--softvol is enabled by default. For most audio outputs, this is a good
thing, as they have either their own (bad) soft volume implementation,
or control the system mixer. With ao_pulse, the situation is a bit
different: it supports per-application volume (i.e. volume control is
not really global). More importantly, ao_pulse uses a rather large audio
buffer, and changing the volume with mplayer's volume filter has a large
delay. With the native ao_pulse volume control, it's instant, because
PulseAudio's audio filtering happens at a later stage in its processing
pipeline (inaccessible for mplayer).
This means native volume control should really be allowed for ao_pulse,
while it's the reverse for other audio outputs. Make --softvol a choice
option, and add a new "auto" choice. This is default and will use PA's
volume control with ao_pulse, and mplayer's volume filter otherwise
(i.e. the old softvol behavior).
PulseAudio is rather high on the auto proving order (to avoid using an
emulated sound API), but it prints an annoying error message if the
PA client library can't connect to a server. On the other hand, we do
want this error message printed if the user explicitly selects the
pulse audio output driver.
Add a flag to indicate that an AO is opened due to auto probing.
ao_pulse checks that flag, and if it's set, do not print if the
initialization error is PA_ERR_CONNECTIONREFUSED, whcih I assume is
the error signalling PulseAudio unavailability. (This error happens
if no PulseAudio server is installed.)
Make mixer support setting the mute attribute at audio driver level,
if one exists separately from volume. As of this commit, no libao2
driver exposes such an attribute yet; that will be added in later
commits.
Since the mute status can now be set externally, it's no longer
completely obvious when the player should automatically disable mute
when uninitializing an audio output. The implemented behavior is to
turn mute off at uninitialization if we turned it on and haven't
noticed it turn off (by external means) since.
MPlayer volume control was originally implemented with the assumption
that it controls a system-wide volume setting which keeps its value
even if a process closes and reopens the audio device. However, this
is not actually true for --softvol mode or some audio output APIs that
only consider volume as a per-client setting for software mixing. This
could have annoying results, as the volume would be reset to a default
value if the AO was closed and reopened, for example whem moving to a
new file or crossing ordered chapter boundaries. Add code to set the
previous volume again after audio reinitialization if the current
audio chain is known to behave this way (softvol active or the AO
driver is known to not keep persistent volume externally).
This also avoids an inconsistency with the mute flag. The frontend
assumed the mute status is persistent across file changes, but it
could be similarly lost.
The audio drivers that are assumed to not keep persistent volume are:
coreaudio, dsound, esd, nas, openal, sdl. None of these changes have
been tested. I'm guessing that ESD and NAS do per-connection
non-persistent volume settings.
Partially based on code by wm4.
Change the audio driver control() command argument from "int" to "enum
aocontrol". Remove unused control types (SET_DEVICE, GET_DEVICE,
QUERY_FORMAT, SET_PLUGIN_DRIVER, SET_PLUGIN_LIST). The QUERY_FORMAT
one looks like there's a possibility such functionality could be
useful in the future, but as ao_oss was the only driver to have an
actual implementation of it, the current code wasn't worth keeping.
For ao_pulse, the current latency is not a good indicator of how soon
the AO requires new data to avoid underflow. Add an internal pipe that
can be used to wake up the input loop from select(), and make the
pulseaudio main loop (which runs in a separate thread) use this
mechanism when pulse requests more data. The wakeup signal currently
contains no information about the reason for the wakup, but audio
buffers are always filled when the event loop wakes up.
Also, request a latency of 1 second from the Pulseaudio server. The
default is normally significantly higher. We don't need low latency,
while higher latency helps prevent underflows reduces need for
wakeups.
Move the buffer storing audio data ready to be fed to the audio output
driver from the audio decoder object to the AO object. This will help
encoding code deal with end of input, and may also be useful to
improve other general gapless audio behavior (as AOs which do not
accept chunks smaller than a certain size may keep them in the buffer
while the decoder changes).
Less data may be dropped now when changing audio filters or switching
timeline parts.
Change ao_pcm to use the new audio output driver API and clean up some
of the code. Rewrite the logic controlling how playback timing works
when using -ao pcm. Deprecate the "fast" suboption; its only effect
now is to print a warning, but it's still accepted so that specifying
it is not an error.
Before, timing with -ao pcm and video enabled had two possible
modes. In the default mode playback speed was rather arbitrary - not
realtime, but not particularly fast. -ao pcm:fast tried to play back
at maximum video playback speed - mostly succeeding, but not quite
guaranteed to work in all cases. Now the default is to play at
realtime speed. The -benchmark option can now be used to get faster
playback (same as the video-only case). In the audio-only case
playback is always maximum speed.
Add a new audio output driver API. The main change is the addition of
a context struct where data can be kept instead of using globals. The
old API remains available. This commit does not yet convert any driver
to use the new API.
Reindent audio_out.c and audio_out.h. Also remove trailing '_s' from
two struct names (which are not currently used anywhere) and make the
audio_out_drivers[] table static.
Make the outside interface of audio output handling similar to the
video output one. An AO object is first created, and then methods
called with ao_[methodname](ao, args...). However internally libao2/
still holds all data in globals, and trying to create multiple
simultaneous AO instances won't work.
truncation of output, set flag AOPLAY_FINAL_CHUNK in play call to tell
ao there will be no more data beyond what's in current buffer.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@18738 b3059339-0415-0410-9bf9-f77b7e298cf2
patch replaces '()' for the correct '(void)' in function
declarations/prototypes which have no parameters. The '()' syntax tell
thats there is a variable list of arguments, so that the compiler cannot
check this. The extra CFLAG '-Wstrict-declarations' shows those cases.
Comments about a similar patch applied to ffmpeg:
That in C++ these mean the same, but in ANSI C the semantics are
different; function() is an (obsolete) K&R C style forward declaration,
it basically means that the function can have any number and any types
of parameters, effectively completely preventing the compiler from doing
any sort of type checking. -- Erik Slagter
Defining functions with unspecified arguments is allowed but bad.
With arguments unspecified the compiler can't report an error/warning
if the function is called with incorrect arguments. -- Måns Rullgård
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@17567 b3059339-0415-0410-9bf9-f77b7e298cf2
wm4
Rudolf Polzer
Uoti Urpala
Uoti Urpala
reimar
diego
Uoti Urpala
uau
rathann
alex
arpi
pontscho
iive
anders
al3x
arpi_esp