| Commit message (Collapse) | Author | Age | Files | Lines |
|
|
|
|
|
|
| |
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.
|
|
|
|
| |
This is more convenient.
|
|
|
|
|
|
| |
Apparently this can happen. So actually only return from waiting if ALSA
excplicitly signals that new output is available, or if we are woken up
externally.
|
|
|
|
|
| |
This means the audio feed thread is woken up exactly at the time new
data is needed, instead of using a time-based heuristic.
|
|
|
|
|
|
|
| |
This function is now always available.
Also remove includes of reorder_ch.h from some AOs (these are just old
relicts).
|
|
|
|
|
|
|
|
|
|
| |
In general, we don't need to have a large hw audio buffer size anymore,
because we can quickly fill it from the soft buffer.
Note that this probably doesn't change much anyway. On my system (dmix
enabled), the buffer size is only 170ms, and ALSA won't give more. Even
when using a hardware device the buffer size seems to be limited to
341ms.
|
|
|
|
|
|
|
|
|
|
|
| |
This AO pretended to support volume operations when in spdif passthrough
mode, but actually did nothing. This is wrong: at least the GET
operations must write their argument. Signal that volume is unsupported
instead.
This was probably a hack to prevent insertion of volume filters or so,
but it didn't work anyway, while recovering after failed volume filter
insertion does work, so this is not needed at all.
|
|
|
|
| |
priv is 0-initialized, can_pause is always overwritten later.
|
|
|
|
|
|
|
|
|
|
| |
It is possible to have ao->reset() called between ao->pause() and
ao->resume() when seeking during the pause. If the underlying PCM
supports pausing, resuming an already reset PCM will produce an error.
Avoid that by explicitly checking PCM state before calling
snd_pcm_pause().
Signed-off-by: wm4 <wm4@nowhere>
|
|
|
|
|
|
|
|
|
|
|
|
| |
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.
|
|
|
|
|
|
| |
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.
|
|
|
|
|
| |
Actually, remove the whole comment, because it's outdated and
get_space() returns the number of free samples now.
|
|
|
|
| |
Well that was dumb.
|
|
|
|
| |
Fixes #324
|
|
|
|
|
|
|
| |
This could output additional, potentially useful error messages. But the
callback is global and not library-safe, and would require us to add
additional state. Remove it, because it's obviously too much of a pain.
Also, it seems ALSA prints stuff to stderr anyway.
|
| |
|
|
|
|
|
|
|
|
|
| |
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.
|
|
|
|
|
| |
If initialization succeeds, p->alsa should always be set. Additional
checks are not needed, and also this wasn't even done consistently.
|
|
|
|
|
|
| |
I have no idea what this code does, but it seems logical it should be
active for all float formats, not just for float with interleaved
access.
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| |
This partially reverts commit 7d152965. It turns out that at least some
ALSA drivers (at least snd-hda-intel) report incorrect audio delay with
non-native sample rates, even if the sample rate is only very slightly
different from the native one.
For example, 48000Hz is fine on my hda-intel system, while both 8000Hz
and 47999Hz lead to a delay off by 40ms (according to mpv's A/V
difference display), which suggests that something in ALSA is
calculating the delay using the wrong sample rate.
As an additional problem, with ALSA resampling enabled, using
48001Hz/float/2ch fails, while 49000Hz/float/2ch or 48001Hz/s16/2ch
work. With resampling disabled, all these cases work obviously, because
our own resampler doesn't just refuse any of these formats.
Since some people want to use the ALSA resampler (because it's highly
configurable, supports multiple backends, etc.), we still allow enabling
ALSA resampling with an ao_alsa suboption.
|
|
|
|
|
|
|
|
| |
Resampling with non-ancient ALSA setups works fine, so there is no
need to keep this around. Furthermore, as of writing, the default
builtin resampler used by many ALSA setups (taken from libspeex)
actually has higher quality than the default resampling modes of
avresample and swresample.
|
|
|
|
|
|
| |
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.
|
|
|
|
|
|
| |
I thought this would always work... how disappointing.
Revert to interleaved format if requesting non-interleaved fails.
|
|
|
|
|
|
|
|
| |
ALSA supports non-interleaved audio natively using a separate API
function for writing audio. (Though you have to tell it about this on
initialization.) ALSA doesn't have separate sample formats for this,
so just pretend to negotiate the interleaved format, and assume that
all non-interleaved formats have an interleaved companion format.
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| |
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.
|
|
|
|
|
|
|
|
|
|
| |
No AO can handle these, so it would be a problem if they get added
later, and non-interleaved formats get accepted erroneously. Let them
gracefully fall back to other formats.
Most AOs actually would fall back, but to an unrelated formats. This is
covered by this commit too, and if possible they should pick the
interleaved variant if a non-interleaved format is requested.
|
|
|
|
|
| |
Also add a corresponding function to audio/format.c, which fills an
audio block with silence.
|
|
|
|
|
| |
I accidentally copied the AES4/ORIGFS constants from the ALSA headers,
instead of the AES3/FS ones. The difference is probably important.
|
|
|
|
| |
Also remove the rather bad/incomplete log calls from ao_alsa and ao_oss.
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| |
I have no idea what these do, but apparently they are needed to inform
ALSA about spdif configuration. First, replace the literal constant "6"
for the AES0 parameter with the symbolic constants from the ALSA
headers (the final value is the same). Second, copy paste some funky
looking parameter setup from VLC's alsa output for setting the AES1,
AES2, AES3 parameters. (The code is actually not literally copy-pasted,
but does exactly the same.)
My small but non-zero hope is that this could make DTS-HD work, or at
least work into that direction. I can't test spdif stuff though, and
for DTS-HD not even opening the ALSA device succeeds on my system.
|
|
|
|
|
|
|
|
|
|
|
| |
Using spdif with alsa requires adding magic parameters to the device
name, and the existing code tried to deal with the situation when the
user wanted to add parameters too.
Rewrite this code, in particular remove the duplicated parameter string
as preparation for the next commit. The new code is a bit stricter, e.g.
it doesn't skip spaces before and after '{' and '}'. (Just don't add
spaces.)
|
|
|
|
| |
Also, remove all af_fmt2str usages.
|
|
|
|
|
| |
No functional change, because the only user of ao_play() ignores return
values below 1.
|
|
|
|
|
|
|
|
| |
It's true that ALSA uses alloca() in some of its API functions, but
since this is hidden behind macros in the ALSA headers, we have no
reason to include alloca.h ourselves.
Might help with portability (FreeBSD).
|
| |
|
|
|
|
| |
Also remove stray include statements from ao_alsa and ao_oss.
|
|
|
|
|
|
|
| |
Use the new MP_ macros for some AOs instead of mp_msg.
Not all AOs are converted, and some only partially. In some cases, some
additional cosmetic changes are made.
|
|
|
|
| |
Followup commit. Fixes all the files references.
|
|
|
|
| |
Same as with VOs in the previous commit.
|
|
|
|
| |
Finally not used by anything anymore. Farewell.
|
|
|
|
|
|
| |
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.
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| |
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 ALSA device was not closed when initialization failed.
The ALSA error handler (set with snd_lib_error_set_handler()) was not
unset when closing ao_alsa. If this is not done, the handler will still
be called when other libraries using ALSA cause errors, even though
ao_alsa was long closed. Since these messages were prefixed with
"[AO_ALSA]", they were misleading and implying ao_alsa was still used.
For some reason, our error handler is still called even after doing
snd_lib_error_set_handler(NULL), which should be impossible. Checking
with the debuggers, inserting printf(), as well as the alsa-lib source
code all suggest our error handler should not be called, but it still
happens. It's a complete mystery.
|
|
|
|
|
|
|
|
|
| |
It turns out that ALSA's 4 channel layout is different from mpv's and
ffmpeg's 4.0 layout. Thus trying to do 4 channel output led to incorrect
remixing via lib{av,sw}resample.
Fix the default layouts for the internal filter chain as well, although
I'm not sure if it matters at all.
|
|
|
|
|
|
| |
The snd_pcm_hw_params_test_format() call actually crashes in alsa-lib if
called with SND_PCM_FORMAT_UNKNOWN, so the already existing fallback
code won't work in this case.
|
|
|
|
|
|
|
|
|
| |
Make all AOs use what has been introduced in the previous commit.
Note that even AOs which can handle all possible layouts (like ao_null)
use the new functions. This might be important if in the future
ao_select_champ() possibly honors global user options about downmixing
and so on.
|
|
|
|
| |
Is unused, is completely pointless.
|
| |
|
| |
|
|
|
|
|
|
|
| |
This allows supporting 5 channel audio (which can be eother 5.0 or 4.1).
Fallback doesn't work yet. It will do nonsense if the channel layout
doesn't match perfectly, even though it's similar.
|
| |
|
| |
|
|
|
|
|
|
|
|
| |
Add a CHECK_ALSA_ERROR macro to report ALSA errors. This is similar to
what vo_vdpau does. This removes lots of boiler plate, it almost gives
me the feeling the ao_alsa initialization code is now readable. This
change is squashed with the reformatting, because both changes are
just as noisy and useless.
|
| |
|
|
|
|
|
|
| |
This actually breaks audio for 5/6/8 channels. There's no reordering
done yet. The actual reordering will be done inside of af_lavrresample
and has to be made part of the format negotiation.
|
|
|
|
|
| |
Currently, internal and alsa order are exactly the same, so this will do
absolutely nothing.
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| |
This causes trouble when a hw device is used:
pcm_hw.c:514:(snd_pcm_hw_delay) SNDRV_PCM_IOCTL_DELAY failed (-77): File descriptor in bad state
when running mpv test.mkv --ao=alsa:device=iec958,alsa and pausing
during playback.
Historically, mplayer usually did not call snd_pcm_delay() (which is
called by get_delay()) while paused, so this problem never showed up.
But at least mpv has changes that cause get_delay() to be called when
updating the status line (see commit 3f949cf).
It's possible that calling snd_pcm_delay() is not always legal when the
audio is paused, and at least fails with the error message mentioned
above is the device is a hardware device. Change get_delay() to return
the last delay before the audio was paused. The intention is to get a
continuous playback status display, even when pausing or frame stepping,
otherwise we could just return the audio buffer fill status in
get_delay() or even just 0 when paused.
|
|
|
|
|
|
| |
This is considered a worthless feature. Note that alaw/mulaw/adpcm input
is unaffected: such data is handed to libavcodec and "decoded" to linear
PCM.
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| |
Do not fall back to 0 for samplerate when parser is not initialized.
Might fix some issues with using -ac spdifenc with audio in MKV
or MP4.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35517 b3059339-0415-0410-9bf9-f77b7e298cf2
Replace outdated list of unsupported formats by list of supported formats.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35534 b3059339-0415-0410-9bf9-f77b7e298cf2
Do not call af_fmt2str on the same data over and over.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35535 b3059339-0415-0410-9bf9-f77b7e298cf2
ad_spdif: use the more specific AF_FORMAT_AC3_LE when
we handle AC3.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35536 b3059339-0415-0410-9bf9-f77b7e298cf2
Make AF_FORMAT_IS_IEC61937 include AF_FORMAT_IS_AC3.
Our AC3 "sample format" is also iec61937.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35537 b30593 |