path: root/audio/out/ao_null.c
Commit message (Collapse)AuthorAgeFilesLines
* mpv_talloc.h: rename from talloc.hDmitrij D. Czarkoff2016-01-111-1/+1
| | | | This change helps avoiding conflict with talloc.h from libtalloc.
* ao_null: add an option for testing channel layout selectionwm42015-04-271-2/+16
* Update license headersMarcin Kurczewski2015-04-131-5/+4
| | | | Signed-off-by: wm4 <wm4@nowhere>
* ao_null: add emulation for certain broken behaviorwm42015-01-301-1/+12
| | | | | I'm not sure how common this behavior possibly is; well whatever. This option will allow reproducing such behavior, and help debugging it.
* audio/out: consistently use double return type for get_delaywm42014-11-091-1/+1
| | | | | ao_get_delay() returns double, but the get_delay callback still returned float.
* audio/out: always round get_space on period sizewm42014-09-061-1/+2
| | | | | | | | | | | 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_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.
* 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_null: fix unit mismatch with latency optionwm42014-05-041-5/+8
| | | | It's in seconds, but the code used it as sample count.
* ao_null: add simulated device latency, simulate EOF problemswm42014-04-171-3/+19
| | | | | This EOF problems happen at least with PulseAudio, but since it's hard to reproduce, let ao_null optionally simulate it.
* audio/out: make draining a separate operationwm42014-03-091-2/+8
| | | | | | | | | | | | 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.
* ao_null: add option for simulated device speedwm42014-03-091-2/+8
| | | | Helps with testing and debugging.
* audio/out: make ao struct opaquewm42014-03-091-0/+1
| | | | | | 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.
* Split mpvcore/ into common/, misc/, bstr/wm42013-12-171-1/+1
* Move options/config related files from mpvcore/ to options/wm42013-12-171-1/+1
| | | | | | | | | 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.
* ao_null: fix simulated buffer sizewm42013-11-191-1/+1
| | | | | The size accidentally defaulted to 200 seconds instead of 200 milliseconds, which had fatal consequences when trying to use it.
* ao_null: properly simulate final chunk, add buffer optionswm42013-11-171-19/+50
| | | | | | | | | | Simulate proper handling of AOPLAY_FINAL_CHUNK. Print when underruns occur (i.e. running out of data). Add some options that control simulated buffer and outburst sizes. All this is useful for debugging and self-documentation. (Note that ao_null always was supposed to simulate an ideal AO, which is the reason why it fools people who try to use it for benchmarking video.)
* ao_null: add untimed sub-optionwm42013-11-131-2/+18
* ao_null: support pausing properlywm42013-11-131-4/+14
| | | | | | ao_null should simulate a "perfect" AO, but framestepping behaved quite badly with it. Framstepping usually exposes problems with AOs dropping their buffers on pause, and that's what happened here.
* ao_null: support non-interleaved audiowm42013-11-121-17/+20
| | | | Simply change internals from using byte counts to sample counts.
* audio/out: prepare for non-interleaved audiowm42013-11-121-3/+4
| | | | | | | | | | | | | | | | | | | This comes with two internal AO API changes: 1. 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.
* audio/out: reject non-interleaved formatswm42013-11-121-0/+2
| | | | | | | | | | No AO can handle these, so it would be a problem if they get added later, and non-interleaved formats get accepted erroneously. Let them gracefully fall back to other formats. Most AOs actually would fall back, but to an unrelated formats. This is covered by this commit too, and if possible they should pick the interleaved variant if a non-interleaved format is requested.
* audio/out: remove useless info struct and redundant fieldswm42013-10-231-6/+2
* audio/out: remove options argument from init()wm42013-07-221-1/+1
| | | | Same as with VOs in the previous commit.
* audio/out: remove ao->outburst/buffersize fieldswm42013-06-161-7/+9
| | | | | | | | | | | | | | | 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->bpswm42013-06-161-1/+0
| | | | | | | 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).
* Replace all calls to GetTimer()/GetTimerMS()wm42013-05-261-4/+4
| | | | | | | | | | | | | | | | | | | | | | GetTimer() is generally replaced with mp_time_us(). Both calls return microseconds, but the latter uses int64_t, us defined to never wrap, and never returns 0 or negative values. GetTimerMS() has no direct replacement. Instead the other functions are used. For some code, switch to mp_time_sec(), which returns the time as double float value in seconds. The returned time is offset to program start time, so there is enough precision left to deliver microsecond resolution for at least 100 years. Unless it's casted to a float (or the CPU reduces precision), which is why we still use mp_time_us() out of paranoia in places where precision is clearly needed. Always switch to the correct time. The whole point of the new timer calls is that they don't wrap, and storing microseconds in unsigned int variables would negate this. In some cases, remove wrap-around handling for time values.
* audio/out: channel map selectionwm42013-05-121-0/+6
| | | | | | | | | Make all AOs use what has been introduced in the previous commit. Note that even AOs which can handle all possible layouts (like ao_null) use the new functions. This might be important if in the future ao_select_champ() possibly honors global user options about downmixing and so on.
* ao: remove ao_driver.is_new fieldwm42013-05-121-1/+0
| | | | Is unused, is completely pointless.
* audio/out: switch to channel mapwm42013-05-121-2/+2
| | | | | | 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.
* 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/+129
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.