Split out some utility functions that use the CoreAudio API but are not related
the main task of the AOs (which is to move data correctly to the ringbuffer).
These are mainly need for the verbosity of the CoreAudio API and are just
obscuring the 'real' code.
Read only the requested amount by the AUHAL (instead of all the buffered data).
No idea what the deal is with pausing the audio units if there is no audio to
play, maybe to avoid underruns of some sort. Anyway from my tests this
condition never occurred so I'm removing it all.
Make the VF/VO/AO option parser available to audio filters. No audio
filter uses this yet, but it's still a quite intrusive change.
In particular, the commands for manipulating filters at runtime
completely change. We delete the old code, and use the same
infrastructure as for video filters. (This forces complete
reinitialization of the filter chain, which hopefully isn't a problem
for any use cases. The old code forced reinitialization too, but it
could potentially allow a filter to cache things; e.g. consider loaded
ladspa plugins and such.)
This code is supposed to run if dynamic filter insertion (such as when
inserting a volume filter in mixer.c) fails. Then it removes all filters
and recreates the default list of filters. But the code just blew up and
entered an endless loop, because it removed even the sentinel in/out
filters. This could happen when trying to use softvol controls while
using spdif, but also other situations. Fix it by calling the correct
code.
Also remove these obnoxious yoda-conditions.
MSDN tells me to multiply the samplerates by 4 (for setting up the S/PDIF
signal frequency), but doesn't mention that I'm only supposed to do it
on the new, NT6.1+ IEC 61937 structs. Works on my Realtek Digital Output,
but as I can't connect any hardware to it I can't hear the result.
Also, always ask for little-endian AC3. I'm not sure if this is supposed
to be LE or NE, but Windows is LE on all platforms, so we go with LE.
Entirely untested as this troper has no S/PDIF hardware.
Refuses trying any other format if we can't use passthrough, or we would
end up sending white noise at the user.
Do an strstr match against the device description and, if we have only
a single match, take it. This works as long as the devices in the system
don't change, but it's not supposed to be reliable; if one wants
reliability, one uses the device ID string.
Formatting.
This could turn valid parameters into syntax errors by the mere presence
or abscence of a device (e.g. USB audio devices), so don't do that.
We do validate that, if the parameter is an integer, it is not negative.
We also respond to the "help" parameter, which does the same as the "list"
suboption but exits after listing.
Demote the validation logging to MSGL_DBG2.
Validates by trying to pick the device using the device enumerator and
aborting with out of range on failure.
Refactors find_and_load_device to not use the wasapi_state; it might be
called during validation. Adds missing CoInitialize/CoUninitialize calls.
Remove unused variables (the SAFE_RELEASE macros keep them referenced so
compiler warnings don't help finding them...).
Remove the IMMDeviceEnumerator from the wasapi_state, it's only needed
during initialization and initialization is now well factored enough to
get rid of it.
Try and connect to unplugged devices as well when using the device ID
string.
Omit "{0.0.0.00000000}." on devices that start with that substring,
re-add when searching for devices by ID.
Log the device ID of the default device.
Log the friendly name of the used device.
Consistently refer to endpoints/devices as devices, as this is more
consistent with mpv terminology.
Uses WASAPI in shared mode by default, add :exclusive flag to choose
exclusive mode (duh). WASAPI works somewhat different in shared mode:
the OS suggests the sample format to use, and the GetBuffer call is
done slightly differently.
The shared mode driver does not consume audio as fast as it notifies
the thread; we need to check how much we're allowed to write. Not doing
this correctly results in spamming the console with
AUDCLNT_E_BUFFER_TOO_LARGE errors.
When guessing formats for exclusive mode, try several sample size and
sample rate combinations instead of just falling back to s16le@44100hz.
If none of the rates are accepted, tries remixing >6 channels to 5.1
channels. Failing that, tries remixing to stereo. Failing everything,
including the CD Red Book format, what else is left to test?
Calculate buffer_block_size based on the configured channels and bytes
per sample; MSDN docs say nBlockAlign is not guaranteed to be set for
anything but integer PCM formats.
Adds the :list suboption to ao_wasapi0, which enumerates the audio endpoints
in the system.
Adds the :device=<n> suboption, which either takes an ID string (as output by
list) or a device number and uses the requested device instead of the system
default.
These two options were supported by ALSA and OSS only. Further, their
values were specific to the respective audio systems, so it doesn't make
sense to keep them as top-level options.
This changes how device names are handled. Before this commit, device
names were mangled in strange ways to avoid clashing with the option
parser syntax. "." was replaced with ",", and "=" with ":" (the user had
to do the inverse to get the correct device name).
The "new" option parser has multiple ways to escape option strings, so
we don't need this confusing hack anymore.
Add an explicit note to the manpage as well.
Seeking calls thread_reset, but doesn't call thread_play. thread_reset
would disable WASAPI events, but they would never get re-enabled unless
the user paused and then unpaused.
Keep track of whether the stream is paused or not (there already was a
field for that, but it was apparently unused), and if it's not paused,
call thread_play after thread_reset. Fixes mpv freezing after seeks.
Fixes format specifies that assume windows TYPEDEFS are as long as they look
like they are.
Remove calls to _beginthreadex and _endthreadex, these are only present on
microsoft's C runtimes. Replace by the otherwise identical CreateThread and
ExitThread calls.
This actually requires fixes to devicetopology.h, but the problem has been
(kinda) reported to mingw-w64:
<Kovensky> I see that those KSJACK* structs are supposedly declared in
devicetopology.h itself, but for some reason (some of?) the decls that use
them aren't seeing them?
<Kovensky> ok, it seems that it expects ks.h and ksmedia.h to declare those
structs, but it doesn't
<Kovensky> the included files declare KDATAFORMAT, KSIDENTIFIER and LUID (and
the associated pointer typedefs)
<Kovensky> but everything else is essentially inside #if 0
<Kovensky> changing the #ifndef _KS_ to only include KDATAFORMAT, KSIDENTIFIER
and LUID (and putting the KSJACK stuff outside that #ifndef) makes the
header compile
<Kovensky> it solves my immediate problem, but if that happened to begin with
there's probably something more wrong with the ks headers :S
Matroska has an output sample rate (OutputSamplingFrequency), which in
theory should be forced instead of whatever the decoder outputs. But it
appears no software (other than mplayer2 and mpv until now) actually
respects this. Even worse, there were broken files around, which played
correctly with (in theory) broken software, but not mplayer2/mpv. Hacks
were added to our code to play these files correctly, but they didn't
catch all cases.
Simplify this by doing what everyone else does, and always use the
decoder's sample rate instead. In particular, we try to handle all
sample rate issues like libavformat's Matroska demuxer does.
It turns out that some code that was removed earlier was still needed.
avcodec_decode_audio4() can decode packets "partially". In that case,
you have to "slice" the packet and call the decode function again.
Codecs which need this are obscure and in low numbers. One sample that
needs it is here:
rsync://fate-suite.ffmpeg.org/fate-suite/lossless-audio/luckynight-partial.shn
(This one decodes in rather small increments.)
The new code is much simpler than what has been removed earlier,
though. The fact that we own the packet returned by the demuxer helps
a lot.
Not sure what should happen if avcodec_decode_audio4() returns 0.
Currently, we throw away the packet in this case. We don't want to be
stuck in an endless loop (could happen if the decoder produces no
output either).
This is not directly related to the handling of format changes itself,
but playing audio normally after the change. This was broken: the output
byte rate was not recalculated, so audio-video sync was simply broken.
Fix this by calculating the byte rate on the fly, instead of storing it
in sh_audio.
Format changes are relatively common (switches between stereo and 5.1
in TV recordings), so this fixes a somewhat critical bug.
pts_bytes can't just be changed at the end. It must be offset to the pts
value, which is reset with each packet read from the demuxer. Make sure
the pts_byte field is always reset after receiving a new PTS, i.e.
increment it after actually writing to the output buffer.
Flush the AVFormatContext's write buffer, because otherwise the audio
PTS will jump around too much: the calculation doesn't use the exact
output buffer size if there's still data in the avio buffer.
Partial packet reads were needed because the video/audio parsers were
working on top of them. So it could happen that a parser read a part of
a packet, and returned that to the decoder. With libavformat/libavcodec,
packets are already parsed, and everything is much simpler.
Most of the simplifications in ad_spdif could have been done earlier.
Remove some other stuff as well, like the questionable slave mode start
time reporting (could be replaced by proper code, but we don't bother).
Remove the unused skip_audio_frame() functionality as well (it was used
by old demuxers). Some functions become private to demux.c, like
demux_fill_buffer(). Introduce new packet read functions, which have
simpler semantics. Packets returned from them are owned by the caller,
and all packets in the demux.c packet queue are considered unread.
Remove special code that dropped subtitle packets with size 0. This
used to be needed because it caused special cases in the old code.
We don't need to deal with partial packet reads, manually using an audio
parser, or having to call the libavcodec decoder multiple times per
packet.
Actually, I'm not sure about the last point. ffplay still does this, but
the ffmpeg demuxing.c example doesn't.
The audio parser was needed only by the "old" demuxers, and
demux_rawaudio. All other demuxers output already parsed packets.
demux_rawaudio is usually for raw audio, so using a parser with it
doesn't usually make sense. But you can also force it to read
compressed formats with fixed packet sizes, in which case the parser
would have been used. This use case is probably broken now, but you
will be able to do the same thing with libavformat demuxers.
Delete demux_avi, demux_asf, demux_mpg, demux_ts. libavformat does
better than them (except in rare corner cases), and the demuxers have
a bad influence on the rest of the code. Often they don't output
proper packets, and require additional audio and video parsing. Most
work only in --no-correct-pts mode.
Remove them to facilitate further cleanups.
The core didn't use these fields, and use of them was inconsistent
accross AOs. Some didn't use them at all. Some only set them; the values
were completely unused by the core. Some made full use of them.
Remove these fields. In places where they are still needed, make them
private AO state.
Remove the --abs option. It set the buffer size for ao_oss and ao_dsound
(being ignored by all other AOs), and was already marked as obsolete. If
it turns out that it's still needed for ao_oss or ao_dsound, their
default buffer sizes could be adjusted, and if even that doesn't help,
AO suboptions could be added in these cases.
Some still do, because they use the value in other places of the init
function. ao_portaudio is tricky and reads ao->bps in the stream
thread, which might be started on initialization (not sure about that,
but better safe than sorry).
Currently every single AO was implementing it's own ringbuffer, many times
with slightly different semantics. This is an attempt to fix the problem.
I stole some good ideas from ao_portaudio's ringbuffer and went from there.
The main difference is this one stores wpos and rpos which are absolute
positions in an "infinite" buffer. To find the actual position for writing /
reading just apply modulo size.
The producer only modifies wpos while the consumer only modifies rpos. This
makes it pretty easy to reason about and make the operations thread safe by
using barriers (thread safety is guaranteed only in the Single-Producer/Single-
Consumer case).
Also adapted ao_coreaudio to use this ringbuffer.
This is hopefully the start of something good. ca_ringbuffer_read and
ca_ringbuffer_write can probably cleaned up from all the NULL checks once
ao_coreaudio.c gets simplyfied.
Conflicts:
audio/out/ao_coreaudio.c
Whatever this was supposed to be originally, it doesn't have much value
anymore. It just forced ad_mpg123 to upmix mono to stereo by default
(the audio chain can do that). As an option, it was mostly useless and
misleading, so get rid of it.
This was overlooked with commit 32a898f, because OSS4 volume control is
typically not available on Linux. BSD does have this feature, so the
broken code broke compilation there.
Fixes crashes when playing with certain numbers of channels. The core
assumes AOs accept data aligned on channels * samplesize, and ao_jack's
play() function broke that assumption:
mpv: core/mplayer.c:2348: fill_audio_out_buffers: Assertion `played % unitsize == 0' failed.
Fix by aligning the buffer and chunk sizes as needed.
Audio and video had their own (very similar) functions to initialize an
AVPacket (ffmpeg's packet struct) from a demux_packet (mplayer's packet
struct). Add a common function for these.
Also use this function for sd_lavc_conv. This is actually a functional
change, as some libavfilter subtitle demuxers add weird out-of-band
stuff as side-data.
