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* 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.
* Improve setting AVOptionswm42014-08-023-23/+12
| | | | | | | | Use OPT_KEYVALUELIST() for all places where AVOptions are directly set from mpv command line options. This allows escaping values, better diagnostics (also no more "pal"), and somehow reduces code size. Remove the old crappy option parser (av_opts.c).
* 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
* audio: ignore (some) decoding errors on initializationwm42014-07-291-0/+1
| | | | | | | | | | | | It probably happens relatively often that the first packet (or even the first N packets) of a stream will fail to decode, but decoding will eventually succeed at a later point. Before commit 261506e3, this was handled by an explicit retry loop (although this was also for other purposes), but with then was changed to abort on the first error. This makes it impossible to decode some audio streams. Change this so that errors are ignored for the first 50 packets, which should make it equivalent to the old code.
* audio: change playback restart and resyncingwm42014-07-285-31/+40
| | | | | | | | | | | | | | | | | | | | | This commit makes audio decoding non-blocking. If e.g. the network is too slow the playloop will just go to sleep, instead of blocking until enough data is available. For video, this was already done with commit 7083f88c. For audio, it's unfortunately much more complicated, because the audio decoder was used in a blocking manner. Large changes are required to get around this. The whole playback restart mechanism must be turned into a statemachine, especially since it has close interactions with video restart. Lots of video code is thus also changed. (For the record, I don't think switching this code to threads would make this conceptually easier: the code would still have to deal with external input while blocked, so these in-between states do get visible [and thus need to be handled] anyway. On the other hand, it certainly should be possible to modularize this code a bit better.) This will probably cause a bunch of regressions.
* 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.
* audio: fix timestampswm42014-07-243-2/+1
| | | | | | | | | Accidentally broken in b6af44d3. For ad_lavc (and in general), the PTS was not updated correctly when filtering only parts of audio frames, and for ad_mpg123 and ad_spdif the PTS was additionally offset by the frame size. This could lead to incorrect time display, and possibly broken A/V sync.
* audio: adjust format change codewm42014-07-241-8/+9
| | | | | | Execute the format change based on whether we logically detected EOF (after filters), instead of when the decode buffer was drained. It's slightly cleaner. (The requirement of len>0 existed before.)
* audio: fix race condition in EOF codewm42014-07-242-3/+3
| | | | | | | | | | Don't return an EOF code if there's still buffered data. Also, don't call demux_stream_eof() in the playloop. There's probably nothing wrong with it, but it's cleaner not to use it. Also give AD_EOF its own value, so that a decoding error doesn't drain audio by causing an EOF condition.
* audio: cosmeticswm42014-07-241-9/+5
| | | | | | | Move a function call, which does not change semantics. Write the extra buffer sample count in a more straight-forward way; the old code was not meaningful in any way (anymore).
* audio: remove unnecessary codewm42014-07-241-3/+0
| | | | | | | | It's true that the decoder can successfully decode, but return no data (for various reasons). We don't need to handle this specially, though. We just let the decoder decode some more data. This doesn't increase the danger of an endless loop either, because audio_decode() already calls this function until enough is decoded.
* 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.
* audio: move initial decode to generic codewm42014-07-216-239/+127
| | | | | | | | | | | | This commit mainly moves the initial decoding of data (done to probe the audio format) to generic code. This will make it easier to make audio decoding non-blocking in a later commit. This commit also changes how decoders return data: instead of having them write the data into a prepared buffer, they return a reference to an internal buffer (by setting dec_audio.decoded). This makes it significantly easier to handle audio format changes, since the decoders don't really need to care anymore.
* ad_lavc: drop questionable fallback codewm42014-07-211-6/+0
| | | | | | | | | | | If the decoder didn't set a samplerate, it was initialized from the container samplerate. This probably didn't make much sense, because it's passed to the decoder on initialization (so it could definitely use it). It's an artifact from commit 66a9eb57 (which removed some Matroska-specific non- sense), and I've never seen it actually happen since it was made into a warning. Just get rid of it.
* audio: remove unused metadata fieldwm42014-07-214-6/+0
| | | | | This was used for replaygain at some point, until replaygain info was passed through explicitly.
* audio: use symbolic constants instead of magic integerswm42014-07-205-12/+18
| | | | Similar to commit 26468743.
* 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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* Remove some mp_msg calls with no trailing \nwm42014-07-131-6/+6
| | | | | | | The final goal is all mp_msg calls produce complete lines. We want this because otherwise, race conditions could corrupt the terminal output, and it's inconvenient for the client API too. This commit works towards this goal. There's still code that has this not fixed yet, though.
* 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.
* build: deal with endian messwm42014-07-101-1/+1
| | | | | | | | | | | | | | | | | | | | There is no standard mechanism for detecting endianess. Doing it at compile time in a portable way is probably hard. Doing it properly with a configure check is probably hard too. Using the endian definitions in <sys/types.h> (usually includes <endian.h>, which is not available everywhere) works under circumstances, but the previous commit broke it on OSX. Ideally all code should be endian dependent, but that is not possible due to the dependencies (such as FFmpeg, some video output APIs, some audio output APIs). Create a header osdep/endian.h, which contains various fallbacks. Note that the last fallback uses libavutil; however, it's not clear whether AV_HAVE_BIGENDIAN is a public symbol, or whether including <libavutil/bswap.h> really makes it visible. And in fact we don't want to pollute the namespace with libavutil definitions either. Thus it's only the last fallback.
* 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.)
* af_volume: fix calculations including replay-gainMohammad Alsaleh2014-06-281-2/+2
| | | | | | | | | | | | | rgain is not an additive value. It's a multiplier/gain. Previous behaviour produced negative level values in some cases (when rgain < 1.0) which caused volume to be louder when its value was lowered. CC: @mpv-player/stable Signed-off-by: Mohammad Alsaleh <CE.Mohammad.AlSaleh@gmail.com> Signed-off-by: wm4 <wm4@nowhere>
* 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.
* ad_lavc: make option struct localwm42014-06-111-9/+23
| | | | Similar to previous commit.
* Add more constwm42014-06-1130-68/+68
| | | | | | | 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.
* player: show "neutral" position markers for OSD barswm42014-06-082-0/+7
| | | | This commit implements them for volume and some video properties.
* 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.
* player: hide audio/video codec and file format messageswm42014-05-311-2/+1
| | | | | None of these are very important usually. For error analysis, the plain log is useless anyway, and this information is still printed with "-v".
* 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 we