Until now, this was for AC3 only. For PCM, we used AudioUnit in
ao_coreaudio, and the only reason ao_coreaudio_exclusive exists
is that there is no other way to passthrough AC3.
PCM support is actually rather simple. The most complicated
issue is that modern OS X versions actually do not support
copying through the data; instead everything must go through
float. So we have to deal with virtual and physical format
being different, which causes some complications.
This possibly also doesn't support some other things correctly.
For one, if the device allows non-interleaved output only, we
will probably fail. (I couldn't test it, so I don't even know
what is required. Supporting it would probably be rather
simple, and we already do it with AudioUnit.)
Mapping of spdif formats was imperfect. Since the first format on the
list is somehow AAC, it was returned first, which is confusing, because
CoreAudio calls all spdif formats AC3. Since the spdif formats have some
rather arbitrary, reverse mapping the formats didn"t actually work
either. Fix by explicitly ignoring these when spdif is used.
Also, don't forget to set the samplerate in ca_asbd_to_mpformat(), or it
will work only in some cases.
May help with (supposedly) bad drivers, which can put the device into
some sort of broken state when trying to set a different physical
format. When the previous format is restored, it apparently recovers.
This might make the change-physical-format suboption more robust.
We can be pretty sure that AudioUnit will remix for us.
Before this commit, we usually upmixed to stereo, because the
stereo and multichannel layouts were the only whitelisted ones.
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".
Audio formats used a semi-clever schema to encode the properties of the
PCM encoding as bitfields into the format integer value.
The af_fmt_change_bits() implementation becomes a bit weird, but it's
an improvement to the rest of the code.
(I've always disliked it, so why not get rid of it.)
This may or may not fix some issues with the format switching
code. Actually, it seems somewhat unlikely, but then checking
the stream type isn't incorrect either, and is probably
something the API user should always be doing.
Originally, this was written for comparing the sample format only, but
ca_change_physical_format_sync() actually expects that the full format
is compared. (For all other uses it doesn't matter.)
The speaker replacement nonsense sometimes made blatantly incorrect
decisions. In this case, it prefered a 7.1(rear) upmix over outputting
5.1(side) as 5.1, which makes no sense at all. This happened because 5.1
and 7.1(rear) appeared equivalent to the final selection, as both of
them lose the sl-sr channels. The old code was too stupid to select the
one with the lower number of channels as well.
Redo this. There's really no reason why there should be a separate final
decision, so move the speaker replacement logic into the
mp_chmap_is_better() function.
Improve some other details. For example, we never should compare the
plain number of channels for deciding upmix/downmix, because due to NA
channels this is essentially meaningless. Remove the NA channels when
doing this comparison. Also, explicitly handle exact matches.
Conceptually this is not necessary, but it avoids that we have to
needlessly shuffle audio data around.
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.
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.
So apparently, this essentially happens when the kernel driver doesn't
implement write accesses in the channel map control. Which doesn't
necessarily mean that the channel map is unsupported, or that there is a
bug - it's just lazyness and a consequence of the terrible ALSA kernel
API for the channel mapping stuff.
In these cases, the channel count implicitly selects the channel map,
and snd_pcm_set_chmap() always fails with ENXIO.
I'm actually not sure what happens if dmix is on top of e.g. HDMI, which
actually lets you change the channel mapping.
I'm also not sure why commit d20e24e5d1614354e9c8195ed0b11fe089c489e4
(alsa-lib git repository) does not take care of this.
MPlayer traditionally had completely separate sh_ structs for
audio/video/subs, without a good way to share fields. This meant that
fields shared across all these headers had to be duplicated. This commit
deduplicates essentially the last remaining duplicated fields.
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.
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.
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.
They are useless. Not only are they actually rarely in use; but
libavcodec doesn't even output them, as libavcodec has no such sample
formats for decoded audio.
Even if it should happen that we actually still need them (e.g. if doing
direct hardware output), there are better solutions. Swapping the sign
is a fast and lossless operation and can be done inplace, so AO actually
needing it could do this directly.
If you wonder why we keep U8 instead of S8: because libavcodec does it.
Channel maps reported by the device as SND_CHMAP_TYPE_VAR can be freely
reordered. We don't use this much (out of laziness), but in this case
it's a simple way to reduce necessary reordering (which would be an
extra libavresample invocation), and to make debug output more readable.
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.
This is just a conceptual issue, since for now every channel count has
an associated standard layout.
But should the max. channel count ever be bumped, some things would stop
function if mp_chmap_from_channels() refused to work for any channel
count within the allowed range.
In the AVFrame-style system (which we inreasingly map our internal data
stuctures on), buffers and plane pointers don't necessarily have a 1:1
correspondence. For example, a single buffer could cover 2 or more
planes, all while other planes are covered by a second buffer, and so
on. They don't need to be ordered in the same way.
Change mp_audio_get_allocated_size() to retrieve the maximum size all
planes provide. This also considers the case of planes not pointing to
buffer start.
Change mp_audio_realloc() to reset all planes, even if corresponding
buffers are not reallocated. (The caller has to be careful anyway if it
wants to be sure the contents are preserved on realloc calls.)
If you try to play surround with dmix, it will advertise surround and
lets you set more than 2 channels, but will report a stereo channel map,
with the extra channels identified as NA. We could handle this now, but
we don't want to (because it's excessively stupid).
Do it only if the channel map is not what we requested, instead of just
acting if it contains NA entries at all. This avoids that we hurt
ourselves in the unlikely but possible case we actually have to use
channel maps with NA entries.
If the audio API takes a while for starting the audio callback, the
current heuristic can be off. In particular, with very short files, it
can happen that the audio callback is not called before playback is
stopped, so no audio is output at all.
Change draining so that it essentially waits for the ringbuffer to
empty. The assumption is that once the audio API has read the data
via the callback, it will always output it, even if the audio API
is stopped right after the callback has returned.
If a frame could only be partially filled with real audio data, the
silence wasn't written at the correct offset. It could have happened
that the remainder of the frame contained garbage.
(This didn't happen in the more common case of playing dummy silence.)
This provides a new method for enabling spdif passthrough. The old
method via --ad (--ad=spdif:ac3 etc.) is deprecated. The deprecated
method will probably stop working at some point.
This also supports PCM fallback. One caveat is that it will lose at
least 1 audio packet in doing so. (I don't care enough to prevent this.)
(This is named after the old S/PDIF connector, because it uses the same
underlying technology as far as the higher level protoco is concerned.
Also, the user should be renamed that passthrough is backwards.)
This deprecates the --ad-spdif-dtshd option, and replaces it with a
pseudo decoder. This means ad_spdif will report two decoders, "dts" and
"dts-hd", of which the second simply enables what the option did.
The --ad-spdif-dtshd option will actually be deprecated in the next
commit.
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.
Listening to kAudioDevicePropertyDeviceHasChanged does not send any
property change notifications when the device dies. Makes no sense,
but I suppose in CoreAudio logic a dead/removed device can't send
any notifications.
This caused the player to essentially pause playback if the audio
device was removed during playback.
Fix by listening to the kAudioHardwarePropertyDevices property too,
which will actually be sent in this specific case. Then, if
querying the already dead device fails, we know we have to reload.
In short, instead of letting the coreaudio property listener set atomic
flags (which are then polled), make the property listeners actually
active.
The format change listener used during audio output now simply calls
ao_request_reload() on its own. All code involved is thread-safe, so
there's no need to do it during this audio callback (we assumed the
callback was never run concurrently with itself).
The listener installed temporarily during ca_change_format() is changed
to post a semaphore. Get rid of the weird retry logic and replace it
with a flat loop + timeout. It appears the maximum wait time could be
2500ms; reduce the total timeout to 500ms instead.
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.
This brings the volume control closer to what is percepted as linear
volume change.
Adjust the --softvol-max default to roughly the old maximum (roughly
doubles the gain).
Now --volume takes an absolute volume, meaning it doesn't depend on
--softvol-max. 0 is still silence, and 100 now always means unchanged
volume. The OSD and the "volume" property are changed accordingly.
Also raise the minimum value of --softvol-max. A value below 100 makes
no sense and breaks the OSD.
Apparently some A/V receivers do not behave well if "normal" DTS is
passed through using the high bitrate spdif format normally used for
DTS-HD (other receivers are fine with it).
Parse the first packet passed to ad_spdif by decoding it with libavcodec
in order to get the profile. Ignore the --ad-spdif-dtshd if it's not
DTS-HD. (If the codec profile changes midstream, the user is out of
luck. But this is probably an insignificant corner case.)
I thought about parsing the bitstream, but let's not. While it probably
wouldn't be that much effort, we are trying to keep it down on codec
details here - otherwise we could just do our own spdif framing instead
of using libavformat's spdif pseudo-muxer.
Another possibility, using the codec parameters signalled by
libavformat, is disregarded. Our builtin Matroska decoder doesn't do
this, and also we do not want on the demuxer having to decode some
packets in order to retrieve codec params (as libavformat does).
Fixes#1949.
There is not much of a reason to have these wrappers around. Use POSIX
standard functions directly, and use a separate utility function to take
care of the timespec calculations. (Course POSIX for using this weird
format for time values.)
Drop mp_chmap_diff() (which is unused too now), and implement
mp_chmap_diffn() in a slightly simpler way. (Too bad there is no
standard function for counting set bits.)
Instead of somehow having 4 different cases with each their own weight,
do it with a single function that decides which channel layout is the
better fallback.
This is simpler, and also introduces new (fixed) semantics. The new test
added to test/chmap_sel.c actually works now. This is a mixed case with
no perfect upmix or downmix, but the better choice is the one which
loses the least channels from the original layout.
One test also changes. If the input is 7.1(wide-side), and the available
layouts are 7.1 and 5.1(side), the latter is now chosen instead of the
former. This makes sense: both layouts contain 6 out of 8 channels from
the original layout, but the 5.1(side) one is smaller. This follows the
general logic. The 7.1 layout has FLC/RLC speakers instead of BL/BR,
and judging by the names, "front left center" is completely different
from "back left". If these should be exchangeable, a separate exception
would have to be added.
Reuse MP_SPEAKER_ID_NA for this. If all mp_chmap entries are set to NA,
the channel layout has special "unknown channel layout" semantics, which
are used to deal with some corner cases.
Sometimes, ALSA will return channel layouts with padded channels (NA
speakers). Use them instead of failing.
This still includes the old "braindeath" code to retry with a layout
without NA channels. This might be helpful for performance, and also the
padded channel layout string looks confusing.
To be fair, I have not encountered a case yet which would really need
this, and for which the old "braindeath" code did not fix it.
volatile barely means anything.
The polling is kind of bad too, but relatively harmless as device
opening/closing is a rare event, and the format change is not expected
to take long.
Remove the pointless talloc call too (must have been a leftover
from previous refactoring).
No reason to keep them separate. It's an artifact from the old
ao_coreaudio.c, which kept usage of two different APIs in the same file.
Removes a forward reference too.
Instead of trying to use af_format_conversion_score() (which tries to be
all kinds of clever), just compare the raw bits as a quality measure. Do
this because otherwise, weird formats like padded 24 bit formats will be
excluded, even though they might be the highest precision formats for
some hardware.
This means that for now, the user would have to check whether the format
is usable at all before calling ca_asbd_is_better(). But since this is
currently only used for ao_coreaudio.c and for the physical format, it
doesn't matter.
If coreaudio-exclusive should get PCM support, the best would be to
revert this change, and to add support for 24 bit formats directly.
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.)
Move all of the channel map retrieval/negotiation code to a separate
file. This will (probably) be helpful when extending
ao_coreaudio_exclusive.c.
Nothing else changes, other than some minor cosmetics and renaming,
and changing some details for decoupling it from the ao_coreaudio.c
internals.
It appears this is the reason coreaudio-exclusive does not work without
explicitly specifying a device, even if the default device maps to
something passthrough-capable.
Instead of always picking a somehow better format over the previous one,
select a format that is equal to or better the requested format, but is
also reasonably close.
Drop the mFormatID comparison - checking the sample format handles this
already.
Make sure to exclude channel counts that can't be used.
If for example the physical format is set to stereo, the reported
multichannel layout will actually be stereo. It fixes itself only after
the physical format is changed.
ao_coreaudio uses AudioUnit - the OSX software mixer. In theory, it
supports multichannel audio just fine. But in practice, this might be
disabled by default, and the user is supposed to select a multichannel
base format in the "Audio MIDI Setup" utility.
This option attempts to change this setting automatically. Some possible
disadvantages and caveats are listed in the manpage additions. It is off
by default, since changing this might be rather bad behavior for a
normal application.
If for example the audio settings are set to 5.1 output, but the
hardware does 8 channels natively (HDMI), the reported channel
layout will have 2 dummy channels. To avoid falling back to stereo,
we have to write audio in this format to the device.
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.)
ca_label_to_mp_speaker_id() checked whether the last entry was >= 0, but
actually this condition was never true, and MP_SPEAKER_ID_UNKNOWN0 is
not negative.
This should for now be equivalent; it's merely more explicit and will
be required if we add PCM support.
Note that the property listeners actually tell you what property
exactly changed, but resolving the current listener mess would be too
hard. So check for changes manually.
Useful with some of the following commits.
ca_fill_asbd() should behave exactly as before.
Instead of actually implementing the inverse function of ca_fill_asbd(),
just loop over the (small) list of mpv functions and check if any mpv
equivalent to a given ASBD exists.
kAudioFormatFlagIsSignedInteger implicates that it's only used with
integer formats. The mpv internal flag on the other hand signals the
presence of a sign, and this is set on float formats.
Until now, this probably worked fine, because at least AudioUnit is
ignoring the uncorrect flag.
We always want to prefer upmix to downmix, as long as it makes sense.
Even if the upmix is not "perfect" (not just adding channels), we want
to prefer the upmix.
Cleanup for commit d3c7fd9d.
As indicated by the added test. In this case, fallback and downmix have
the same score, but fallback happens to give better results. So prefer
fallback over downmix.
(This is probably not a correct solution.)
Remove the old implementation for these properties. It was never very
good, often returned very innaccurate values or just 0, and was static
even if the source was variable bitrate. Replace it with the
implementation of "packet-video-bitrate". Mark the "packet-..."
properties as deprecated. (The effective difference is different
formatting, and returning the raw value in bits instead of kilobits.)
Also extend the documentation a little.
It appears at least some decoders (sipr?) need the
AVCodecContext.bit_rate field set, so this one is still passed through.
mp_chmap_from_channels_alsa() doesn't always succeed - there are a bunch
of channel counts for which no defined ALSA layout exists. Fallback to
stereo in this case. (Normally, this code path shouldn't happen at all.)
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.
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.
Silence the usually user-visible warning about unsupported channel maps.
This might be an ALSA bug, but ALSA will never fix this behavior anyway.
(Or maybe it's a feature.)
Log some other information that might be useful.
The message log level shouldn't get to decide whether something fails
or not. So replace the fatal error check on the verbose output code
path with a warning.
* unify passthrough and pcm exclusive mode format setting/testing
* set passthrough format parameters correctly
* support all of mpv's existing passthrough formats
* automatically test passthrough with exclusive mode and enable
exclusive if it succeeds, even if it was not explictly requested.
this obviates the need for --ao=wasapi,wasapi=exclusive
* if passthrough fails (such as the device doesn't support the
format), fallback to either exclusive pcm or shared mode depending
on what the user specified. Right now this isn't very useful as
it still fails due to the decoder path remainin stuck on spdif.
fixes#1742
wm4
Marcin Kurczewski
Kevin Mitchell