif PCM fallback.
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Reduces the dependency of the filter/output code on the decoder.
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Regression caused by commit 3b95dd47. Also see commit 4c25b000. We can
either use video_next_pts and add "delay", or we just use video_pts. Any
other combination breaks. The reason why the assumption that delay==0 at
this point was wrong exactly because after displaying the first video
frame (usually done before audio resync) a new frame might be "added"
immediately, resulting in a new video_next_pts and "delay", which will
still amount to video_pts.
Fixes #2770. (The reason why display-sync was blamed in this issue is
because enabling display-sync in the options forces a prefetch by 2
instead of 1 frames for seeks/playback restart, which triggers the
issue, even if display-sync is not actually enabled. In this case,
display-sync is never enabled because the frames have a unusually high
frame duration. This is also what exposed the initial desync issue.)
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This codes tries to deal with broken PTS timestamps, but since commit
271cabe6 it didn't always overwrite the previous timestamp as it should
have. This mattered only if there were broken timestamps in the video
stream.
Also remove the pointless prev_codec_pts variables, since the decoder
doesn't overwrite these fields anymore.
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This also makes it refcounted, i.e. the new AVFrame will reference the
mp_audio buffers, instead of potentially forcing the consumer of the
AVFrame to copy the data.
All the extra code is for handling the >8 channels case, which requires
very messy dealing with the extended_ fields (not our fault).
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Only triggers if ao=null. This is required for ao=wasapi to recover from periods
with no available devices.
fixes #2738
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Oops, this was forgotten earlier. Enables automatic use of the VC-1
hardware decoder on RPI.
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Fixes DTS handling with certain container formats broken in commit
b53cb8de (when using vaapi-copy or dxva2-copy).
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Correctly avoid a reload if the current device was specified by the user through
--audio-device. Previously, we only recognized if the user had specified
--ao=wasapi:device=.
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oops, forgot to change this when I made get_deviceID a more proper function.
state->deviceID is not set or read here - that's for the caller to do.
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If cover art is re-enabled during playback, the covert art picture
(which has pts==0) will be discarded. Add another corner case to
the list.
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wm4
Uros Vampl
Alexis Nootens
Stefano Pigozzi
archenemies
Tracerneo
Kevin Mitchell