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Commit Graph

233 Commits

Author SHA1 Message Date
wm4
0025030cef af: don't attempt to remove last filter for spdif filter removal
Some time ago, a mechanism was added for automatically removing PCM-only
filters if the input format is spdif.

This could cause an infinite loop if the AO did not support spdif, but
was falling back to some PCM format. Then this code tried to remove the
last filter, which is a dummy filter for receiving and queuing filter
output. af_remove() simply fails gracefully in this case, so this
happens over and over again.

Fix by explicitly checking whether the filter to remove is a dummy
filter. (af_remove() also fails only if the dummy filters are attempted
to be removed - checking this directly is simpler.)
2015-05-05 21:47:48 +02:00
wm4
f719b8164d af_lavrresample: remove dead undefs 2015-05-05 01:11:16 +02:00
wm4
06050aed99 audio: introduce support for padding channels
Some audio APIs explicitly require you to add dummy channels. These are
not rendered, and only exist for the sake of the audio API or hardware
strangeness. At least ALSA, Sndio, and CoreAudio seem to have them.

This commit is preparation for using them with ao_coreaudio.

The result is a bit messy. libavresample/libswresample don't have good
API for this; avresample_set_channel_mapping() is pretty useless.
Although in theory you can use it to add and remove channels, you
can't set the channel counts. So we do the ordering ourselves by making
sure the audio data is planar, and by swapping the plane pointers. This
requires lots of messiness to get the conversions in place. Also, the
input reordering is still done with the "old" method, and doesn't
support padded channels - hopefully this will never be needed. (I tried
to come up with cleaner solutions, but compared to my other attempts,
the final commit is not that bad.)
2015-05-05 01:11:16 +02:00
wm4
d8dd4b6c39 af_lavrresample: fix draining
configure_lavrr() clears s->pending, so we have to assign it after that
call.
2015-04-18 13:39:40 +02:00
Marcin Kurczewski
f43017bfe9 Update license headers
Signed-off-by: wm4 <wm4@nowhere>
2015-04-13 12:10:01 +02:00
wm4
ab2a27ae01 af_lavrresample: minor simplification
The in/out pointers usually have not much meaning outside of
AF_CONTROL_REINIT. Also remove the redundant casts.
2015-04-12 18:07:05 +02:00
wm4
f8a98fc133 af_lavrresample: allow resetting output sample format
It must be allowed to set format==0.
2015-04-12 18:07:05 +02:00
wm4
e466a735a3 audio/filter: fully renegotiate audio formats on every reconfig
It could happen that a lavrresample filter would keep its old output
format when the decoder changed its output format. This simply happened
because the output format was never reset.

Normally, this was not an issue, because lavrresample filters only
inserted for format conversion were removed on format changes. But if
--no-audio-pitch-correction is set and playback speed is changed, then
there is a "permanent" lavrresample filter in the filter chain, which
shows this behavior.

Fix by explicitly resetting output formats for all filters which support
it.

Note: this can crash with libswresample in some cases. I'm not sure if
this is mpv's fault or libswresample's, but since it works with
libavresample, I'm going to assume it's not our's.
2015-04-12 18:06:23 +02:00
wm4
36ae8a6cab audio: automatically deatch filters if spdif prevents their use
Fixes #1743 and partially #1780.
2015-04-07 21:38:39 +02:00
wm4
579c4dac34 audio: change a detail about filter insertion
The af_add() function has a problem: if the inserted filter returns
AF_DETACH during init, the function will have a dangling pointer. Until
now this was avoided by making sure none of the used filters actually
return AF_DETACH, but it's getting infeasible.

Solve this by requiring passing an unique label to af_add(), which is
then used instead of the pointer.
2015-04-07 21:24:22 +02:00
Kevin Mitchell
46b9df9f9e audio: make all format query shortcuts macros
af_fmt_is_float and af_fmt_is_planar were previously inconsistent with
AF_FORAMT_IS_SPECIAL/AF_FORMAT_IS_IEC61937
2015-04-03 15:40:01 -07:00
wm4
bf69edb1c2 af_lavrresample: always normalize (libswresample is stupid)
libswresample doesn't normalize when remixing to a float format. This
will cause clipping due to float samples being out of the allowed range.
Fortunately this extremely bad default can be changed.

This does not happen with libavresample: it normalizes by default.

Fixes #1752.
2015-04-02 00:42:54 +02:00
wm4
f5603cba23 af: remove unused functions 2015-04-01 21:39:40 +02:00
wm4
7205e75079 af_bs2b: fix option default value
--af=bs2b:help abort()ed because the default value of the "profile"
option is not represented by any choice. Fix it by adding an "unset"
choice. (It's a bit odd because there's already a "default" choice,
which is not default, but I don't care enough about this filter.)

Fixes #1712.
2015-03-22 13:28:20 +01:00
wm4
775a02aab5 af_lavfi: handle seeking
To handle seeking correctly, we need to flush the filter. libavfilter
does not support flushing, so we destroy and recreate it. We also need
to handle resume-after-EOF, because the mpv audio code sends an EOF
before and after seeking (the latter happens because the player drains
the filter chain in a generic way, which "causes" EOF).
2015-03-17 22:31:05 +01:00
wm4
89db92398e audio: refuse to change playback speed with spdif
Handle the failure gracefully, instead of exploding and disabling audio.
Just set the speed back to 1.0.

Also remove the AF_DETACH from af_scaletempo. This actually created a
dangling pointer in af_add(), a tricky consequence of af_add()
reconfiguring the filter chain and the newly added filter using
AF_DETACH. Fortunately the AF_DETACH is not needed (and probably never
worked - it comes from MPlayer times, and MPlayer also disables audio
when trying to change speed with spdif).
2015-03-07 20:34:05 +01:00
wm4
ddbecd09b0 af_scaletempo: minor simplification 2015-03-06 21:51:18 +01:00
wm4
c30d5f79b5 af_scaletempo: restore confusing mplayer behavior
This matters only when setting obscure scaletempo suboptions.

See #1653.

(But what we really should do is figuring out how to do this in a sane
way.)
2015-03-06 21:48:41 +01:00
wm4
89bc2975e9 audio: change playback speed directly in resampler
Although the libraries we use for resampling (libavresample and
libswresample) do not support changing sampelrate on the fly, this makes
it easier to make sure no audio buffers are implicitly dropped. In fact,
this commit adds additional code to drain the resampler explicitly.

Changing speed twice without feeding audio in-between made it crash
with libavresample inc ertain cases (libswresample is fine). This is
probably a libavresample bug. Hopefully this will be fixed, and also I
attempted to workaround the situation that crashes it. (It seems to
point in direction of random memory corruption, though.)
2015-03-02 19:09:44 +01:00
Martin Herkt
a17ea73636 af_rubberband: actually fix deadlock
371e5d0 missed this one
2015-02-12 10:15:12 +01:00
wm4
371e5d0665 af_rubberband: fix filter error deadlock
rubberband_available() can return a negative value, which we assigned to
a size_t variable, leading to the frame allocation to fail. This could
spam "Error filtering frame.". (That it spams this instead of exiting
should probably also be considered a bug.)

At least in the realtime mode and in our case, a negative return value
should not have any different meaning from a 0 return value, in
particular because we call rubberband_get_samples_required() or set the
"final" parameter for rubberband_process() to continue/stop processing.
2015-02-12 09:47:01 +01:00
Martin Herkt
2dc49ea866 af_rubberband: change defaults
After some testing, I am fairly convinced that these defaults sound
better than the previous settings. This also eliminates some issue
with random crackling and noise.

Also remove the `stretch` option since it has no effect in
realtime mode.
2015-02-12 00:58:40 +01:00
wm4
6299da2047 af_rubberband: fix breakage
The previous commit on this filter accidentally removed the
RubberBandOptionProcessRealTime option. Without it, the lib prints a
warning and passes the audio through.

Also add the RubberBandOptionSmoothingOn option back. Though for some
reason the output sounds still very wrong.
2015-02-11 21:32:01 +01:00
wm4
df5548a754 af_rubberband: make all librubberband options configurable
librubberband exports a big load of options. Normally, the default
settings (whether they're librubberband defaults or our defaults) should
be sufficient, but since I'm not so sure about this, making it
configurable allows others to figure it out for me.
2015-02-11 17:11:05 +01:00
wm4
6f24a61d84 af_rubberband: attempt to fix audio position calculation
The problem here is that librubberband can buffer an arbitrary amount
of data, but at the same time doesn't provide a way to query how much
data is buffered. So we keep track of this manually, assuming that
librubberband tries to reach the requested time ratio for input and
output (which is probably true).

The disadvantage is that rounding errors could accumulate over time.
(Maybe it should try to round towards keeping the time ratio.)
2015-02-11 16:32:40 +01:00
wm4
76501f4f57 af_rubberband: always calculate and set delay
Basically, add an if and reindent the block instead of exiting early.
2015-02-11 16:32:40 +01:00
wm4
d85aa35ffb af: account for queued frames in audio position calculation
af_rubberband exposed this issue.
2015-02-11 16:32:40 +01:00
wm4
8c055f873f af_rubberband: improve EOF handling
In theory it could happen that draining on EOF happens incrementally,
and then the unconditional reset could have dropped the remaining
buffered audio.
2015-02-11 16:31:35 +01:00
wm4
b6ab34fc98 af_rubberband: pitch correction with librubberband
If "--af=rubberband" is used, librubberband will be used to speed up or
slow down audio with pitch correction.

This still has some problems: the audio delay is not calculated
correctly, so the audio position jitters around by a few milliseconds.
This will probably ruin video timing.
2015-02-11 00:29:12 +01:00
wm4
81d8c5d519 af_scaletempo: allow changing speed at runtime without reinit
Staring at the code a bit, it turns out that changing speed without
losing state is quite easy. The initialization code is big and
complicated, but most of it is specific only to the configured audio
format, not the speed.

Refactor the code so that changing speed at runtime could work. (It's
not actually used yet - the player code still does a complete reinit.
This will be fixed in the next commit.)

The "if (s->speed_tempo == s->speed_pitch)" looks a bit strange, but
does the same thing as the code did before: speed can be changed only if
exactly one flag is set. If both are set or none, speed can't be
changed.
2015-02-10 22:34:07 +01:00
wm4
2a3d19a9df af_scaletempo: drop detaching or skipping init on speed=1
This code skipped initialization if no speed/pitch change was to be
applied.

It also didn't force conversion of the audio to a supported format,
which is probably the most important case in context of compatibility.
With this change applied, af_scaletempo will always force format
conversion.

To make the change less disruptive, make the filter detach if
unconvertable formats are used. Some users use spdif and also have
"af=scaletempo" in their config, so better not completely break this.

In the case the filter was added with the "speed=both" suboption, the
filter also detached itself in this case; but it's an obscure case, so I
don't care about that.
2015-02-10 22:14:26 +01:00
wm4
ae641d200a af: remove old filter compatibility hack 2015-01-15 20:13:15 +01:00
wm4
388cf6dc96 audio/filter: switch remaining filters to refcounting
All of these filters are very similar in frame management, and copy data
to a new frame during filtering.
2015-01-15 20:13:14 +01:00
wm4
87fe7d8788 audio/filter: switch remaining in-place filters to refcounting
Adds about 7 lines of boilerplate per filter. This could be avoided by
providing a different entrypoint (something like af->filter_inplace),
which would basically mirror the old interface exactly for this kind of
filter. But I feel like it would just be a hack to support all those
old, useless filters better. (The ideal solution would be using a
language that can do closures to provide a compat. wrapper, but
whatever.)

af_bs2b has terribly repetitious code for setting up filter functions
for each format (most of them useless, in addition to bs2b being
useless), so I did something terrible with macros.

af_sinesuppress had commented code for float filtering (maybe it was
broken; it has been commented every since it was added in 2006). Remove
this code.
2015-01-15 20:13:12 +01:00
wm4
ba0e8b754c af: verify filter input formats
Just to make sure all filters get the correct format. Together wih the
check in af_add_output_frame(), this asserts that

    af->prev->fmt_out == af->fmt_in

This also requires setting the "in" pseudo-filter (s->first) formats
correctly. Before this commit, the fmt_in/fmt_out fields weren't used
for this filter.
2015-01-15 20:10:46 +01:00
wm4
e865d255d0 af_lavcac3enc: use refcounted frames 2015-01-14 22:16:30 +01:00
wm4
5d972491bb af_lavfi: use refcounted frames 2015-01-14 22:15:56 +01:00
wm4
9c974b2a1b audio/filter: actually set fmt_in/fmt_out fields 2015-01-14 22:15:51 +01:00
wm4
f6a0a1554c af_scaletempo: use refcounted frames 2015-01-14 22:15:39 +01:00
wm4
218c749a16 af_lavrresample: use refcounted frames 2015-01-14 22:15:31 +01:00
wm4
ecca64e182 af_convert24: use refcounted frames
This requires allocating a fully new frame. 32->24 could be in-place,
but this is not possible for 24->32.
2015-01-13 20:17:08 +01:00
wm4
983f5efa3c audio/filters: use refcounted frames for some in-place filters
These are also quite simple, but require requesting write access to the
frames. The error handling (for OOM) is a bit annoying.
2015-01-13 20:17:03 +01:00
wm4
1fde40732e audio/filters: use refcounted frames for some simple filters
These are read-only, and very trivial to convert.
2015-01-13 20:16:59 +01:00
wm4
772c42a95c af_volume: use refcounted frames 2015-01-13 20:15:53 +01:00
wm4
5e25a3d216 audio: use refcounted frames in the filter chain
The goal is switching the whole audio chain to using refcounted frames.
This brings the architecture closer to FFmpeg, enables better
integration with libavfilter, will reduce useless copying somewhat, and
will probably allow better timestamp tracking.

For now, every filter goes through a semi-awful wrapper in
af_do_filter(), though. This will be fixed step by step, and the wrapper
should eventually be removed. Another thing that will have to be done is
improving the timestamp handling and avoiding extra copies for the AO.

Some of the new code is rather similar to the video filter code (the
core filter code basically just has types replaced). Such code
duplication is normally very unwanted, but in this case there's probably
no other choice. On the other hand, this code is pretty simple (even if
somewhat tricky). Maybe there will be unified filter code in the future,
but this is still far away.
2015-01-13 20:15:43 +01:00
wm4
0bbd65b09c audio/filter: remove unused af_calc_filter_multiplier()
The purpose of this function was to filter only as much audio input as
needed to produce a certain amount of audio output. This could (in
theory) avoid excessive buffering when e.g. changing playback speed with
resampling.

Use of this was already removed in commit 5fd8a1e0. No problems were
experienced, so let's assume this feature is practically worthless.
(Though it's possible that it was quite useful over a decade ago, or in
some cornercases with evil files.)
2015-01-13 20:14:02 +01:00
wm4
fda44ecc92 af_volume: dump applied replaygain in verbose mode 2015-01-04 01:35:48 +01:00
wm4
3fdb6be316 win32: add mmap() emulation
Makes all of overlay_add work on windows/mingw.

Since we now don't explicitly check for mmap() anymore (it's always
present), this also requires us to make af_export.c compile, but I
haven't tested it.
2014-12-26 17:30:10 +01:00
reimar
13b4fb9d28 af_hrtf: Fix out-of-range read.
Based on patch by Yuriy Kaminskiy [yumkam gmail].

git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@37330 b3059339-0415-0410-9bf9-f77b7e298cf2
Signed-off-by: wm4 <wm4@nowhere>
2014-12-06 17:09:57 +01:00
wm4
7d6e58471f audio: make mp_audio_config_to_str return a stack-allocated string
Simpler overall.
2014-11-25 11:11:31 +01:00