Remove the unnecessary indirection through ao fields.
Also fix the inverted result of AOCONTROL_HAS_TEMP_VOLUME. Hopefully the
change is equivalent. But actually, it looks like the old code did it
wrong.
With --gapless-audio=no, changing from one file to the next apparently
made it hang, until the player was woken up by unrelated events like
input. The reason was that the AO doesn't notify the player of EOF
properly. the played was querying ao_eof_reached(), and then just went
to sleep, without anything waking it up.
Make it event-based: the AO wakes up the playloop if the EOF state
changes.
We could have fixed this in a simpler way by synchronously draining the
AO in these cases. But I think proper event handling is preferable.
Fixes: #1069
CC: @mpv-player/stable (perhaps)
Logic for this was missing from pull.c. For push.c it was missing if the
driver didn't support it. But even if the driver supported it (such as
with ao_alsa), strange behavior was observed by users. See issue #933.
Always check explicitly whether the AO is in paused mode, and if so,
don't drain.
Possibly fixes#933.
CC: @mpv-player/stable
The mplayer1/2/mpv CoreAudio audio output historically contained both usage
of AUHAL APIs (these go through the CoreAudio audio server) and the Device
based APIs (used only for output of compressed formats in exclusive mode).
The latter is a very unwieldy and low level API and pretty much forces us to
write a lot of code for little workr. Also with the widespread of HDMI, the
actual need for outputting compressed audio directly to the device is getting
lower (it was very useful with S/PDIF for bandwidth constraints not allowing
a number if channels transmitted in LPCM).
Considering how invasive it is (uses hog/exclusive mode), the new AO
(`ao_coreaudio_device`) is not going to be autoprobed but the user will have
to select it.
In most places where af_fmt2bits is called to get the bits/sample, the
result is immediately converted to bytes/sample. Avoid this by getting
bytes/sample directly by introducing af_fmt2bps.
For some reason, the buffered_audio variable was used to "cache" the
ao_get_delay() result. But I can't really see any reason why this should
be done, and it just seems to complicate everything.
One reason might be that the value should be checked only if the AO
buffers have been recently filled (as otherwise the delay could go low
and trigger an accidental EOF condition), but this didn't work anyway,
since buffered_audio is set from ao_get_delay() anyway at a later point
if it was unset. And in both cases, the value is used _after_ filling
the audio buffers anyway.
Simplify it. Also, move the audio EOF condition to a separate function.
(Note that ao_eof_reached() probably could/should whether the last
ao_play() call had AOPLAY_FINAL_CHUNK set to avoid accidental EOF on
underflows, but for now let's keep the code equivalent.)
Since the addition of the AO feed thread, 200ms of latency (MIN_BUFFER)
was added to all push-based AOs. This is not so nice, because even AOs
with relatively small buffering (e.g. ao_alsa on my system with ~170ms
of buffer size), the additional latency becomes noticable when e.g.
toggling mute with softvol.
Fix this by trying to keep not only 200ms minimum buffer, but also 200ms
maximum buffer. In other words, never buffer beyond 200ms in total. Do
this by estimating the AO's buffer fill status using get_space and the
initially known AO buffer size (the get_space return value on
initialization, before any audio was played). We limit the maximum
amount of data written to the soft buffer so that soft buffer size and
audio buffer size equal to 200ms (MIN_BUFFER).
To avoid weird problems with weird AOs, we buffer beyond MIN_BUFFER if
the AO's get_space requests more data than that, and as long as the soft
buffer is large enough.
Note that this is just a hack to improve the latency. When the audio
chain gains the ability to refilter data, this won't be needed anymore,
and instead we can introduce some sort of buffer replacement function in
order to update data in the soft buffer.
Until now, this was always conflated with uninit. This was ugly, and
also many AOs emulated this manually (or just ignored it). Make draining
an explicit operation, so AOs which support it can provide it, and for
all others generic code will emulate it.
For ao_wasapi, we keep it simple and basically disable the internal
draining implementation (maybe it should be restored later).
Tested on Linux only.
This has 2 goals:
- Ensure that AOs have always enough data, even if the device buffers
are very small.
- Reduce complexity in some AOs, which do their own buffering.
One disadvantage is that performance is slightly reduced due to more
copying.
Implementation-wise, we don't change ao.c much, and instead "redirect"
the driver's callback to an API wrapper in push.c.
Additionally, we add code for dealing with AOs that have a pull API.
These AOs usually do their own buffering (jack, coreaudio, portaudio),
and adding a thread is basically a waste. The code in pull.c manages
a ringbuffer, and allows callback-based AOs to read data directly.
We want to move the AO to its own thread. There's no technical reason
for making the ao struct opaque to do this. But it helps us sleep at
night, because we can control access to shared state better.
Always pass around mp_log contexts in the option parser code. This of
course affects all users of this API as well.
In stream.c, pass a mp_null_log, because we can't do it properly yet.
This will be fixed later.
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.
The tmsg stuff was for the internal gettext() based translation system,
which nobody ever attempted to use and thus was removed. mp_gtext() and
set_osd_tmsg() were also for this.
mp_dbg was once enabled in debug mode only, but since we have log level
for enabling debug messages, it seems utterly useless.
There are some use cases for this. For example, you can use it to set
defaults of automatically inserted filters (like af_lavrresample). It's
also useful if you have a non-trivial VO configuration, and want to use
--vo to quickly change between the drivers without repeating the whole
configuration in the --vo argument.
Replace the code that used a single buffer with mp_audio_buffer. This
also enables non-interleaved output operation, although it's still
disabled, and no AO supports it yet.
This comes with two internal AO API changes:
1. ao_driver.play now can take non-interleaved audio. For this purpose,
the data pointer is changed to void **data, where data[0] corresponds to
the pointer in the old API. Also, the len argument as well as the return
value are now in samples, not bytes. "Sample" in this context means the
unit of the smallest possible audio frame, i.e. sample_size * channels.
2. ao_driver.get_space now returns samples instead of bytes. (Similar to
the play function.)
Change all AOs to use the new API.
The AO API as exposed to the rest of the player still uses the old API.
It's emulated in ao.c. This is purely to split the commits changing all
AOs and the commits adding actual support for outputting N-I audio.
ao_lavc.c accesses ao->buffer, which I consider internal. The access was
done in ao_lavc.c/uninit(), which tried to get the left-over audio in
order to write the last (possibly partial) audio frame. The play()
function didn't accept partial frames, because the AOPLAY_FINAL_CHUNK
flag was not correctly set, and handling it otherwise would require an
internal FIFO.
Fix this by making sure that with gapless audio (used with encoding),
the AOPLAY_FINAL_CHUNK is set only once, instead when each file ends.
Basically, move the hack in ao_lavc's uninit to uninit_player.
One thing can not be entirely correctly handled: if gapless audio is
active, we don't know really whether the AO is closed because the file
ended playing (i.e. we want to send the buffered remainder of the audio
to the AO), or whether the user is quitting the player. (The stop_play
flag is overwritten, fixing that is perhaps not worth it.) Handle this
by adding additional code to drain the AO and the buffers when playback
is quit (see play_current_file() change).
Test case: mpv avdevice://lavfi:sine=441 avdevice://lavfi:sine=441 -length 0.2267 -gapless-audio
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
Was disabled by default, was never used, internal support was
inconsistent and poor, and there has been virtually no interest in
creating translations.
And I don't even think that a terminal program should be translated.
This is something for (hypothetical) GUIs.
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).
The point is selecting a minimal fallback. The AOs will call this
through the AO API, so it will be possible to add options affecting
the general channel layout selection.
It provides the following mechanism to AOs:
- forcing the correct channel order
- downmixing to stereo if no layout is available
- allow 5.1 <-> 5.1(side) fallback
- handling "unknown" channel layouts
This is quite weak and lots of code/complexity for little gain. All AOs
already made sure the channel order was correct, and the fallback is of
little value, and could perhaps be done in the frontend instead, like
stereo downmixing with --channels=2 is handled. But I'm not really sure
how this stuff should _really_ work, and the new code will hopefully
provides enough flexibility to make radical changes to channel layout
negotiation easier.
wm4
Stefano Pigozzi
Marcoen Hirschberg
Christian Neukirchen
Diogo Franco (Kovensky)
Jonathan Yong