| Commit message (Collapse) | Author | Age | Files | Lines |
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The --cache-pause feature (enabled by default) will pause playback for a
while if network runs out of data. If this is not done, then playback
will go on frame-wise (as packets are slowly read from the network and
then instantly decoded and displayed). This feature is actually useless,
as you won't get nice playback no matter what if network is too slow,
but I guess I still prefer this behavior for some reason.
This commit changes this behavior from using the demuxer cache state
only, to trying to use underrun information from the AO/VO. This means
if you have a very large audio buffer, then cache-pausing will trigger
once that buffer is depleted, which will be some time _after_ the
demuxer cache has run out.
This requires explicit support from the AO. Otherwise, the behavior
should be mostly the same as before this commit.
This does not care about the AO buffer. In theory, the AO may underrun,
then the player will write some data to the AO buffer, then the AO will
recover and play this bit of data, then the player will probably trigger
the cache-pause behavior. The probability of this happening should be
pretty low, so I will hold off fixing this until the next refactor of
the AO chain (if ever).
The VO underflow detection was devised and tested in 5 minutes, and may
not be correct. At least I'm fairly sure that the combination of all the
factors should make incorrect behavior relatively unlikely, but problems
are possible.
Also, the demux_reader_state.underrun field may be inaccurate. It's only
the present state at the time demux_get_reader_state() was called, and
may exclude past underruns. In theory, this could cause "close" cases to
be missed. Then you might get an audio underrun without cache-pausing
acting on it. If the stars align, this could happen multiple times in
the row, effectively making this feature not work.
The most user-visible consequence of this change is that the user
will now see an AO underrun warning every time the cache runs out.
Maybe this cache-pause feature should just be removed...
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AOs can now call ao_underrun_event() (in any context) if an underrun has
happened. It will print a message.
This will be used in the following commits. But for now, audio.c only
clears the underrun bit, so that subsequent underruns still print the
warning message.
Since the underrun flag will be used in fragile ways by the playback
state machine, there is the "reports_underruns" field that signals
strong support for underrun reporting. (Otherwise, underrun events will
not be used by it.)
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Make it a warning. This is such an intrusive and shitty hack (but of
course my fault) that it should not be hidden.
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In this case, gapless will most likely not work. It will result in (very
slight) desync, or (more commonly with small buffer sizes), in an
underflow.
I think it would be legitimate to disable gapless at end of playback
completely if video is enabled at all. But this would need an exception
for cover art mode, so I guess the current solution is OK as well.
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In spdif mode, there are hacks that try to cut audio on frame boundaries
(blame spdif, which is a hack in itself). The "alignment" is used in a
bunch of places, but --end does not respect it. This leads to some audio
that can't be pushed because the alignment is off (I don't know why, not
do I care), which puts audio into an underrun state forever.
Fix this by discarding unusable extra samples if no new data can be
expected.
Fixes: #6935
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reinit_audio_filters_and_output() can fully shutdown the audio chain on
failure. Specifically, it will deallocate mpctx->ao_chain. The value of
that field was cached in ao_c. The code after the call did not account
that the audio chain can be shutdown, and used the stale ao_c value.
Fixes: #6808
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Track switching doesn't run reset_playback_state(), so a track enabled
at runtime during backward playback would lead to a messed up state.
This commit just does a bad code monkey fix to this. It feels like there
needs to be a much better way to propagate this state.
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We need to transform the timestamp returned by get_play_end_pts().
I considered making it return the transformed timestamp directly. There
are 4 callers; 2 need a transformed timestamps, 2 don't. So I guess it
doesn't matter.
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The buffer is written to in `audio_config_to_str_buf` using `snprintf`
with a `%s` formatting of a 128-byte buffer. This can overflow the
target buffer of 80 causing a truncated output.
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Makes it easier to not break the build by confusing the ifdeffery.
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Until recently, ao_lavc and vo_lavc started encoding whenever the core
happened to send them data. Since audio and video are not initialized at
the same time, and the muxer was not necessarily opened when the first
encoder started to produce data, the resulting packets were put into a
queue. As soon as the muxer was opened, the queue was flushed.
Change this to make the core wait with sending data until all encoders
are initialized. This has the advantage that we don't need to queue up
the packets.
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It stopped and did nothing, instead of terminating (or just letting
video play, if there was any video). Regression due to recent filter
changes.
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Until recently, the AO was reinitialized strictly only on decoder format
changes. But the commit for simplifying audio format negotiation removed
this. Now the AO is recreated for any format change.
This is sort of annoying if you change playback speed. The
insertion/removal of af_scaletempo can change the sample format. For
example, the acompressor filter will convert output to double, so
toggling scaletempo will force the format back to float. This recreates
the AO under the --gapless-audio=weak default. This likely affects a lot
of other filters too.
Work this around by allowing sample format changes, and keeping the
current AO format in these cases. This is probably not a big problem.
Most audio APIs force the output format to float anyway.
This means you actually have to worry about what the default gapless
mode does to your audio. If you start with a file that uses 8 bit per
sample, and then continue playing a 24 bit FLAC, it will be converted
down to 8 bit per sample. (Assuming they are played in a way that uses
the gapless logic.)
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Just bail out immediately (and disable audio) if format probing has no
result, instead of doing nothing and then apparently freezing.
This can happen with bogus filters, cases where the first audio frame is
essentially dropped by filters (can happen with large resampling
factors), and if the audio track contains no packets at all, or all
packets fail to decode.
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The recent changes to player/audio.c moved PTS jump detection to after
audio filtering. This was mostly done for convenience, because dataflow
between decoder and filters was made "automatic", and jump detection
would have to be done as filter. Now move it back to after decoders,
again out of convenience. The future direction is to make the dataflow
between filters and AO automatic, so this is a bit in the way. Another
reason is that speed changes tend to cause jumps - these are legitimate,
but get annoying quickly.
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Use the decoder wrapper that was introduced for video. This removes all
code duplication the old audio decoder wrapper had with the video code.
(The audio wrapper was copy pasted from the video one over a decade ago,
and has been kept in sync ever since by the power of copy&paste. Since
the original copy&paste was possibly done by someone who did not answer
to the LGPL relicensing, this should also remove all doubts about
whether any of this code is left, since we now completely remove any
code that could possibly have been based on it.)
There is some complication with spdif handling, and a minor behavior
change (it will restrict the list of codecs to spdif if spdif is to be
used), but there should not be any difference in practice.
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Move dec_video.c to filters/f_decoder_wrapper.c. It essentially becomes
a source filter. vd.h mostly disappears, because mp_filter takes care of
the dataflow, but its remains are in struct mp_decoder_fns.
One goal is to simplify dataflow by letting the filter framework handle
it (or more accurately, using its conventions). One result is that the
decode calls disappear from video.c, because we simply connect the
decoder wrapper and the filter chain with mp_pin_connect().
Another goal is to eventually remove the code duplication between the
audio and video paths for this. This commit prepares for this by trying
to make f_decoder_wrapper.c extensible, so it can be used for audio as
well later.
Decoder framedropping changes a bit. It doesn't seem to be worse than
before, and it's an obscure feature, so I'm content with its new state.
Some special code that was apparently meant to avoid dropping too many
frames in a row is removed, though.
I'm not sure how the source code tree should be organized. For one,
video/decode/vd_lavc.c is the only file in its directory, which is a bit
annoying.
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lavfi.c is not necessary anymore, because f_lavfi.c (which was actually
converted from it) can be used now.
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Use the new filtering code for audio too.
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If you play a video with an external audio track, and do backwards
keyframe seeks, then audio can be missing. This is because a backwards
seek can end up way before the seek target (this is just how this seek
mode works). The audio file will be seeked at the correct seek target
(since audio usually has a much higher seek granularity), which results
in silence being played until the video reaches the originally intended
seek target.
There was a hack in audio.c to deal with this. Replace it with a
different hack. The new hack probably works about as well as the old
hack, except it doesn't add weird crap to the audio resync path (which
is some of the worst code here, so this is some nice preparation for
rewriting it). As a more practical advantage, it doesn't discard the
audio demuxer packet cache. The old code did, which probably ruined
seeking in youtube DASH streams.
A non-hacky solution would be handling external files in the demuxer
layer. Then chaining the seeks would be pretty easy. But we're pretty
far from that, because it would either require intrusive changes to the
demuxer layer, or wouldn't be flexible enough to load/unload external
files at runtime. Maybe later.
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The future direction might be not having such a user-visible filter at
all, similar to how vf_scale went away (or actually, redirects to
libavfilter's vf_scale).
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No idea why this wasn't done earlier. This makes playback start in audio
only tracks closer to video-only or video/audio restart. It has the
consequence that --cache-pause-initial now works for audio-only streams
too.
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A release has been made, so drop options deprecated for that release.
Also drop some options which have been deprecated a much longer time
before.
Also fix a typo in client-api-changes.rst.
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This is never updated after the AO inits, so there are several cases
where the volume would stay at 100%, even if it shouldn't. This affects
initial volume as well as track switching or switching between files.
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This does what af_volume used to do. Since we couldn't relicense it,
just rewrite it. Since we don't have a new filter mechanism yet, and the
libavfilter is too inconvenient, do applying the volume gain in ao.c
directly. This is done before handling the audio data to the driver.
Since push.c runs a separate thread, and pull.c is called asynchronously
from the audio driver's thread, the volume value needs to be
synchronized. There's no existing central mutex, so do some shit with
atomics. Since there's no atomic_float type predefined (which is at
least needed when using the legacy wrapper), do some nonsense about
reinterpret casting the float value to an int for the purpose of atomic
access. Not sure if using memcpy() is undefined behavior, but for now I
don't care.
The advantage of not using a filter is lower complexity (no filter auto
insertion), and lower latency (gain processing is done after our
internal audio buffer of at least 200ms).
Disavdantages include inability to use native volume control _before_
other filters with custom filter chains, and the need to add new
processing for each new sample type.
Since this doesn't reuse any of the old GPL code, nor does indirectly
rely on it, volume and replaygain handling now works in LGPL mode.
How to process the gain is inspired by libavfilter's af_volume (LGPL).
In particular, we use exactly the same rounding, and we quantize
processing for integer sample types by 256 steps. Some of libavfilter's
copyright may or may not apply, but I think not, and it's the same
license anyway.
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These couldn't be relicensed, and won't survive the LGPL transition. The
other existing filters are mostly LGPL (except libaf glue code).
This remove the deprecated pan option. I guess it could be restored by
inserting a libavfilter filter (if there's one), but for now let it be
gone.
This temporarily breaks volume control (and things related to it, like
replaygain).
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Most commonly happened with --end, in which case that field tends to be
set.
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See "Copyright" file for caveats.
This changes the remaining "almost LGPL" files to LGPL, because we think
that the conditions the author set for these was finally fulfilled.
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This code could not be relicensed. The intention was to write new filter
code (which could handle both audio and video), but that's a bit of
work. Write some code that can do audio conversion (resampling,
downmixing, etc.) without the old audio filter chain code in order to
speed up the LGPL relicensing.
If you build with --disable-libaf, nothing in audio/filter/* is compiled
in. It breaks a few features, such as --volume, --af, pitch correction
on speed changes, replaygain.
Most likely this adds some bugs, even if --disable-libaf is not used.
(How the fuck does EOF notification work again anyway?)
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Just reimplement it in some way, as mp_audio is GPL-only.
Actually I wanted to get rid of audio_buffer.c completely (and instead
have a list of mp_aframes), but to do so would require rewriting some
more player core audio code. So to get this LGPL relicensing over
quickly, just do some extra work.
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Ever since the mp_aframe change, audio mid-stream format changes crash.
I have no idea why the recent mp_aframe change triggers this. Didn't
look too deeply into it either. It appears to work now, though.
Fixes #4828.
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This is pretty pointless, but I believe it allows us to claim that the
new code is not affected by the copyright of the old code. This is
needed, because the original mp_audio struct was written by someone who
has disagreed with LGPL relicensing (it was called af_data at the time,
and was defined in af.h).
The "GPL'ed" struct contents that surive are pretty trivial: just the
data pointer, and some metadata like the format, samplerate, etc. - but
at least in this case, any new code would be extremely similar anyway,
and I'm not really sure whether it's OK to claim different copyright. So
what we do is we just use AVFrame (which of course is LGPL with 100%
certainty), and add some accessors around it to adapt it to mpv
conventions.
Also, this gets rid of some annoying conventions of mp_audio, like the
struct fields that require using an accessor to write to them anyway.
For the most part, this change is only dumb replacements of mp_audio
related functions and fields. One minor actual change is that you can't
allocate the new type on the stack anymore.
Some code still uses mp_audio. All audio filter code will be deleted, so
it makes no sense to convert this code. (Audio filters which are LGPL
and which we keep will have to be ported to a new filter infrastructure
anyway.) player/audio.c uses it because it interacts with the old filter
code. push.c has some complex use of mp_audio and mp_audio_buffer, but
this and pull.c will most likely be rewritten to do something else.
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Refresh seeks are automatically issued when changing filters, which
improves user experience if these filters change buffering or such.
The refresh seek could actually overwrite a previously ongoing seek:
set pause yes
set time-pos 10
set vf ""
Here, the video code issued a refresh seek to the previous video
position, which could be different from the previously triggered (and
still ongoing) seek, this overwriting the seek.
Factor all refresh seek handling into a new function, and make it handle
ongoing seeks correctly.
Remove the weird new canonical_pts field, which actually had no use.
Fixes #4757.
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This oddly triggers bogus EOF when switching filter graphs between two
audio files (in this case, "[vid1]f[vo];[aid2]f[ao]"->"[aid1]f[ao]",
with aid2 being an external audio source).
This commit also fixes desync when seeking with an external file
connected via --lavfi-complex.
(Yes, the audio resync code is cursed.)
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Tends to be somewhat glitchy if subtitles are enabled, and you enable
and disable tracks.
On error, this will disable --lavfi-complex, which will result in
whatever behavior.
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Commit 0e0b87b6f3297 fixed that dropped packets did not trigger further
work correctly. But it also made trivial --lavfi-complex freeze. The
reason is that the meaning if DATA_AGAIN was overloaded: the decoders
meant that they should be called again, while lavfi.c meant that other
outputs needed to be checked again. Rename the latter meaning to
DATA_STARVE, which means that the current input will deliver no more
data, until "other" work has been done (like reading other outputs, or
feeding input).
The decoders never return DATA_STARVE, because they don't get input from
the player core (instead, they get it from the demuxer directly, which
is why they still can return DATA_WAIT).
Also document the DATA_* semantics in the enum.
Fixes #4746.
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Just the audio resync code in its normal state: buggy. This time,
AD_NO_PROGRESS was handled about the same as AD_WAIT. But it means the
decoder didn't output data, even though input is still readily
available.
This happened in particular when the timeline code was used (potentially
skipping many packets), and thus should fix #4688.
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Causea a simple integer overflow.
Fixes #4650.
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These files have all in common that they were fully or mostly taken from
mplayer.c. (mplayer.c was a huge file that contains almost all of the
playback core, until it was split into multiple parts.) This was
probably the hardest part to relicense, because so much code was moved
around all the time.
player/audio.c still does not compile. We'll have to redo audio
filtering. Once that is done, we can probably actually provide an
actual LGPL configure switch.
Here is a relatively detailed list of potential issues:
8d190244: author did not reply, parts were made GPL-only in a previous
commit.
7882ea9b: author could not be reached, but the code is gone. wscript
still has --datadir switch, but I don't think this is relevant to
copyright.
f197efd5: unclear origin, but I consider the code gone anyway (replaced
with generic OSD mechanisms).
8337d9c2: author did not reply, but only the option still exists (under
a different name), other code was removed.
d8fd7131: did not reply. Disabled in a previous commit.
05258251: same author as above. Both fields actually seem to have
vanished (even when tracking renames), so no action taken.
d459e644, 268b2c1a: author did not reply, but we reuse only the options
(with different names and slightly or fully different semantics, and
completely different implementations), so I don't think this is relevant
for copyright.
09e742fe, 17c39c4e: same as above.
e8a173de, bff4b3ee: author could not be reached. The commands were
reworked to properties, and the code outside of the TV code were moved
back to the TV code. So I don't think copyright applies to the current
command.c parts (mp_property_tv_color, mp_property_tv_freq,
mp_property_tv_scan). The TV parts remain GPL.
0810e427: could not be reached. Disabled in a previous commit.
43744a2d: unknown author, but this was replaced by dynamic alloc (if the
change is even copyrightable).
116ca0c7: unknown author; reasoning see input.c relicensing commit.
e7e4d1d8: these semantics still exist, but as generic code, and this
code was fully removed.
f1175cd9: the author of the cited patch is unknown, and upon inspection
it turns out that I was only using the idea to pause the player on EOF,
so I claim it's not copyright relevant.
25affdcc: author could not be reached (yet) - but it's only a function
rename, not copyrightable.
5728504c was committed by Arpi (who agreed), but hints that it might be
by a different author. In fact it seems to be mostly this patch:
http://lists.mplayerhq.hu/pipermail/mplayer-dev-eng/2001-November/002041.html
The author did not respond, but it all seems to have been removed later.
It's a terrible mess though. Arpi reverted the A-V sync code at first,
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