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* Fix various typos in log messagesNicolas F2017-12-031-1/+1
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* audio: add audio softvol processing to AOwm42017-11-291-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This does what af_volume used to do. Since we couldn't relicense it, just rewrite it. Since we don't have a new filter mechanism yet, and the libavfilter is too inconvenient, do applying the volume gain in ao.c directly. This is done before handling the audio data to the driver. Since push.c runs a separate thread, and pull.c is called asynchronously from the audio driver's thread, the volume value needs to be synchronized. There's no existing central mutex, so do some shit with atomics. Since there's no atomic_float type predefined (which is at least needed when using the legacy wrapper), do some nonsense about reinterpret casting the float value to an int for the purpose of atomic access. Not sure if using memcpy() is undefined behavior, but for now I don't care. The advantage of not using a filter is lower complexity (no filter auto insertion), and lower latency (gain processing is done after our internal audio buffer of at least 200ms). Disavdantages include inability to use native volume control _before_ other filters with custom filter chains, and the need to add new processing for each new sample type. Since this doesn't reuse any of the old GPL code, nor does indirectly rely on it, volume and replaygain handling now works in LGPL mode. How to process the gain is inspired by libavfilter's af_volume (LGPL). In particular, we use exactly the same rounding, and we quantize processing for integer sample types by 256 steps. Some of libavfilter's copyright may or may not apply, but I think not, and it's the same license anyway.
* audio_buffer: remove dependency on mp_audiowm42017-09-211-46/+43
| | | | | | | | | Just reimplement it in some way, as mp_audio is GPL-only. Actually I wanted to get rid of audio_buffer.c completely (and instead have a list of mp_aframes), but to do so would require rewriting some more player core audio code. So to get this LGPL relicensing over quickly, just do some extra work.
* Avoid calling close(-1)wm42017-06-291-2/+5
| | | | | | | | | | While this is perfectly OK on Unix, it causes annoying valgrind warnings, and might be otherwise confusing to others. On Windows, the runtime can actually abort the process if this is called. push.c part taken from a patch by Pedro Pombeiro.
* audio/out: require AO drivers to report period size and correct bufferwm42017-06-251-2/+13
| | | | | | | | | | | | | | | | Before this change, AOs could have internal alignment, and play() would not consume the trailing data if the size passed to it is not aligned. Change this to require AOs to report their alignment (via period_size), and make sure to always send aligned data. The buffer reported by get_space() now always has to be correct and reliable. If play() does not consume all data provided (which is bounded by get_space()), an error is printed. This is preparation for potential further AO changes. I casually checked alsa/lavc/null/pcm, the other AOs might or might not work.
* audio/out/push: merge if branches with same conditionwm42017-01-091-4/+3
| | | | Cosmetic change.
* audio/out/push: play silence on --audio-stream-silencewm42016-11-241-7/+34
| | | | | | | | | | | | | | | | | | | Until now, this was only implemented for ao_alsa and AOs not using push.c. ao_alsa.c relied on enabling funny underrun semantics for avoiding resets on lower levels, while other AOs using push.c didn't do anything. Change this and at least make push.c copy silent data to the AO. This still isn't perfect as keeping track of how much silence was played when seems complex, so we don't do it. The consequence is that frame-stepping will essentially randomize the A/V offset (it'll recover immediately when unpausing, but still ugly). Also, in order to empty the currently buffered audio on seeks etc., we still call ao_driver->reset and so on, so the AO driver will still need to handle this specially. The intent is to make behavior with ALSA less weird (for one we can remove the code in ao_alsa.c that tries to trigger an initial underflow). Also might help with #3754.
* audio: fix --audio-stream-silence with ao_alsawm42016-11-211-2/+3
| | | | | | | ao_alsa.c calls this before the common code sets ao->sstride. Other than this, I'm still not sure whether this works. Seems like no, or depends.
* audio/out: prevent underruns with spdif under certain conditionswm42016-10-041-0/+2
| | | | | | | | | | | | | | | | | The player tries to avoid splitting frames with spdif (sample alignment stuff). This can in certain corner cases with certain drivers lead to the situation that ao_get_space() returns a number higher than 0 and lower than the audio frame size. The playloop will round this down to 0 bytes and do nothing, leading to a missed wakeup. This can lead to underruns or playback completely getting stuck. It can be reproduced by playing AC3 passthrough with no video and: --ao=null --ao-null-buffer=0.256 --ao-null-outburst=6100 This commit attempts to fix it by allowing the playloop to write some additional data (to get a complete frame), that will be buffered within the AO ringbuffer even if the audio device doesn't want it.
* player, ao, vo: don't call mp_input_wakeup() directlywm42016-09-161-2/+2
| | | | | | | | | | | | | Currently, calling mp_input_wakeup() will wake up the core thread (also called the playloop). This seems odd, but currently the core indeed calls mp_input_wait() when it has nothing more to do. It's done this way because MPlayer used input_ctx as central "mainloop". This is probably going to change. Remove direct calls to this function, and replace it with mp_wakeup_core() calls. ao and vo are changed to use opaque callbacks and not use input_ctx for this purpose. Other code already uses opaque callbacks, or has legitimate reasons to use input_ctx directly (such as sending actual user input).
* osdep: rename atomics.h to atomic.hwm42016-09-071-1/+1
| | | | | The standard header is stdatomic.h, so the extra "s" freaks me out every time I look at it.
* osdep/io: introduce mp_flush_wakeup_pipe()Rostislav Pehlivanov2016-07-301-4/+2
| | | | Makes a fairly common occurence with wakeup_pipes easier to handle.
* audio: apply an upper bound timeout when drainingwm42016-06-121-3/+13
| | | | | | | | | | | | | | | | | | | | | | | | This helps with shitty APIs and even shittier drivers (I'm looking at you, ALSA). Sometimes they won't send proper wakeups. This can be fine during playback, when for example playing video, because mpv still will wakeup the AO outside of its own wakeup mechanisms when sending new data to it. But when draining, it entirely relies on the driver's wakeup mechanism. So when the driver wakeup mechanism didn't work, it could hard freeze while waiting for the audio thread to play the rest of the data. Avoid this by waiting for an upper bound. We set this upper bound at the total mpv audio buffer size plus 1 second. We don't use the get_delay value, because the audio API could return crap for it, and we're being paranoid here. I couldn't confirm whether this works correctly, because my driver issue fixed itself. (In the case that happened to me, the driver somehow stopped getting interrupts. aplay froze instead of playing audio, and playing audio-only files resulted in a chop party. Video worked, for reasons mentioned above, but drainign froze hard. The driver problem was solved when closing all audio output streams in the system. Might have been a dmix related problem too.)
* audio: do not wake up core during EOFwm42016-06-121-3/+4
| | | | | | | | | | | | | When we're draining, don't wakeup the core on every buffer fill, since unlike during normal playback, we won't actually get more data. The wakeup here conceptually works like wakeups with condition variables, so redundant wakeups do not hurt, so this is just a minor change and nothing of consequence. (Final EOF also requires waking up the core, but there is separate code to send this notification.) Also dump the p->still_playing field in trace logging.
* build: silence -Wunused-resultNiklas Haas2016-06-071-2/+2
| | | | | | | | For clang, it's enough to just put (void) around usages we are intentionally ignoring the result of. Since GCC does not seem to want to respect this decision, we are forced to disable the warning globally.
* Relicense some non-MPlayer source files to LGPL 2.1 or laterwm42016-01-191-7/+7
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This covers source files which were added in mplayer2 and mpv times only, and where all code is covered by LGPL relicensing agreements. There are probably more files to which this applies, but I'm being conservative here. A file named ao_sdl.c exists in MPlayer too, but the mpv one is a complete rewrite, and was added some time after the original ao_sdl.c was removed. The same applies to vo_sdl.c, for which the SDL2 API is radically different in addition (MPlayer supports SDL 1.2 only). common.c contains only code written by me. But common.h is a strange case: although it originally was named mp_common.h and exists in MPlayer too, by now it contains only definitions written by uau and me. The exceptions are the CONTROL_ defines - thus not changing the license of common.h yet. codec_tags.c contained once large tables generated from MPlayer's codecs.conf, but all of these tables were removed. From demux_playlist.c I'm removing a code fragment from someone who was not asked; this probably could be done later (see commit 15dccc37). misc.c is a bit complicated to reason about (it was split off mplayer.c and thus contains random functions out of this file), but actually all functions have been added post-MPlayer. Except get_relative_time(), which was written by uau, but looks similar to 3 different versions of something similar in each of the Unix/win32/OSX timer source files. I'm not sure what that means in regards to copyright, so I've just moved it into another still-GPL source file for now. screenshot.c once had some minor parts of MPlayer's vf_screenshot.c, but they're all gone.
* audio: fix format function consistency issueswm42015-06-261-1/+1
| | | | | | | | | | | 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".
* threads: use utility+POSIX functions instead of weird wrapperswm42015-05-111-3/+6
| | | | | | | 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.)
* audio/out/push: fix off-by-one errorwm42014-11-211-1/+1
| | | | | | Needs 1 additional free entry. Found by Coverity.
* audio/out: consistently use double return type for get_delaywm42014-11-091-2/+2
| | | | | ao_get_delay() returns double, but the get_delay callback still returned float.
* audio/out/push: when using audio wait fallback, recheck conditionwm42014-11-061-1/+2
| | | | | | | | | | | If calling ao->driver->wait() fails, we need to fallback to timeout- based waiting. But it could be that at this point, the mutex was already released (and then re-acquired). So we need to recheck the condition in order to avoid missed wakeups. This probably wasn't an actually occurring problem, but still could cause a small race-condition window if the dynamic fallback is actually used.
* Set thread name for debuggingwm42014-10-191-0/+1
| | | | | | | | | | Especially with other components (libavcodec, OSX stuff), the thread list can get quite populated. Setting the thread name helps when debugging. Since this is not portable, we check the OS variants in waf configure. old-configure just gets a special-case for glibc, since doing a full check here would probably be a waste of effort.
* audio/out/push: reset projected EOF time on new datawm42014-10-141-1/+4
| | | | | Seems like this could theoretically happen in low buffer situations, but I haven't spotted this behavior in the wild.
* audio/out/push: make draining slightly more robustwm42014-10-101-1/+1
| | | | | | | | Don't wait after the audio thread has pushed the remaining audio to the AO. Avoids hard hangs if the heuristic fails completely (could still happen if get_delay returns absurd values). CC: @mpv-player/stable
* audio/out/push: fix EOF heuristicwm42014-10-101-18/+14
| | | | | | | | | | | | | Since the internal AO driver API has no proper way to determine EOF, we need to guess by querying get_delay. But some AOs (e.g. ao_pulse with no-latency-hacks set) may never reach 0, maybe because they naively add the latency to the buffer level. In this case our heuristic can break. Fix by always using the delay to estimate the EOF time. It's not even that important - it's mostly used to avoid blocking draining. So this should be ok. CC: @mpv-player/stable (maybe)
* audio/out/push: fix some AOs freezing on exitwm42014-10-051-1/+1
| | | | Caused by a dumb deadlock.
* audio/out/push: make draining more robustwm42014-10-051-20/+15
| | | | | | It was more complicated than it had to be: the audio thread already determines whether audio has ended, so we can use that. Remove the separate logic for draining.
* audio/out/push: limit fallback sleep time to reasonable limitswm42014-10-051-2/+4
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* audio/out/push: clean up properly on init errorwm42014-09-271-9/+16
| | | | Close the wakeup pipes, free the mutex and condition var.
* audio/out: check device buffer size for push.c onlywm42014-09-271-0/+5
| | | | Should fix #1125.
* audio/out: fix active waiting during pause againwm42014-09-061-1/+2
| | | | | This was fixed in commit 8432eaefa, and commit 39609fc1 of course broke it again. This was pretty stupid.
* audio/out/push: redo audio waitingwm42014-09-061-45/+37
| | | | | | | | | | | | | | | | | | | | | | | | | | | Improve the logic how the audio thread decides how to wait until the AO is ready for new data. The previous commit makes some of this easier, although it turned out that it wasn't required, and we still can handle AOs with bad get_space implementation (although the new code prints an error message, and it might fail in obscure situations). The new code is pretty similar to the old one, and the main thing that changes is that complicated conditions are tweaked. AO waiting is now used better (mainly instead of max>0, r>0 is used). Whether to wakeup is reevaluated every time, instead of somehow doing the wrong thing and compensating for it with a flag. This fixes the specific situation when the device buffer is full, and we don't want to buffer more data. In the old code, this wasn't handled correctly: the AO went to sleep forever, because it prevented proper wakeup by the AO driver, and as consequence never asked the core for new data. Commit 4fa3ffeb was a hack-fix against this, and now that we have a proper solution, this hack is removed as well. Also make the refill threshold consistent and always use 1/4 of the buffer. (The threshold is used for situations when an AO doesn't support proper waiting or chunked processing.) This commit will probably cause a bunch of regressions again.
* audio/out: make EOF handling properly event-basedwm42014-09-051-5/+40
| | | | | | | | | | | | | | | | | 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)
* audio: make buffer size configurablewm42014-09-051-2/+2
| | | | Really only for testing.
* audio/out: prevent burning CPU when seeking while pausedwm42014-08-311-1/+2
| | | | | | | | | The audio/video sync code in player/audio.c calls ao_reset() each time audio decoding is entered, but the player is paused, and there would be more than 1 sample to skip to make audio start match with video start. This caused a wakeup feedback loop with push.c. CC: @mpv-player/stable
* Move compat/ and bstr/ directory contents somewhere elsewm42014-08-291-1/+1
| | | | | | | | | bstr.c doesn't really deserve its own directory, and compat had just a few files, most of which may as well be in osdep. There isn't really any justification for these extra directories, so get rid of them. The compat/libav.h was empty - just delete it. We changed our approach to API compatibility, and will likely not need it anymore.
* audio: limit on low (and not high) buffer sizewm42014-08-211-1/+1
| | | | | | | The original intention was probably to avoid unnecessarily high numbers of wakeups. Change it to wait at most 25% of buffer time instead of 75% until refilling. Might help with the dsound problems in issue #1024, but I don't know if success is guaranteed.
* audio/out: fix initialization failure with win32wm42014-07-261-2/+1
| | | | | | | mp_make_wakeup_pipe() always fails on win32. If this call fails on Linux (and e.g. ao_alsa is used), this will probably burn CPU since poll() won't work on the invalid file descriptor, but whatever, the failure case is obscure enough.
* audio, client API: check mp_make_wakeup_pipe() return valuewm42014-07-251-5/+7
| | | | Could fail e.g. due to FD exhaustion.
* audio: don't wait for draining if pausedwm42014-07-131-1/+6
| | | | | | | | | | | | | 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
* audio: more detailed debugging outputwm42014-06-121-0/+2
| | | | Dump what the AO does on driver->play().
* audio: don't wait when draining and pausedwm42014-06-121-1/+1
| | | | | A corner case that could possibly lead to infinite waiting. Though I'm not aware that this actually happened in practice.
* audio/out/push: don't attempt to fill AO buffer when pausedwm42014-06-031-2/+3
| | | | Doing so will implicitly resume playback. Broken in commit 5929dc45.
* audio/out/push: keep some extra bufferwm42014-05-311-6/+4
| | | | | | | | | | | | | | | | | | | So the device buffer can be refilled quickly. Fixes dropouts in certain cases: if all data is moved from the soft buffer to the audio device buffer, the waiting code thinks it has to enter the mode in which it waits for new data from the decoder. This doesn't work, because the get_space() logic tries to keep the total buffer size down. get_space() will return 0 (or a very low value) because the device buffer is full, and the decoder can't refill the soft buffer. But this means if the AO buffer runs out, the device buffer can't be refilled from the soft buffer. I guess this mess happened because the code is trying to deal with both AOs with proper event handling, and AOs with arbitrary behavior. Unfortunately this increases latency, as the total buffered audio becomes larger. There are other ways to fix this again, but not today. Fixes #818.
* ao_alsa: reduce spurious wakeupswm42014-05-301-2/+4
| | | | | | 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.
* audio/out/push: handle draining correctlywm42014-05-301-7/+22
| | | | | | | | | This did not flush remaining audio in the buffer correctly (in case an AO has an internal block size). So we have to make the audio feed thread to write the remaining audio, and wait until it's done. Checking the avoid_ao_wait variable should be enough to be sure that all data that can be written was written to the AO driver.
* audio: change handling of an EOF corner casewm42014-05-301-5/+9
| | | | | | This code handles buggy AOs (even if all AOs are bug-free, it's good for robustness). Move handling of it to the AO feed thread. Now this check doesn't require magic numbers and does exactly what's it supposed to do.
* audio/out/push: add a way to wait for the audio device with poll()wm42014-05-301-3/+64
| | | | Will be used for ALSA.
* audio/out/push: add mechanism for event-based waitingwm42014-05-301-74/+87
| | | | | | | | | | | | | | | | Until now, we've always calculated a timeout based on a heuristic when to refill the audio buffers. Allow AOs to do it completely event-based by providing wait and wakeup callbacks. This also shuffles around the heuristic used for other AOs, and there is a minor possibility that behavior slightly changes in real-world cases. But in general it should be much more robust now. ao_pulse.c now makes use of event-based waiting. It already did before, but the code for time-based waiting was also involved. This commit also removes one awkward artifact of the PulseAudio API out of the generic code: the callback asking for more data can be reentrant, and thus requires a separate lock for waiting (or a recursive mutex).
* threads: use mpv time for mpthread_cond_timedwait wrapperwm42014-05-181-1/+1
| | | | | | Use the time as returned by mp_time_us() for mpthread_cond_timedwait(), instead of calculating the struct timespec value based on a timeout. This (probably) makes it easier to wait for a specific deadline.
* audio/out: fix previous commitwm42014-05-111-9/+11
| | | | | | | | | This didn't quite work. The main issue was that get_space tries to be clever to reduce overall buffering, so it will cause the playloop to decode and queue only as much audio as is needed to refill the AO in reasonable time. Also, even if ignoring the problem, the logic of the previous commit was slightly broken. (This required a few retries, because I couldn't reproduce the issue on my own machine.)
* audio/out: avoid wakeup feedback loopwm42014-05-111-2/+7
| | | | | | | | | | | | | When the audio buffer went low, but could not be refilled yet, it could happen that the AO playback thread and the decode thread could enter a wakeup feedback loop, causing up to 100% CPU usage doing nothing. This happened because the decoder thread would wake up the AO thread when writing 0 bytes of newly decoded data, and the AO thread in reaction wakes up the decoder thread after writing 0 by