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path: root/audio/out/ao_lavc.c
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* mpv_talloc.h: rename from talloc.hDmitrij D. Czarkoff2016-01-111-1/+1
| | | | This change helps avoiding conflict with talloc.h from libtalloc.
* audio: use AVFrames with more than 8 channels correctlywm42015-10-261-0/+3
| | | | | | | Requires messy dealing with the extended_ fields. Don't bother with af_lavfi and ao_lavc for now. There are probably no valid use-cases for these.
* ao_lavc: minor simplificationwm42015-09-111-2/+2
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* ao_lavc: use new sample format determination codewm42015-09-101-20/+15
| | | | | | | This is just a refactor, which makes it use the previously introduced function, and allows us to make af_format_conversion_score() private. (We drop 2 unlikely warning messages too... who cares.)
* audio: remove unused legacy libavutil headerwm42015-08-071-1/+0
| | | | It was never used, but is a leftover from old times.
* audio: fix format function consistency issueswm42015-06-261-3/+3
| | | | | | | | | | | 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".
* Update license headersMarcin Kurczewski2015-04-131-4/+3
| | | | Signed-off-by: wm4 <wm4@nowhere>
* audio: make all format query shortcuts macrosKevin Mitchell2015-04-031-2/+2
| | | | | af_fmt_is_float and af_fmt_is_planar were previously inconsistent with AF_FORAMT_IS_SPECIAL/AF_FORMAT_IS_IEC61937
* ao_lavc: fix setting up AVFrame pointerswm42014-11-211-3/+4
| | | | | | The caller set up the "start" pointer array using the number of planes, the encode() function used the number of channels. This copied uninitialized values for packed formats, which makes Coverity warn.
* ao_lavc: fix dangling pointerswm42014-11-211-1/+1
| | | | Found by Coverity.
* ao_lavc, vo_lavc: Fix crashes in case of multiple init attempts.Rudolf Polzer2014-11-121-0/+8
| | | | | | | | | | | | When initialization failed, vo_lavc may cause an irrecoverable state in the ffmpeg-related structs. Therefore, we reject additional initialization attempts at least until we know a better way to clean up the mess. ao_lavc currently cannot be initialized more than once, yet it's good to do consistent changes there as well. Also, clean up uninit-after-failure handling to be less spammy.
* Move compat/ and bstr/ directory contents somewhere elsewm42014-08-291-1/+0
| | | | | | | | | 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.
* 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_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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* 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.
* af_fmt2bits: change to af_fmt2bps (bytes/sample) where appropriateMarcoen Hirschberg2014-05-281-1/+1
| | | | | | In most places where af_fmt2bits is called to get the bits/sample, the result is immediately converted to bytes/sample. Avoid this by getting bytes/sample directly by introducing af_fmt2bps.
* encode: fix PTS unit mismatchwm42014-05-101-5/+5
| | | | | | This used MP_NOPTS_VALUE to compare with ffmpeg-style int64_t PTS values. This probably happened to work, because both constants use the same value.
* encode: add a missing \n to a log callwm42014-04-101-1/+1
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* ao_lavc: set AVFrame.formatwm42014-03-161-0/+1
| | | | | Seems kind of wrong that this wasn't done, although it didn't have any bad consequences.
* encode: use new AVFrame APIwm42014-03-161-5/+3
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* audio/out: make draining a separate operationwm42014-03-091-1/+7
| | | | | | | | | | | | 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.
* encode: add lockingwm42014-03-091-4/+19
| | | | | | | | | | Since the AO will run in a thread, and there's lots of shared state with encoding, we have to add locking. One case this doesn't handle correctly are the encode_lavc_available() calls in ao_lavc.c and vo_lavc.c. They don't do much (and usually only to protect against doing --ao=lavc with normal playback), and changing it would be a bit messy. So just leave them.
* 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.
* encode: don't access ao->ptswm42014-03-071-1/+4
| | | | | | | | | | This field will be moved out of the ao struct. The encoding code was basically using an invalid way of accessing this field. Since the AO will be moved into its own thread too and will do its own buffering, the AO and the playback core might not even agree which sample a PTS timestamp belongs to. Add some extrapolation code to handle this case.
* audio/fmt-conversion.c: remove unknown audio format messageswm42013-12-211-0/+5
| | | | Same deal as with video/fmt-conversion.c.
* Split mpvcore/ into common/, misc/, bstr/wm42013-12-171-3/+3
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* 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_lavc: use af_format_conversion_score()wm42013-11-161-26/+29
| | | | | | | | | | This should allow it to select better fallback formats, instead of picking the first encoder sample format if ao->format is not equal to any of the encoder sample formats. Not sure what is supposed to happen if the encoder provides no compatible sample format (or no sample format list at all), but in this case ao_lavc.c still fails gracefully.
* ao_lavc: write the final audio chunks from uninit()Rudolf Polzer2013-11-161-7/+10
| | | | | | | | | These must be written even if there was no "final frame", e.g. due to the player being exited with "q". Although the issue is mostly of theoretical nature, as most audio codecs don't need the final encoding calls with NULL data. Maybe will be more relevant in the future.
* ao_lavc: fix crash with interleaved audio outputs.Rudolf Polzer2013-11-161-2/+4
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* ao_lavc: support non-interleaved audiowm42013-11-131-193/+42
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* Merge branch 'planar_audio'wm42013-11-121-10/+12
|\ | | | | | | | | Conflicts: audio/out/ao_lavc.c
| * audio/out: prepare for non-interleaved audiowm42013-11-121-11/+12
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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.
| * audio/out: reject non-interleaved formatswm42013-11-121-0/+1
| | | | | | | | | | | | | | | | | | | | 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.
* | ao_lavc: remove audio offset hack to ease supporting planar audio.Rudolf Polzer2013-11-111-66/+11
|/ | | | | | | | Now to shift audio pts when outputting to e.g. avi, you need an explicit facility to insert/remove initial samples, to avoid initial regions of the video to be sped up/slowed down. One such facility is the delay filter in libavfilter.
* audio: don't let ao_lavc access frontend internals, change gapless audiowm42013-11-081-28/+31
| | | | | | | | | | | | | | | | | | | | | | | 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
* audio/out: remove useless info struct and redundant fieldswm42013-10-231-6/+2
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* audio/out: do some mp_msg conversionswm42013-08-221-44/+30
| | | | | | | Use the new MP_ macros for some AOs instead of mp_msg. Not all AOs are converted, and some only partially. In some cases, some additional cosmetic changes are made.
* core: move contents to mpvcore (2/2)Stefano Pigozzi2013-08-061-4/+4
| | | | Followup commit. Fixes all the files references.
* 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 fieldswm42013-06-161-6/+4
| | | | | | | | | | | | | | | 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).
* encode_lavc strings: use new option syntaxRudolf Polzer2013-06-161-1/+1
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* Merge branch 'audio_changes'wm42013-05-121-54/+31
|\ | | | | | | | | Conflicts: audio/out/ao_lavc.c
| * audio/out: channel map selectionwm42013-05-121-0/+4
| | | | | | | | | | | | | | | | | | 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 mapwm42013-05-121-56/+30
| | | | | | | | | | | | 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.
* | encoding: fix final audio frame syncRudolf Polzer2013-04-281-10/+22
|/ | | | | When --ocopyts was used, the final audio frame got improper pts. Fixed by now using the play() logic to play the final frame too.
* audio/out, video/out: hide encoding VO/AOwm42013-02-061-0/+1
| | | | | | 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.
* sdl, encode_lavc: fix copyright headersRudolf Polzer2012-12-281-2/+2
| | | | | | Some of them had changes in 2012; extend their header. Fix project name.
* ao_lavc: stop using av_get_alt_sample_fmtRudolf Polzer2012-12-131-6/+6
| | | | Use av_get_planar_sample_fmt instead.
* af_lavcac3enc, encode: support planar formatsRudolf Polzer2012-12-031-3/+55
| | | | | | | This fixes operation with current ffmpeg releases. Note that this planarization is slow and should be reverted once proper planar audio support is there in mpv.
* Improve compatibility with Libav 0.8.4 and ffmpeg 0.11.2wm42012-11-141-0/+1
| | | | | | | | | | | Libav 0.8.4 is ridiculously old (in relative terms), so I don't know how many things are broken silently. Encoding is disabled, because the required API hasn't been added yet. (On the other hand, the old API can't be used in newer versions.) This should improve compatibility with ffmpeg 0.11.2 as well, which didn't define AV_CODEC_ID_SUBRIP yet.
* Rename directories, move files (step 2 of 2)wm42012-11-121-8/+7
| | | | | | | | | | | | 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/+621
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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