| Commit message (Collapse) | Author | Age | Files | Lines |
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Small adjustments to the playback speed use swr_set_compensation()
to stretch the audio as it is required. But since large adjustments
are now handled by actually reinitializing libswresample, the small
adjustments get rounded off completely with typical frame sizes.
Compensate for this by accounting for the rounding error and keeping
track of fractional samples that should have been output to achieve
the correct ratio.
This fixes display sync mode behavior, which requires these adjustments
to be relatively accurate.
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swr/avresample_set_compensation() was made for small speed adjustments.
Non-documentation says it should be used for changes not larger than 1%,
so reinitialize the sampler if the change is larger than that.
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swr_set_compensation() changes the apparent sample rate on the fly (who
would have guessed). It is thus very well-suited for adjusting audio
speed on the fly during playback (like needed by the display-sync mode).
It skips the relatively slow resampler reinitialization.
If this doesn't work (libswresample soxr backend), then fall back to the
old method.
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Not sure why struct af_resample_opts even exists. It seems useful to
group the fields set by user options. But storing the current format
conversion parameters doesn't seem very elegant, and having a separate
instance in the "ctx" field isn't helpful either.
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Apparently, this broke compilation with Libav under some circumstances.
Looking at it again, it shouldn't have, but this change doesn't hurt
anyway.
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It was never used, but is a leftover from old times.
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This was a minor optimization to potentially avoid resampler
reconfiguration when the filter is reinitialized. But filter
reinitialization is a rare event, and the case when no reconfiguration
is needed is even rarer. As such, this is an unnecessary micro-
optimization and only adds potential for bugs.
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This message bloats verbose log output if e.g. audio speed is frequently
readjusted, such as when syncing audio to video. So don't print the
message if only speed is changed. (This case requires reconfiguration,
but can't change the input/output channel maps.)
Also do not print the message if no remixing is done at all.
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With all the reordering etc. that can go on in this filter, it's useful
to see what upmix/downmix it's actually performing.
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Replace all the check macros with function calls. Give them all the
same case and naming schema.
Drop af_fmt2bits(). Only af_fmt2bps() survives as af_fmt_to_bytes().
Introduce af_fmt_is_pcm(), and use it in situations that used
!AF_FORMAT_IS_SPECIAL. Nobody really knew what a "special" format
was. It simply meant "not PCM".
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This avoids keeping "bad" state from previous reconfig calls, such as
the internal_sample_format option (which is set only on the first
reconfig call).
There's no advantage to keeping the resample contexts around anyway.
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mp_format is not a libavresample input format here, and the comment was
more confusing than it helped.
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Basically, af_fix_format_conversion() behaves stupid you insert a
conversion filter that won't work, and adding back the conversion test
function is the simplest fix to it.
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libswresample verbosely complains.
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The 24 bit conversion code needs the relevant preprocessor symbols.
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This attempted to find a minimal filter graph for a format conversion
involving multiple conversion filters. With the last 2 commits it
becomes dead code - remove it.
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Now af_lavrresample can output 24 bit samples directly, by doing the
conversion "inline". Luckily, S32->S24 can be done in-place, so this
isn't too much work. But the output conversion logic (which seems to be
adding up) gets slightly more complicated again.
Normally this is done by af_convert24. But having multiple conversion
filters complicates some aspects of the filter chain. S24 output is the
only thing the code for multiple conversion filters is still needed for,
and getting rid of that is preferable.
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If the code path for additional output conversion is active,
reorder_planes() is always called, even if the reorder_out array wasn't
filled. This is obviously wrong - always fill this array.
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Until now, we didn't do this, because it required some effort, and
didn't seem to be necessary. It probably still isn't, but it sounds
like a good idea not to output arbitrary data on these channels.
The situation is complicated by the fact that just adding new channels
to a planar frame would require messing with buffers. So we would have
to allocate new buffers and add them to the frame. We could have to
maintain an extra buffer pool for this. Avoid this by being "clever",
and just allocate a frame with enough channels in the first place.
libav/swresample won't know about these channels and won't write to
them, but we can grab them in reorder_planes() and use them for the
NA channels.
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On libavresample, don't ignore the buffered output data.
On libswresample, don't round the total buffer size to the input
samplerate.
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It was recently added to libswresample, and it does exactly what we
need.
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This is better, because now we call swr_get_delay() with the output
samplerate, instead of with the input samplerate and then multiplying it
with the ratio and rounding it up.
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This also drops the unused get_drain_samples() function.
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This manually retrieved the remaining audio from the resampler. It
subtly missed a conversion which could leave to an unsubtle crash.
This could happen if reorder_planes() was supposed to insert NA
channels, and the resampler/actual output format were different.
Simplify it by reusing the normal drain path. One oddness is that
the filter will add an output frame outside of normal filtering,
but that should be fine.
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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.)
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configure_lavrr() clears s->pending, so we have to assign it after that
call.
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Signed-off-by: wm4 <wm4@nowhere>
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The in/out pointers usually have not much meaning outside of
AF_CONTROL_REINIT. Also remove the redundant casts.
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It must be allowed to set format==0.
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af_fmt_is_float and af_fmt_is_planar were previously inconsistent with
AF_FORAMT_IS_SPECIAL/AF_FORMAT_IS_IEC61937
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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.
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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.)
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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.)
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The filter output size can be 0. Due to how filtering works, this is
nothing unusual, but avresample_convert() will return 0. The same case
is already handling with "normal" resampling (this commit fixes the
reordering code).
Additionally, don't use an assert(). avresample_convert() failing is
unusual, but might also happen due to e.g. internal out of memory
conditions, so we shouldn't just crash on it.
Curiously observed with --ao=oss --audio-channels=5.1 when changing
speed.
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Use OPT_KEYVALUELIST() for all places where AVOptions are directly set
from mpv command line options. This allows escaping values, better
diagnostics (also no more "pal"), and somehow reduces code size.
Remove the old crappy option parser (av_opts.c).
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While I'm not very fond of "const", it's important for declarations
(it decides whether a symbol is emitted in a read-only or read/write
section). Fix all these cases, so we have writeable global data only
when we really need.
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This function is now always available.
Also remove includes of reorder_ch.h from some AOs (these are just old
relicts).
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These are now appended automatically, so you'd get them twice before
this commit.
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Since m_option.h and options.h are extremely often included, a lot of
files have to be changed.
Moving path.c/h to options/ is a bit questionable, but since this is
mainly about access to config files (which are also handled in
options/), it's probably ok.
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The new signature is actually closer to how it actually works, and
someone who is not familiar to the API and how it works might make fewer
fatal mistakes with the new signature than the old one. Pretty weird.
Do this to sneak in a flags parameter, which will later be used to flush
remaining data of at least vf_lavfi.
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This could lead to (barely) audible artifacts with --af=scaletempo and
modified playback speed.
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The AF control commands used an elaborate and unnecessary organization
for the command constants. Get rid of all that and convert the
definitions to a simple enum. Also remove the control commands that
were not really needed, because they were not used outside of the
filters that implemented them.
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Regression introduced with commit a89549e8.
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You get the native format by not appending any suffix to the format.
This change includes user-facing names, e.g. for the --format option.
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Before this commit, the af_instance->mul/delay values were in bytes.
Using bytes is confusing for non-interleaved audio, so switch mul to
samples, and delay to seconds. For delay, seconds are more intuitive
than bytes or samples, because it's used for the latency calculation.
We also might want to replace the delay mechanism with real PTS
tracking inside the filter chain some time in the future, and PTS
will also require time-adjustments to be done in seconds.
For most filters, we just remove the redundant mul=1 initialization.
(Setting this used to be required, but not anymore.)
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Allocate af_instance->data in generic code before filter initialization.
Every filter needs af->data (since it contains the output
configuration), so there's no reason why every filter should allocate
and free it.
Remove RESIZE_LOCAL_BUFFER(), and replace it with mp_audio_realloc_min().
Interestingly, most code becomes simpler, because the new function takes
the size in samples, and not in bytes. There are larger change in
af_scaletempo.c and af_lavcac3enc.c, because these had copied and
modified versions of the RESIZE_LOCAL_BUFFER macro/function.
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Based on earlier work by Stefano Pigozzi.
There are 2 changes:
1. Instead of mp_audio.audio, mp_audio.planes[0] must be used.
2. mp_audio.len used to contain the size of the audio in bytes. Now
mp_audio.samples must be used. (Where 1 sample is the smallest unit
of audio that covers all channels.)
Also, some filters need changes to reject non-interleaved formats
properly.
Nothing uses the non-interleaved features yet, but this is needed so
that things don't just break when doing so.
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My main problem with this is that the output format will be incorrect.
(This doesn't matter right, because there are no samples output.)
This assumes all audio filters can deal with len==0 passed in for
filtering (though I wouldn't see why not).
A filter can still signal an error by returning NULL.
af_lavrresample has to be fixed, since resampling 0 samples makes
libavresample fail and return a negative error code. (Even though it's
not documented to return an error code!)
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If filter initialization fails anyway, we don't need to reconfigure
libavresample.
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Helps with readability. Also remove the ctx_opt_set_* helper macros and
use av_opt_set_* directly (I think these macros were used because the
lines ended up too long, but this commit removes two indentation levels,
giving more space).
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The configure followed 5 different convetions of defines because the next guy
always wanted to introduce a new better way to uniform it[1]. For an
hypothetic feature 'hurr' you could have had:
* #define HAVE_HURR 1 / #undef HAVE_DURR
* #define HAVE_HURR / #undef HAVE_DURR
* #define CONFIG_HURR 1 / #undef CONFIG_DURR
* #define HAVE_HURR 1 / #define HAVE_DURR 0
* #define CONFIG_HURR 1 / #define CONFIG_DURR 0
All is now uniform and uses:
* #define HAVE_HURR 1
* #define HAVE_DURR 0
We like definining to 0 as opposed to `undef` bcause it can help spot typos
and is very helpful when doing big reorganizations in the code.
[1]: http://xkcd.com/927/ related
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Drop the author and comment fields. They were completely unused - not
even printed in verbose mode, just dead weight.
Also use designated initializers and drop redundant flags.
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Fixes #304.
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Followup commit. Fixes all the files references.
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Also add a "o" suboption, which should allow fine control over
libavresample.
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