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* ao_oss: fix audio device leak on errorwm42014-09-111-21/+25
| | | | | Close the audio device if it was already opened, but the rest of initialization failed.
* ao_oss: use poll(), drop --disable-audio-select supportwm42014-09-111-32/+17
| | | | | | | | | | | | | | Replace select() usage with poll() (and reduce code duplication). Also, while we're at it, drop --disable-audio-select, since it has the wrong name anyway. And I have doubts that this is needed anywhere. If it is, it should probably fallback to doing the right thing by default, instead of requiring the user to do it manually. Since nobody has done that yet, and since this configure option has been part of MPlayer ever since ao_oss was added, it's probably safe to say it's not needed. The '#ifdef SNDCTL_DSP_GETOSPACE' was pointless, since it's already used unconditionally in another place.
* ao_pulse: dump library version etc.wm42014-09-101-0/+6
| | | | | | | Might help with debugging. Unfortunately, there doesn't seem to be a way to get the actual pulseaudio server version.
* ao_pulse: fix typo in error messagewm42014-09-081-1/+1
| | | | Closes #1076.
* 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: always round get_space on period sizewm42014-09-064-4/+6
| | | | | | | | | | | Round get_space() results in the same way play() rounds the input size. Some audio APIs do this for various reasons. This affects only "push" based AOs. Some of these need no change, because they either do it already right (like ao_openal), or they seem not to have any such requirements (like ao_pulse). Needed for the following commit.
* ao_sndio: fix a commentwm42014-09-061-2/+2
| | | | | Whether this code was written with the correct assumptions in mind, I don't know.
* ao_pcm: minor simplificationwm42014-09-061-1/+1
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* ao_oss: minor simplificationwm42014-09-061-3/+1
| | | | Equivalent code.
* audio/out: remove old thingswm42014-09-0610-29/+25
| | | | | | | | 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.
* audio/out: make EOF handling properly event-basedwm42014-09-054-6/+52
| | | | | | | | | | | | | | | | | 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 configurablewm42014-09-053-6/+5
| | | | 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 elsewm42014-08-295-5/+4
| | | | | | | | | 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.
* coreaudio_device: fix overwriting of user inputStefano Pigozzi2014-08-251-9/+8
| | | | Fixes #1030
* ao_dsound: raise default buffer size to 200ms, make it configurablewm42014-08-221-3/+5
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* 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.
* ao_dsound: reduce default buffer sizewm42014-08-081-1/+1
| | | | | | Reduce from 1000ms to 100ms. Since there is an audio thread updating AOs quickly enough now, requesting such a large buffer size makes no sense anymore.
* ao_alsa: disable use of non-interleaved formats by defaultwm42014-07-301-0/+6
| | | | | | | | Some ALSA plugins take non-interleaved audio, but treat it as interleaved, which results in various funny bugs. Users keep hitting this issue, and it just doesn't seem worth the trouble. CC: @mpv-player/stable
* ao_pulse: allow disabling timing bug workaroundswm42014-07-261-3/+38
| | | | | | | | | | | | | | | | Add an option that enables using native PulseAudio auto-updated timing information, instead of the manual calculations added in mplayer2 times. You can use --ao=pulse:no-latency-hacks to enable the new code. The code is almost the same as the code that was removed with commit de435ed5, but I didn't readd some bits I didn't understand. Likewise, the option will disable the code added with that commit. In my tests this seemed to work well, though the A/V sync display looks funny when seeking. The default is still the old behavior. See issue #959.
* ao_pulse: remove hacks for ancient PulseAudio versionswm42014-07-261-21/+0
| | | | | | | | | | | This was needed by very old (0.9) versions only. Get rid of it. Unfortunately, I can't cross-check with the original bug report, since the bug URL leads to this: Internal Server Error TracError: IOError: [Errno 2] No such file or directory: '/home/lennart/svn/trac/pulseaudio/VERSION'
* ao_null: never fail at initializationwm42014-07-261-1/+1
| | | | | | | | | ao_null is used to stop autoprobing (if all AOs before fail to init). After it come things like ao_pcm, which should never be automatically selected. Remove a certain theoretically possible failure case, and force "some" fallback.
* 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.
* encode: deal even more with codec->time_base deprecation.Rudolf Polzer2014-07-231-6/+5
| | | | I assume this works too with Libav 10 and FFmpeg d3e51b41.
* ao_pulse: fix potential compilation problemwm42014-07-221-2/+2
| | | | | | It seems at least on some platforms (OSX 10.9), the POSIX wait() function becomes visible, and conflicts with this unrelated function. Just rename it.
* ao_lavc: Fix design of audio pts handling.Rudolf Polzer2014-07-161-2/+5
| | | | | | | | | There was confusion about what should go into audio pts calculation and what not (mainly due to the audio push thread). This has been fixed by using the playing - not written - audio pts (which properly takes into account the ao's buffer), and incrementing the samples count only by the amount of samples actually taken from the buffer (unfortunately this now forces us to keep the lock too long for my taste).
* ao_lavc: Add a missing newline for the log.Rudolf Polzer2014-07-161-1/+1
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* ao_lavc: Fix advancing of audio pts.Rudolf Polzer2014-07-161-1/+1
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* audio: don't wait for draining if pausedwm42014-07-134-15/+16
| | | | | | | | | | | | | 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
* build: include <strings.h> for strcasecmp()wm42014-07-101-0/+1
| | | | | | | It happens to work without strings.h on glibc or with _GNU_SOURCE, but the POSIX standard requires including <strings.h>. Hopefully fixes OSX build.
* ao_null: disable latency emulationwm42014-07-071-1/+0
| | | | | | | | | Doesn't work quite right, and will pause for the latency duration after seeking. Some users use --ao=null to disable audio (even though they should probably use --no-audio), and this use-case is broken by this issue too. CC: @mpv-player/stable
* ao_pulse: set icon nameatomnuker2014-07-051-0/+2
| | | | Will replace the generic XDG video icon inherited from media role.
* ao_coreaudio: report hardware latency to ao_read_dataStefano Pigozzi2014-07-032-3/+43
| | | | | Commit a6a4cd2c88 added reporting of playout latency, this commit also adds support for reporting hardware and constant audio unit latency.
* ao_coreaudio: report latency more correctlyStefano Pigozzi2014-07-021-1/+19
| | | | | | | | | | Previous code was completly wrong. This still doesn't report the device latency, but we report the buffer latency (as before the AO refactoring) and the AudioUnit's latency (this is a new 'feature'). Apparently we can also report the device actual latency and we should also calculate the actual sample rate of the audio device instead of using the nominal sample rate, but I'll leave this for a later commit.
* ao_coreaudio: move channel mapping away from utilsStefano Pigozzi2014-07-023-126/+128
| | | | | Channel mapping functions are only used in the AUHAL based coreaudio, so move them there.
* ao_coreaudio: use mpv's internal pull APIStefano Pigozzi2014-07-021-82/+9
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* ao_coreaudio: remove useless commentsStefano Pigozzi2014-07-021-5/+3
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* ao_coreaudio: rename init_lpcm -> init_audiounitStefano Pigozzi2014-07-021-6/+9
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* ao_coreaudio: fill asbd with an helper functionStefano Pigozzi2014-07-024-39/+29
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* ao_coreaudio: split control to helper functionsStefano Pigozzi2014-07-021-23/+30
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* ao_coreaudio: move device related functions to the new AOStefano Pigozzi2014-07-023-277/+255
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* ao_coreaudio: remove useless call to print_asbdStefano Pigozzi2014-07-022-4/+0
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* ao_coreaudio: move spdif code to a new AOStefano Pigozzi2014-07-025-433/+595
| | | | | | | | | | | | | | | | 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.
* Audit and replace all ctype.h useswm42014-07-012-3/+2
| | | | | | | | | | | | | | | | Something like "char *s = ...; isdigit(s[0]);" triggers undefined behavior, because char can be signed, and thus s[0] can be a negative value. The is*() functions require unsigned char _or_ EOF. EOF is a special value outside of unsigned char range, thus the argument to the is*() functions can't be a char. This undefined behavior can actually trigger crashes if the implementation of these functions e.g. uses lookup tables, which are then indexed with out-of-range values. Replace all <ctype.h> uses with our own custom mp_is*() functions added with misc/ctype.h. As a bonus, these functions are locale-independent. (Although currently, we _require_ C locale for other reasons.)
* ao_pcm: fix message stringsAmos Onn2014-06-151-2/+2
| | | | Signed-off-by: wm4 <wm4@nowhere>
* encode: get rid of the recursion that led to a deadlock.Rudolf Polzer2014-06-121-23/+28
| | | | | Instead, the recursive call has been flattened away by instead overwriting a parameter and continuing.
* 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.
* Add more constwm42014-06-112-3/+3
| | | | | | | 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.
* 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: prefer dsound over wasapiwm42014-06-011-3/+3
| | | | | ao_wasapi has too many subtle failures that were reported, but there's nobody to fix them. ao_dsound seems to be more robust; so prefer it.
* ao_alsa: make device the first sub optionwm42014-05-311-1/+1
| | | | This is more convenient.
* 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 wakeupswm42014-05-302-10/+18
| | | | | | 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.
* ao_alsa: use poll() to wait for devicewm42014-05-301-0/+30
| | | | | This means the audio feed thread is woken up exactly at the time new data is needed, instead of using a time-based heuristic.
* audio/out/push: add a way to wait for the audio device with poll()wm42014-05-302-3/+68
| | | | Will be used for ALSA.
* audio/out/push: add mechanism for event-based waitingwm42014-05-303-76/+143
| | | | | | | | | | | | | | | | 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).
* audio/out: adjust documentation commentswm42014-05-301-11/+19
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* audio/out/pull: remove race conditionswm42014-05-296-57/+68
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | There were subtle and minor race conditions in the pull.c code, and AOs using it (jack, portaudio, sdl, wasapi). Attempt to remove these. There was at least a race condition in the ao_reset() implementation: mp_ring_reset() was called concurrently to the audio callback. While the ringbuffer uses atomics to allow concurrent access, the reset function wasn't concurrency-safe (and can't easily be made to). Fix this by stopping the audio callback before doing a reset. After that, we can do anything without needing synchronization. The callback is resumed when resuming playback at a later point. Don't call driver->pause, and make driver->resume and driver->reset start/stop the audio callback. In the initial state, the audio callback must be disabled. JackAudio of course is different. Maybe there is no way to suspend the audio callback without "disconnecting" it (what jack_deactivate() would do), so I'm not trying my luck, and implemented a really bad hack doing active waiting until we get the audio callback into a state where it won't interfere. Once the callback goes from AO_STATE_WAIT to NONE, we can be sure that the callback doesn't access the ringbuffer or anything else anymore. Since both sched_yield() and pthread_yield() apparently are not always available, use mp_sleep_us(1) to avoid burning CPU during active waiting. The ao_jack.c change also removes a race condition: apparently we didn't initialize _all_ ao fields before starting the audio callback. In ao_wasapi.c, I'm not sure whether reset really waits for the audio callback to return. Kovensky says it's not guaranteed, so disable the reset callback - for now the behavior of ao_wasapi.c is like with ao_jack.c, and active waiting is used to deal with the audio callback.
* ao_wasapi: simplify nAvgBytesPerSec calculationMarcoen Hirschberg2014-05-281-4/+3
| | | | | Calculate nBlockAlign seperately to reuse in the calculation of nAvgBytesPerSec.
*