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* player: unrangle one aspect of audio EOF handlingwm42014-04-171-0/+5
| | | | | | | | | | | | | | | | | | For some reason, the buffered_audio variable was used to "cache" the ao_get_delay() result. But I can't really see any reason why this should be done, and it just seems to complicate everything. One reason might be that the value should be checked only if the AO buffers have been recently filled (as otherwise the delay could go low and trigger an accidental EOF condition), but this didn't work anyway, since buffered_audio is set from ao_get_delay() anyway at a later point if it was unset. And in both cases, the value is used _after_ filling the audio buffers anyway. Simplify it. Also, move the audio EOF condition to a separate function. (Note that ao_eof_reached() probably could/should whether the last ao_play() call had AOPLAY_FINAL_CHUNK set to avoid accidental EOF on underflows, but for now let's keep the code equivalent.)
* ao: remove redundant get_delay checkwm42014-04-171-4/+0
| | | | It did nothing; the real check is in push.c.
* ao: print (estimated) device buffer size on init in verbose modewm42014-03-141-1/+3
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* audio/out: reduce amount of audio bufferingwm42014-03-101-3/+0
| | | | | | | | | | | | | | | | | | | | | | | | | Since the addition of the AO feed thread, 200ms of latency (MIN_BUFFER) was added to all push-based AOs. This is not so nice, because even AOs with relatively small buffering (e.g. ao_alsa on my system with ~170ms of buffer size), the additional latency becomes noticable when e.g. toggling mute with softvol. Fix this by trying to keep not only 200ms minimum buffer, but also 200ms maximum buffer. In other words, never buffer beyond 200ms in total. Do this by estimating the AO's buffer fill status using get_space and the initially known AO buffer size (the get_space return value on initialization, before any audio was played). We limit the maximum amount of data written to the soft buffer so that soft buffer size and audio buffer size equal to 200ms (MIN_BUFFER). To avoid weird problems with weird AOs, we buffer beyond MIN_BUFFER if the AO's get_space requests more data than that, and as long as the soft buffer is large enough. Note that this is just a hack to improve the latency. When the audio chain gains the ability to refilter data, this won't be needed anymore, and instead we can introduce some sort of buffer replacement function in order to update data in the soft buffer.
* audio/out: make draining a separate operationwm42014-03-091-7/+15
| | | | | | | | | | | | 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.
* audio/out: feed AOs from a separate threadwm42014-03-091-25/+45
| | | | | | | | | | | | | | | | | | This has 2 goals: - Ensure that AOs have always enough data, even if the device buffers are very small. - Reduce complexity in some AOs, which do their own buffering. One disadvantage is that performance is slightly reduced due to more copying. Implementation-wise, we don't change ao.c much, and instead "redirect" the driver's callback to an API wrapper in push.c. Additionally, we add code for dealing with AOs that have a pull API. These AOs usually do their own buffering (jack, coreaudio, portaudio), and adding a thread is basically a waste. The code in pull.c manages a ringbuffer, and allows callback-based AOs to read data directly.
* ao: remove opts fieldwm42014-03-091-1/+0
| | | | Apparently unused.
* audio/out: make ao struct opaquewm42014-03-091-2/+36
| | | | | | 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.
* ao: document some functionswm42014-02-281-0/+23
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* build: fix usage of HAVE_SDL1 defineStefano Pigozzi2014-01-251-1/+1
| | | | This is needed after fd1f8ed49.
* msg: rename mp_msg_log -> mp_msgwm42013-12-211-1/+1
| | | | Same for companion functions.
* m_option, m_config: mp_msg conversionswm42013-12-211-1/+1
| | | | | | | | Always pass around mp_log contexts in the option parser code. This of course affects all users of this API as well. In stream.c, pass a mp_null_log, because we can't do it properly yet. This will be fixed later.
* ao: some missing mp_msg conversionswm42013-12-211-13/+17
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* Split mpvcore/ into common/, misc/, bstr/wm42013-12-171-2/+2
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* Move options/config related files from mpvcore/ to options/wm42013-12-171-2/+2
| | | | | | | | | 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.
* Replace mp_tmsg, mp_dbg -> mp_msg, remove mp_gtext(), remove set_osd_tmsgwm42013-12-161-2/+2
| | | | | | | | | The tmsg stuff was for the internal gettext() based translation system, which nobody ever attempted to use and thus was removed. mp_gtext() and set_osd_tmsg() were also for this. mp_dbg was once enabled in debug mode only, but since we have log level for enabling debug messages, it seems utterly useless.
* options: add options that set defaults for af/vf/ao/vowm42013-12-011-0/+2
| | | | | | | | There are some use cases for this. For example, you can use it to set defaults of automatically inserted filters (like af_lavrresample). It's also useful if you have a non-trivial VO configuration, and want to use --vo to quickly change between the drivers without repeating the whole configuration in the --vo argument.
* audio: switch output to mp_audio_bufferwm42013-11-121-5/+7
| | | | | | Replace the code that used a single buffer with mp_audio_buffer. This also enables non-interleaved output operation, although it's still disabled, and no AO supports it yet.
* audio/out: prepare for non-interleaved audiowm42013-11-121-7/+10
| | | | | | | | | | | | | | | | | | | 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.
* ao: add ao_play_silence, use for ao_alsa and ao_osswm42013-11-101-0/+12
| | | | | Also add a corresponding function to audio/format.c, which fills an audio block with silence.
* ao: print requested audio format on initwm42013-11-091-0/+4
| | | | Also remove the rather bad/incomplete log calls from ao_alsa and ao_oss.
* audio: don't let ao_lavc access frontend internals, change gapless audiowm42013-11-081-6/+0
| | | | | | | | | | | | | | | | | | | | | | | ao_lavc.c accesses ao->buffer, which I consider internal. The access was done in ao_lavc.c/uninit(), which tried to get the left-over audio in order to write the last (possibly partial) audio frame. The play() function didn't accept partial frames, because the AOPLAY_FINAL_CHUNK flag was not correctly set, and handling it otherwise would require an internal FIFO. Fix this by making sure that with gapless audio (used with encoding), the AOPLAY_FINAL_CHUNK is set only once, instead when each file ends. Basically, move the hack in ao_lavc's uninit to uninit_player. One thing can not be entirely correctly handled: if gapless audio is active, we don't know really whether the AO is closed because the file ended playing (i.e. we want to send the buffered remainder of the audio to the AO), or whether the user is quitting the player. (The stop_play flag is overwritten, fixing that is perhaps not worth it.) Handle this by adding additional code to drain the AO and the buffers when playback is quit (see play_current_file() change). Test case: mpv avdevice://lavfi:sine=441 avdevice://lavfi:sine=441 -length 0.2267 -gapless-audio
* configure: uniform the defines to #define HAVE_xxx (0|1)Stefano Pigozzi2013-11-031-13/+13
| | | | | | | | | | | | | | | | | | | | | The configure followed 5 different convetions of defines because the next guy always wanted to introduce a new better way to uniform it[1]. For an hypothetic feature 'hurr' you could have had: * #define HAVE_HURR 1 / #undef HAVE_DURR * #define HAVE_HURR / #undef HAVE_DURR * #define CONFIG_HURR 1 / #undef CONFIG_DURR * #define HAVE_HURR 1 / #define HAVE_DURR 0 * #define CONFIG_HURR 1 / #define CONFIG_DURR 0 All is now uniform and uses: * #define HAVE_HURR 1 * #define HAVE_DURR 0 We like definining to 0 as opposed to `undef` bcause it can help spot typos and is very helpful when doing big reorganizations in the code. [1]: http://xkcd.com/927/ related
* audio/out: remove useless info struct and redundant fieldswm42013-10-231-3/+3
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* mp_msg: remove gettext() supportwm42013-10-181-1/+1
| | | | | | | | | Was disabled by default, was never used, internal support was inconsistent and poor, and there has been virtually no interest in creating translations. And I don't even think that a terminal program should be translated. This is something for (hypothetical) GUIs.
* audio/out: add sndio supportChristian Neukirchen2013-10-031-0/+4
| | | | Based on an earlier patch for mplayer by Alexandre Ratchov <alex@caoua.org>
* ao: remove some leftoverswm42013-08-221-5/+0
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* core: move contents to mpvcore (2/2)Stefano Pigozzi2013-08-061-4/+4
| | | | Followup commit. Fixes all the files references.
* audio/out: add support for new logging APIStefano Pigozzi2013-08-011-7/+12
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* audio/out: remove options argument from init()wm42013-07-221-2/+3
| | | | Same as with VOs in the previous commit.
* ao_wasapi: Make default on Windows.Diogo Franco (Kovensky)2013-07-221-3/+3
| | | | Ahead of OSS because cygwin provides OSS.
* ao_wasapi0: Rename to ao_wasapiDiogo Franco (Kovensky)2013-07-221-3/+3
| | | | | Nobody knows what the 0 was for. There's no "WASAPI version 0". Just take it out.
* options: hide encoding AO/VO in help outputwm42013-07-211-0/+1
| | | | | These can't be used manually. Encoding is enabled with -o instead, and the encoding AO/VO is selected using internal mechanisms.
* options: use new option code for --aowm42013-07-211-86/+80
| | | | This requires completely refactoring the AO creation code too.
* ao_wasapi0: add new wasapi event mode aoJonathan Yong2013-06-181-0/+4
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* audio/out: remove ao->outburst/buffersize fieldswm42013-06-161-2/+1
| | | | | | | | | | | | | | | 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.
* audio/out: don't require AOs to set ao->bpswm42013-06-161-9/+8
| | | | | | | 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).
* audio/out: remove wrapper for old AOswm42013-06-161-54/+0
| | | | It's unused now.
* audio: add channel map selection functionwm42013-05-121-0/+10
| | | | | | | | | | | | | | | | | | | | The point is selecting a minimal fallback. The AOs will call this through the AO API, so it will be possible to add options affecting the general channel layout selection. It provides the following mechanism to AOs: - forcing the correct channel order - downmixing to stereo if no layout is available - allow 5.1 <-> 5.1(side) fallback - handling "unknown" channel layouts This is quite weak and lots of code/complexity for little gain. All AOs already made sure the channel order was correct, and the fallback is of little value, and could perhaps be done in the frontend instead, like stereo downmixing with --channels=2 is handled. But I'm not really sure how this stuff should _really_ work, and the new code will hopefully provides enough flexibility to make radical changes to channel layout negotiation easier.
* audio/out: switch to channel mapwm42013-05-121-1/+1
| | | | | | 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.
* audio/out, video/out: hide encoding VO/AOwm42013-02-061-8/+17
| | | | | | mpv -ao help and mpv -vo help shouldn't show the encoding outputs (named "lavc" on both cases). Also make it impossible to select these manually when not encoding.
* audio/out: prefer ao_dsound over ao_portaudiowm42013-02-061-3/+3
| | | | | | On Linux, ao_portaudio has weird freezing issues (possibly specific to the ALSA backend, though). Also ao_dsound is more likely to get multi- channel audio output right, and ao_portaudio probably mangles these.
* vo/ao: SDL 1.2+ audio driver, SDL 2.0+ accelerated video driverRudolf Polzer2012-12-281-0/+4
| | | | | | | | | | | This mainly serves as a fallback for platforms where nothing better is available; also as a debugging help. Both the audio and video driver are not first class - the audio driver lacks delay detection, and the video driver only supports a single YUV color space. Configure options: --disable-sdl2 to disable SDL 2.0+ detection, --disable-sdl to disable SDL 1.2+ detection. Both options need to be specified to turn off SDL support entirely.
* Rename directories, move files (step 2 of 2)wm42012-11-121-2/+2
| | | | | | | | | | | | Finish renaming directories and moving files. Adjust all include statements to make the previous commit compile. The two commits are separate, because git is bad at tracking renames and content changes at the same time. Also take this as an opportunity to remove the separation between "common" and "mplayer" sources in the Makefile. ("common" used to be shared between mplayer and mencoder.)
* Rename directories, move files (step 1 of 2) (does not compile)wm42012-11-121-0/+294
Tis drops the silly lib prefixes, and attempts to organize the tree in a more logical way. Make the top-level directory less cluttered as well. Renames the following directories: libaf -> audio/filter libao2 -> audio/out libvo -> video/out libmpdemux -> demux Split libmpcodecs: vf* -> video/filter vd*, dec_video.* -> video/decode mp_image*, img_format*, ... -> video/ ad*, dec_audio.* -> audio/decode libaf/format.* is moved to audio/ - this is similar to how mp_image.* is located in video/. Move most top-level .c/.h files to core. (talloc.c/.h is left on top- level, because it's external.) Park some of the more annoying files in compat/. Some of these are relicts from the time mplayer used ffmpeg internals. sub/ is not split, because it's too much of a mess (subtitle code is mixed with OSD display and rendering). Maybe the organization of core is not ideal: it mixes playback core (like mplayer.c) and utility helpers (like bstr.c/h). Should the need arise, the playback core will be moved somewhere else, while core contains all helper and common code.
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