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Author SHA1 Message Date
wm4 463b877ec0 Release 0.1.2 2013-08-19 01:12:43 +02:00
wm4 72c81aa111 vdpau: don't try to match codec profiles
Instead, do what MPlayer did all these years. It worked for them, so
there's probably no reason to change this.

Apparently fixes playback with some files, where the VDPAU decoder does
not formally support a profile, but decoding works with a more powerful
profile anyway.

Though note that MPlayer did this because it couldn't do it in a better
way (no decoder reported profiles available when creating the VDPAU
decoder), so it's not entirely clear whether this is a good idea. An
alterbative implementation might try to map the profiles exactly, and
do some fall backs if the exact profile is not available. But this
hack-solution works too.

Conflicts:
	video/decode/vdpau.c
2013-08-19 01:09:39 +02:00
wm4 9f7c014930 changes: add some things for completeness 2013-08-19 01:07:32 +02:00
wm4 97eddff865 vo_vdpau: fix screenshots if panscan is used
When panscan was used, i.e. the video is cropped to make the video fill
the screen if video and screen aspects don't match, screenshots
contained only the visible region of the source video, stretched to
original video size.
2013-08-18 05:58:33 +02:00
wm4 cf880b9070 mplayer: don't make restored options from quit_watch_later per-file local
Consider:

    mpv file1.mkv file2.mkv

and file1.mkv is restored from an earlier session when quit_watch_later
was used. Then all restored options were reset when file2.mkv is played,
even if the user changed them during playback. This affects for example
the fullscreen setting.

Make it so that after finishing a resumed file, the previously restored
settings are not reset again. (Which means only resuming will forcefully
overwrite the settings.)
2013-08-18 05:58:28 +02:00
wm4 3478575419 manpage: fix incorrect ao_alsa example, extend changes.rst
One example in ao.rst used the old syntax with mangled device names.
Fix it. Mention some shell related caveats. Explicitly mention the
change of device name syntax in changes.rst, because it seems to be
a common issue.
2013-08-15 23:46:18 +02:00
wm4 88a53309cd vdpau: fix compilation on Libav
Libav's <libavcodec/vdpau.h> header uses some libavocdec symbols without
forward-declaring them and without including the headers declaring them.
FFmpeg's header for this is fine.
2013-08-15 23:46:07 +02:00
wm4 da76eb9f14 sub: fix accidental subtitle overlaps
The fix_overlaps_and_gaps() function in dec_sub.c fixes small gaps or
overlaps between subtitle events. However, sometimes it could happen
that the corrected subtitle events could overlap by 1ms due to bad
rounding, making libass shift subtitles to reduce collisions. (The
second subtitle will be shown above the previous one, even if both
subtitles are visible only for 1ms.)

sd_ass.c rounds the timestamps when converting to integers for unknown
reasons. I think it would work fine without that rounding, but since
I have no clue why it rounds, and since it could be needed to ensure
correct timestamps with ASS subtitles demuxed from Matroska, I'd rather
not touch it. So the solution is to use already rounded timestamps to
calculate the new subtitle duration in fix_overlaps_and_gaps().

See github issue #182.
2013-08-15 23:43:36 +02:00
wm4 df6901b653 x11: fix --ontop
Apply it after mapping.
2013-08-15 23:43:33 +02:00
Stefano Pigozzi 53e4ba26b5 changes.rst: document some OSX and windows changes 2013-08-15 23:43:29 +02:00
wm4 f10a3cac0f demux: move demux_mf before demux_subreader
demux_subreader is quite aggressive, and sometimes detects random
strings in EXIF as subtitle text.
2013-08-15 23:43:11 +02:00
wm4 7370ad2120 demux_subreader: report what subtitle format has been found 2013-08-15 23:43:05 +02:00
Noble Huang ed2d819ea5 stream_bluray: fix bd:// url segfault introduced by commit bc1d61 2013-08-15 23:43:00 +02:00
wm4 4f728cfb99 Add VERSION for release
No idea what the correct workflow is for these things. In any case, we
somehow have to make the github release system to include the correct
version in its tarball. I don't think you can provide your own tarballs
either, so it has to be in git.
2013-08-12 00:18:22 +02:00
614 changed files with 67677 additions and 84304 deletions

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@@ -2,7 +2,6 @@
*.a
*.d
*.exe
*.pyc
.depend
/config.h
@@ -11,26 +10,13 @@
/mpv
/mpv.app
/version.h
/input/input_conf.h
/mpvcore/input/input_conf.h
/tags
/TAGS
/video/out/gl_video_shaders.h
/video/out/x11_icon.inc
/video/out/vdpau_template.c
/demux/ebml_defs.c
/demux/ebml_types.h
/sub/osd_font.h
/player/lua/*.inc
/DOCS/man/mpv.1
/DOCS/man/mpv.aux
/DOCS/man/mpv.log
/DOCS/man/mpv.out
/DOCS/man/mpv.pdf
/DOCS/man/mpv.toc
/DOCS/man/*/mpv.1
/waf
/build
/.waf*
/.lock-waf_*
/old_build
/Makefile
+10 -20
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@@ -1,8 +1,5 @@
language: c
os:
- linux
- osx
env:
- LIBAV=libav-stable
- LIBAV=libav-git
@@ -10,31 +7,24 @@ env:
- LIBAV=ffmpeg-git
compiler:
- clang
branches:
only:
- master
- ci
matrix:
exclude:
- os: osx
env: LIBAV=libav-stable
- os: osx
env: LIBAV=libav-git
- os: osx
env: LIBAV=ffmpeg-git
before_install: ./travis-deps libass-stable $LIBAV
script:
- ./bootstrap.py
- ./waf configure
- ./waf build
script: ./configure && make
notifications-policy: &notifications-policy
on_success: change
on_failure: always
notifications:
email: false
email:
recipients:
- mpv-team@googlegroups.com
<<: *notifications-policy
irc:
channels:
- "irc.freenode.org#mpv-player-dev"
on_success: change
on_failure: always
<<: *notifications-policy
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+30 -14
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@@ -1,19 +1,35 @@
mpv is a fork of mplayer2, which is a fork of MPlayer.
Original MPlayer Copyright file follows below.
mpv as a whole is licensed as GPL version 2 or later (see LICENSE). Most source
files are GPLv2+, but some files are available under a more liberal license,
such as LGPLv2+, BSD, MIT, ISC, and possibly others. Look at the copyright
header of each source file, and grep the sources for "Copyright" if you need
to know details. Files without Copyright notice are licensed as LGPLv2+.
MPlayer was originally written by Árpád Gereöffy and has been extended and
worked on by many more since then, see the AUTHORS file for an (incomplete)
list. MPlayer as a whole is copyrighted by the MPlayer team. Individual
copyright notices can be found in file headers.
For information about authors and contributors, consult the git log, which
contains the complete SVN and CVS history as well.
Most files in MPlayer are licensed under the GNU General Public License,
version 2 with an explicit "or later" clause or a compatible license
(LGPL 2.1+, MIT/BSD). An exception are the files talloc.c and talloc.h,
which are LGPL 3+.
Thus MPlayer as a whole can be redistributed under terms compatible with
the GNU General Public License, version 3; see the LICENSE file for details.
Note that mplayer2 as a whole is licensed under GPLv3+. This is because it uses
a copy of talloc (part of Samba), which is LGPLv3+, and the next compatible
license for this mix is GPLv3+.
MPlayer includes code from several external sources. General information and
license details are listed below:
MPlayer as a whole is licensed under GPLv2 (incompatible to GPLv3!), because
some files are licensed to GPLv2 (and _not_ any later version of the license).
In particular, this affects the file libmpdemux/demux_ty_osd.c. It is disabled
under mplayer2, and has been removed from mpv.
Name: a couple of files copied from FFmpeg
Version: updated to different versions
URL: http://www.ffmpeg.org
Directory: compat/x86_cpu.h
Copyright: Many, see individual files for copyright notices.
License: GNU Lesser General Public License
Name: dvbstream
Version: 0.4.3-pre3 (CVS checkout)
URL: http://sourceforge.net/projects/dvbtools/
Directory: stream/dvbin.h stream/stream_dvb.c
Copyright: 2001-2002 Dave Chapman <dave@dchapman.com>
License: GNU General Public License
This is not a complete list. grep the sources for "Copyright".
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1. Input layer, supported devices, methods:
- plain file, with seeking
- STDIN, without seeking backward
- network streaming (currently plain wget-like HTTP and MMS (.asx))
- VCD (Video CD) track, by direct CDROM device access (not requires mounting disc)
- DVD titles using .IFO structure, by direct DVD device access (not requires mounting disc)
- DVD titles using menu navigation (experimental/alpha, not yet finished!!!)
- CDDA - raw audio from audio CD-ROM discs (using cdparanoia libs)
- RTP streaming (mpeg-ps over multicast only)
- LIVE555 streaming - support SDP/RTSP (using the LIVE555 libs)
- SMB - file access over samba (experimental)
2. Demuxer/parser layer, supported file/media formats:
- MPEG streams (ES,PES,PS. no TS support yet)
note: mpeg demuxer silently ignore non-mpeg content, and find mpeg packets
in arbitrary streams. it means you can play directly VCD images (for example
CDRwin's .BIN files) without extracting mpeg files first (with tools like vcdgear)
It accepts all PES variants, including files created by VDR.
Note: VOB (video object) is simple mpeg stream, but it usually has 01BD
packets which may contain subtitles and non-mpeg audio. Usually found on DVD discs.
Headers: mpeg streams has no global header. each frame sequence (also called GOP,
group of pictures) contains an sequence header, it describes that block.
In normal mpeg 1/2 content there are groups of 12-15 frames (24/30 fps).
It means you can freely seek in mpeg streams, and even can cut it to
small parts with standard file tools (dd, cut) without destroying it.
Codecs: video is always mpeg video (mpeg1, mpeg2 or mpeg4).
audio is usually mpeg audio (any layer allowed, but it's layer 2 in most files)
but 01BD packets may contain AC3, DTS or LPCM too.
FPS: mpeg2 content allows variable framerate, in form of delayed frames.
It's mostly used for playback 24fps content at 29.97/30 fps (NTSC) rate.
(so called Telecine or 3:2 pulldown effect)
It means you see 30 frames per second, but there are only 24 different
pictures and some of them are shown longer to fill 30 frame time.
If you encode such files with mencoder, using -ofps 24 or -ofps 23.976
is recommended.
- AVI streams.
Two kind of RIFF AVI files exists:
1. interleaved: audio and video content is interleaved. it's faster and
requires only 1 reading thread, so it's recommended (and mostly used).
2. non-interleaved: audio and video aren't interleaved, i mean first come
whole video followed by whole audio. it requires 2 reading process or
1 reading with lots of seeking. very bad for network or cdrom.
3. badly interleaved streams: mplayer detects interleaving at startup and
enables -ni option if it finds non-interleaved content. but sometimes
the stream seems to be interleaved, but with bad sync so it should be
played as non-interleaved otherwise you get a-v desync or buffer overflow.
MPlayer supports 2 kind of timing for AVI files:
- bps-based: it is based on bitrate/samplerate of video/audio stream.
this method is used by most players, including avifile and wmp.
files with broken headers, and files created with VBR audio but not
vbr-compliant encoder will result a-v desync with this method.
(mostly at seeking).
- interleaving-based: note: it can't be used togethwer with -ni
it doesn't use bitrate stuff of header, it uses the relative position
of interleaved audio and video chunks. makes some badly encoded file
with vbr audio playable.
Headers: AVI files has a mandatory header at the begin of the file,
describing video parameters (resolution, fps) and codecs. Optionally
they have an INDEX block at the end of the file. It's optional, but
most files has such block, because it's REQUIRED for seeking.
Btw usually it can be rebuilt from file content, mplayer does it with
the -idx switch. MPlayer can recreate broken index blocks using -forceidx.
As AVI files needs index for random access, broken files with no index
are usually unplayable.
Of course, cutting/joining AVI files needs special programs.
Codecs: any audio and video codecs allowed, but I note that VBR audio is
not well supported by most players. The file format makes it possible to
use VBR audio, but most players expect CBR audio and fails with VBR,
as VBR is unusual, and Microsoft's AVI specs only describe CBR audio.
I also note, that most AVI encoders/multiplexers create bad files if
using VBR audio. only 2 exception (known by me): NaNDub and MEncoder.
FPS: only constant framerate allowed, but it's possible to skip frames.
- ASF streams:
ASF (active streaming format) comes from Microsoft. they developed two
variant of ASF, v1.0 and v2.0. v1.0 is used by their media tools (wmp and
wme) and v2.0 is published and patented :). of course, they differ,
no compatibility at all. (it's just a legality game)
MPlayer supports only v1.0, as nobody ever seen v2.0 files :)
Note, that .ASF files are nowdays come with extension .WMA or .WMV.
UPDATE: MS recently released the ASF v1.0 specs too, but it has some
restrictions making it illegal to read by us :)
Headers: Stream headers (codecs parameters) can be everywhere (in theory),
but all files i've seen had it at the beginning of the file.
Asf uses fixed packet size, so it is seekable without any INDEX block,
and broken files are playable well.
Codecs: video is mostly microsoft's mpeg4 variants: MP42, MP43 (aka DivX),
WMV1 and WMV2. but any codecs allowed.
audio is usually wma or voxware, sometimes mp3, but any codecs allowed.
FPS: no fixed fps, every video frame has an exact timestamp instead.
I've got stream with up to 3 sec frame display times.
- QuickTime / MOV files:
They come from Mac users, usually with .mov or .qt extension, but as
MPEG Group chose quicktime as recommended file format for MPEG4,
sometimes you meet quicktime files with .mpg or .mp4 extension.
At first look, it's a mixture of ASF and AVI.
It requires INDEX block for random access and seeking, and even for
playback, like AVI, but uses timestamps instead of constant framerate
and has more flexible stream options (including network stuff) like ASF.
Headers: header can be placed at the beginning or at the end of file.
About half of my files have it at the beginning, others have it at the end.
Broken files are only playable if they have header at the beginning!
Codecs: any codecs allowed, both CBR and VBR.
Note: most new mov files use Sorenson video and QDesign Music audio,
they are patented, closed, secret, (TM)-ed etc formats, only Apple's
quicktime player is able to playback these files (on win/mac only).
- VIVO files:
They are funny streams. They have a human-readable ascii header at
the beginning, followed by interleaved audio and video chunks.
It has no index block, has no fixed packetsize or sync bytes, and most
files even has no keyframes, so forget seeking!
Video is standard h.263 (in vivo/2.0 files it's modified, non-standard
h.263), audio is either standard g.723 or Vivo Siren codec.
Note, that microsoft licensed vivo stuff, and included in their netshow
v2.0 program, so there are VfW/ACM codecs for vivo video and audio.
- RealMedia files:
A mixture of AVI and ASF features. It has mandatory headers at the
beginning and an optional INDEX (missing in most files).
The file is constructed of variable size chunks, with small header
telling the stream ID, timestamp, flags (keyframe...) and size.
But it has some features found in ASF files:
The video is actually double-muxed, the video chunks are really
appended fragments of the video frame. RV30+ supports B frames, so
you have to parse some bits of the first fragment to get the real PTS.
The audio frames are fixed size (CBR) but using the same scrambling
(out-of-order interleaving) as in the ASF files.
Codecs: Audio is either COOK(er), SIPR(o), ATRAC3 or DNET.
The DNET is actually a byte-swapped low-bitrate Dolby AC3 variant :)
Video is RV10 (h263 variant), RV20 (rp G2), RV30 (rp v8) or RV40 (rp v9).
FPS: variable, just like in ASF.
Note, that similarity of real and asf has some background - they worked
together on the (never finished/used) ASF v2 spec for some time.
- GIF files:
The GIF format is a common format for web graphics that supports
animation. These are read through libungif or compatible library.
Variable frame delays are supported, but seeking is not supported.
Seeking will be supported once an index of gif frames can be built.
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So, I'll describe how this stuff works.
The main modules:
1. stream.c: this is the input layer, this reads the input media (file, stdin,
vcd, dvd, network etc). what it has to know: appropriate buffering by
sector, seek, skip functions, reading by bytes, or blocks with any size.
The stream_t (stream.h) structure describes the input stream, file/device.
There is a stream cache layer (cache2.c), it's a wrapper for the stream
API. It does fork(), then emulates stream driver in the parent process,
and stream user in the child process, while proxying between them using
preallocated big memory chunk for FIFO buffer.
2. demuxer.c: this does the demultiplexing (separating) of the input to
audio, video or dvdsub channels, and their reading by buffered packages.
The demuxer.c is basically a framework, which is the same for all the
input formats, and there are parsers for each of them (mpeg-es,
mpeg-ps, avi, avi-ni, asf), these are in the demux_*.c files.
The structure is the demuxer_t. There is only one demuxer.
2.a. demux_packet_t, that is DP.
Contains one chunk (avi) or packet (asf,mpg). They are stored in memory as
in linked list, cause of their different size.
2.b. demuxer stream, that is DS.
Struct: demux_stream_t
Every channel (a/v/s) has one. This contains the packets for the stream
(see 2.a). For now, there can be 3 for each demuxer :
- audio (d_audio)
- video (d_video)
- DVD subtitle (d_dvdsub)
2.c. stream header. There are 2 types (for now): sh_audio_t and sh_video_t
This contains every parameter essential for decoding, such as input/output
buffers, chosen codec, fps, etc. There are each for every stream in
the file. At least one for video, if sound is present then another,
but if there are more, then there'll be one structure for each.
These are filled according to the header (avi/asf), or demux_mpg.c
does it (mpg) if it founds a new stream. If a new stream is found,
the ====> Found audio/video stream: <id> messages is displayed.
The chosen stream header and its demuxer are connected together
(ds->sh and sh->ds) to simplify the usage. So it's enough to pass the
ds or the sh, depending on the function.
For example: we have an asf file, 6 streams inside it, 1 audio, 5
video. During the reading of the header, 6 sh structs are created, 1
audio and 5 video. When it starts reading the packet, it chooses the
stream for the first found audio & video packet, and sets the sh
pointers of d_audio and d_video according to them. So later it reads
only these streams. Of course the user can force choosing a specific
stream with
-vid and -aid switches.
A good example for this is the DVD, where the english stream is not
always the first, so every VOB has different language :)
That's when we have to use for example the -aid 128 switch.
Now, how this reading works?
- demuxer.c/demux_read_data() is called, it gets how many bytes,
and where (memory address), would we like to read, and from which
DS. The codecs call this.
- this checks if the given DS's buffer contains something, if so, it
reads from there as much as needed. If there isn't enough, it calls
ds_fill_buffer(), which:
- checks if the given DS has buffered packages (DP's), if so, it moves
the oldest to the buffer, and reads on. If the list is empty, it
calls demux_fill_buffer() :
- this calls the parser for the input format, which reads the file
onward, and moves the found packages to their buffers.
Well it we'd like an audio package, but only a bunch of video
packages are available, then sooner or later the:
DEMUXER: Too many (%d in %d bytes) audio packets in the buffer
error shows up.
2.d. video.c: this file/function handle the reading and assembling of the
video frames. each call to video_read_frame() should read and return a
single video frame, and it's duration in seconds (float).
The implementation is splitted to 2 big parts - reading from mpeg-like
streams and reading from one-frame-per-chunk files (avi, asf, mov).
Then it calculates duration, either from fixed FPS value, or from the
PTS difference between and after reading the frame.
2.e. other utility functions: there are some useful code there, like
AVI muxer, or mp3 header parser, but leave them for now.
So everything is ok 'till now. It can be found in libmpdemux/ library.
It should compile outside of mplayer tree, you just have to implement few
simple functions, like mp_msg() to print messages, etc.
See libmpdemux/test.c for example.
See also formats.txt, for description of common media file formats and their
implementation details in libmpdemux.
Now, go on:
3. mplayer.c - ooh, he's the boss :)
Its main purpose is connecting the other modules, and maintaining A/V
sync.
The given stream's actual position is in the 'timer' field of the
corresponding stream header (sh_audio / sh_video).
The structure of the playing loop :
while(not EOF) {
fill audio buffer (read & decode audio) + increase a_frame
read & decode a single video frame + increase v_frame
sleep (wait until a_frame>=v_frame)
display the frame
apply A-V PTS correction to a_frame
handle events (keys,lirc etc) -> pause,seek,...
}
When playing (a/v), it increases the variables by the duration of the
played a/v.
- with audio this is played bytes / sh_audio->o_bps
Note: i_bps = number of compressed bytes for one second of audio
o_bps = number of uncompressed bytes for one second of audio
(this is = bps*samplerate*channels)
- with video this is usually == 1.0/fps, but I have to note that
fps doesn't really matters at video, for example asf doesn't have that,
instead there is "duration" and it can change per frame.
MPEG2 has "repeat_count" which delays the frame by 1-2.5 ...
Maybe only AVI and MPEG1 has fixed fps.
So everything works right until the audio and video are in perfect
synchronity, since the audio goes, it gives the timing, and if the
time of a frame passed, the next frame is displayed.
But what if these two aren't synchronized in the input file?
PTS correction kicks in. The input demuxers read the PTS (presentation
timestamp) of the packages, and with it we can see if the streams
are synchronized. Then MPlayer can correct the a_frame, within
a given maximal bounder (see -mc option). The summary of the
corrections can be found in c_total .
Of course this is not everything, several things suck.
For example the soundcards delay, which has to be corrected by
MPlayer! The audio delay is the sum of all these:
- bytes read since the last timestamp:
t1 = d_audio->pts_bytes/sh_audio->i_bps
- if Win32/ACM then the bytes stored in audio input buffer
t2 = a_in_buffer_len/sh_audio->i_bps
- uncompressed bytes in audio out buffer
t3 = a_buffer_len/sh_audio->o_bps
- not yet played bytes stored in the soundcard's (or DMA's) buffer
t4 = get_audio_delay()/sh_audio->o_bps
From this we can calculate what PTS we need for the just played
audio, then after we compare this with the video's PTS, we have
the difference!
Life didn't get simpler with AVI. There's the "official" timing
method, the BPS-based, so the header contains how many compressed
audio bytes or chunks belong to one second of frames.
In the AVI stream header there are 2 important fields, the
dwSampleSize, and dwRate/dwScale pairs:
- If the dwSampleSize is 0, then it's VBR stream, so its bitrate
isn't constant. It means that 1 chunk stores 1 sample, and
dwRate/dwScale gives the chunks/sec value.
- If the dwSampleSize is >0, then it's constant bitrate, and the
time can be measured this way: time = (bytepos/dwSampleSize) /
(dwRate/dwScale) (so the sample's number is divided with the
samplerate). Now the audio can be handled as a stream, which can
be cut to chunks, but can be one chunk also.
The other method can be used only for interleaved files: from
the order of the chunks, a timestamp (PTS) value can be calculated.
The PTS of the video chunks are simple: chunk number * fps
The audio is the same as the previous video chunk was.
We have to pay attention to the so called "audio preload", that is,
there is a delay between the audio and video streams. This is
usually 0.5-1.0 sec, but can be totally different.
The exact value was measured until now, but now the demux_avi.c
handles it: at the audio chunk after the first video, it calculates
the A/V difference, and take this as a measure for audio preload.
3.a. audio playback:
Some words on audio playback:
Not the playing is hard, but:
1. knowing when to write into the buffer, without blocking
2. knowing how much was played of what we wrote into
The first is needed for audio decoding, and to keep the buffer
full (so the audio will never skip). And the second is needed for
correct timing, because some soundcards delay even 3-7 seconds,
which can't be forgotten about.
To solve this, the OSS gives several possibilities:
- ioctl(SNDCTL_DSP_GETODELAY): tells how many unplayed bytes are in
the soundcard's buffer -> perfect for timing, but not all drivers
support it :(
- ioctl(SNDCTL_DSP_GETOSPACE): tells how much can we write into the
soundcard's buffer, without blocking. If the driver doesn't
support GETODELAY, we can use this to know how much the delay is.
- select(): should tell if we can write into the buffer without
blocking. Unfortunately it doesn't say how much we could :((
Also, doesn't/badly works with some drivers.
Only used if none of the above works.
4. Codecs. Consists of libmpcodecs/* and separate files or libs,
for example libmpeg2, loader, mp3lib.
mplayer.c doesn't call them directly, but through the dec_audio.c and
dec_video.c files, so the mplayer.c doesn't have to know anything about
the codecs.
libmpcodecs contains wrapper for every codecs, some of them include the
codec function implementation, some calls functions from other files
included with mplayer, some calls optional external libraries.
file naming convention in libmpcodecs:
ad_*.c - audio decoder (called through dec_audio.c)
vd_*.c - video decoder (called through dec_video.c)
ve_*.c - video encoder (used by mencoder)
vf_*.c - video filter (see option -vf)
On this topic, see also:
libmpcodecs.txt - The structure of the codec-filter path, with explanation
dr-methods.txt - Direct rendering, MPI buffer management for video codecs
codecs.conf.txt - How to write/edit codec configuration file (codecs.conf)
codec-devel.txt - Mike's hints about codec development - a bit OUTDATED
hwac3.txt - about SP/DIF audio passthrough
5. libvo: this displays the frame.
for details on this, read libvo.txt
6. libao2: this control audio playing
6.a audio plugins
for details on this, read libao2.txt
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mails by A'rpi and Marcus Blomenkamp <Marcus.Blomenkamp@epost.de>
describing how this ac3-passtrough hack work under linux and mplayer...
-----------------------------------------------------------------------
Hi,
> I received the following patch from Steven Brookes <stevenjb@mda.co.uk>.
> He is working on fixing the digital audio output of the dxr3 driver and
> told me he fixed some bugs in mplayer along the way. I don't know shit
> about hwac3 output so all I did was to make sure the patch applied
> against latest cvs.
> This is from his e-mail to me:
>
> "Secondly there is a patch to dec_audio.c and
> ac3-iec958 to fix the -ac hwac3 codec stuff and to use liba52 to sync it.
> Seems to work for everything I've thrown at and maintains sync for all audio
> types through the DXR3."
patch applied (with some comments added and an unwanted change (in software
a52 decoder) removed)
now i understand how this whole hwac3 mess work.
it's very very tricky. it virtually decodes ac3 to LPCM packets, but really
it keeps the original compressed data padded by zeros. this way it's
constant bitrate, and sync is calculated just like for stereo PCM.
(so it bypass LPCM-capable media converters...)
so, every ac3 frame is translated to 6144 byte long tricky LPCM packet.
6144 = 4*(6*256) = 4 * samples_per_ac3_frame = LPCM size of uncompressed ac3
frame.
i wanna know if it works for sblive and other ac3-capable cards too?
(i can't test it, lack of ac3 decoder)
A'rpi / Astral & ESP-team
-----------------------------------------------------------------------
Hi folks.
I spend some time fiddling with ac3 passthrough in mplayer. The
traditional way of setting the output format to AFMT_AC3 was no ideal
solution since not all digital io cards/drivers supported this format or
honoured it to set the spdif non-audio bit. To make it short, it only
worked with oss sblive driver IIRC.
Inspired by alsa's ac3dec program I found an alternative way by
inspecting to which format the alsa device had been set. Suprise: it was
simple 16bit_le 2_channel pcm. So setting the non-audio bit doesn't
necessarily mean the point. The only important thing seems to be
bit-identical output at the correct samplerate. Modern AV-Receivers seem
to be quite tolerant/compatible.
So I changed the output format of hwac3 from
AFMT_AC3 channels=1
to
AFMT_S16_LE channels=2
and corrected the absolute time calculation. That was all to get it
running for me.
-----------------------------------------------------------------------
Hi there.
Perhaps I can clear up some mystification about AC3 passthrough in
general and mplayer in special:
To get the external decoder solution working, it must be fed with data
which is bitidentical to the chunks in the source ac3 file (compressed
data is very picky about bit errors). Additionally - or better to say
'historically' - the non-audio bit should be set in the spdif status
fields to prevent old spdif hardware from reproducing ugly scratchy
noise. Note: for current decoders (probably those with DTS capability)
this safety bit isn't needed anymore. At least I can state that for my
Sherwood RVD-6095RDS. I think it is due to DTS because DTS sound can
reside on a ordinary AudioCD and an ordinary AudioCD-Player will always
have it's audio-bit set.
The sample format of the data must be 2channel 16bit (little endian
IIRC). Samplerates are 48kHz - although my receiver also accepts
44100Hz. I do not know if this is due to an over-compatability of my
receiver or if 44100 is also possible in the ac3 specs. For safety's
sake lets keep this at 48000Hz. AC3 data chunks are inserted into the
stream every 0x1600 bytes (don't bite me on that, look into
'ac3-iec958.c': 'ac3_iec958_build_burst').
To come back to the problem: data must be played bit-identically through
the soundcard at the correct samplerate and should optionally have it's
non-audio bit set. There are two ways to accomplish this:
1) Some OSS guy invented the format AFMT_AC3. Soundcard drivers
implementing this format should therefore adjust it's mixers and
switches to produce the desired output. Unfortunately some soundcard
drivers do not support this format correctly and most do not even
support it at all (including ALSA).
2) The alternative approach currently in mplayer CVS is to simply set
the output format to 48kHz16bitLE and rely on the user to have the
soundcard mixers adjusted properly.
I do have two soundcards with digital IO facilities (CMI8738 and
Trident4DWaveNX based) plus the mentioned decoder. I'm currently running
Linux-2.4.17. Following configurations are happily running here:
1. Trident with ALSA drivers (OSS does not support Hoontech's dig. IO)
2. CMI with ALSA drivers
3. CMI with OSS drivers
For Linux I'd suggest using ALSA because of it's cleaner architecture
and more consitent user interface. Not to mention that it'll be the
standard sound support in Linux soon.
For those who want to stick to OSS drivers: The CMI8738 drivers works
out-of-the-box, if the PCM/Wave mixer is set to 100%.
For ALSA I'd suggest using its OSS emulation. More on that later.
ALSA-0.9 invented the idea of cards, devices and dubdevices. You can
reach the digital interface of all supported cards consitently by using
the device 'hw:x,2' (x counting from 0 is the number of your soundcard).
So most people would end up at 'hw:0,2'. This device can only be opened
in sample formats and rates which are directly supported in hardware
hence no samplerate conversion is done keeping the stream as-is. However
most consumer soundcards do not support 44kHz so it would definitively
be a bad idea to use this as your standard device if you wanted to
listen to some mp3s (most of them are 44kHz due to CD source). Here the
OSS comes to play again. You can configure which OSS device (/dev/dsp
and /dev/adsp) uses which ALSA device. So I'd suggest pointing the
standard '/dev/dsp' to standard 'hw:0,0' which suports mixing and
samplerate conversion. No further reconfiguration would be needed for
your sound apps. For movies I'd point '/dev/adsp' to 'hw:0,2' and
configure mplayer to use adsp instead of dsp. The samplerate constrain
is no big deal here since movies usually are in 48Khz anyway. The
configuration in '/etc/modules.conf' is no big deal also:
alias snd-card-0 snd-card-cmipci # insert your card here
alias snd-card-1 snd-pcm-oss # load OSS emulation
options snd-pcm-oss snd_dsp_map=0 snd_adsp_map=2 # do the mapping
This works flawlessly in combination with alsa's native
SysVrc-init-script 'alsasound'. Be sure to disable any distribution
dependent script (e.g. Mandrake-8.1 has an 'alsa' script which depends
on ALSA-0.5).
Sorry for you *BSD'lers out there. I have no grasp on sound support there.
HTH Marcus
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6. libao2: this control audio playing
As in libvo (see 5.) also here are some drivers, based on the same API:
static int control(int cmd, int arg);
This is for reading/setting driver-specific and other special parameters.
Not really used for now.
static int init(int rate,int channels,int format,int flags);
The init of driver, opens device, sets sample rate, channels, sample format
parameters.
Sample format: usually AFMT_S16_LE or AFMT_U8, for more definitions see
dec_audio.c and linux/soundcards.h files!
static void uninit(void);
Guess what.
Ok I help: closes the device, not (yet) called when exit.
static void reset(void);
Resets device. To be exact, it's for deleting buffers' contents,
so after reset() the previously received stuff won't be output.
(called if pause or seek)
static int get_space(void);
Returns how many bytes can be written into the audio buffer without
blocking (making caller process wait). MPlayer occasionally checks the
remaining space and tries to fill the buffer with play() if there's free
space. The buffer size used should be sane; a buffer that is too small
could run empty before MPlayer tries filling it again (normally once per
video frame), a buffer that is too big would force MPlayer decode the file
far ahead trying to find enough audio data to fill it.
static int play(void* data,int len,int flags);
Plays a bit of audio, which is received throught the "data" memory area, with
a size of "len". It has to copy the data, because they can be overwritten
after the call is made. Doesn't have to use all the bytes; it has to
return the number of bytes used used (copied to buffer). If
flags|AOPLAY_FINAL_CHUNK is true then this is the last audio in the file.
The purpose of this flag is to tell aos that round down the audio played
from "len" to a multiple of some chunksize that this "len" should not be
rounded down to 0 or the data will never be played (as MPlayer will never
call play() with a larger len).
static float get_delay(void);
Returns how long time it will take to play the data currently in the
output buffer. Be exact, if possible, since the whole timing depends
on this! In the worst case, return the maximum delay.
!!! Because the video is synchronized to the audio (card), it's very important
!!! that the get_delay function is correctly implemented!
static void audio_pause(void);
Pause playing but do not delete buffered data if possible.
static void audio_resume(void);
Continue playing after audio_pause().
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libvo --- the library to handle video output by A'rpi, 2002.04
============================================
Note: before start on this, read colorspaces.txt !
The constants for different pixelformats are defined in img_format.h,
their usage is mandatory.
WARNING: Please keep in mind that some of this information may be out-dated,
so if you are working on a new vo, consider submitting preliminary patches
very early on. Currently vo_gl is one of the more up-to-date VOs to use
as reference if you are unsure about something and do not want to ask on the
list.
vo_vdpau and vo_direct3d may be a good choice, too, they use different
approaches closer to the sometimes convoluted way DirectX works.
Each vo driver _has_ to implement these:
preinit():
init the video system (to support querying for supported formats)
uninit():
Uninit the whole system, this is on the same "level" as preinit.
control():
Current controls (VOCTRL_QUERY_FORMAT must be implemented,
VOCTRL_DRAW_IMAGE, VOCTRL_FULLSCREEN, VOCTRL_UPDATE_SCREENINFO
should be implemented):
VOCTRL_QUERY_FORMAT - queries if a given pixelformat is supported.
It also returns various flags decsirbing the capabilities
of the driver with teh given mode. for the flags, see
file vfcaps.h !
the most important flags, every driver must properly report
these:
0x1 - supported (with or without conversion)
0x2 - supported without conversion (define 0x1 too!)
0x100 - driver/hardware handles timing (blocking)
also SET sw/hw scaling and osd support flags, and flip,
and accept_stride if you implement VOCTRL_DRAW_IMAGE (see bellow)
NOTE: VOCTRL_QUERY_FORMAT may be called _before_ first config()
but is always called between preinit() and uninit()
VOCTRL_GET_IMAGE
libmpcodecs Direct Rendering interface
You need to update mpi (mp_image.h) structure, for example,
look at vo_x11, vo_sdl, vo_xv or mga_common.
VOCTRL_DRAW_IMAGE
replacement for the current draw_slice/draw_frame way of
passing video frames. by implementing SET_IMAGE, you'll get
image in mp_image struct instead of by calling draw_*.
unless you return VO_TRUE for VOCTRL_DRAW_IMAGE call, the
old-style draw_* functils will be called!
Note: draw_slice is still mandatory, for per-slice rendering!
VOCTRL_RESET - reset the video device
This is sent on seeking and similar and is useful if you are
using a device which prebuffers frames that need to flush them
before refilling audio/video buffers.
VOCTRL_PAUSE
VOCTRL_RESUME
VOCTRL_GUISUPPORT
return true only if driver supports co-operation with
MPlayer's GUI (not yet used by GUI)
VOCTRL_SET_EQUALIZER
set the video equalizer to the given values
two arguments are provided: item and value
item is a string, the possible values are (currently):
brightness, contrast, saturation, hue
VOCTRL_GET_EQUALIZER
get the current video equalizer values
two arguments are provided: item and value
item is a string, the possible values are (currently):
brightness, contrast, saturation, hue
VOCTRL_ONTOP
Makes the player window stay-on-top. Only supported (currently)
by drivers which use X11, except SDL, as well as directx and
gl2 under Windows.
VOCTRL_BORDER
Makes the player window borderless.
VOCTRL_FULLSCREEN
Switch from and to fullscreen mode
VOCTRL_GET_PANSCAN
VOCTRL_SET_PANSCAN
Needed to implement pan-scan support ('w' and 'e' keys during
playback in fullscreen mode)
VOCTRL_START_SLICE
Called before the first draw_slice of each frame, useful if
you need to do some set-up work.
VOCTRL_DRAW_EOSD
Required for EOSD (ASS subtitle) support. Provides all
information necessary to draw the EOSD for the current video
frame.
VOCTRL_GET_EOSD_RES
Required for EOSD (ASS subtitle) support. Informs the ASS
renderer about the properties of the drawing area (size,
borders).
VOCTRL_SET_DEINTERLACE
VOCTRL_GET_DEINTERLACE
Get or set deinterlacing status for VOs that support some kind
of deinterlacing.
VOCTRL_UPDATE_SCREENINFO
Should set the xinerama_x, xinerama_y, vo_screenwidth and
vo_screenheight appropriately for the currently used
monitor and -xineramascreen option.
Usually should simply call the w32_update_xinerama_info or
update_xinerama_info function.
By supporting this, the VO also requests the newer API
that sets vo_dx, vo_dy etc. appropriately before config()
is called.
config():
Set up the video system. You get the dimensions and flags.
width, height: size of the source image
d_width, d_height: wanted scaled/display size (it's a hint)
Flags:
0x01 - force fullscreen (-fs)
0x02 - allow mode switching (-vm)
0x04 - allow software scaling (-zoom)
0x08 - flipping (-flip)
They're defined as VOFLAG_* (see libvo/video_out.h)
IMPORTANT NOTE: config() may be called 0 (zero), 1 or more (2,3...)
times between preinit() and uninit() calls. You MUST handle it, and
you shouldn't crash at second config() call or at uninit() without
any config() call! To make your life easier, vo_config_count is
set to the number of previous config() call, counted from preinit().
It's set by the caller (vf_vo.c), you don't have to increase it!
So, you can check for vo_config_count>0 in uninit() when freeing
resources allocated in config() to avoid crash!
You should implement VOCTRL_UPDATE_SCREENINFO so that vo_dx, vo_dy,
vo_dwidth and vo_dheight are already pre-set to values that take
aspect and -geometry into account. It is also necessary to properly
support multi-monitor setups (if based on x11_common, w32_common).
draw_slice(): this displays YV12 pictures (3 planes, one full sized that
contains brightness (Y), and 2 quarter-sized which the colour-info
(U,V). MPEG codecs (libmpeg2, opendivx) use this. This doesn't have
to display the whole frame, only update small parts of it.
If this is not supported, it must be signaled in QUERY_FORMAT with
VOCAP_NOSLICES.
draw_frame(): this is the older interface, this displays only complete
frames, and can do only packed format (YUY2, RGB/BGR).
Win32 codecs use this (DivX, Indeo, etc).
If you implement VOCTRL_DRAW_IMAGE, you do not need to implement draw_frame.
draw_osd(): this displays subtitles and OSD.
It's a bit tricky to use it, since it's a callback-style stuff.
It should call vo_draw_text() with screen dimension and your
draw_alpha implementation for the pixelformat (function pointer).
The vo_draw_text() checks the characters to draw, and calls
draw_alpha() for each. As a help, osd.c contains draw_alpha for
each pixelformats, use this if possible!
Note that if you do not draw directly onto the video you should
use vo_draw_text_ext() which allows you to specify the border
values etc. needed to draw DVD menu highlights at the correct place.
If you do not want to implement this, you can still use -vf
expand=osd=1 to draw the OSD, or even implement code to insert
this filter automatically.
Make sure you set VFCAP_OSD depending on whether you implemented it
or not.
NOTE: This one will be obsolete soon! But it's still useful when
you want to do tricks, like rendering osd _after_ hardware scaling
(tdfxfb) or render subtitles under of the image (vo_mpegpes, sdl)
NOTE2: above NOTE is probably wrong, there are currently no plans to
obsolete draw_osd, though there is the more advanced EOSD support for
ASS subtitles.
flip_page(): this is called after each frame, this displays the buffer for
real. This is 'swapbuffers' when doublebuffering.
Try to do as little work here as possible, since that affect jitter/
A-V sync.
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TITLE=Weird Al Yankovic - The Saga Begins (clip)
# just to be informative (maybe future
# use?)
FILE=14233792,7972c637f619e01854ecb453b7339e3f # size,md5
AUTHOR=Gabucino
TYPE=VIDEO # maybe AUDIO in future (for displaying
# mp3 lyrics while playing or whatever)
FORMAT=TIME # this file contains measures based on
# TIME, not frames.
# if FORMAT is a number, then it's frame based, i.e.:
# FORMAT=25
NOTE=MPlayerSub test file v0.1
# first number : wait this much after previous subtitle disappeared
# second number : display the current subtitle for this many seconds
15 3
A long, long time ago...
0 3
in a galaxy far away...
0 3
Naboo was under an attack.
1 2.5
And I thought me and
Qui-Gon Jinn could
0 2.5
talk the Federation into
0 4
...maybe cutting them a
little slack.
2 3
But their response, it
didn't thrill us,
0 3
They locked the doors,
and tried to kill us.
0 2.5
We escaped from that gas,
0 3.5
then met Jar-jar and
Boss-Nass.
1 6
We took a bongo from the
scene and we went to
Theed to see the Queen.
0 5
We all wound' up on
Tatooine.
0 6
That's where, we've found
this boy.
0 4
Oh my, my this here
Anakin guy,
0 5
maybe Vader someday
later now he's just
a small fry.
0 4
And he left his home and
kissed his mommy goodbye,
0 5
singing "Soon I'm gonna be
a Jedi!"
6 6
Did you know this junkyard
slave isn't even old enough
to shave,
0 3
but he can use the Force,
they say.
1 6
Ahh, do you see him hitting
on the queen though he's
just nine and she's fourteen
0 6
yeah, he's probably gonna
marry her, someday!
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software scaler filter docs
-ssf cgb=<float> chrominance gaussian blur
useful for videos with some chroma (color) noise (TV captures, VHS, ...)
-ssf lgb=<float> -> luminance gaussian blur
hmm, mostly special effect, could be used as lame noise reducer though
-ssf chs=<int> -> shift chrominance horizontally
-ssf cvs=<int> -> shift chrominance vertically
useful if the luma & chroma are not aligned exactly
-ssf cs=<float> -> chrominance sharpen
-ssf ls=<float> -> luminance sharpen
simple sharpening filter
note: these will only work if the swscaler is used at all (use -vf scale)
example: mplayer foobar.avi -vf scale -ssf lgb=3.0
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artifact types:
---------------------------------------------------------------------
ringing
wave or noise like patterns around sharp edges
bad: sinc, lanczos (with high filter length)
acceptable: lanczos (with low filter length), cubic, spline
ok: area, (fast)blinear, gauss, point
blur
loss of detail / high frequency
bad: gauss with high variance
acceptable: (fast)bilinear, area
ok: others
aliasing (only downscale)
straight lines look like staircases
areas of high detail look worse
regular patterns turn into a funny looking mess (moire patterns)
bad: fast bilinear, point, gauss with low variance
acceptable: area
ok: others
blocky (upscale only)
looks like the image is made of rectangular blocks like a mosaic
bad: point
ok: others
recommendations: (based upon my personal opinion many ppl might disagree ...)
---------------------------------------------------------------------
the recommended scalers for upscaling:
fast_bilinear, point if speed is important
cubic, spline, lanczos if quality is important
the recommended scalers for downscaling:
fast_bilinear, point if speed is important
gauss, bilinear if quality is important
cubic, spline, lanczos if a sharper picture is important
note: when encoding at a limited amount of bits (not constant quantizer)
then a slightly blurred input might look better after decoding than a slightly
sharpened one, especially for lower bitrates
sws numbers / names / artifact types
-------------------------------------------------------------------------
r ringing
b blurry
a aliasing (downscale only)
m mosaic (blocky) (upscale only)
-sws 0 fast bilinear bA
-sws 1 bilinear b
-sws 2 bicubic high sharpness r
low sharpness b
-sws 3 experimental ????
-sws 4 nearest neighbour AM
-sws 5 area ba
-sws 6 luma bicubic / chroma bilinear r
-sws 7 gauss low sharpness B
high sharpness AM
-sws 8 sinc R
-sws 9 lanczos long filter R
short filter b
-sws 10 bicubic spline r
Notes:
area upscale is identical to bilinear
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Introduction
============
This file lists all changes that can cause compatibility issues when using
mpv through the client API (libmpv and ``client.h``). Since the client API
interfaces to input handling (commands, properties) as well as command line
options, this list is interesting for other uses of mpv, such as the Lua
scripting interface, key bindings in ``input.rst``, or plain command line
usage.
Normally, changes to the C API that are incompatible to previous iterations
receive a major version bump (i.e. the first version number is increased),
while C API additions bump the minor version (i.e. the second number is
increased). Changes to properties/commands/options may also lead to a minor
version bump, in particular if they are incompatible.
The version number is the same as used for MPV_CLIENT_API_VERSION (see
``client.h`` how to convert between major/minor version numbers and the flat
32 bit integer).
Also, read the section ``Compatibility`` in ``client.h``.
API changes
===========
::
--- mpv 0.5.2 is released ---
1.4 - subtle change in X11 and "--wid" behavior
--- mpv 0.5.0 is released ---
1.3 - add MPV_MAKE_VERSION()
1.2 - remove "stream-time-pos" property (no replacement)
1.1 - remap dvdnav:// to dvd://
- add "--cache-file", "--cache-file-size"
- add "--colormatrix-primaries" (and property)
- add "primaries" sub-field to image format properties
- add "playback-time" property
- extend the "--start" option; a leading "+", which was previously
insignificant is now significant
- add "cache-free" and "cache-used" properties
- OSX: the "coreaudio" AO spdif code is split into a separate AO
--- mpv 0.4.0 is released ---
1.0 - the API is declared stable
-136
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// License: pick one of: public domain, WTFPL, ISC, Ms-PL, AGPLv3
// This example can be built with: qmake && make
#include <sstream>
#include <QFileDialog>
#include <QStatusBar>
#include <QMenuBar>
#include <QMenu>
#include <QGridLayout>
#include <QApplication>
#include "qtexample.h"
static void wakeup(void *ctx)
{
// This callback is invoked from any mpv thread (but possibly also
// recursively from a thread that is calling the mpv API). Just notify
// the Qt GUI thread to wake up (so that it can process events with
// mpv_wait_event()), and return as quickly as possible.
MainWindow *mainwindow = (MainWindow *)ctx;
QCoreApplication::postEvent(mainwindow, new QEvent(QEvent::User));
}
MainWindow::MainWindow(QWidget *parent) :
QMainWindow(parent)
{
QMenu *menu = menuBar()->addMenu(tr("&File"));
QAction *on_open = new QAction(tr("&Open"), this);
on_open->setShortcuts(QKeySequence::Open);
on_open->setStatusTip(tr("Open a file"));
connect(on_open, SIGNAL(triggered()), this, SLOT(on_file_open()));
menu->addAction(on_open);
statusBar();
mpv = mpv_create();
if (!mpv)
throw "can't create mpv instance";
// Create a video child window. Force Qt to create a native window, and
// pass the window ID to the mpv wid option. This doesn't work on OSX,
// because Cocoa doesn't support this form of embedding.
mpv_container = new QWidget(this);
setCentralWidget(mpv_container);
mpv_container->setAttribute(Qt::WA_NativeWindow);
int64_t wid = mpv_container->winId();
mpv_set_option(mpv, "wid", MPV_FORMAT_INT64, &wid);
// Enable default bindings, because we're lazy. Normally, a player using
// mpv as backend would implement its own key bindings.
mpv_set_option_string(mpv, "input-default-bindings", "yes");
// Let us receive property change events with MPV_EVENT_PROPERTY_CHANGE if
// this property changes.
mpv_observe_property(mpv, 0, "time-pos", MPV_FORMAT_DOUBLE);
// From this point on, the wakeup function will be called. The callback
// can come from any thread, so we use the Qt QEvent mechanism to relay
// the wakeup in a thread-safe way.
mpv_set_wakeup_callback(mpv, wakeup, this);
if (mpv_initialize(mpv) < 0)
throw "mpv failed to initialize";
}
void MainWindow::handle_mpv_event(mpv_event *event)
{
switch (event->event_id) {
case MPV_EVENT_PROPERTY_CHANGE: {
mpv_event_property *prop = (mpv_event_property *)event->data;
if (strcmp(prop->name, "time-pos") == 0) {
if (prop->format == MPV_FORMAT_DOUBLE) {
double time = *(double *)prop->data;
std::stringstream ss;
ss << "At: " << time;
statusBar()->showMessage(QString::fromStdString(ss.str()));
} else if (prop->format == MPV_FORMAT_NONE) {
// The property is unavailable, which probably means playback
// was stopped.
statusBar()->showMessage("");
}
}
break;
}
case MPV_EVENT_SHUTDOWN: {
mpv_terminate_destroy(mpv);
mpv = NULL;
break;
}
default: ;
// Ignore uninteresting or unknown events.
}
}
bool MainWindow::event(QEvent *event)
{
// QEvent::User is sent by wakeup().
if (event->type() == QEvent::User) {
// Process all events, until the event queue is empty.
while (mpv) {
mpv_event *event = mpv_wait_event(mpv, 0);
if (event->event_id == MPV_EVENT_NONE)
break;
handle_mpv_event(event);
}
return true;
}
return QMainWindow::event(event);
}
void MainWindow::on_file_open()
{
QString filename = QFileDialog::getOpenFileName(this, "Open file");
if (mpv) {
const QByteArray c_filename = filename.toUtf8();
const char *args[] = {"loadfile", c_filename.data(), NULL};
mpv_command_async(mpv, 0, args);
}
}
MainWindow::~MainWindow()
{
if (mpv)
mpv_terminate_destroy(mpv);
}
int main(int argc, char *argv[])
{
QApplication a(argc, argv);
MainWindow w;
w.show();
return a.exec();
}
-30
View File
@@ -1,30 +0,0 @@
#ifndef QTEXAMPLE_H
#define QTEXAMPLE_H
#include <QMainWindow>
#include <mpv/client.h>
class MainWindow : public QMainWindow
{
Q_OBJECT
public:
explicit MainWindow(QWidget *parent = 0);
~MainWindow();
protected:
virtual bool event(QEvent *event);
private slots:
void on_file_open();
private:
QWidget *mpv_container;
mpv_handle *mpv;
void create_player();
void handle_mpv_event(mpv_event *event);
};
#endif // QTEXAMPLE_H
-12
View File
@@ -1,12 +0,0 @@
QT += core gui
greaterThan(QT_MAJOR_VERSION, 4): QT += widgets
TARGET = qtexample
TEMPLATE = app
CONFIG += link_pkgconfig
PKGCONFIG = mpv
SOURCES += qtexample.cpp
HEADERS += qtexample.h
-10
View File
@@ -1,10 +0,0 @@
#include <stdio.h>
#include <stdlib.h>
static inline void check_error(int status)
{
if (status < 0) {
printf("mpv API error: %s\n", mpv_error_string(status));
exit(1);
}
}
-44
View File
@@ -1,44 +0,0 @@
#include <stddef.h>
#include <stdio.h>
#include <stdlib.h>
#include "libmpv/client.h"
#include "shared.h"
int main(int argc, char *argv[])
{
if (argc != 2) {
printf("pass a single media file as argument\n");
return 1;
}
mpv_handle *ctx = mpv_create();
if (!ctx) {
printf("failed creating context\n");
return 1;
}
// Enable default key bindings, so the user can actually interact with
// the player (and e.g. close the window).
check_error(mpv_set_option_string(ctx, "input-default-bindings", "yes"));
int val = 1;
check_error(mpv_set_option(ctx, "osc", MPV_FORMAT_FLAG, &val));
// Done setting up options.
check_error(mpv_initialize(ctx));
// Play this file.
const char *cmd[] = {"loadfile", argv[1], NULL};
check_error(mpv_command(ctx, cmd));
// Let it play, and wait until the user quits.
while (1) {
mpv_event *event = mpv_wait_event(ctx, 10000);
printf("event: %s\n", mpv_event_name(event->event_id));
if (event->event_id == MPV_EVENT_SHUTDOWN)
break;
}
mpv_terminate_destroy(ctx);
return 0;
}
-147
View File
@@ -1,147 +0,0 @@
How to contribute
=================
General
-------
The main contact for mpv development is IRC, specifically #mpv-player and
#mpv-player-dev on Freenode.
Sending patches
---------------
- Make a github pull request, or send a link to a plaintext patch created with
``git format-patch``. diffs posted as pastebins (especially if the http link
returns HTML) just cause extra work for everyone, because they lack commit
message and authorship information.
- Write informative commit messages. Use present tense to describe the
situation with the patch applied, and past tense for the situation before
the change.
- The subject line (the first line in a commit message) should contain a
prefix identifying the sub system, followed by a short description what
impact this commit has. This subject line shouldn't be longer than 72
characters, because it messes up the output of many git tools otherwise.
For example, you fixed a crash in af_volume.c:
Bad: ``fixed the bug (wtf?)``
Good: ``af_volume: fix crash due to null pointer access``
Having a prefix gives context, and is especially useful when trying to find
a specific change by looking at the history, or when running ``git blame``.
- The body of the commit message (everything else after the subject line) should
be as informative as possible and contain everything that isn't obvious. Don't
hesitate to dump as much information as you can - it doesn't cost you
anything. Put some effort into it. If someone finds a bug months or years
later, and finds that it's caused by your commit (even though your commit was
supposed to fix another bug), it would be bad if there wasn't enough
information to test the original bug. The old bug might be reintroduced while
fixing the new bug.
The commit message should be wrapped on 72 characters per line, because git
tools usually do not break text automatically. On the other hand, you do not
need to break text that would be unnatural to break (like data for test cases,
or long URLs).
Important: put an empty line between the subject line and the commit message.
If this is missing, it will break display in common git tools.
- Try to separate cosmetic and functional changes. It's ok to make a few
additional cosmetic changes in the same file you're working on. But don't do
something like reformatting a whole file, and hiding an actual functional
change in the same commit.
- If you add a new command line option, document it in options.rst. If you
add a new input property, document it in input.rst.
Code formatting
---------------
mpv uses C99 with K&R formatting, with some exceptions.
- Use the K&R indent style.
- Use 4 spaces of indentation, never use tabs (except in Makefiles).
- Add a single space between keywords and binary operators. There are some other
cases where spaces should be added. Example:
```C
if ((a * b) > c) {
// code
some_function(a, b, c);
}
```
- Break lines on 80 columns. There is a hard limit of 85 columns. You may ignore
this limit if there's a strong case that not breaking the line will increase
readability. Going over 85 columns might provoke endless discussions about
whether such a limit is needed or not, so avoid it.
- If the body of an if/for/while statement has more than 1 physical lines, then
always add braces, even if they're technically redundant.
Bad:
```C
if (a)
// do something if b
if (b)
do_something();
```
Good:
```C
if (a) {
// do something if b
if (b)
do_something();
}
```
- If the body of an if statement uses braces, the else branch should also
use braces (and reverse).
Example:
```C
if (a) {
// do something
something();
something_else();
} else {
one_line();
}
```
- If an if condition spans multiple physical lines, then put the opening brace
for the if body on the next physical line. (Also, preferably always add a
brace, even if technically none is needed.)
Example:
```C
if (very_long_condition_a &&
very_long_condition_b)
{
code();
}
```
- Remove any trailing whitespace.
- If the file you're editing uses formatting different from from what is
described here, it's probably an old file from times when nobody followed a
consistent style. You're free to use the existing style, or the new style, or
to send a patch to reformat the file to the new style before making functional
changes.
General coding
--------------
- Use C99. Also freely make use of C99 features if it's appropriate, such as
stdbool.h.
- Don't use GNU-only features. In some cases they may be warranted, if they
are optional (such as attributes enabling printf-like format string checks).
But in general, standard C99 should be used.
- The same applies to libc functions. We have to be Windows-compatible too. Use
functions guaranteed by C99 or POSIX only, unless your use is guarded by a
configure check.
- Prefer fusing declaration and initialization, rather than putting declarations
on the top of a block. Obvious data flow is more important than avoiding
mixing declarations and statements, which is just a C90 artifact.
- tech-overview.txt might help to get an overview how mpv is structured.
- If you add features that require intrusive changes, discuss them on the dev
channel first. There might be a better way to add a feature and it can avoid
wasted work.
-74
View File
@@ -1,74 +0,0 @@
Cross Compiling to Windows
==========================
Cross compiling mpv to Windows is supported with MinGW-w64. This can be used to
produce both 32 bit and 64 bit executables. MinGW-w64 is available from
http://mingw-w64.sourceforge.net.
You have to run mpv's configure with these arguments:
```bash
DEST_OS=win32 TARGET=i686-w64-mingw32 ./waf configure
```
While building a complete MinGW-w64 toolchain yourself is possible, people have
created scripts to help ease the process. These are the two recommended methods:
- Using [mingw-w64-cmake](https://github.com/lachs0r/mingw-w64-cmake) to setup
a MinGW-w64 environment. We recommend you to try this first before MXE.
mingw-w64-cmake will also build mpv and its dependencies.
- Alternatively, you can use MXE: http://mxe.cc. With MXE, you have to manually
set the target to MinGW-w64 (even if you compile to 32 bit). A working example
below.
**Warning**: the original MinGW (http://www.mingw.org) is unsupported.
Note that MinGW environments included in Linux distributions are often broken,
outdated and useless, and usually don't use MinGW-w64.
Example with MXE
----------------
```bash
# Download MXE. Note that compiling the required packages requires about 1.4 GB
# or more!
cd /opt
git clone https://github.com/mxe/mxe mingw
cd mingw
# Set build options.
# The JOBS environment variable controls threads to use when building. DO NOT
# use the regular `make -j4` option with MXE as it will slow down the build.
# Alternatively, you can set this in the make command by appending "JOBS=4"
# to the end of command:
echo "JOBS := 4" >> settings.mk
# The MXE_TARGET environment variable builds MinGW-w64 for 32 bit targets.
# Alternatively, you can specify this in the make command by appending
# "MXE_TARGETS=i686-w64-mingw32" to the end of command:
echo "MXE_TARGETS := i686-w64-mingw32" >> settings.mk
# If you want to build 64 bit version, use this:
# echo "MXE_TARGETS := x86_64-w64-mingw32" >> settings.mk
# Build required packages. The following provide a minimum required to build
# mpv.
make gcc ffmpeg libass jpeg pthreads
# Add MXE binaries to $PATH
export PATH=/opt/mingw/usr/bin/:$PATH
# Build mpv. The target will be used to automatically select the name of the
# build tools involved (e.g. it will use i686-w64-mingw32-gcc).
cd ..
git clone https://github.com/mpv-player/mpv.git
cd mpv
DEST_OS=win32 TARGET=i686-w64-mingw32 ./waf configure
# Or, if 64 bit version,
# DEST_OS=win32 TARGET=x86_64-w64-mingw32 ./waf configure
./waf build
```
+60
View File
@@ -0,0 +1,60 @@
Cross compiling mpv to Windows is supported with MinGW-w64. This can be used to
produce both 32 bit and 64 bit executables. MinGW-w64 is available from
http://mingw-w64.sourceforge.net.
You have to run mpv's configure with these arguments:
./configure --enable-cross-compile --target=i686-w64-mingw32
Using MXE to setup a MinGW-w64 environment is recommended: http://mxe.cc
Currently, using an experimental branch of MXE is required:
https://github.com/tonytheodore/mxe.git
Warning: the original MinGW (http://www.mingw.org) is unsupported.
Note that MinGW environments included in Linux distributions are often broken,
outdated and useless, and usually don't use MinGW-w64.
Example with MXE
================
# Download MXE. Note that compiling the required packages requires about 1 GB
# or more! We are using an experimental repository, which supports MinGW-w64.
# The official MXE supports classic MinGW only, which doesn't work with mpv.
cd /opt
git clone https://github.com/tonytheodore/mxe.git mingw
cd mingw
# Edit the MXE target, so that MinGW-w64 for 32 bit targets is built.
echo "MXE_TARGETS := i686-w64-mingw32" > settings.mk
# Build required packages. The following provide a minimum required to build
# mpv. (Not all of the following packages are strictly required.)
make gcc
make ffmpeg
make libass
make jpeg
make pthreads
# Older MXE and MinGW-w64 snapshots don't provide up-to-date OpenGL headers.
wget http://www.opengl.org/registry/api/glext.h
mv glext.h usr/i686-w64-mingw32/include/GL/
# Build mpv. The target will be used to automatically select the name of the
# build tools involved (e.g. it will use i686-w64-mingw32-gcc).
git clone https://github.com/mpv-player/mpv.git
cd mpv
export PATH=/opt/mingw/usr/bin/:$PATH
./configure --enable-cross-compile --target=i686-w64-mingw32
make
# This should work. For some reason, the resulting .exe will depend on the DLL
# /opt/mingw/usr/i686-w64-mingw32/bin/libwinpthread-1.dll because pthreads-w32
# folks don't like static linking. How to enable static linking is left as
# exercise to the reader. Passsing --disable-pthreads to mpv's configure will
# get rid of the dependency, but may disable some features (such as the stream
# cache).
-121
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@@ -1,121 +0,0 @@
EDL files
=========
EDL files basically concatenate ranges of video/audio from multiple source
files into a single continuous virtual file. Each such range is called a
segment, and consists of source file, source offset, and segment length.
For example::
# mpv EDL v0
f1.mkv,10,20
f2.mkv
f1.mkv,40,10
This would skip the first 10 seconds of the file f1.mkv, then play the next
20 seconds, then switch to the file f2.mkv and play all of it, then switch
back to f1.mkv, skip to the 40 second mark, and play 10 seconds, and then
stop playback. The difference to specifying the files directly on command
line (and using ``--{ --start=10 --length=20 f1.mkv --}`` etc.) is that the
virtual EDL file appears as a virtual timeline (like a single file), instead
as a playlist.
The general simplified syntax is::
# mpv EDL v0
<filename>
<filename>,<start in seconds>,<length in seconds>
If the start time is omitted, 0 is used. If the length is omitted, the
estimated duration of the source file is used.
Note::
mpv can't use ordered chapter files or libquvi-resolved URLs in EDL
entries. Usage of relative or absolute paths as well as any protocol
prefixes is prevented for security reasons.
Syntax of mpv EDL files
=======================
Generally, the format is relatively strict. No superfluous whitespace (except
empty lines and commented lines) are allowed. You must use UNIX line breaks.
The first line in the file must be ``# mpv EDL v0``. This designates that the
file uses format version 0, which is not frozen yet and may change any time.
(If you need a stable EDL file format, make a feature request. Likewise, if
you have suggestions for improvements, it's not too late yet.)
The rest of the lines belong to one of these classes:
1) An empty or commented line. A comment starts with ``#``, which must be the
first character in the line. The rest of the line (up until the next line
break) is ignored. An empty line has 0 bytes between two line feed bytes.
2) A segment entry in all other cases.
Each segment entry consists of a list of named or unnamed parameters.
Parameters are separated with ``,``. Named parameters consist of a name,
followed by ``=``, followed by the value. Unnamed parameters have only a
value, and the name is implicit from the parameter position.
Syntax::
segment_entry ::= <param> ( <param> ',' )*
param ::= [ <name> '=' ] ( <value> | '%' <number> '%' <valuebytes> )
The ``name`` string can consist of any characters, except ``=%,;\n``. The
``value`` string can consist of any characters except of ``,;\n``.
The construct starting with ``%`` allows defining any value with arbitrary
contents inline, where ``number`` is an integer giving the number of bytes in
``valuebytes``. If a parameter value contains disallowed characters, it has to
be guarded by a length specifier using this syntax.
The parameter name defines the meaning of the parameter:
1) ``file``, the source file to use for this segment.
2) ``start``, a time value that specifies the start offset into the source file.
3) ``length``, a time value that specifies the length of the segment.
See the section below for the format of timestamps.
Unnamed parameters carry implicit names. The parameter position determines
which of the parameters listed above is set. For example, the second parameter
implicitly uses the name ``start``.
Example::
# mpv EDL v0
%18%filename,with,.mkv,10,length=20,param3=%13%value,escaped,param4=value2
this sets ``file`` to ``filename,with,.mkv``, ``start`` to ``10``, ``length``
to ``20``, ``param3`` to ``value,escaped``, ``param4`` to ``value2``.
Instead of line breaks, the character ``;`` can be used. Line feed bytes and
``;`` are treated equally.
Timestamp format
================
Currently, time values are floating point values in seconds.
As an extension, you can set the ``timestamps=chapters`` option. If this option
is set, timestamps have to be integers, and refer to chapter numbers, starting
with 0.
Example::
# mpv EDL v0
file.mkv,2,4,timestamps=chapters
Plays chapter 3 and ends with the start of chapter 7 (4 chapters later).
Syntax of EDL URIs
==================
mpv accepts inline EDL data in form of ``edl://`` URIs. Other than the
header, the syntax is exactly the same. It's far more convenient to use ``;``
instead of line breaks, but that is orthogonal.
Example: ``edl://f1.mkv,length=5,start=10;f2.mkv,30,20;f3.mkv``
+139
View File
@@ -0,0 +1,139 @@
Format of mplayer2 EDL files
============================
The first line in the file must be ``mplayer EDL file, version 2``.
The rest of the lines belong to one of these classes:
1) lines specifying source files
2) empty lines
3) lines specifying timeline segments.
Lines beginning with ``<`` specify source files. These lines first
contain an identifier used to refer to the source file later, then the
filename separated by whitespace. The identifier must start with a
letter. Filenames that start or end with whitespace or contain
newlines are not supported.
On other lines ``#`` characters delimit comments. Lines that contain
only whitespace after comments have been removed are ignored.
Timeline segments must appear in the file in chronological order. Each
segment has the following information associated with it:
- duration
- output start time
- output end time (= output start time + duration)
- source id (specifies the file the content of the segment comes from)
- source start time (timestamp in the source file)
- source end time (= source start time + duration)
The output timestamps must form a continuous timeline from 0 to the
end of the last segment, such that each new segment starts from the
time the previous one ends at. Source files and times may change
arbitrarily between segments.
The general format for lines specifying timeline segments is
[output time info] source_id [source time info]
source_id must be an identifier defined on a ``<`` line. Both the time
info parts consists of zero or more of the following elements:
1) ``timestamp``
2) ``-timestamp``
3) ``+duration``
4) ``*``
5) ``-*``
, where ``timestamp`` and ``duration`` are decimal numbers (computations
are done with nanosecond precision). Whitespace around ``+`` and ``-`` is
optional. 1) and 2) specify start and end time of the segment on
output or source side. 3) specifies duration; the semantics are the
same whether this appears on output or source side. 4) and 5) are
ignored on the output side (they're always implicitly assumed). On the
source side 4) specifies that the segment starts where the previous
segment *using this source* ended; if there was no previous segment
time 0 is used. 5) specifies that the segment ends where the next
segment using this source starts.
Redundant information may be omitted. It will be filled in using the
following rules:
- output start for first segment is 0
- two of [output start, output end, duration] imply third
- two of [source start, source end, duration] imply third
- output start = output end of previous segment
- output end = output start of next segment
- if ``*``, source start = source end of earlier segment
- if ``-*``, source end = source start of a later segment
As a special rule, a last zero-duration segment without a source
specification may appear. This will produce no corresponding segment
in the resulting timeline, but can be used as syntax to specify the
end time of the timeline (with effect equal to adding -time on the
previous line).
Examples::
mplayer EDL file, version 2
< id1 filename
0 id1 123
100 id1 456
200 id1 789
300
All segments come from the source file ``filename``. First segment
(output time 0-100) comes from time 123-223, second 456-556, third
789-889.
::
mplayer EDL file, version 2
< f filename
f 60-120
f 600-660
f 30- 90
Play first seconds 60-120 from the file, then 600-660, then 30-90.
::
mplayer EDL file, version 2
< id1 filename1
< id2 filename2
+10 id1 *
+10 id2 *
+10 id1 *
+10 id2 *
+10 id1 *
+10 id2 *
This plays time 0-10 from filename1, then 0-10 from filename1, then
10-20 from filename1, then 10-20 from filename2, then 20-30 from
filename1, then 20-30 from filename2.
::
mplayer EDL file, version 2
< t1 filename1
< t2 filename2
t1 * +2 # segment 1
+2 t2 100 # segment 2
t1 * # segment 3
t2 *-* # segment 4
t1 3 -* # segment 5
+0.111111 t2 102.5 # segment 6
7.37 t1 5 +1 # segment 7
This rather pathological example illustrates the rules for filling in
implied data. All the values can be determined by recursively applying
the rules given above, and the full end result is this::
+2 0-2 t1 0-2 # segment 1
+2 2-4 t2 100-102 # segment 2
+0.758889 4-4.758889 t1 2-2.758889 # segment 3
+0.5 4.4758889-5.258889 t2 102-102.5 # segment 4
+2 5.258889-7.258889 t1 3-5 # segment 5
+0.111111 7.258889-7.37 t2 102.5-102.611111 # segment 6
+1 7.37-8.37 t1 5-6 # segment 7
+8 -12
View File
@@ -25,7 +25,7 @@ from the input video. Note that not all codecs and not all formats support VFR
encoding, and some which do have bugs when a target bitrate is specified - use
-ofps or -oautofps to force CFR encoding in these cases.
Of course, the options can be stored in a profile, like this .config/mpv/mpv.conf
Of course, the options can be stored in a profile, like this .mpv/config
section::
[myencprofile]
@@ -38,17 +38,12 @@ section::
oac = aac
oacopts-add = b=96k
It's also possible to define default encoding options by putting them into
the section named ``[encoding]``. (This behavior changed after mpv 0.3.x. In
mpv 0.3.x, config options in the default section / no section were applied
to encoding. This is not the case anymore.)
One can then encode using this profile using the command::
mpv infile -o outfile.mp4 -profile myencprofile
Some example profiles are provided in a file
etc/encoding-profiles.conf; as for this, see below.
etc/encoding-example-profiles.conf; as for this, see below.
Encoding examples
@@ -94,12 +89,13 @@ Device targets
As the options for various devices can get complex, profiles can be used.
An example profile file for encoding is provided in
etc/encoding-profiles.conf in the source tree. This file is installed and loaded
by default (if libavfilter is enabled at compilation). If you want to modify
it, you can replace and it with your own copy by doing::
etc/encoding-example-profiles.conf in the source tree. You can include it into
your configuration by doing::
mkdir -p ~/.mpv
cp /etc/mpv/encoding-profiles.conf ~/.mpv/encoding-profiles.conf
curl https://raw.github.com/mpv-player/mpv/master/etc/encoding-example-profiles.conf \
> ~/.mpv/encoding-profiles.conf
echo "include = $HOME/.mpv/encoding-profiles.conf" >> ~/.mpv/config
Refer to the top of that file for more comments - in a nutshell, the following
options are added by it::
@@ -139,9 +135,9 @@ What works
* Deinterlacing/Inverse Telecine with any of mpv's filters for that
* Audio file converting: mpv -o outfile.mp3 infile.flac -no-video -oac
libmp3lame -oacopts ab=320k
* inverse telecine filters (confirmed working: detc, pullup, filmdint)
What does not work yet
======================
* 3-pass encoding (ensuring constant total size and bitrate constraints while
having VBR audio; mencoder calls this "frameno")
-383
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@@ -1,383 +0,0 @@
CHANGES FROM OTHER VERSIONS OF MPLAYER
======================================
**mpv** is based on mplayer2, which in turn is based on the original MPlayer
(also called mplayer, mplayer-svn, mplayer1). Many changes have been made, a
large part of which is incompatible or completely changes how the player
behaves. Although there are still many similarities to its ancestors, **mpv**
should generally be treated as a completely different program.
.. note::
These lists are incomplete.
General Changes for MPlayer to mpv
----------------------------------
* Switch to GPLv2+. (Technically speaking, MPlayer as a whole seems to be
GPLv2-only, while mplayer2 is GPLv3+ - see ``Copyright`` file for details.)
* Removal of the internal GUI, MEncoder, OSD menu, video kernel drivers for
Linux 2.4 (including VIDIX)
* Large internal cleanups
* Removal of support for dead platforms
* New build system
* No embedded copy of FFmpeg and other libraries
* Better pause handling (do not unpause on a command)
* Better MKV support (such as ordered chapters)
* vo_vdpau improvements
* Precise seeking support
* Native OpenGL backend for OS X
* General OS X improvements
* Improvements in audio/video sync handling
* Cleaned up terminal output
* Gapless audio support (``--gapless-audio``)
* Improved responsiveness on user input
* Support for modifier keys (alt, shift, ctrl) in input.conf
* OSS4 volume control
* More correct color reproduction (color matrix generation)
* Use libass for subtitle rendering by default (better quality)
* Generally preferring FFmpeg/Libav over internal demuxers and decoders
* Improvements when playing multiple files (``--fixed-vo``)
* Screenshot improvements (instant screenshots without 1-frame delay, allow
taking screenshots even with hardware decoding)
* Improved support for PulseAudio
* Generally improved MS Windows support (dealing with Unicode file names,
improved ``--vo=direct3d``, improved window handling)
* Better OSD rendering (using libass). This has full Unicode support, and
languages like Arabic should be better supported.
* Cleaned up terminal output (nicer status line, less useless noise)
* Support for playing URLs of popular streaming sites directly
(e.g. ``mpv https://www.youtube.com/watch?v=...``)
* Improved OpenGL output (``--vo=opengl-hq``)
* Make ``--softvol`` default (**mpv** is not a mixer control panel)
* Improved support for .cue files
* Screenshot improvements (can save screenshots as JPEG or PNG, configurable
file names, support for taking screenshots with or without subtitles - the
``screenshot`` video filter is not needed anymore, and should not be put
into the mpv config file)
* Removal of teletext support
* Removal of most built-in demuxers, using libavformat instead
* Removal of built-in network support, using libavformat instead (also,
support https via libavformat)
* Replace image VOs (``--vo=jpeg`` etc.) with ``--vo=image``
* Do not lose settings when playing a new file in the same player instance
* New location for config files, new name for the binary.
* Slave mode compatibility broken (see below)
* Encoding functionality (replacement for MEncoder, see ``DOCS/encoding.rst``)
* Remove ``--vo=gif89a``, ``--vo=md5sum``, ``--vo=yuv4mpeg``, as encoding can
handle these use cases. For yuv4mpeg, for example, use:
``mpv input.mkv -o output.y4m --no-audio --oautofps --oneverdrop``.
* Image subtitles (DVDs etc.) are rendered in color and use more correct
positioning (color can be disabled with ``--sub-gray``)
* Wayland support
* Support for precise scrolling which scales the parameter of commands. If the
input doesn't support precise scrolling the scale factor stays 1.
* OS X: Cocoa event loop is independent from MPlayer's event loop, so user
actions like accessing menus and live resizing do not block the playback.
* OS X: Apple Remote support.
* OS X: Media Keys support.
* Windows: Added WASAPI audio output.
* New OSD bar with chapter marks and not positioned in the middle of the video
(though this can be customized with the ``--osd-bar-align-y`` option).
* Allow customizing whether a key binding for seeking shows the video time, the
OSD bar, or nothing (see section `Input Command Prefixes`_).
* Display list of chapters and audio/subtitle tracks on OSD (see section
`Properties`_).
* ``--keep-open`` option to stop the player from closing the window and
exiting after playback ends.
* Re-enable screensaver while the player is paused.
* Matroska edition switching at runtime.
* Support for libavfilter (for video->video and audio->audio). This allows
using most of FFmpeg's filters, which improve greatly on the old MPlayer
filters in features, performance, and correctness.
* Improved downmixing and output of surround audio. Instead of using hardcoded
pan filters to do remixing, use libavresample. Channel maps are used to
identify the channel layout, so e.g. ``3.0`` and ``2.1`` audio can be
distinguished.
* Allow resuming playback at a later point with ``Shift+q``, also see
``quit_watch_later`` input command.
* Support mapping multiple commands to one key.
* Allow changing/adjusting video filters at runtime. (This is also used to make
the ``D`` key insert vf_yadif if deinterlacing is not supported otherwise.)
* Native VAAPI support
* OS X: VDA support using libavcodec hwaccel API instead of FFmpeg's decoder. Up
to 2-2.5x reduction in CPU usage.
* Make hardware decoding in general work with the ``opengl`` video output.
* Lua scripting (see `LUA SCRIPTING`_)
* A client API, that allows embedding **mpv** into applications
(see ``libmpv/client.h`` in the sources)
* General bug fixes and removal of long-standing issues
* General code cleanups (including refactoring or rewrites of many parts)
* Many more changes
Detailed Listing of User-visible Changes
----------------------------------------
This listing is about changed command line switches, slave commands, and similar
things. Completely removed features are not listed.
Command Line Switches
~~~~~~~~~~~~~~~~~~~~~
* There is a new command line syntax, which is generally preferred over the old
syntax. ``-optname optvalue`` becomes ``--optname=optvalue``.
The old syntax will not be removed. However, the new syntax is mentioned in
all documentation and so on, and unlike the old syntax is not ambiguous,
so it is a good thing to know about this change.
* In general, negating switches like ``-noopt`` now have to be written as
``-no-opt`` or ``--no-opt``.
* Per-file options are not the default anymore. You can explicitly specify
file-local options. See ``Usage`` section.
* Many options have been renamed, removed or changed semantics. Some options
that are required for a good playback experience with MPlayer are now
superfluous or even worse than the defaults, so make sure to read the manual
before trying to use your existing configuration with **mpv**.
* Table of renamed/replaced switches:
=========================== ========================================
Old New
=========================== ========================================
``-no<opt>`` ``--no-<opt>`` (add a dash)
``-a52drc level`` ``--ad-lavc-ac3drc=level``
``-ac spdifac3`` ``--ad=spdif:ac3`` (see ``--ad=help``)
``-af volnorm`` ``--af=drc`` (renamed)
``-afm hwac3`` ``--ad=spdif:ac3,spdif:dts``
``-ao alsa:device=hw=0.3`` ``--ao=alsa:device=[hw:0,3]``
``-aspect`` ``--video-aspect``
``-ass-bottom-margin`` ``--vf=sub=bottom:top``
``-ass`` ``--sub-ass``
``-audiofile-cache`` (removed; the main cache settings are used)
``-audiofile`` ``--audio-file``
``-benchmark`` ``--untimed`` (no stats)
``-capture`` ``--stream-capture=<filename>``
``-channels`` ``--audio-channels`` (changed semantics)
``-cursor-autohide-delay`` ``--cursor-autohide``
``-delay`` ``--audio-delay``
``-dumpstream`` ``--stream-dump=<filename>``
``-dvdangle`` ``--dvd-angle``
``-endpos`` ``--length``
``-font`` ``--osd-font``
``-forcedsubsonly`` ``--sub-forced-only``
``-forceidx`` ``--index``
``-format`` ``--audio-format``
``-fsmode-dontuse`` (removed)
``-fstype`` ``--x11-netwm`` (changed semantics)
``-hardframedrop`` ``--framedrop=hard``
``-identify`` (removed; use TOOLS/mpv_identify.sh)
``-idx`` ``--index``
``-lavdopts ...`` ``--vd-lavc-...``
``-lavfdopts`` ``--demuxer-lavf-...``
``-lircconf`` ``--input-lirc-conf``
``-loop 0`` ``--loop=inf``
``-mixer-channel`` AO suboptions (``alsa``, ``oss``)
``-mixer`` AO suboptions (``alsa``, ``oss``)
``-mouse-movements`` ``--input-cursor``
``-msgcolor`` ``--msg-color``
``-msglevel`` ``--msg-level`` (changed semantics)
``-msgmodule`` ``--msg-module``
``-name`` ``--x11-name``
``-noar`` ``--no-input-appleremote``
``-noautosub`` ``--no-sub-auto``
``-noconsolecontrols`` ``--no-input-terminal``
``-nojoystick`` ``--no-input-joystick``
``-nosound`` ``--no-audio``
``-osdlevel`` ``--osd-level``
``-panscanrange`` ``--video-zoom``, ``--video-pan-x/y``
``-playing-msg`` ``--term-playing-msg``
``-pp ...`` ``'--vf=pp=[...]'``
``-pphelp`` ``--vf=pp:help``
``-rawaudio ...`` ``--demuxer-rawaudio-...``
``-rawvideo ...`` ``--demuxer-rawvideo-...``
``-spugauss`` ``--sub-gauss``
``-srate`` ``--audio-samplerate``
``-ss`` ``--start``
``-ssf <sub>`` ``--sws-...``
``-stop-xscreensaver`` ``--stop-screensaver``
``-sub-fuzziness`` ``--sub-auto``
``-sub`` ``--sub-file``
``-subcp`` ``--sub-codepage``
``-subdelay`` ``--sub-delay``
``-subfile`` ``--sub-file``
``-subfont-*`` ``--sub-text-*``, ``--osd-*``
``-subfont-text-scale`` ``--sub-scale``
``-subfont`` ``--sub-text-font``
``-subfps`` ``--sub-fps``
``-subpos`` ``--sub-pos``
``-sws`` ``--sws-scaler``
``-tvscan`` ``--tv-scan``
``-use-filename-title`` ``--title='${filename}'``
``-vc ffh264vdpau`` (etc.) ``--hwdec=vdpau``
``-vobsub`` ``--sub-file`` (pass the .idx file)
``-x W``, ``-y H`` ``--geometry=WxH`` + ``--no-keepaspect``
``-xineramascreen`` ``--screen`` (different values)
``-xy W`` ``--autofit=W``
``-zoom`` Inverse available as ``--video-unscaled``
``dvdnav://`` ``dvdnav://menu``
``dvd://1`` ``dvd://0`` (0-based offset)
=========================== ========================================
.. note::
``-opt val`` becomes ``--opt=val``.
.. note::
Quite some video filters, video outputs, audio filters, audio outputs, had
changes in their option parsing. These aren't mentioned in the table above.
Also, some video and audio filters have been removed, and you have to use
libavfilter (using ``--vf=lavfi=[...]`` or ``--af=lavfi=[...]``) to get
them back.
input.conf and Slave Commands
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
* Table of renamed input commands:
This lists only commands that are not always gracefully handled by the
internal legacy translation layer. If an input.conf contains any legacy
commands, a warning will be printed when starting the player. The warnings
also show the replacement commands.
Properties containing ``_`` to separate words use ``-`` instead.
+--------------------------------+----------------------------------------+
| Old | New |
+================================+========================================+
| ``pt_step 1 [0|1]`` | ``playlist_next [weak|force]`` |
| | (translation layer cannot deal with |
| | whitespace) |
+--------------------------------+----------------------------------------+
| ``pt_step -1 [0|1]`` | ``playlist_prev [weak|force] (same)`` |
+--------------------------------+----------------------------------------+
| ``switch_ratio [<ratio>]`` | ``set video-aspect <ratio>`` |
| | |
| | ``set video-aspect 0`` (reset aspect) |
+--------------------------------+----------------------------------------+
| ``step_property_osd <prop>`` | ``cycle <prop> <step>`` (wraps), |
| ``<step> <dir>`` | ``add <prop> <step>`` (clamps). |
| | ``<dir>`` parameter unsupported. Use |
| | a negative ``<step>`` instead. |
+--------------------------------+----------------------------------------+
| ``step_property <prop>`` | Prefix ``cycle`` or ``add`` with |
| ``<step> <dir>`` | ``no-osd``: ``no-osd cycle <prop>`` |
| | ``<step>`` |
+--------------------------------+----------------------------------------+
| ``osd_show_property_text`` | ``show_text <text>`` |
| ``<text>`` | The property expansion format string |
| | syntax slightly changed. |
+--------------------------------+----------------------------------------+
| ``osd_show_text`` | Now does the same as |
| | ``osd_show_property_text``. Use the |
| | ``raw`` prefix to disable property |
| | expansion. |
+--------------------------------+----------------------------------------+
| ``show_tracks`` | ``show_text ${track-list}`` |
+--------------------------------+----------------------------------------+
| ``show_chapters`` | ``show_text ${chapter-list}`` |
+--------------------------------+----------------------------------------+
| ``af_switch``, ``af_add``, ... | ``af set|add|...`` |
+--------------------------------+----------------------------------------+
| ``tv_start_scan`` | ``set tv-scan yes`` |
+--------------------------------+----------------------------------------+
| ``tv_set_channel <val>`` | ``set tv-channel <val>`` |
+--------------------------------+----------------------------------------+
| ``tv_step_channel`` | ``cycle tv-channel`` |
+--------------------------------+----------------------------------------+
| ``dvb_set_channel <v1> <v2>`` | ``set dvb-channel <v1>-<v2>`` |
+--------------------------------+----------------------------------------+
| ``dvb_step_channel`` | ``cycle dvb-channel`` |
+--------------------------------+----------------------------------------+
| ``tv_set_freq <val>`` | ``set tv-freq <val>`` |
+--------------------------------+----------------------------------------+
| ``tv_step_freq`` | ``cycle tv-freq`` |
+--------------------------------+----------------------------------------+
| ``tv_set_norm <norm>`` | ``set tv-norm <norm>`` |
+--------------------------------+----------------------------------------+
| ``tv_step_norm`` | ``cycle tv-norm`` |
+--------------------------------+----------------------------------------+
.. note::
Due to lack of hardware and users using the TV/DVB/PVR features, and
due to the need to cleanup the related command code, it's possible
that the new commands are buggy or behave worse. This can be improved
if testers are available. Otherwise, some of the TV code will be
removed at some point.
Slave mode
~~~~~~~~~~
* Slave mode was removed. A proper slave mode application needed tons of code
and hacks to get
it right. The main problem is that slave mode is a bad and incomplete
interface, and to get around that, applications parsed output messages
intended for users. It is hard to know which messages exactly are parsed by
slave mode applications. This makes it virtually impossible to improve
terminal output intended for users without possibly breaking something.
This is absolutely insane, and since initial improvements to **mpv** quickly
made slave mode incompatible to most applications, it was removed as useless
cruft. The client API (see below) is provided instead.
``--identify`` was replaced by the ``TOOLS/mpv_identify.sh`` wrapper script.
* For some time (until including release 0.4.x), mpv supported a
``--slave-broken`` option. The following options are equivalent:
::
--input-file=/dev/stdin --input-terminal=no
Assuming the system supports ``/dev/stdin``.
(The option was readded in 0.5.1 and sets exactly these options.)
* A JSON RPC protocol giving access to the client API is planned, but nothing
has emerged yet.
* **mpv** also provides a client API, which can be used to embed the player
by loading it as shared library. (See ``libmpv/client.h`` in the sources.)
It might also be possible to implement a custom slave mode-like protocol
using Lua scripting.
Policy for Removed Features
---------------------------
**mpv** is in active development. If something is in the way of more important
development (such as fixing bugs or implementing new features), we sometimes
remove features. Usually this happens only with old features that either seem
to be useless, or are not used by anyone. Often these are obscure, or
"inherited", or were marked experimental, but never received any particular
praise by any users.
Sometimes, features are replaced by something new. The new code will be either
simpler or more powerful, but doesn't necessarily provide everything the old
feature did.
We can not exclude that we accidentally remove features that are actually
popular. Generally, we do not know how much a specific functionality is used.
If you miss a feature and think it should be re-added, please open an issue
on the mpv bug tracker. Hopefully, a solution can be found. Often, it turns
out that re-adding something is not much of a problem, or that there are
better alternatives.
Why this Fork?
--------------
mplayer2 is practically dead, and mpv started out as a branch containing
new/experimental development. (Some of it was merged right *after* the fork
was made public, seemingly as an acknowledgment that development, or at
least merging, should have been more active.)
MPlayer is focused on not breaking anything, but is stuck with a horrible
codebase resistant to cleanup. (Unless you do what mpv did - merciless and
consequent pruning of bad, old code.) Cleanup and keeping broken things
conflict, so the kind of development mpv strives for can't be done within
MPlayer due to clashing development policies.
Additionally, mplayer2 already had lots of changes over MPlayer, which would
have needed to be backported to the MPlayer codebase. This would not only
have been hard (several years of diverging development), but also would have
been impossible due to the aforementioned MPlayer development policy.
+93 -136
View File
@@ -11,12 +11,6 @@ syntax is:
To get a full list of available audio filters, see ``--af=help``.
You can also set defaults for each filter. The defaults are applied before the
normal filter parameters.
``--af-defaults=<filter1[=parameter1:parameter2:...],filter2,...>``
Set defaults for each filter.
Audio filters are managed in lists. There are a few commands to manage the
filter list:
@@ -41,7 +35,7 @@ Available filters are:
This filter is automatically enabled if the audio output does not support
the audio configuration of the file being played.
It supports only the following sample formats: u8, s16, s32, float.
It supports only the following sample formats: u8, s16ne, s32ne, floatne.
``filter-size=<length>``
Length of the filter with respect to the lower sampling rate. (default:
@@ -56,9 +50,9 @@ Available filters are:
entries. (default: no)
``no-detach``
Do not detach if input and output audio format/rate/channels match.
(If you just want to set defaults for this filter that will be used
even by automatically inserted lavrresample instances, you should
prefer setting them with ``--af-defaults=lavrresample:...``.)
You should add this option if you specify additional parameters, as
automatically inserted lavrresample instances will use the default
settings.
``o=<string>``
Set AVOptions on the SwrContext or AVAudioResampleContext. These should
be documented by FFmpeg or Libav.
@@ -70,18 +64,17 @@ Available filters are:
outputting to S/PDIF. If the input sample rate is not 48 kHz, 44.1 kHz or
32 kHz, it will be resampled to 48 kHz.
``tospdif=<yes|no>``
Output raw AC-3 stream if ``no``, output to S/PDIF for
pass-through if ``yes`` (default).
``bitrate=<rate>``
The bitrate use for the AC-3 stream. Set it to 384 to get 384 kbps.
``<tospdif>``
Output raw AC-3 stream if zero or not set, output to S/PDIF for
passthrough when ``<tospdif>`` is set non-zero.
``<bitrate>``
The bitrate use for the AC-3 stream. Set it to either 384 or 384000
to get 384 kbps.
Valid values: 32, 40, 48, 56, 64, 80, 96, 112, 128,
160, 192, 224, 256, 320, 384, 448, 512, 576, 640.
The special value ``default`` selects a default bitrate based on the
input channel number:
Default bitrate is based on the input channel number:
:1ch: 96
:2ch: 192
@@ -90,7 +83,7 @@ Available filters are:
:5ch: 448
:6ch: 448
``minchn=<n>``
``<minchn>``
If the input channel number is less than ``<minchn>``, the filter will
detach itself (default: 5).
@@ -101,7 +94,7 @@ Available filters are:
Sine function delta, use very low values to hear the sweep.
``sinesuppress[=freq:decay]``
Remove a sine at the specified frequency. Useful to get rid of the 50/60 Hz
Remove a sine at the specified frequency. Useful to get rid of the 50/60Hz
noise on low quality audio equipment. It only works on mono input.
``<freq>``
@@ -147,7 +140,7 @@ Available filters are:
0 no matrix decoding (default)
==== ===================================
``equalizer=g1:g2:g3:...:g10``
``equalizer=[g1:g2:g3:...:g10]``
10 octave band graphic equalizer, implemented using 10 IIR band-pass
filters. This means that it works regardless of what type of audio is
being played back. The center frequencies for the 10 bands are:
@@ -182,9 +175,9 @@ Available filters are:
``mpv --af=equalizer=11:11:10:5:0:-12:0:5:12:12 media.avi``
Would amplify the sound in the upper and lower frequency region
while canceling it almost completely around 1 kHz.
while canceling it almost completely around 1kHz.
``channels=nch[:routes]``
``channels=nch[:nr:from1:to1:from2:to2:from3:to3:...]``
Can be used for adding, removing, routing and copying audio channels. If
only ``<nch>`` is given, the default routing is used. It works as follows:
If the number of output channels is greater than the number of input
@@ -195,53 +188,46 @@ Available filters are:
``<nch>``
number of output channels (1-8)
``<routes>``
List of ``,`` separated routes, in the form ``from1-to1,from2-to2,...``.
Each pair defines where to route each channel. There can be at most
8 routes. Without this argument, the default routing is used. Since
``,`` is also used to separate filters, you must quote this argument
with ``[...]`` or similar.
``<nr>``
number of routes (1-8)
``<from1:to1:from2:to2:from3:to3:...>``
Pairs of numbers between 0 and 7 that define where to route each
channel.
.. admonition:: Examples
``mpv --af=channels=4:[0-1,1-0,0-2,1-3] media.avi``
``mpv --af=channels=4:4:0:1:1:0:2:2:3:3 media.avi``
Would change the number of channels to 4 and set up 4 routes that
swap channel 0 and channel 1 and leave channel 2 and 3 intact.
Observe that if media containing two channels were played back,
channels 2 and 3 would contain silence but 0 and 1 would still be
swapped.
``mpv --af=channels=6:[0-0,0-1,0-2,0-3] media.avi``
``mpv --af=channels=6:4:0:0:0:1:0:2:0:3 media.avi``
Would change the number of channels to 6 and set up 4 routes that
copy channel 0 to channels 0 to 3. Channel 4 and 5 will contain
silence.
.. note::
You should probably not use this filter. If you want to change the
output channel layout, try the ``format`` filter, which can make mpv
automatically up- and downmix standard channel layouts.
``format=format:srate:channels:out-format:out-srate:out-channels``
``force=in-format:in-srate:in-channels:out-format:out-srate:out-channels``
Force a specific audio format/configuration without actually changing the
audio data. Keep in mind that the filter system might auto-insert actual
conversion filters before or after this filter if needed.
All parameters are optional. The first 3 parameters restrict what the filter
accepts as input. The ``out-`` parameters change the audio format, without
All parameters are optional. The ``in-`` variants restrict what the filter
accepts as input. The ``out-`` variants change the audio format, without
actually doing a conversion. The data will be 'reinterpreted' by the
filters or audio outputs following this filter.
``<format>``
Force conversion to this format. Use ``--af=format=format=help`` to get
a list of valid formats.
``<in-format>``
Force conversion to this format. See ``format`` filter for valid audio
format values.
``<srate>``
``<in-srate>``
Force conversion to a specific sample rate. The rate is an integer,
48000 for example.
``<channels>``
Force mixing to a specific channel layout. See ``--audio-channels`` option
``<in-channels>``
Force mixing to a specific channel layout. See ``--channels`` option
for possible values.
``<out-format>``
@@ -250,71 +236,55 @@ Available filters are:
``<out-channels>``
See also ``--audio-format``, ``--audio-samplerate``, and
``--audio-channels`` for related options. Keep in mind that
``--audio-channels`` does not actually force the number of
channels in most cases, while this filter can do this.
``format[=format]``
Convert between different sample formats. Automatically enabled when
needed by the audio output or another filter. See also ``--format``.
*NOTE*: this filter used to be named ``force``. Also, unlike the old
``format`` filter, this does not do any actual conversion anymore.
Conversion is done by other, automatically inserted filters.
``<format>``
Sets the desired format. The general form is 'sbe', where 's' denotes
the sign (either 's' for signed or 'u' for unsigned), 'b' denotes the
number of bits per sample (16, 24 or 32) and 'e' denotes the
endianness ('le' means little-endian, 'be' big-endian and 'ne' the
endianness of the computer mpv is running on). Valid values (amongst
others) are: 's16le', 'u32be' and 'u24ne'. Exceptions to this rule that
are also valid format specifiers: u8, s8, floatle, floatbe, floatne,
mpeg2, and ac3.
``convert24``
Filter for internal use only. Converts between 24-bit and 32-bit sample
formats.
``convertsignendian``
Filter for internal use only. Converts between signed/unsigned formats
and formats with different endian.
``volume[=<volumedb>[:...]]``
``volume[=v[:sc[:fast]]]``
Implements software volume control. Use this filter with caution since it
can reduce the signal to noise ratio of the sound. In most cases it is
best to use the *Master* volume control of your sound card or the volume
knob on your amplifier.
This filter has a second feature: It measures the overall maximum sound
level and prints out that level when mpv exits. This feature currently
only works with floating-point data.
*NOTE*: This filter is not reentrant and can therefore only be enabled
once for every audio stream.
``<volumedb>``
Sets the desired gain in dB for all channels in the stream from -200 dB
to +60 dB, where -200 dB mutes the sound completely and +60 dB equals a
``<v>``
Sets the desired gain in dB for all channels in the stream from -200dB
to +60dB, where -200dB mutes the sound completely and +60dB equals a
gain of 1000 (default: 0).
``replaygain-track``
Adjust volume gain according to the track-gain replaygain value stored
in the file metadata.
``replaygain-album``
Like replaygain-track, but using the album-gain value instead.
``replaygain-preamp``
Pre-amplification gain in dB to apply to the selected replaygain gain
(default: 0).
``replaygain-clip=yes|no``
Prevent clipping caused by replaygain by automatically lowering the
gain (default). Use ``replaygain-clip=no`` to disable this.
``softclip``
Turns soft clipping on. Soft-clipping can make the
``<sc>``
Turns soft clipping on (1) or off (0). Soft-clipping can make the
sound more smooth if very high volume levels are used. Enable this
option if the dynamic range of the loudspeakers is very low.
*WARNING*: This feature creates distortion and should be considered a
last resort.
``s16``
Force S16 sample format if set. Lower quality, but might be faster
``<fast>``
Force S16 sample format if set to 1. Lower quality, but might be faster
in some situations.
``detach``
Remove the filter if the volume is not changed at audio filter config
time. Useful with replaygain: if the current file has no replaygain
tags, then the filter will be removed if this option is enabled.
(If ``--softvol=yes`` is used and the player volume controls are used
during playback, a different volume filter will be inserted.)
.. admonition:: Example
``mpv --af=volume=10.1 media.avi``
Would amplify the sound by 10.1 dB and hard-clip if the sound level
``mpv --af=volume=10.1:0 media.avi``
Would amplify the sound by 10.1dB and hard-clip if the sound level
is too high.
``pan=n:[<matrix>]``
``pan=n[:L00:L01:L02:...L10:L11:L12:...Ln0:Ln1:Ln2:...]``
Mixes channels arbitrarily. Basically a combination of the volume and the
channels filter that can be used to down-mix many channels to only a few,
e.g. stereo to mono, or vary the "width" of the center speaker in a
@@ -326,33 +296,29 @@ Available filters are:
``<n>``
Number of output channels (1-8).
``<matrix>``
A list of values ``[L00,L01,L02,...,L10,L11,L12,...,Ln0,Ln1,Ln2,...]``,
where each element ``Lij`` means how much of input channel i is mixed
into output channel j (range 0-1). So in principle you first have n
numbers saying what to do with the first input channel, then n numbers
that act on the second input channel etc. If you do not specify any
numbers for some input channels, 0 is assumed.
Note that the values are separated by ``,``, which is already used
by the option parser to separate filters. This is why you must quote
the value list with ``[...]`` or similar.
``<Lij>``
How much of input channel i is mixed into output channel j (0-1). So
in principle you first have n numbers saying what to do with the first
input channel, then n numbers that act on the second input channel
etc. If you do not specify any numbers for some input channels, 0 is
assumed.
.. admonition:: Examples
``mpv --af=pan=1:[0.5,0.5] media.avi``
``mpv --af=pan=1:0.5:0.5 media.avi``
Would downmix from stereo to mono.
``mpv --af=pan=3:[1,0,0.5,0,1,0.5] media.avi``
``mpv --af=pan=3:1:0:0.5:0:1:0.5 media.avi``
Would give 3 channel output leaving channels 0 and 1 intact, and mix
channels 0 and 1 into output channel 2 (which could be sent to a
subwoofer for example).
.. note::
.. note::
If you just want to force remixing to a certain output channel layout,
it is easier to use the ``format`` filter. For example,
``mpv '--af=format=channels=5.1' '--audio-channels=5.1'`` would always force
remixing audio to 5.1 and output it like this.
If you just want to force remixing to a certain output channel
layout, it is easier to use the ``force`` filter. For example,
``mpv '--af=force=channels=5.1' '--channels=5.1'`` would always
force remixing audio to 5.1 and output it like this.
``sub[=fc:ch]``
Adds a subwoofer channel to the audio stream. The audio data used for
@@ -368,8 +334,8 @@ Available filters are:
to the subwoofer.
``<fc>``
cutoff frequency in Hz for the low-pass filter (20 Hz to 300 Hz)
(default: 60 Hz) For the best result try setting the cutoff frequency
cutoff frequency in Hz for the low-pass filter (20Hz to 300Hz)
(default: 60Hz) For the best result try setting the cutoff frequency
as low as possible. This will improve the stereo or surround sound
experience.
``<ch>``
@@ -380,8 +346,8 @@ Available filters are:
.. admonition:: Example
``mpv --af=sub=100:4 --audio-channels=5 media.avi``
Would add a subwoofer channel with a cutoff frequency of 100 Hz to
``mpv --af=sub=100:4 --channels=5 media.avi``
Would add a subwoofer channel with a cutoff frequency of 100Hz to
output channel 4.
``center``
@@ -408,16 +374,16 @@ Available filters are:
.. admonition:: Example
``mpv --af=surround=15 --audio-channels=4 media.avi``
Would add surround sound decoding with 15 ms delay for the sound to
``mpv --af=surround=15 --channels=4 media.avi``
Would add surround sound decoding with 15ms delay for the sound to
the rear speakers.
``delay[=[ch1,ch2,...]]``
``delay[=ch1:ch2:...]``
Delays the sound to the loudspeakers such that the sound from the
different channels arrives at the listening position simultaneously. It is
only useful if you have more than 2 loudspeakers.
``[ch1,ch2,...]``
``ch1,ch2,...``
The delay in ms that should be imposed on each channel (floating point
number between 0 and 1000).
@@ -436,11 +402,11 @@ Available filters are:
.. admonition:: Example
``mpv --af=delay=[10.5,10.5,0,0,7,0] media.avi``
Would delay front left and right by 10.5 ms, the two rear channels
and the subwoofer by 0 ms and the center channel by 7 ms.
``mpv --af=delay=10.5:10.5:0:0:7:0 media.avi``
Would delay front left and right by 10.5ms, the two rear channels
and the subwoofer by 0ms and the center channel by 7ms.
``export=mmapped_file:nsamples]``
``export[=mmapped_file[:nsamples]]``
Exports the incoming signal to other processes using memory mapping
(``mmap()``). Memory mapped areas contain a header::
@@ -452,7 +418,7 @@ Available filters are:
The rest is payload (non-interleaved) 16-bit data.
``<mmapped_file>``
File to map data to (required)
File to map data to (default: ``~/.mpv/mpv-af_export``).
``<nsamples>``
number of samples per channel (default: 512).
@@ -472,7 +438,7 @@ Available filters are:
``drc[=method:target]``
Applies dynamic range compression. This maximizes the volume by compressing
the audio signal's dynamic range. (Formerly called ``volnorm``.)
the audio signal's dynamic range.
``<method>``
Sets the used method.
@@ -493,7 +459,7 @@ Available filters are:
This filter can cause distortion with audio signals that have a very
large dynamic range.
``ladspa=file:label:[<control0>,<control1>,...]``
``ladspa=file:label[:controls...]``
Load a LADSPA (Linux Audio Developer's Simple Plugin API) plugin. This
filter is reentrant, so multiple LADSPA plugins can be used at once.
@@ -509,21 +475,12 @@ Available filters are:
one filter, but others contain many of them. Entering 'help' here
will list all available filters within the specified library, which
eliminates the use of 'listplugins' from the LADSPA SDK.
``[<control0>,<control1>,...]``
Controls are zero or more ``,`` separated floating point values that
determine the behavior of the loaded plugin (for example delay,
threshold or gain).
``<controls>``
Controls are zero or more floating point values that determine the
behavior of the loaded plugin (for example delay, threshold or gain).
In verbose mode (add ``-v`` to the mpv command line), all
available controls and their valid ranges are printed. This eliminates
the use of 'analyseplugin' from the LADSPA SDK.
Note that ``,`` is already used by the option parser to separate
filters, so you must quote the list of values with ``[...]`` or
similar.
.. admonition:: Example
``mpv --af=ladspa='/usr/lib/ladspa/delay.so':delay_5s:[0.5,0.2] media.avi``
Does something.
``karaoke``
Simple voice removal filter exploiting the fact that voice is usually
@@ -588,9 +545,9 @@ Available filters are:
tempo.
``mpv --af=scaletempo=stride=30:overlap=.50:search=10 media.ogg``
Would tweak the quality and performance parameters.
Would tweak the quality and performace parameters.
``mpv --af=format=float,scaletempo media.ogg``
``mpv --af=format=floatne,scaletempo media.ogg``
Would make scaletempo use float code. Maybe faster on some
platforms.
@@ -600,7 +557,7 @@ Available filters are:
1.2x.
``lavfi=graph``
Filter audio using FFmpeg's libavfilter.
Filter audio using ffmpeg's libavfilter.
``<graph>``
Libavfilter graph. See ``lavfi`` video filter for details - the graph
+11 -101
View File
@@ -10,53 +10,35 @@ syntax is:
If the list has a trailing ',', mpv will fall back on drivers not contained
in the list. Suboptions are optional and can mostly be omitted.
You can also set defaults for each driver. The defaults are applied before the
normal driver parameters.
``--ao-defaults=<driver1[:parameter1:parameter2:...],driver2,...>``
Set defaults for each driver.
.. note::
See ``--ao=help`` for a list of compiled-in audio output drivers. The
driver ``--ao=alsa`` is preferred. ``--ao=pulse`` is preferred on systems
where PulseAudio is used. On Windows, ``--ao=wasapi`` is preferred,
though it might cause trouble sometimes, in which case ``--ao=dsound``
should be used. On BSD systems, ``--ao=oss`` or `--ao=sndio`` may work
(the latter being experimental). On OS X systems, use ``--ao=coreaudio``.
See ``--ao=help`` for a list of compiled-in audio output drivers.
.. admonition:: Examples
- ``--ao=alsa,oss,`` Try the ALSA driver, then the OSS driver, then others.
- ``--ao=alsa:no-block:device=[hw:0,3]`` Sets noblock-mode and the
- ``--ao=alsa:noblock:device=[hw:0,3]`` Sets noblock-mode and the
device-name as first card, fourth device.
Available audio output drivers are:
``alsa`` (Linux only)
ALSA audio output driver
ALSA 0.9/1.x audio output driver
``no-block``
Sets noblock-mode.
``device=<device>``
Sets the device name. For ac3 output via S/PDIF, use an "iec958" or
"spdif" device, unless you really know how to set it correctly.
``no-block``
Sets noblock-mode.
``resample=yes``
Enable ALSA resampling plugin. (This is disabled by default, because
some drivers report incorrect audio delay in some cases.)
``mixer-device=<device>``
Set the mixer device used with ``--no-softvol`` (default: ``default``).
``mixer-name=<name>``
Set the name of the mixer element (default: ``Master``). This is for
example ``PCM`` or ``Master``.
``mixer-index=<number>``
Set the index of the mixer channel (default: 0). Consider the output of
Set the index of the mixer channel (default: 0). Consider the output
"``amixer scontrols``", then the index is the number that follows the
name of the element.
``non-interleaved``
Allow output of non-interleaved formats (if the audio decoder uses
this format). Currently disabled by default, because some popular
ALSA plugins are utterly broken with non-interleaved formats.
.. note::
@@ -90,6 +72,9 @@ Available audio output drivers are:
``name=<client>``
Client name that is passed to JACK (default: ``mpv``). Useful
if you want to have certain connections established automatically.
``(no-)estimate``
Estimate the audio delay, supposed to make the video playback smoother
(default: enabled).
``(no-)autostart``
Automatically start jackd if necessary (default: disabled). Note that
this tends to be unreliable and will flood stdout with server messages.
@@ -114,19 +99,7 @@ Available audio output drivers are:
with fixed setups.
``coreaudio`` (Mac OS X only)
Native Mac OS X audio output driver using the AudioUnits and the CoreAudio
sound server.
``device_id=<id>``
ID of output device to use (0 = default device)
``help``
List all available output devices with their IDs.
``coreaudio_device`` (Mac OS X only)
Native Mac OS X audio output driver using direct device access and
exclusive mode (bypasses the sound server).
Supports only compressed formats (AC3 and DTS).
Native Mac OS X audio output driver
``device_id=<id>``
ID of output device to use (0 = default device)
@@ -136,9 +109,6 @@ Available audio output drivers are:
``openal``
Experimental OpenAL audio output driver
.. note:: This driver is not very useful. Playing multi-channel audio with
it is slow.
``pulse``
PulseAudio audio output driver
@@ -147,26 +117,10 @@ Available audio output drivers are:
string uses a local connection, "localhost" uses network transfer
(most likely not what you want).
``buffer=<1-2000|native>``
Set the audio buffer size in milliseconds. A higher value buffers
more data, and has a lower probability of buffer underruns. A smaller
value makes the audio stream react faster, e.g. to playback speed
changes. Default: 250.
``latency-hacks=<yes|no>``
Enable hacks to workaround PulseAudio timing bugs (default: yes). If
enabled, mpv will do elaborate latency calculations on its own. If
disabled, it will use PulseAudio automatically updated timing
information. Disabling this might help with e.g. networked audio.
``portaudio``
PortAudio audio output driver. This works on all platforms, and has
extensive MS Windows support.
.. note:: This driver is not very useful. It was added in the hope of
providing portable audio API across all platforms, but turned
out semi-broken and underfeatured.
``device``
Specify the subdevice to use. Giving ``help`` as device name lists all
devices found by PortAudio. Devices can be given as numeric values,
@@ -175,8 +129,6 @@ Available audio output drivers are:
``dsound`` (Windows only)
DirectX DirectSound audio output driver
.. note:: This driver is for compatibility with old systems.
``device=<devicenum>``
Sets the device number to use. Playing a file with ``-v`` will show a
list of available devices.
@@ -186,10 +138,6 @@ Available audio output drivers are:
1.2, but may require the ``SDL_AUDIODRIVER`` environment variable to be set
appropriately for your system.
.. note:: This driver is for compatibility with extremely foreign
environments, such as systems where none of the other drivers
are available.
``buflen=<length>``
Sets the audio buffer length in seconds. Is used only as a hint by the
sound system. Playing a file with ``-v`` will show the requested and
@@ -202,30 +150,7 @@ Available audio output drivers are:
``null``
Produces no audio output but maintains video playback speed. Use
``--ao=null:untimed`` for benchmarking.
``untimed``
Do not simulate timing of a perfect audio device. This means audio
decoding will go as fast as possible, instead of timing it to the
system clock.
``buffer``
Simulated buffer length in seconds.
``outburst``
Simulated chunk size in samples.
``speed``
Simulated audio playback speed as a multiplier. Usually, a real audio
device will not go exactly as fast as the system clock. It will deviate
just a little, and this option helps simulating this.
``latency``
Simulated device latency. This is additional to EOF.
``broken-eof``
Simulate broken audio drivers, which always add the fixed device
latency to the reported audio playback position.
``--no-audio`` for benchmarking.
``pcm``
Raw PCM/WAVE file writer audio output
@@ -241,10 +166,6 @@ Available audio output drivers are:
``rsound``
Audio output to an RSound daemon
.. note:: Completely useless, unless you intend to run RSound. Not to be
confused with RoarAudio, which is something completely
different.
``host=<name/path>``
Set the address of the server (default: localhost). Can be either a
network hostname for TCP connections or a Unix domain socket path
@@ -253,17 +174,6 @@ Available audio output drivers are:
Set the TCP port used for connecting to the server (default: 12345).
Not used if connecting to a Unix domain socket.
``sndio``
Audio output to the OpenBSD sndio sound system
.. note:: Experimental. There are known bugs and issues.
(Note: only supports mono, stereo, 4.0, 5.1 and 7.1 channel
layouts.)
``device=<device>``
sndio device to use (default: ``$AUDIODEVICE``, resp. ``snd0``).
``wasapi``
Audio output to the Windows Audio Session API.
+273
View File
@@ -0,0 +1,273 @@
CHANGES FROM OTHER VERSIONS OF MPLAYER
======================================
**mpv** is based on mplayer2, which in turn is based on the original
MPlayer (also called mplayer, mplayer-svn, mplayer1). Many changes
have been made. Some changes are incompatible, or completely change how the
player behaves.
General Changes for MPlayer to mplayer2
---------------------------------------
* Removal of the internal GUI, MEncoder, OSD menu
* Better pause handling (do not unpause on a command)
* Better MKV support (such as ordered chapters)
* vo_vdpau improvements
* Precise seeking support
* No embedded copy of ffmpeg and other libraries
* Native OpenGL backend for OSX
* General OSX improvements
* Improvements in audio/video sync handling
* Cleaned up terminal output
* Gapless audio support (``--gapless-audio``)
* Improved responsiveness on user input
* Support for modifier keys (alt, shift, ctrl) in input.conf
* OSS4 volume control
* More correct color reproduction (color matrix generation)
* Use libass for subtitle rendering by default (better quality)
* Generally preferring ffmpeg/libav over internal demuxers and decoders
* Improvements when playing multiple files (``--fixed-vo``)
* Screenshot improvements (instant screenshots without 1-frame delay, allow
taking screenshots even with hardware decoding)
* Improved support for PulseAudio
* General code cleanups
* Many more changes
General Changes for mplayer2 to mpv
-----------------------------------
* Removal of lots of unneeded code to encourage developer activity (less
obscure scary zombie code that kills any desire for hacking the codebase)
* Removal of dust and dead bodies (code-wise), such as kernel drivers for
decades old hardware
* Removal of support for dead platforms
* Generally improved MS Windows support (dealing with unicode filenames,
improved ``--vo=direct3d``, improved window handling)
* Better OSD rendering (using libass). This has full unicode support, and
languages like Arabic should be better supported.
* Cleaned up terminal output (nicer status line, less useless noise)
* Support for playing URLs of popular streaming sites directly
(e.g. ``mpv https://www.youtube.com/watch?v=...``)
* Improved OpenGL output (``--vo=opengl-hq``)
* Make ``--softvol`` default (**mpv** is not a mixer control panel)
* Improved support for .cue files
* Screenshot improvements (can save screenshots as JPG or PNG, configurable
filenames, support for taking screenshots with or without subtitles - the
``screenshot`` video filter is not needed anymore, and should not be put
into the mpv config file)
* Removal of teletext support
* Removal of most builtin demuxers, using libavformat instead
* Removal of builtin network support, using libavformat instead (also,
support https via libavformat)
* Replace image VOs (``--vo=jpeg`` etc.) with ``--vo=image``
* Do not lose settings when playing a new file in the same player instance
* New location for config files, new name for the binary.
* Slave mode compatibility broken (see below)
* Encoding functionality (replacement for mencoder, see ``DOCS/encoding.rst``)
* Remove ``--vo=gif89a``, ``--vo=md5sum``, ``--vo=yuv4mpeg``, as encoding can
handle these use cases. For yuv4mpeg, for example, use:
``mpv input.mkv -o output.y4m --no-audio --oautofps --oneverdrop``.
* Image subtitles (DVDs etc.) are rendered in color and use more correct
positioning (color can be disabled with ``--sub-gray``)
* Wayland support
* Support for precise scrolling which scales the parameter of commands. If the
input doesn't support precise scrolling the scale factor stays 1.
* OSX: Cocoa event loop is independent from mplayer's event loop, so user
actions like accessing menus and live resizing do not block the playback.
* OSX: Apple Remote support.
* OSX: Media Keys support.
* Windows: Added WASAPI audio output.
* New OSD bar with chapter marks and not positioned in the middle of the video
(though this can be customized with the ``--osd-bar-align-y`` option).
* Allow customizing whether a key binding for seeking shows the video time, the
OSD bar, or nothing (see section `Input Command Prefixes`_).
* Display list of chapters and audio/subtitle tracks on OSD (see section
`Properties`_).
* ``--keep-open`` option to stop the player from closing the window and
exiting after playback ends.
* Re-enable screensaver while the player is paused.
* Matroska edition switching at runtime.
* Support for libavfilter (for video->video and audio->audio). This allows
using most of FFmpeg's filters, which improve greatly on the old MPlayer
filters in features, performance, and correctness.
* Improved downmixing and output of surround audio. Instead of using hardcoded
pan filters to do remixing, use libavresample. Channel maps are used to
identify the channel layout, so e.g. ``3.0`` and ``2.1`` audio can be
distinguished.
* Allow resuming playback at a later point with ``Shift+q``, also see
``quit_watch_later`` input command.
* Support mapping multiple commands to one key.
* Allow changing/adjusting video filters at runtime. (This is also used to make
the ``D`` key insert vf_yadif if deinterlacing is not supported otherwise.)
* General bug fixes and removal of long-standing issues
* General code cleanups (including refactoring or rewrites of many parts)
* Many more changes
Detailed Listing of User-visible Changes
----------------------------------------
This listing is about changed command line switches, slave commands, and similar
things. Completely removed features are not listed.
Command Line Switches
~~~~~~~~~~~~~~~~~~~~~
* There is a new command line syntax, which is generally preferred over the old
syntax. ``-optname optvalue`` becomes ``--optname=optvalue``.
The old syntax will not be removed in the near future. However, the new
syntax is mentioned in all documentation and so on, so it is a good thing to
know about this change.
(The new syntax was introduced in mplayer2.)
* In general, negating switches like ``-noopt`` now have to be written as
``-no-opt``, or better ``--no-opt``.
* Per-file options are not the default anymore. You can explicitly specify
file-local options. See ``Usage`` section.
* Table of renamed/replaced switches:
=========================== ========================================
Old New
=========================== ========================================
``-no<opt>`` ``--no-<opt>`` (add a dash)
``-nosound`` ``--no-audio``
``-use-filename-title`` ``--title="${filename}"``
``-loop 0`` ``--loop=inf``
``-hardframedrop`` ``--framedrop=hard``
``-osdlevel`` ``--osd-level``
``-delay`` ``--audio-delay``
``-subdelay`` ``--sub-delay``
``-subpos`` ``--sub-pos``
``-forcedsubsonly`` ``--sub-forced-only``
``-benchmark`` ``--untimed`` (no stats)
``-xineramascreen`` ``--screen`` (different values)
``-ss`` ``--start``
``-endpos`` ``--length``
``--cursor-autohide-delay`` ``--cursor-autohide``
``-sub-fuzziness`` ``--autosub-match``
``-subfont`` ``--sub-text-font``
``-font`` ``--osd-font``
``-subfont-*`` ``--sub-text-*``, ``--osd-*``
``-subfont-text-scale`` ``--sub-scale``
``-spugauss`` ``--sub-gauss``
``-vobsub`` ``--sub`` (pass the .idx file)
``-ass-bottom-margin`` ``--vf=sub=bottom:top``
``-vc ffh264vdpau`` (etc.) ``--hwdec=vdpau``
``-ac spdifac3`` ``--ad=spdif:ac3`` (see ``--ad=help``)
``-afm hwac3`` ``--ad=spdif:ac3,spdif:dts``
``-x W``, ``-y H`` ``--geometry=WxH`` + ``--no-keepaspect``
``-xy W`` ``--autofit=W``
``-a52drc level`` ``--ad-lavc-ac3drc=level``
``-dumpstream`` ``--stream-dump=<filename>``
``-capture`` ``--stream-capture=<filename>``
``-stop-xscreensaver`` ``--stop-screensaver``
``-subfile`` ``--sub``
``-lavdopts ...`` ``--vd-lavc-...``
``-lavfdopts`` ``--demuxer-lavf-...``
``-rawaudio ...`` ``--demuxer-rawaudio-...``
``-rawvideo ...`` ``--demuxer-rawvideo-...``
``--mixer`` AO suboptions (``alsa``, ``oss``)
``--mixer-channel`` AO suboptions (``alsa``, ``oss``)
``--ao=alsa:device=hw=0.3`` ``--ao=alsa:device=[hw:0,3]``
=========================== ========================================
.. note::
``-opt val`` becomes ``--opt=val``.
input.conf and Slave Commands
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
* Table of renamed input commands:
This lists only commands that are not always gracefully handled by the
internal legacy translation layer. If an input.conf contains any legacy
commands, they will be displayed with ``-v`` when it is loaded, and show
and the replacement commands.
Properties containing ``_`` to separate words use ``-`` instead.
+--------------------------------+----------------------------------------+
| Old | New |
+================================+========================================+
| ``pt_step 1 [0|1]`` | ``playlist_next [weak|force]`` |
| | (translation layer cannot deal with |
| | whitespace) |
+--------------------------------+----------------------------------------+
| ``pt_step -1 [0|1]`` | ``playlist_prev [weak|force] (same)`` |
+--------------------------------+----------------------------------------+
| ``switch_ratio [<ratio>]`` | ``set aspect <ratio>`` |
| | |
| | ``set aspect 0`` (to reset aspect) |
+--------------------------------+----------------------------------------+
| ``step_property_osd <prop>`` | ``cycle <prop> <step>`` (wraps), |
| ``<step> <dir>`` | ``add <prop> <step>`` (clamps). |
| | ``<dir>`` parameter unsupported. Use |
| | a negative ``<step>`` instead. |
+--------------------------------+----------------------------------------+
| ``step_property <prop>`` | Prefix ``cycle`` or ``add`` with |
| ``<step> <dir>`` | ``no-osd``: ``no-osd cycle <prop>`` |
| | ``<step>`` |
+--------------------------------+----------------------------------------+
| ``osd_show_property_text`` | ``show_text <text>`` |
| ``<text>`` | The property expansion format string |
| | syntax slightly changed. |
+--------------------------------+----------------------------------------+
| ``osd_show_text`` | Now does the same as |
| | ``osd_show_property_text``. Use the |
| | ``raw`` prefix to disable property |
| | expansion. |
+--------------------------------+----------------------------------------+
| ``show_tracks`` | ``show_text ${track-list}`` |
+--------------------------------+----------------------------------------+
| ``show_chapters`` | ``show_text ${chapter-list}`` |
+--------------------------------+----------------------------------------+
| ``af_switch``, ``af_add``, ... | ``af [set|add|...]`` |
+--------------------------------+----------------------------------------+
Other
~~~~~
* The playtree has been removed. **mpv**'s internal playlist is a simple and
flat list now. This simplifies the code and makes **mpv** usage less
confusing.
* Slave mode is broken. This mode is entirely insane in the ``old`` versions of
MPlayer. A proper slave mode application needed tons of code and hacks to get
it right. The main problem is that slave mode is a bad and incomplete
interface, and to get around that, applications parsed output messages
intended for users. It is hard to know which messages exactly are parsed by
slave mode applications. This makes it virtually impossible to improve
terminal output intended for users without possibly breaking something.
This is absolutely insane, and **mpv** will not try to keep slave mode
compatible. If you are a developer of a slave mode application, contact us,
and a new and better protocol can be developed.
Policy for Removed Features
---------------------------
Features are a good thing, because they make users happy. As such, it is
attempted to preserve useful features as far as possible. But if a feature is
likely to be not used by many, and causes problems otherwise, it will be
removed. Developers should not be burdened with fixing or cleaning up code that
has no actual use.
It is always possible to add back removed features. File a feature request if a
feature you relied on has been removed, and you want it back. Though it might be
rejected in the worst case, it is much more likely that it will be either added
back, or that a better solution will be implemented.
Why this Fork?
--------------
* MPlayer wants to maintain old code, even if it is very bad code. It seems
mplayer2 was forked because MPlayer developers refused to get rid of all the
cruft. The mplayer2 and MPlayer codebases also deviated enough to make a
reunification unlikely.
* mplayer2 development is slow, and it is hard to get in changes. Details
withheld as to not turn this into a rant.
* MPlayer rarely merged from mplayer2, and mplayer2 practically stopped
merging from MPlayer (not even code cleanups or new features are merged)
* **mpv** intends to continuously merge from mplayer-svn and mplayer2, while
speeding up development. There is willingness for significant changes, even
if this means breaking compatibility.
@@ -3,7 +3,7 @@ ENCODING
You can encode files from one format/codec to another using this facility.
``--o=<filename>``
``-o <filename>``
Enables encoding mode and specifies the output file name.
``--of=<format>``
@@ -38,7 +38,7 @@ You can encode files from one format/codec to another using this facility.
``--oautofps``
Sets the output format time base to the guessed frame rate of the input
video (simulates MEncoder behavior, useful for AVI; may cause frame drops).
video (simulates mencoder behavior, useful for AVI; may cause frame drops).
Note that not all codecs and not all formats support VFR encoding, and some
which do have bugs when a target bitrate is specified - use ``--ofps`` or
``--oautofps`` to force CFR encoding in these cases.
@@ -74,7 +74,7 @@ You can encode files from one format/codec to another using this facility.
.. admonition:: Example
"``--oac=libmp3lame --oacopts=b=128000``"
selects 128 kbps MP3 encoding.
selects 128kbps MP3 encoding.
Options are managed in lists. There are a few commands to manage the
options list.
@@ -149,7 +149,3 @@ You can encode files from one format/codec to another using this facility.
container formats, e.g. AVI). In this mode, discontinuities are not fixed
and all pts are passed through as-is. Never seek backwards or use multiple
input files in this mode!
``--no-ometadata``
Turns off copying of metadata from input files to output files when
encoding (which is enabled by default).
+547
View File
@@ -0,0 +1,547 @@
INPUT.CONF
==========
The input.conf file consists of a list of key bindings, for example::
s screenshot # take a screenshot with the s key
Each line maps a key to an input command. Keys are specified with their literal
value (upper case if combined with ``Shift``), or a name for special keys. For
example, ``a`` maps to the ``a`` key without shift, and ``A`` maps to ``a``
with shift.
A list of special keys can be obtained with
``mpv --input-keylist``
In general, keys can be combined with ``Shift``, ``Ctrl`` and ``Alt``::
ctrl+q quit
**mpv** can be started in input test mode, which displays key bindings and the
commands they're bound to on the OSD, instead of executing the commands::
mpv --input-test --demuxer=rawvideo --demuxer-rawvideo=w=1280:h=720 /dev/zero
(Commands which normally close the player will not work in this mode, and you
must kill **mpv** externally to make it exit.)
General Input Command Syntax
----------------------------
``[Shift+][Ctrl+][Alt+][Meta+]<key> [{<section>}] [<prefixes>] <command> (<argument>)* [; <command>]``
Newlines always start a new binding. ``#`` starts a comment (outside of quoted
string arguments). To bind commands to the ``#`` key, ``SHARP`` can be used.
``<key>`` is either the literal character the key produces (ASCII or Unicode
character), or a symbol name.
``<section>`` (braced with ``{`` and ``}``) is the input section for this
command.
Arguments are separated by whitespace. This applies even to string arguments.
For this reason, string arguments should be quoted with ``"``. Inside quotes,
C-style escaping can be used.
Optional arguments can be skipped with ``-``.
You can bind multiple commands to one key. For example:
| a show_text "command 1" ; show_text "command 2"
Note that some magic is disabled for keys: seek commands inside lists are not
coalesced (seeking will appear slower), and no check is done for abort commands
(so these commands can't be used to abort playback if the network cache is
stuck).
List of Input Commands
----------------------
``ignore``
Use this to "block" keys that should be unbound, and do nothing. Useful for
disabling default bindings, without disabling all bindings with
``--no-input-default-bindings``.
``seek <seconds> [relative|absolute|absolute-percent|- [default-precise|exact|keyframes]]``
Change the playback position. By default, seeks by a relative amount of
seconds.
The second argument sets the seek mode:
relative (default)
Seek relative to current position (a negative value seeks backwards).
absolute
Seek to a given time.
absolute-percent
Seek to a given percent position.
The third argument defines how exact the seek is:
default-precise (default)
Follow the default behavior as set by ``--hr-seek``, which by default
does imprecise seeks (like ``keyframes``).
exact
Always do exact/hr/precise seeks (slow).
keyframes
Always restart playback at keyframe boundaries (fast).
``frame_step``
Play one frame, then pause.
``frame_back_step``
Go back by one frame, then pause. Note that this can be very slow (it tries
to be precise, not fast), and sometimes fails to behave as expected. How
well this works depends on whether precise seeking works correctly (e.g.
see the ``--hr-seek-demuxer-offset`` option). Video filters or other video
postprocessing that modifies timing of frames (e.g. deinterlacing) should
usually work, but might make backstepping silently behave incorrectly in
corner cases.
This does not work with audio-only playback.
``set <property> "<value>"``
Set the given property to the given value.
``add <property> [<value>]``
Add the given value to the property. On overflow or underflow, clamp the
property to the maximum. If ``<value>`` is omitted, assume ``1``.
``cycle <property> [up|down]``
Cycle the given property. ``up`` and ``down`` set the cycle direction. On
overflow, set the property back to the minimum, on underflow set it to the
maximum. If ``up`` or ``down`` is omitted, assume ``up``.
``speed_mult <value>``
Multiply the ``speed`` property by the given value.
``screenshot [subtitles|video|window|- [single|each-frame]]``
Take a screenshot.
First argument:
<subtitles> (default)
Save the video image, in its original resolution, and with subtitles.
Some video outputs may still include the OSD in the output under certain
circumstances.
<video>
Like ``subtitles``, but typically without OSD or subtitles. The exact
behavior depends on the selected video output.
<window>
Save the contents of the mpv window. Typically scaled, with OSD and
subtitles. The exact behavior depends on the selected video output, and
if no support is available, this will act like ``video``.
Second argument:
<single> (default)
Take a single screenshot.
<each-frame>
Take a screenshot each frame. Issue this command again to stop taking
screenshots.
``screenshot_to_file "<filename>" [subtitles|video|window]``
Take a screenshot and save it to a given file. The format of the file will
be guessed by the extension (and ``--screenshot-format`` is ignored - the
behavior when the extension is missing or unknown is arbitrary).
The second argument is like the first argument to ``screenshot``.
This command tries to never overwrite files. If the file already exists,
it fails.
Like all input command parameters, the filename is subject to property
expansion as described in `Property Expansion`_.
``playlist_next [weak|force]``
Go to the next entry on the playlist.
weak (default)
If the last file on the playlist is currently played, do nothing.
force
Terminate playback if there are no more files on the playlist.
``playlist_prev [weak|force]``
Go to the previous entry on the playlist.
weak (default)
If the first file on the playlist is currently played, do nothing.
force
Terminate playback if the first file is being played.
``loadfile "<file>" [replace|append]``
Load the given file and play it.
Second argument:
<replace> (default)
Stop playback of the current file, and play the new file immediately.
<append>
Append the file to the playlist.
``loadlist "<playlist>" [replace|append]``
Load the given playlist file (like ``--playlist``).
``playlist_clear``
Clear the playlist, except the currently played file.
``playlist_remove <index>``
Remove the playlist entry at the given index. Index values start counting
with 0. You cannot remove the entry for the currently played file.
``playlist_move <index1> <index2>``
Move the playlist entry at index1, so that it takes the place of the
entry index2. (Paradoxically, the moved playlist entry will not have
the index value index2 after moving if index1 was lower than index2,
because index2 refers to the target entry, not the index the entry
will have after moving.)
``run "<command>"``
Run the given command with ``/bin/sh -c``. The string is expanded like in
`Property Expansion`_.
``quit [<code>]``
Exit the player using the given exit code.
``quit_watch_later``
Exit player, and store current playback position. Playing that file later
will seek to the previous position on start.
``sub_add "<file>"``
Load the given subtitle file. It is not selected as current subtitle after
loading.
``sub_remove [<id>]``
Remove the given subtitle track. If the ``id`` argument is missing, remove
the current track. (Works on external subtitle files only.)
``sub_reload [<id>]``
Reload the given subtitle tracks. If the ``id`` argument is missing, remove
the current track. (Works on external subtitle files only.)
This works by unloading and re-adding the subtitle track.
``sub_step <skip>``
Change subtitle timing such, that the subtitle event after the next
``<skip>`` subtitle events is displayed. ``<skip>`` can be negative to step
backwards.
``osd [<level>]``
Toggle OSD level. If ``<level>`` is specified, set the OSD mode
(see ``--osd-level`` for valid values).
``print_text "<string>"``
Print text to stdout. The string can contain properties (see
`Property Expansion`_).
``show_text "<string>" [<duration>|- [<level>]]``
Show text on the OSD. The string can contain properties, which are expanded
as described in `Property Expansion`_. This can be used to show playback
time, filename, and so on.
<duration>
The time in ms to show the message for. By default, it uses the same
value as ``--osd-duration``.
<level>
The minimum OSD level to show the text at (see ``--osd-level``).
``show_progress``
Show the progress bar, the elapsed time and the total duration of the file
on the OSD.
Input Commands that are Possibly Subject to Change
--------------------------------------------------
``af set|add|toggle|del|clr "filter1=params,filter2,..."``
Change audio filter chain. See ``vf`` command.
``vf set|add|toggle|del|clr "filter1=params,filter2,..."``
Change video filter chain.
The first argument decides what happens:
set
Overwrite the previous filter chain with the new one.
add
Append the new filter chain to the previous one.
toggle
Check if the given filter (with the exact parameters) is already
in the video chain. If yes, remove the filter. If no, add the filter.
(If several filters are passed to the command, this is done for
each filter.)
del
Remove the given filters from the video chain. Unlike in the other
cases, the second parameter is a comma separated list of filter names
or integer indexes. ``0`` would denote the first filter. Negative
indexes start from the last filter, and ``-1`` denotes the last
filter.
clr
Remove all filters. Note that like the other sub-commands, this does
not control automatically inserted filters.
You can assign labels to filter by prefixing them with ``@name:`` (where
``name`` is a user-chosen arbitrary identifier). Labels can be used to
refer to filters by name in all of the filter chain modification commands.
For ``add``, using an already used label will replace the existing filter.
The ``vf`` command shows the list of requested filters on the OSD after
changing the filter chain. This is roughly equivalent to
``show_text ${vf}``. Note that auto-inserted filters for format conversion
are not shown on the list, only what was requested by the user.
.. admonition:: Example for input.conf
- ``a vf set flip`` turn video upside-down on the ``a`` key
- ``b vf set ""`` remove all video filters on ``b``
- ``c vf toggle lavfi=gradfun`` toggle debanding on ``c``
``enable_section "<section>" [default|exclusive]``
Enable all key bindings in the named input section.
The enabled input sections form a stack. Bindings in sections on the top of
the stack are preferred to lower sections. This command puts the section
on top of the stack. If the section was already on the stack, it is
implicitly removed beforehand. (A section cannot be on the stack more than
once.)
If ``exclusive`` is specified as second argument, all sections below the
newly enabled section are disabled. They will be re-enabled as soon as
all exclusive sections above them are removed.
``disable_section "<section>"``
Disable the named input section. Undoes ``enable_section``.
Undocumented commands: ``tv_start_scan``, ``tv_step_channel``, ``tv_step_norm``,
``tv_step_chanlist``, ``tv_set_channel``, ``tv_last_channel``, ``tv_set_freq``,
``tv_step_freq``, ``tv_set_norm``, ``dvb_set_channel``, ``radio_step_channel``,
``radio_set_channel``, ``radio_set_freq``, ``radio_step_freq`` (all of these
should be replaced by properties), ``stop`` (questionable use), ``get_property``
(?), ``vo_cmdline`` (experimental).
Input Command Prefixes
----------------------
``osd-auto`` (default)
Use the default behavior for this command.
``no-osd``
Do not use any OSD for this command.
``osd-bar``
If possible, show a bar with this command. Seek commands will show the
progress bar, property changing commands may show the newly set value.
``osd-msg``
If possible, show an OSD message with this command. Seek command show
the current playback time, property changing commands show the newly set
value as text.
``osd-msg-bar``
Combine osd-bar and osd-msg.
``raw``
Do not expand properties in string arguments. (Like ``"${property-name}"``.)
``expand-properties`` (default)
All string arguments are expanded as described in `Property Expansion`_.
All of the osd prefixes are still overridden by the global ``--osd-level``
settings.
Undocumented prefixes: ``pausing``, ``pausing_keep``, ``pausing_toggle``,
``pausing_keep_force``. (Should these be made official?)
Input Sections
--------------
Input sections group a set of bindings, and enable or disable them at once.
In ``input.conf``, each key binding is assigned to an input section, rather
than actually having explicit text sections.
Also see ``enable_section`` and ``disable_section`` commands.
Predefined bindings:
``default``
Bindings without input section are implicitly assigned to this section. It
is enabled by default during normal playback.
``encode``
Section which is active in encoding mode. It is enabled exclusively, so
that bindings in the ``default`` sections are ignored.
Properties
----------
Properties are used to set mpv options during runtime, or to query arbitrary
information. They can be manipulated with the ``set``/``add``/``cycle``
commands, and retrieved with ``show_text``, or anything else that uses property
expansion. (See `Property Expansion`_.)
The ``W`` column indicates whether the property is generally writable. If an
option is referenced, the property should take/return exactly the same values
as the option.
=============================== = ==================================================
Name W Comment
=============================== = ==================================================
``osd-level`` x see ``--osd-level``
``osd-scale`` x osd font size multiplicator, see ``--osd-scale``
``loop`` x see ``--loop``
``speed`` x see ``--speed``
``filename`` currently played file (path stripped)
``path`` currently played file (full path)
``media-title`` filename or libquvi ``QUVIPROP_PAGETITLE``
``demuxer``
``stream-path`` filename (full path) of stream layer filename
``stream-pos`` x byte position in source stream
``stream-start`` start byte offset in source stream
``stream-end`` end position in bytes in source stream
``stream-length`` length in bytes (``${stream-end} - ${stream-start}``)
``stream-time-pos`` x time position in source stream (also see ``time-pos``)
``length`` length of the current file in seconds
``avsync`` last A/V synchronization difference
``percent-pos`` x position in current file (0-100)
``ratio-pos`` x position in current file (0.0-1.0)
``time-pos`` x position in current file in seconds
``time-remaining`` estimated remaining length of the file in seconds
``chapter`` x current chapter number
``edition`` x current MKV edition number
``titles`` number of DVD titles
``chapters`` number of chapters
``editions`` number of MKV editions
``angle`` x current DVD angle
``metadata`` metadata key/value pairs
``metadata/<key>`` value of metadata entry ``<key>``
``pause`` x pause status (bool)
``cache`` network cache fill state (0-100)
``pts-association-mode`` x see ``--pts-association-mode``
``hr-seek`` x see ``--hr-seek``
``volume`` x current volume (0-100)
``mute`` x current mute status (bool)
``audio-delay`` x see ``--audio-delay``
``audio-format`` audio format (string)
``audio-codec`` audio codec selected for decoding
``audio-bitrate`` audio bitrate
``samplerate`` audio samplerate
``channels`` number of audio channels
``aid`` x current audio track (similar to ``--aid``)
``audio`` x alias for ``aid``
``balance`` x audio channel balance
``fullscreen`` x see ``--fullscreen``
``deinterlace`` x deinterlacing, if available (bool)
``colormatrix`` x see ``--colormatrix``
``colormatrix-input-range`` x see ``--colormatrix-input-range``
``colormatrix-output-range`` x see ``--colormatrix-output-range``
``ontop`` x see ``--ontop``
``border`` x see ``--border``
``framedrop`` x see ``--framedrop``
``gamma`` x see ``--gamma``
``brightness`` x see ``--brightness``
``contrast`` x see ``--contrast``
``saturation`` x see ``--saturation``
``hue`` x see ``--hue``
``panscan`` x see ``--panscan``
``video-format`` video format (string)
``video-codec`` video codec selected for decoding
``video-bitrate`` video bitrate
``width`` video width (container or decoded size)
``height`` video height
``fps`` container FPS (may contain bogus values)
``dwidth`` video width (after filters and aspect scaling)
``dheight`` video height
``aspect`` x video aspect
``vid`` x current video track (similar to ``--vid``)
``video`` x alias for ``vid``
``program`` x switch TS program (write-only)
``sid`` x current subtitle track (similar to ``--sid``)
``sub`` x alias for ``sid``
``sub-delay`` x see ``--sub-delay``
``sub-pos`` x see ``--sub-pos``
``sub-visibility`` x whether current subtitle is rendered
``sub-forced-only`` x see ``--sub-forced-only``
``sub-scale`` x subtitle font size multiplicator
``ass-use-margins`` x see ``--ass-use-margins``
``ass-vsfilter-aspect-compat`` x see ``--ass-vsfilter-aspect-compat``
``ass-style-override`` x see ``--ass-style-override``
``stream-capture`` x a filename, see ``--capture``
``tv-brightness`` x
``tv-contrast`` x
``tv-saturation`` x
``tv-hue`` x
``playlist-pos`` current position on playlist
``playlist-count`` number of total playlist entries
``playlist`` playlist, current entry marked
``track-list`` list of audio/video/sub tracks, cur. entr. marked
``chapter-list`` list of chapters, current entry marked
``quvi-format`` x see ``--quvi-format``
``af`` x see ``--af``
``vf`` x see ``--vf``
``options/name`` read-only access to value of option ``--name``
=============================== = ==================================================
Property Expansion
------------------
All string arguments to input commands as well as certain options (like
``--playing-msg``) are subject to property expansion.
.. admonition:: Example for input.conf
``i show_text "Filename: ${filename}"``
shows the filename of the current file when pressing the ``i`` key
Within ``input.conf``, property expansion can be inhibited by putting the
``raw`` prefix in front of commands.
The following expansions are supported:
``${NAME}``
Expands to the value of the property ``NAME``. If retrieving the property
fails, expand to an error string. (Use ``${NAME:}`` with a trailing
``:`` to expand to an empty string instead.)
If ``NAME`` is prefixed with ``=``, expand to the raw value of the property
(see below).
``${NAME:STR}``
Expands to the value of the property ``NAME``, or ``STR`` if the
property cannot be retrieved. ``STR`` is expanded recursively.
``${!NAME:STR}``
Expands to ``STR`` (recursively) if the property ``NAME`` cannot be
retrieved.
``${?NAME:STR}``
Expands to ``STR`` (recursively) if the property ``NAME`` is available.
``$$``
Expands to ``$``.
``$}``
Expands to ``}``. (To produce this character inside recursive
expansion.)
``$>``
Disable property expansion and special handling of ``$`` for the rest
of the string.
In places where property expansion is allowed, C-style escapes are often
accepted as well. Example:
- ``\n`` becomes a newline character
- ``\\`` expands to ``\``
Raw and Formatted Properties
----------------------------
Normally, properties are formatted as human-readable text, meant to be
displayed on OSD or on the terminal. It is possible to retrieve an unformatted
(raw) value from a property by prefixing its name with ``=``. These raw values
can be parsed by other programs and follow the same conventions as the options
associated with the properties.
.. admonition:: Examples
- ``${time-pos}`` expands to ``00:14:23`` (if playback position is at 14
minutes 23 seconds)
- ``${=time-pos}`` expands to ``863.4`` (same time, plus 400 milliseconds -
milliseconds are normally not shown in the formatted case)
Sometimes, the difference in amount of information carried by raw and formatted
property values can be rather big. In some cases, raw values have more
information, like higher precision than seconds with ``time-pos``. Sometimes
it is the other way around, e.g. ``aid`` shows track title and language in the
formatted case, but only the track number if it is raw.
+115 -170
View File
@@ -1,11 +1,20 @@
..
Hint: To generate a nicely formatted XeLaTeX document from this file, use
rst2xetex --use-latex-docinfo --use-latex-toc --hyperlink-color=MidnightBlue --latex-preamble="\usepackage[usenames,dvipsnames]{xcolor}\usepackage{fullpage}\setmainfont{Droid Sans}\setsansfont{Linux Biolinum O}\setmonofont[HyphenChar=None]{DejaVu Sans Mono}" mpv.rst
You might want to put the .. contents:: directive below to generate a TOC.
mpv
###
##############
a media player
a movie player
##############
:Copyright: GPLv2+
:Author: wm4
:Date: 2013-07-08
:Copyright: GPLv3
:Version: git
:Manual section: 1
:Manual group: multimedia
@@ -15,11 +24,23 @@ SYNOPSIS
| **mpv** [options] [file|URL|-]
| **mpv** [options] --playlist=PLAYLIST
| **mpv** [options] files
| **mpv** [options] {group of files and options}
| **mpv** [bd]://[title][/device] [options]
| **mpv** dvd://[title|[start\_title]-end\_title][/device] [options]
| **mpv** \vcd://[/device]
| **mpv** \tv://[channel][/input_id] [options]
| **mpv** radio://[channel|frequency][/capture] [options]
| **mpv** \pvr:// [options]
| **mpv** \dvb://[card\_number@]channel [options]
| **mpv** \mf://[filemask|\@listfile] [-mf options] [options]
| **mpv** cdda://track[-endtrack][:speed][/device] [options]
| **mpv** [file|mms[t]|http|httpproxy|rt[s]p|ftp|udp|smb]:// [user:pass\@]URL[:port] [options]
DESCRIPTION
===========
**mpv** is a media player based on MPlayer and mplayer2. It supports a wide variety of video
**mpv** is a movie player based on MPlayer and mplayer2. It supports a wide variety of video
file formats, audio and video codecs, and subtitle types. Special input URL
types are available to read input from a variety of sources other than disk
files. Depending on platform, a variety of different video and audio output
@@ -40,13 +61,16 @@ Keyboard Control
----------------
LEFT and RIGHT
Seek backward/forward 5 seconds. Shift+arrow does a 1 second exact seek
Seek backward/forward 10 seconds. Shift+arrow does a 1 second exact seek
(see ``--hr-seek``).
UP and DOWN
Seek forward/backward 1 minute. Shift+arrow does a 5 second exact seek (see
``--hr-seek``).
PGUP and PGDWN
Seek forward/backward 10 minutes.
[ and ]
Decrease/increase current playback speed by 10%.
@@ -66,13 +90,9 @@ p / SPACE
Pause (pressing again unpauses).
\.
Step forward. Pressing once will pause, every consecutive press will
Step forward. Pressing once will pause movie, every consecutive press will
play one frame and then go into pause mode again.
,
Step backward. Pressing once will pause, every consecutive press will
play one frame in reverse and then go into pause mode again.
q / ESC
Stop playing and quit.
@@ -80,12 +100,21 @@ Q
Like ``q``, but store the current playback position. Playing the same file
later will resume at the old playback position if possible.
U
Stop playing (and quit if ``--idle`` is not used).
\+ and -
Adjust audio delay by +/- 0.1 seconds.
/ and *
Decrease/increase volume.
9 and 0
Decrease/increase volume.
( and )
Adjust audio balance in favor of left/right channel.
m
Mute sound.
@@ -95,6 +124,9 @@ m
\#
Cycle through the available audio tracks.
TAB (MPEG-TS and libavformat only)
Cycle through the available programs.
f
Toggle fullscreen (see also ``--fs``).
@@ -104,10 +136,7 @@ T
w and e
Decrease/increase pan-and-scan range.
o (also P)
Show progression bar, elapsed time and total duration on the OSD.
O
o
Toggle OSD states: none / seek / seek + timer / seek + timer + total time.
d
@@ -120,6 +149,9 @@ v
j and J
Cycle through the available subtitles.
F
Toggle displaying "forced subtitles".
x and z
Adjust subtitle delay by +/- 0.1 seconds.
@@ -140,14 +172,21 @@ S
I
Show filename on the OSD.
PGUP and PGDWN
Seek to the beginning of the previous/next chapter. In most cases,
"previous" will actually go to the beginning of the current chapter; see
``--chapter-seek-threshold``.
P
Show progression bar, elapsed time and total duration on the OSD.
D
! and @
Seek to the beginning of the previous/next chapter.
D (``--vo=vdpau``, ``--vf=yadif`` only)
Activate/deactivate deinterlacer.
A
Cycle through the available DVD angles.
c
Change YUV colorspace.
(The following keys are valid only when using a video output that supports the
corresponding adjustment, or the software equalizer (``--vf=eq``).)
@@ -163,24 +202,23 @@ corresponding adjustment, or the software equalizer (``--vf=eq``).)
7 and 8
Adjust saturation.
(The following keys are valid only on OS X.)
(The following keys are valid only when using the corevideo video output
driver.)
command + 0
Resize video window to half its original size.
(On other platforms, you can bind keys to change the ``window-scale``
property.)
Resize movie window to half its original size.
command + 1
Resize video window to its original size.
Resize movie window to its original size.
command + 2
Resize video window to double its original size.
Resize movie window to double its original size.
command + f
Toggle fullscreen (see also ``--fs``).
command + [ and command + ]
Set video window alpha.
Set movie window alpha.
(The following keys are valid if you have a keyboard with multimedia keys.)
@@ -194,11 +232,17 @@ PREVIOUS and NEXT
Seek backward/forward 1 minute.
(The following keys are only valid if you compiled with TV or DVB input
support.)
support and will take precedence over the keys defined above.)
h and k
Select previous/next channel.
n
Change norm.
u
Change channel list.
Mouse Control
-------------
@@ -208,6 +252,24 @@ button 3 and button 4
button 5 and button 6
Decrease/increase volume.
Joystick Control
----------------
left and right
Seek backward/forward 10 seconds.
up and down
Seek forward/backward 1 minute.
button 1
Pause.
button 2
Toggle OSD states: none / seek / seek + timer / seek + timer + total time.
button 3 and button 4
Decrease/increase volume.
USAGE
=====
@@ -237,28 +299,6 @@ It has the following format::
``mpv --ao=pcm:file=%`expr length "$NAME"`%"$NAME" test.avi``
Paths
-----
Some care must be taken when passing arbitrary paths and filenames to mpv. For
example, paths starting with ``-`` will be interpreted as options. Likewise,
if a path contains the sequence ``://``, the string before that might be
interpreted as protocol prefix, even though ``://`` can be part of a legal
UNIX path. To avoid problems with arbitrary paths, you should be sure that
absolute paths passed to mpv start with ``/``, and relative paths with ``./``.
The name ``-`` itself is interpreted as stdin, and will cause mpv to disable
console controls. (Which makes it suitable for playing data piped to stdin.)
For paths passed to suboptions, the situation is further complicated by the
need to escape special characters. To work this around, the path can be
additionally wrapped in the ``%n%string_of_length_n`` syntax (see above).
Some mpv options interpret paths starting with ``~``. Currently, the prefix
``~~/`` expands to the mpv configuration directory (usually ``~/.mpv/``).
``~/`` expands to the user's home directory. (The trailing ``/`` is always
required.)
Per-File Options
----------------
@@ -274,8 +314,6 @@ file1.mkv ``--a --b --c``
file2.mkv ``--a --b --c``
=============== ===========================
(This is different from MPlayer and mplayer2.)
Also, if any option is changed at runtime (via input commands), they are not
reset when a new file is played.
@@ -308,7 +346,7 @@ Location and Syntax
You can put all of the options in configuration files which will be read every
time mpv is run. The system-wide configuration file 'mpv.conf' is in your
configuration directory (e.g. ``/etc/mpv`` or ``/usr/local/etc/mpv``), the
user-specific one is ``~/.config/mpv/mpv.conf``.
user-specific one is ``~/.mpv/config``.
User-specific options override system-wide options and options given on the
command line override either. The syntax of the configuration files is
``option=<value>``; everything after a *#* is considered a comment. Options
@@ -343,8 +381,8 @@ File-specific Configuration Files
---------------------------------
You can also write file-specific configuration files. If you wish to have a
configuration file for a file called 'video.avi', create a file named
'video.avi.conf' with the file-specific options in it and put it in
configuration file for a file called 'movie.avi', create a file named
'movie.avi.conf' with the file-specific options in it and put it in
``~/.mpv/``. You can also put the configuration file in the same directory
as the file to be played, as long as you give the ``--use-filedir-conf``
option (either on the command line or in your global config file). If a
@@ -370,7 +408,7 @@ profile name ``default`` to continue with normal options.
::
[vo.vdpau]
# Use hardware decoding (might break playback of some h264 files)
# Use hardware decoding (this might break playback of some h264 files)
hwdec=vdpau
[protocol.dvd]
@@ -380,7 +418,7 @@ profile name ``default`` to continue with normal options.
[extension.flv]
profile-desc="profile for .flv files"
vf=flip
flip=yes
[ao.alsa]
device=spdif
@@ -404,84 +442,8 @@ taking screenshots is handled by the VOs, and adding the screenshot filter will
break hardware decoding. (The filter may still be useful for taking screenshots
at a certain point within the video chain when using multiple video filters.)
PROTOCOLS
=========
``http://...``, ``https://``, ...
Many network protocols are supported, but the protocol prefix must always
be specified. mpv will never attempt to guess whether a filename is
actually a network address. A protocol prefix is always required.
``-``
Play data from stdin.
``smb://PATH``
Play a path from Samba share.
``bd://[title][/device]`` ``--bluray-device=PATH``
Play a Blu-Ray disc. Currently, this does not accept ISO files. Instead,
you must mount the ISO file as filesystem, and point ``--bluray-device``
to the mounted directly.
``bdnav://[title][/device]``
Play a Blu-Ray disc, with navigation features enabled. This feature is
permanently experimental.
``dvd://[title|[starttitle]-endtitle][/device]`` ``--dvd-device=PATH``
Play a DVD. If you want dvdnav menus, use ``dvd://menu``. If no title
is given, the longest title is auto-selected.
``dvdnav://`` is an old alias for ``dvd://`` and does exactly the same
thing.
``dvdread://...:``
Play a DVD using the old libdvdread code. This is what MPlayer and older
mpv versions use for ``dvd://``. Use is discouraged. It's provided only
for compatibility and for transition.
``tv://[channel][/input_id]`` ``--tv-...``
Analogue TV via V4L. Also useful for webcams. (Linux only.)
``pvr://`` ``--pvr-...``
PVR. (Linux only.)
``dvb://[cardnumber@]channel`` ``--dvbin-...``
Digital TV via DVB. (Linux only.)
``mf://[filemask|@listfile]`` ``--mf-...``
Play a series of images as video.
``cdda://track[-endtrack][:speed][/device]`` ``--cdrom-device=PATH`` ``--cdda-...``
Play CD.
``lavf://...``
Access any FFmpeg/Libav libavformat protocol. Basically, this passed the
string after the ``//`` directly to libavformat.
``av://type:options``
This is intended for using libavdevice inputs. ``type`` is the libavdevice
demuxer name, and ``options`` is the (pseudo-)filename passed to the
demuxer.
For example, ``mpv av://lavfi:mandelbrot`` makes use of the libavfilter
wrapper included in libavdevice, and will use the ``mandelbrot`` source
filter to generate input data.
``avdevice://`` is an alias.
``file://PATH``
A local path as URL. Might be useful in some special use-cases. Note that
``PATH`` itself should start with a third ``/`` to make the path an
absolute path.
``edl://[edl specification as in edl-mpv.rst]``
Stitch together parts of multiple files and play them.
``null://``
Simulate an empty file.
``memory://data``
Use the ``data`` part as source data.
OPTIONS
=======
.. include:: options.rst
@@ -497,10 +459,6 @@ PROTOCOLS
.. include:: input.rst
.. include:: osc.rst
.. include:: lua.rst
.. include:: changes.rst
ENVIRONMENT VARIABLES
@@ -509,28 +467,23 @@ ENVIRONMENT VARIABLES
There are a number of environment variables that can be used to control the
behavior of mpv.
``HOME``, ``XDG_CONFIG_HOME``
Used to determine mpv config directory. If ``XDG_CONFIG_HOME`` is not set,
``$HOME/.config/mpv`` is used.
``$HOME/.mpv`` is always added to the list of config search paths with a
lower priority.
``XDG_CONFIG_DIRS``
If set, XDG-style system configuration directories are used. Otherwise,
the UNIX convention (``PREFIX/etc/mpv/``) is used.
``HOME``
Used to determine mpv config directory: ``$HOME/.mpv``
``TERM``
Used to determine terminal type.
``MPV_HOME``
Directory where mpv looks for user settings. Overrides ``HOME``, and mpv
will try to load the config file as ``$MPV_HOME/mpv.conf``.
will try to load the config file as ``$MPV_HOME/config``.
``MPV_VERBOSE`` (see also ``-v`` and ``--msg-level``)
``MPV_LOCALEDIR``
Directory where mpv looks for gettext translation files (if enabled).
``MPV_VERBOSE`` (see also ``-v`` and ``--msglevel``)
Set the initial verbosity level across all message modules (default: 0).
This is an integer, and the resulting verbosity corresponds to the number
of ``--v`` options passed to the command line.
The resulting verbosity corresponds to that of ``--msglevel=5`` plus the
value of ``MPV_VERBOSE``.
``MPV_LEAK_REPORT``
If set to ``1``, enable internal talloc leak reporting. Note that this can
@@ -545,7 +498,7 @@ behavior of mpv.
FFmpeg/Libav:
This library accesses various environment variables. However, they are not
centrally documented, and documenting them is not our job. Therefore, this
centrally documented, and documenting them is not our job. Therefore, the
list is incomplete.
Notable environment variables:
@@ -638,31 +591,23 @@ FILES
``/usr/local/etc/mpv/mpv.conf``
mpv system-wide settings (depends on ``--prefix`` passed to configure)
``~/.config/mpv/mpv.conf``
``~/.mpv/config``
mpv user settings
``~/.config/mpv/input.conf``
key bindings (see `INPUT.CONF`_ section)
``~/.mpv/input.conf``
input bindings (see ``--input-keylist`` for the full list)
``~/.config/mpv/lua/``
All files in this directly are loaded as if they were passed to the
``--lua`` option. They are loaded in alphabetical order, and sub-directories
and files with no ``.lua`` extension are ignored. The ``--load-scripts=no``
option disables loading these files.
Note that the environment variables ``$XDG_CONFIG_HOME`` and ``$MPV_HOME`` can
override the standard directory ``~/.config/mpv/``.
Also, the old config location at ``~/.mpv/`` is still read, and if the XDG
variant does not exist, will still be preferred.
EXAMPLES OF MPV USAGE
=====================
Blu-ray playback:
Quickstart Blu-ray playing:
- ``mpv bd:////path/to/disc``
- ``mpv bd:// --bluray-device=/path/to/disc``
Quickstart DVD playing:
``mpv dvd://1``
Play in Japanese with English subtitles:
``mpv dvd://1 --alang=ja --slang=en``
@@ -672,8 +617,8 @@ Play only chapters 5, 6, 7:
Play only titles 5, 6, 7:
``mpv dvd://5-7``
Play a multi-angle DVD:
``mpv dvd://1 --dvd-angle=2``
Play a multiangle DVD:
``mpv dvd://1 --dvdangle=2``
Play from a different DVD device:
``mpv dvd://1 --dvd-device=/dev/dvd2``
@@ -698,6 +643,6 @@ mpv is a MPlayer fork based on mplayer2, which in turn is a fork of MPlayer.
MPlayer was initially written by Arpad Gereoffy. See the ``AUTHORS`` file for
a list of some of the many other contributors.
MPlayer is (C) 2000-2013 The MPlayer Team
MPlayer is (C) 2000-2012 The MPlayer Team
This man page was written mainly by Gabucino, Jonas Jermann and Diego Biurrun.
File diff suppressed because it is too large. Load diff
+123 -237
View File
@@ -7,11 +7,10 @@ syntax is:
``--vf=<filter1[=parameter1:parameter2:...],filter2,...>``
Setup a chain of video filters.
You can also set defaults for each filter. The defaults are applied before the
normal filter parameters.
``--vf-defaults=<filter1[=parameter1:parameter2:...],filter2,...>``
Set defaults for each filter.
Many parameters are optional and set to default values if omitted. To
explicitly use a default value, set a parameter to '-1'. Parameters ``w:h``
means width x height in pixels, ``x:y`` means x;y position counted from the
upper left corner of the bigger image.
.. note::
@@ -47,7 +46,7 @@ Available filters are:
``crop[=w:h:x:y]``
Crops the given part of the image and discards the rest. Useful to remove
black bands from widescreen videos.
black bands from widescreen movies.
``<w>,<h>``
Cropped width and height, defaults to original width and height.
@@ -55,7 +54,7 @@ Available filters are:
Position of the cropped picture, defaults to center.
``expand[=w:h:x:y:aspect:round]``
Expands (not scales) video resolution to the given value and places the
Expands (not scales) movie resolution to the given value and places the
unscaled original at coordinates x, y.
``<w>,<h>``
@@ -75,7 +74,7 @@ Available filters are:
.. admonition:: Example
``expand=800::::4/3``
``expand=800:::::4/3``
Expands to 800x600, unless the source is higher resolution, in
which case it expands to fill a 4/3 aspect.
@@ -83,19 +82,24 @@ Available filters are:
Rounds up to make both width and height divisible by <r> (default: 1).
``flip``
Flips the image upside down.
Flips the image upside down. See also ``--flip``.
``mirror``
Mirrors the image on the Y axis.
``rotate[=0|90|180|270]``
Rotates the image by a multiple of 90 degrees clock-wise.
``rotate[=<0-7>]``
Rotates the image by 90 degrees and optionally flips it. For values
between 4-7 rotation is only done if the movie geometry is portrait and
not landscape.
``scale[=w:h:param:param2:chr-drop:noup:arnd``
:0: Rotate by 90 degrees clockwise and flip (default).
:1: Rotate by 90 degrees clockwise.
:2: Rotate by 90 degrees counterclockwise.
:3: Rotate by 90 degrees counterclockwise and flip.
``scale[=w:h[:interlaced[:chr_drop[:par[:par2[:noup[:arnd]]]]]]]``
Scales the image with the software scaler (slow) and performs a YUV<->RGB
color space conversion (see also ``--sws``).
All parameters are optional.
colorspace conversion (see also ``--sws``).
``<w>,<h>``
scaled width/height (default: original width/height)
@@ -109,7 +113,15 @@ Available filters are:
:-(n+8): Like -n above, but rounding the dimension to the closest
multiple of 16.
``<param>[:<param2>]`` (see also ``--sws``)
``<chr_drop>``
chroma skipping
:0: Use all available input lines for chroma.
:1: Use only every 2. input line for chroma.
:2: Use only every 4. input line for chroma.
:3: Use only every 8. input line for chroma.
``<par>[:<par2>]`` (see also ``--sws``)
Set some scaling parameters depending on the type of scaler selected
with ``--sws``::
@@ -120,17 +132,9 @@ Available filters are:
0.33:0.33 Mitchell-Netravali spline
1.00:0.00 cubic B-spline
--sws=7 (Gaussian): sharpness (0 (soft) - 100 (sharp))
--sws=7 (gaussian): sharpness (0 (soft) - 100 (sharp))
--sws=9 (Lanczos): filter length (1-10)
``<chr-drop>``
chroma skipping
:0: Use all available input lines for chroma (default).
:1: Use only every 2. input line for chroma.
:2: Use only every 4. input line for chroma.
:3: Use only every 8. input line for chroma.
--sws=9 (lanczos): filter length (1-10)
``<noup>``
Disallow upscaling past the original dimensions.
@@ -146,7 +150,7 @@ Available filters are:
:0: Disable accurate rounding (default).
:1: Enable accurate rounding.
``dsize[=w:h:aspect-method:r:aspect]``
``dsize[=aspect|w:h:aspect-method:r]``
Changes the intended display size/aspect at an arbitrary point in the
filter chain. Aspect can be given as a fraction (4/3) or floating point
number (1.33). Alternatively, you may specify the exact display width and
@@ -195,9 +199,6 @@ Available filters are:
Rounds up to make both width and height divisible by ``<r>``
(default: 1).
``<aspect>``
Force an aspect ratio.
``format[=fmt[:outfmt]]``
Restricts the color space for the next filter without doing any conversion.
Use together with the scale filter for a real conversion.
@@ -222,8 +223,8 @@ Available filters are:
====================== =====================
``noformat[=fmt]``
Restricts the color space for the next filter without doing any conversion.
Unlike the format filter, this will allow any color space except the one
Restricts the colorspace for the next filter without doing any conversion.
Unlike the format filter, this will allow any colorspace except the one
you specify.
.. note:: For a list of available formats, see ``noformat=fmt=help``.
@@ -231,8 +232,8 @@ Available filters are:
``<fmt>``
Format name, e.g. rgb15, bgr24, 420p, etc. (default: 420p).
``pp[=[filter1[:option1[:option2...]]/[-]filter2...]]``
Enables the specified chain of post-processing subfilters. Subfilters must
``pp[=filter1[:option1[:option2...]]/[-]filter2...]``
Enables the specified chain of postprocessing subfilters. Subfilters must
be separated by '/' and can be disabled by prepending a '-'. Each
subfilter and some options have a short and a long name that can be used
interchangeably, i.e. ``dr``/``dering`` are the same. All subfilters share
@@ -249,12 +250,7 @@ Available filters are:
.. note::
``--vf=pp:help`` shows a list of available subfilters.
.. note::
Unlike in MPlayer or in earlier versions, you must quote the pp string
if it contains ``:`` characters, e.g. ``'--vf=pp=[...]'``.
``--pphelp`` shows a list of available subfilters.
Available subfilters are:
@@ -336,7 +332,7 @@ Available filters are:
filtering every second line with a (-1 4 2 4 -1) filter.
``l5/lowpass5``
Vertically applied FIR low-pass deinterlacing filter that deinterlaces
Vertically applied FIR lowpass deinterlacing filter that deinterlaces
the given block by filtering all lines with a (-1 2 6 2 -1) filter.
``fq/forceQuant[:quantizer]``
@@ -356,7 +352,7 @@ Available filters are:
.. note::
This filter is only available if FFmpeg/Libav has been compiled
This filter is only available if FFmpeg/libav has been compiled
with libpostproc enabled.
.. admonition:: Examples
@@ -368,14 +364,14 @@ Available filters are:
``--vf=pp=de/-al``
default filters without brightness/contrast correction
``--vf=pp=[default/tmpnoise:1:2:3]``
``--vf=pp=default/tmpnoise:1:2:3``
Enable default filters & temporal denoiser.
``--vf=pp=[hb:y/vb:a]``
``--vf=pp=hb:y/vb:a``
Horizontal deblocking on luminance only, and switch vertical
deblocking on or off automatically depending on available CPU time.
``lavfi=graph[:sws-flags[:o=opts]]``
``lavfi=graph[:sws_flags[:o=opts]]``
Filter video using FFmpeg's libavfilter.
``<graph>``
@@ -409,7 +405,7 @@ Available filters are:
``'--vf=lavfi=graph="gradfun=radius=30:strength=20,vflip"'``
Same as before, but uses named parameters for everything.
``<sws-flags>``
``<sws_flags>``
If libavfilter inserts filters for pixel format conversion, this
option gives the flags which should be passed to libswscale. This
option is numeric and takes a bit-wise combination of ``SWS_`` flags.
@@ -424,30 +420,21 @@ Available filters are:
``'--vf=lavfi=yadif:o="threads=2,thread_type=slice"'``
forces a specific threading configuration.
``noise[=<strength>[:averaged][:pattern][:temporal][:uniform][:hq]``
``noise[=luma[u][t|a][h][p]:chroma[u][t|a][h][p]]``
Adds noise.
``strength``
Set the noise for all components. If you want different strength
values for luma and chroma, use libavfilter's noise filter directly
(using ``--vf=lavfi=[noise=...]``), or tell the libavfilter developers
to stop being stupid.
:<0-100>: luma noise
:<0-100>: chroma noise
:u: uniform noise (gaussian otherwise)
:t: temporal noise (noise pattern changes between frames)
:a: averaged temporal noise (smoother, but a lot slower)
:h: high quality (slightly better looking, slightly slower)
:p: mix random noise with a (semi)regular pattern
``averaged``
averaged temporal noise (smoother, but a lot slower)
.. note::
``pattern``
mix random noise with a (semi)regular pattern
``temporal``
temporal noise (noise pattern changes between frames)
``uniform``
uniform noise (Gaussian otherwise)
``hq``
high quality (slightly better looking, slightly slower) - not available
when using libavfilter
Deprecated. Use libavfilter's ``noise`` filter through ``--vf=lavfi``
instead.
``hqdn3d[=luma_spatial:chroma_spatial:luma_tmp:chroma_tmp]``
This filter aims to reduce image noise producing smooth images and making
@@ -463,6 +450,11 @@ Available filters are:
chroma temporal strength (default:
``luma_tmp*chroma_spatial/luma_spatial``)
.. note::
Deprecated. Use libavfilter's ``hqdn3d`` filter through ``--vf=lavfi``
instead.
``eq[=gamma:contrast:brightness:saturation:rg:gg:bg:weight]``
Software equalizer that uses lookup tables (slow), allowing gamma correction
in addition to simple brightness and contrast adjustment. The parameters are
@@ -502,26 +494,39 @@ Available filters are:
:0: nearest-neighbor sampling, fast but incorrect
:1: linear interpolation (default)
``unsharp[=lx:ly:la:cx:cy:ca]``
unsharp mask / Gaussian blur
``unsharp[=l|cWxH:amount[:l|cWxH:amount]]``
unsharp mask / gaussian blur
``l`` is for the luma component, ``c`` for the chroma component. ``x``/``y``
is the filter size. ``a`` is the amount.
``l``
Apply effect on luma component.
``lx``, ``ly``, ``cx``, ``cy``
``c``
Apply effect on chroma components.
``<width>x<height>``
width and height of the matrix, odd sized in both directions (min =
3:3, max = 13:11 or 11:13, usually something between 3:3 and 7:7)
3x3, max = 13x11 or 11x13, usually something between 3x3 and 7x7)
``la``, ``ca``
``amount``
Relative amount of sharpness/blur to add to the image (a sane range
should be -1.5-1.5).
:<0: blur
:>0: sharpen
.. note::
Deprecated. Use libavfilter's ``unsharp`` filter through ``--vf=lavfi``
instead.
``swapuv``
Swap U & V plane.
.. note::
Deprecated. Use libavfilter's ``swapuv`` filter through ``--vf=lavfi``
instead.
``pullup[=jl:jr:jt:jb:sb:mp]``
Pulldown reversal (inverse telecine) filter, capable of handling mixed
hard-telecine, 24000/1001 fps progressive, and 30000/1001 fps progressive
@@ -546,7 +551,7 @@ Available filters are:
frames in the output.
``mp`` (metric plane)
This option may be set to ``u`` or ``v`` to use a chroma plane instead of the
This option may be set to 1 or 2 to use a chroma plane instead of the
luma plane for doing ``pullup``'s computations. This may improve accuracy
on very clean source material, but more likely will decrease accuracy,
especially if there is chroma noise (rainbow effect) or any grayscale
@@ -617,7 +622,7 @@ Available filters are:
``phase[=t|b|p|a|u|T|B|A|U][:v]``
Delay interlaced video by one field time so that the field order changes.
The intended use is to fix PAL videos that have been captured with the
The intended use is to fix PAL movies that have been captured with the
opposite field order to the film-to-video transfer. The options are:
``t``
@@ -665,36 +670,41 @@ Available filters are:
``v``
Verbose operation. Prints the selected mode for each frame and the
average squared difference between fields for ``t``, ``b``, and ``p``
alternatives. (Ignored when libavfilter is used.)
alternatives.
``yadif=[mode[:enabled=yes|no]]``
Yet another deinterlacing filter
``<mode>``
:frame: Output 1 frame for each frame.
:field: Output 1 frame for each field.
:frame-nospatial: Like ``frame`` but skips spatial interlacing check.
:field-nospatial: Like ``field`` but skips spatial interlacing check.
:0: Output 1 frame for each frame.
:1: Output 1 frame for each field.
:2: Like 0 but skips spatial interlacing check.
:3: Like 1 but skips spatial interlacing check.
``<enabled>``
:yes: Filter is active (default).
:no: Filter is not active, but can be activated with the ``D`` key
(or any other key that toggles the ``deinterlace`` property).
This filter, is automatically inserted when using the ``D`` key (or any
other key that toggles the ``deinterlace`` property or when using the
``--deinterlace`` switch), assuming the video output does not have native
.. note::
Deprecated. Use libavfilter's ``yadif`` filter through ``--vf=lavfi``
instead.
This filter, or libavfilter's implementation if available, is automatically
inserted when using the ``D`` key (or any other key that toggles the
``deinterlace`` property), assuming the video output does not have native
deinterlacing support.
If you just want to set the default mode, put this filter and its options
into ``--vf-defaults`` instead, and enable deinterlacing with ``D`` or
``--deinterlace``.
``down3dright[=lines]``
Reposition and resize stereoscopic images. Extracts both stereo fields and
places them side by side, resizing them to maintain the original movie
aspect.
Also note that the ``D`` key is stupid enough to insert an interlacer twice
when inserting yadif with ``--vf``, so using the above methods is
recommended.
``<lines>``
number of lines to select from the middle of the image (default: 12)
``delogo[=x:y:w:h:t:show]``
``delogo[=x:y:w:h:t]``
Suppresses a TV station logo by a simple interpolation of the surrounding
pixels. Just set a rectangle covering the logo and watch it disappear (and
sometimes something even uglier appear - your mileage may vary).
@@ -707,8 +717,15 @@ Available filters are:
Thickness of the fuzzy edge of the rectangle (added to ``w`` and
``h``). When set to -1, a green rectangle is drawn on the screen to
simplify finding the right ``x``,``y``,``w``,``h`` parameters.
``show``
Draw a rectangle showing the area defined by x/y/w/h.
``file=<file>``
You can specify a text file to load the coordinates from. Each line
must have a timestamp (in seconds, and in ascending order) and the
``x:y:w:h:t`` coordinates (``t`` can be omitted).
.. note::
Deprecated. Use libavfilter's ``delogo`` or ``removelogo`` filters
through ``--vf=lavfi`` instead.
``screenshot``
Optional filter for screenshot support. This is only needed if the video
@@ -769,8 +786,8 @@ Available filters are:
anaglyph red/cyan color (red filter on left eye, cyan filter on
right eye)
``arcd`` or ``anaglyph_red_cyan_dubois``
anaglyph red/cyan color optimized with the least-squares
projection of Dubois (red filter on left eye, cyan filter on right
anaglyph red/cyan color optimized with the least squares
projection of dubois (red filter on left eye, cyan filter on right
eye)
``agmg`` or ``anaglyph_green_magenta_gray``
anaglyph green/magenta gray (green filter on left eye, magenta
@@ -801,24 +818,29 @@ Available filters are:
``mr`` or ``mono_right``
mono output (right eye only)
``gradfun[=strength[:radius|:size=<size>]]``
.. note::
Deprecated. Use libavfilter's ``stereo3d`` filter through ``--vf=lavfi``
instead.
``gradfun[=strength[:radius]]``
Fix the banding artifacts that are sometimes introduced into nearly flat
regions by truncation to 8-bit color depth. Interpolates the gradients that
regions by truncation to 8bit color depth. Interpolates the gradients that
should go where the bands are, and dithers them.
``<strength>``
Maximum amount by which the filter will change any one pixel. Also the
threshold for detecting nearly flat regions (default: 1.5).
threshold for detecting nearly flat regions (default: 1.2).
``<radius>``
Neighborhood to fit the gradient to. Larger radius makes for smoother
gradients, but also prevents the filter from modifying pixels near
detailed regions (default: disabled).
detailed regions (default: 16).
``<size>``
size of the filter in percent of the image diagonal size. This is
used to calculate the final radius size (default: 1).
.. note::
Deprecated. Use libavfilter's ``gradfun`` filter through ``--vf=lavfi``
instead.
``dlopen=dll[:a0[:a1[:a2[:a3]]]]``
Loads an external library to filter the image. The library interface
@@ -839,139 +861,3 @@ Available filters are:
``a3=<string>``
Specify the fourth parameter to pass to the library.
``vapoursynth=file:buffered-frames:concurrent-frames``
Loads a VapourSynth filter script. This is intended for streamed
processing: mpv actually provides a source filter, instead of using a
native VapourSynth video source. The mpv source will answer frame
requests only within a small window of frames (the size of this window
is controlled with the ``buffered-frames`` parameter), and requests outside
of that will return errors. As such, you can't use the full power of
VapourSynth, but you can use certain filters.
If you just want to play video generated by a VapourSynth (i.e. using
a native VapourSynth video source), it's better to use ``vspipe`` and a
FIFO to feed the video to mpv. The same applies if the filter script
requires random frame access (see ``buffered-frames`` parameter).
This filter is experimental. If it turns out that it works well and is
used, it will be ported to libavfilter. Otherwise, it will be just removed.
``file``
Filename of the script source. Currently, this is always a python
script. The variable ``video_in`` is set to the mpv video source,
and it is expected that the script reads video from it. (Otherwise,
mpv will decode no video, and the video packet queue will overflow,
eventually leading to audio being stopped.) The script is also
expected to pass through timestamps using the ``_DurationNum`` and
``_DurationDen`` frame properties.
.. admonition:: Example:
::
import vapoursynth as vs
core = vs.get_core()
core.std.AddBorders(video_in, 10, 10, 20, 20).set_output()
.. warning::
The script will be reloaded on every seek. This is done to reset
the filter properly on discontinuities.
``buffered-frames``
Maximum number of decoded video frames that should be buffered before
the filter (default: 4). This specifies the maximum number of frames
the script can requests backwards. E.g. if ``buffered-frames=5``, and
the script just requested frame 15, it can still request frame 10, but
frame 9 is not available anymore. If it requests frame 30, mpv will
decode 15 more frames, and keep only frames 25-30.
The actual number of buffered frames also depends on the value of the
``concurrent-frames`` option. Currently, both option values are
multiplied to get the final buffer size.
(Normally, VapourSynth source filters must provide random access, but
mpv was made for playback, and does not provide frame-exact random
access. The way this video filter works is a compromise to make simple
filters work anyway.)
``concurrent-frames``
Number of frames that should be requested in parallel (default: 2). The
level of concurrency depends on the filter and how quickly mpv can
decode video to feed the filter. This value should probably be
proportional to the number of cores on your machine. Most time,
making it higher than the number of cores can actually make it
slower.
``vavpp``
VA-AP-API video post processing. Works with ``--vo=vaapi`` and ``--vo=opengl``
only. Currently deinterlaces. This filter is automatically inserted if
deinterlacing is requested (either using the ``D`` key, by default mapped to
the command ``cycle deinterlace``, or the ``--deinterlace`` option).
``deint=<method>``
Select the deinterlacing algorithm.
no
Don't perform deinterlacing.
first-field
Show only first field (going by ``--field-dominance``).
bob
bob deinterlacing (default).
``vdpaupp``
VDPAU video post processing. Works with ``--vo=vdpau`` and ``--vo=opengl``
only. This filter is automatically inserted if deinterlacing is requested
(either using the ``D`` key, by default mapped to the command
``cycle deinterlace``, or the ``--deinterlace`` option). When enabling
deinterlacing, it is always preferred over software deinterlacer filters
if the ``vdpau`` VO is used, and also if ``opengl`` is used and hardware
decoding was activated at least once (i.e. vdpau was loaded).
``sharpen=<-1-1>``
For positive values, apply a sharpening algorithm to the video, for
negative values a blurring algorithm (default: 0).
``denoise=<0-1>``
Apply a noise reduction algorithm to the video (default: 0; no noise
reduction).
``deint=<yes|no>``
Whether deinterlacing is enabled (default: no). If enabled, it will use
the mode selected with ``deint-mode``.
``deint-mode=<first-field|bob|temporal|temporal-spatial>``
Select deinterlacing mode (default: temporal).
All modes respect ``--field-dominance``.
Note that there's currently a mechanism that allows the ``vdpau`` VO to
change the ``deint-mode`` of auto-inserted ``vdpaupp`` filters. To avoid
confusion, it's recommended not to use the ``--vo=vdpau`` suboptions
related to filtering.
first-field
Show only first field.
bob
Bob deinterlacing.
temporal
Motion-adaptive temporal deinterlacing. May lead to A/V desync
with slow video hardware and/or high resolution.
temporal-spatial
Motion-adaptive temporal deinterlacing with edge-guided spatial
interpolation. Needs fast video hardware.
``chroma-deint``
Makes temporal deinterlacers operate both on luma and chroma (default).
Use no-chroma-deint to solely use luma and speed up advanced
deinterlacing. Useful with slow video memory.
``pullup``
Try to apply inverse telecine, needs motion adaptive temporal
deinterlacing.
``hqscaling=<0-9>``
0
Use default VDPAU scaling (default).
1-9
Apply high quality VDPAU scaling (needs capable hardware).
``buffer=<num>``
Buffer ``<num>`` frames in the filter chain. This filter is probably pretty
useless, except for debugging. (Note that this won't help smoothing out
latencies with decoding, because the filter will never output a frame if
the buffer isn't full, except on EOF.)
+64 -232
View File
@@ -10,19 +10,10 @@ syntax is:
If the list has a trailing ',', mpv will fall back on drivers not contained
in the list. Suboptions are optional and can mostly be omitted.
You can also set defaults for each driver. The defaults are applied before the
normal driver parameters.
``--vo-defaults=<driver1[:parameter1:parameter2:...],driver2,...>``
Set defaults for each driver.
.. note::
See ``--vo=help`` for a list of compiled-in video output drivers.
The recommended output drivers are ``--vo=vdpau`` and ``--vo=opengl-hq``.
All other drivers are just for compatibility or special purposes.
.. admonition:: Example
``--vo=opengl,xv,``
@@ -31,43 +22,44 @@ normal driver parameters.
Available video output drivers are:
``xv`` (X11 only)
Uses the XVideo extension to enable hardware-accelerated display. This is
Uses the XVideo extension to enable hardware-accelerated playback. This is
the most compatible VO on X, but may be low-quality, and has issues with
OSD and subtitle display.
.. note:: This driver is for compatibility with old systems.
For information about what colorkey is used and how it is drawn, run
mpv with the ``-v`` option and look out for the lines tagged with ``[xv
common]`` at the beginning.
``adaptor=<number>``
Select a specific XVideo adapter (check xvinfo results).
Select a specific XVideo adaptor (check xvinfo results).
``port=<number>``
Select a specific XVideo port.
``ck=<cur|use|set>``
Select the source from which the color key is taken (default: cur).
Select the source from which the colorkey is taken (default: cur).
cur
The default takes the color key currently set in Xv.
The default takes the colorkey currently set in Xv.
use
Use but do not set the color key from mpv (use the ``--colorkey``
Use but do not set the colorkey from mpv (use the ``--colorkey``
option to change it).
set
Same as use but also sets the supplied color key.
Same as use but also sets the supplied colorkey.
``ck-method=<man|bg|auto>``
Sets the color key drawing method (default: man).
Sets the colorkey drawing method (default: man).
man
Draw the color key manually (reduces flicker in some cases).
Draw the colorkey manually (reduces flicker in some cases).
bg
Set the color key as window background.
Set the colorkey as window background.
auto
Let Xv draw the color key.
Let Xv draw the colorkey.
``colorkey=<number>``
Changes the color key to an RGB value of your choice. ``0x000000`` is
Changes the colorkey to an RGB value of your choice. ``0x000000`` is
black and ``0xffffff`` is white.
``no-colorkey``
Disables color-keying.
Disables colorkeying.
``x11`` (X11 only)
Shared memory video output driver without hardware acceleration that works
@@ -79,36 +71,21 @@ Available video output drivers are:
Uses the VDPAU interface to display and optionally also decode video.
Hardware decoding is used with ``--hwdec=vdpau``.
.. note::
Earlier versions of mpv (and MPlayer, mplayer2) provided sub-options
to tune vdpau post-processing, like ``deint``, ``sharpen``, ``denoise``,
``chroma-deint``, ``pullup``, ``hqscaling``. These sub-options are
deprecated, and you should use the ``vdpaupp`` video filter instead.
``sharpen=<-1-1>``
(Deprecated. See note about ``vdpaupp``.)
For positive values, apply a sharpening algorithm to the video, for
negative values a blurring algorithm (default: 0).
``denoise=<0-1>``
(Deprecated. See note about ``vdpaupp``.)
Apply a noise reduction algorithm to the video (default: 0; no noise
reduction).
``deint=<-4-4>``
(Deprecated. See note about ``vdpaupp``.)
Select deinterlacing mode (default: 0). In older versions (as well as
MPlayer/mplayer2) you could use this option to enable deinterlacing.
This doesn't work anymore, and deinterlacing is enabled with either
the ``D`` key (by default mapped to the command ``cycle deinterlace``),
or the ``--deinterlace`` option. Also, to select the default deint mode,
you should use something like ``--vf-defaults=vdpaupp:deint-mode=temporal``
instead of this sub-option.
Select deinterlacing mode (default: -3). Positive values choose mode
and enable deinterlacing. Corresponding negative values select the
same deinterlacing mode, but do not enable deinterlacing on startup
(useful in configuration files to specify which mode will be enabled by
the "D" key). All modes respect ``--field-dominance``.
0
Pick the ``vdpaupp`` video filter default, which corresponds to 3.
Same as -3.
1
Show only first field.
2
@@ -120,19 +97,13 @@ Available video output drivers are:
Motion-adaptive temporal deinterlacing with edge-guided spatial
interpolation. Needs fast video hardware.
``chroma-deint``
(Deprecated. See note about ``vdpaupp``.)
Makes temporal deinterlacers operate both on luma and chroma (default).
Use no-chroma-deint to solely use luma and speed up advanced
deinterlacing. Useful with slow video memory.
``pullup``
(Deprecated. See note about ``vdpaupp``.)
Try to apply inverse telecine, needs motion adaptive temporal
deinterlacing.
``hqscaling=<0-9>``
(Deprecated. See note about ``vdpaupp``.)
0
Use default VDPAU scaling (default).
1-9
@@ -162,18 +133,8 @@ Available video output drivers are:
``output_surfaces=<2-15>``
Allocate this many output surfaces to display video frames (default:
3). See below for additional information.
``colorkey=<#RRGGBB|#AARRGGBB>``
Set the VDPAU presentation queue background color, which in practice
is the colorkey used if VDPAU operates in overlay mode (default:
``#020507``, some shade of black). If the alpha component of this value
is 0, the default VDPAU colorkey will be used instead (which is usually
green).
``force-yuv``
Never accept RGBA input. This means mpv will insert a filter to convert
to a YUV format before the VO. Sometimes useful to force availability
of certain YUV-only features, like video equalizer or deinterlacing.
Using the VDPAU frame queuing functionality controlled by the queuetime
Using the VDPAU frame queueing functionality controlled by the queuetime
options makes mpv's frame flip timing less sensitive to system CPU load and
allows mpv to start decoding the next frame(s) slightly earlier, which can
reduce jitter caused by individual slow-to-decode frames. However, the
@@ -198,9 +159,6 @@ Available video output drivers are:
``direct3d_shaders`` (Windows only)
Video output driver that uses the Direct3D interface.
.. note:: This driver is for compatibility with systems that don't provide
proper OpenGL drivers.
``prefer-stretchrect``
Use ``IDirect3DDevice9::StretchRect`` over other methods if possible.
@@ -257,15 +215,11 @@ Available video output drivers are:
Same as ``direct3d_shaders``, but with the options ``disable-textures``
and ``disable-shaders`` forced.
.. note:: This driver is for compatibility with old systems.
``corevideo`` (Mac OS X 10.6 and later)
Mac OS X CoreVideo video output driver. Uses the CoreVideo APIs to fill
PixelBuffers and generate OpenGL textures from them (useful as a fallback
for ``opengl``).
.. note:: This driver is for compatibility with old systems.
``opengl``
OpenGL video output driver. It supports extended scaling methods, dithering
and color management.
@@ -280,11 +234,6 @@ Available video output drivers are:
Some features are available with OpenGL 3 capable graphics drivers only
(or if the necessary extensions are available).
Hardware decoding over OpenGL-interop is supported to some degree. Note
that in this mode, some corner case might not be gracefully handled, and
color space conversion and chroma upsampling is generally in the hand of
the hardware decoder APIs.
``lscale=<filter>``
``bilinear``
@@ -293,13 +242,11 @@ Available video output drivers are:
``lanczos2``
Lanczos scaling with radius=2. Provides good quality and speed.
This is the default when using ``opengl-hq``.
``lanczos3``
Lanczos with radius=3.
``spline36``
This is the default when using ``opengl-hq``.
``bicubic_fast``
Bicubic filter. Has a blurring effect on the image, even if no
scaling is done.
@@ -318,9 +265,7 @@ Available video output drivers are:
There are some more filters. For a complete list, pass ``help`` as
value, e.g.::
mpv --vo=opengl:lscale=help
value, e.g.: ``mpv --vo=opengl:lscale=help``
``lparam1=<value>``
Set filter parameters. Ignored if the filter is not tunable. These are
@@ -338,9 +283,8 @@ Available video output drivers are:
disabled.
``stereo=<value>``
Select a method for stereo display. You may have to use
``--video-aspect`` to fix the aspect value. Experimental, do not expect
too much from it.
Select a method for stereo display. You may have to use ``--aspect`` to
fix the aspect value. Experimental, do not expect too much from it.
no
Normal 2D display
@@ -349,30 +293,26 @@ Available video output drivers are:
green-magenta
Convert side by side input to full-color green-magenta stereo.
quadbuffer
Convert side by side input to quad buffered stereo. Only supported
Convert side by side input to quadbuffered stereo. Only supported
by very few OpenGL cards.
``srgb``
Convert and color correct the output to sRGB before displaying it on
the screen. This option enables linear light scaling. It also forces
the options ``indirect`` and ``gamma``.
Enable gamma-correct scaling by working in linear light. This
makes use of sRGB textures and framebuffers.
This option forces the options ``indirect`` and ``gamma``.
This option is equivalent to using ``icc-profile`` with an sRGB ICC
profile, but it is implemented without a 3DLUT and does not require
LittleCMS 2. If both ``srgb`` and ``icc-profile`` are present, the
latter takes precedence, as they are somewhat redundant.
.. note::
Note: When playing back BT.2020 content with this option enabled, out
of gamut colors will be numerically clipped, which can potentially
change the hue and/or luminance. If this is not desired, it is
recommended to use ``icc-profile`` with an sRGB ICC profile instead,
when playing back wide-gamut BT.2020 content.
for YUV colorspaces, gamma 1/0.45 (2.222) is assumed. RGB input is
always assumed to be in sRGB.
This option is not really useful, as gamma-correct scaling has not much
influence on typical video playback. Most visible effect comes from
slightly different gamma.
``pbo``
Enable use of PBOs. This is slightly faster, but can sometimes lead to
sporadic and temporary image corruption (in theory, because reupload
is not retried when it fails), and perhaps actually triggers slower
paths with drivers that don't support PBOs properly.
Enable use of PBOs. This is faster, but can sometimes lead to sporadic
and temporary image corruption.
``dither-depth=<N|no|auto>``
Set dither target depth to N. Default: no.
@@ -452,7 +392,7 @@ Available video output drivers are:
auto
auto-select (default)
cocoa
Cocoa/OS X
Cocoa/OSX
win
Win32/WGL
x11
@@ -473,96 +413,56 @@ Available video output drivers are:
Selects the internal format of textures used for FBOs. The format can
influence performance and quality of the video output. (FBOs are not
always used, and typically only when using extended scalers.)
``fmt`` can be one of: rgb, rgba, rgb8, rgb10, rgb10_a2, rgb16, rgb16f,
rgb32f, rgba12, rgba16, rgba16f, rgba32f.
``fmt`` can be one of: rgb, rgba, rgb8, rgb10, rgb16, rgb16f, rgb32f,
rgba12, rgba16, rgba16f, rgba32f.
Default: rgb.
``gamma=<0.0..10.0>``
Set a gamma value. If gamma is adjusted in other ways (like with
the ``--gamma`` option or key bindings and the ``gamma`` property), the
value is multiplied with the other gamma value.
Setting this value to 1.0 can be used to always enable gamma control.
(Disables delayed enabling.)
``gamma``
Always enable gamma control. (Disables delayed enabling.)
``icc-profile=<file>``
Load an ICC profile and use it to transform linear RGB to screen output.
Needs LittleCMS 2 support compiled in. This option overrides the ``srgb``
property, as using both is somewhat redundant. It also enables linear
light scaling.
``icc-profile-auto``
Automatically select the ICC display profile currently specified by
the display settings of the operating system.
NOTE: Only implemented on OS X with Cocoa.
Needs LittleCMS2 support compiled in.
``icc-cache=<file>``
Store and load the 3D LUT created from the ICC profile in this file.
This can be used to speed up loading, since LittleCMS 2 can take a while
This can be used to speed up loading, since LittleCMS2 can take a while
to create the 3D LUT. Note that this file contains an uncompressed LUT.
Its size depends on the ``3dlut-size``, and can be very big.
``icc-intent=<value>``
Specifies the ICC Intent used for transformations between color spaces.
This affects the rendering when using ``icc-profile`` or ``srgb`` and
also affects the way DCP XYZ content gets converted to RGB.
0
perceptual
1
relative colorimetric (default)
relative colorimetric
2
saturation
3
absolute colorimetric
``approx-gamma``
Approximate the actual gamma function as a pure power curve of
1.95. A number of video editing programs and studios apparently use this
for mastering instead of the true curve. Most notably, anything in the
Apple ecosystem uses this approximation - including all programs
compatible with it. It's a sound idea to try enabling this flag first
when watching videos and shows to see if things look better that way.
This only affects the output when using either ``icc-profile`` or ``srgb``.
absolute colorimetric (default)
``3dlut-size=<r>x<g>x<b>``
Size of the 3D LUT generated from the ICC profile in each dimension.
Default is 128x256x64.
Sizes must be a power of two, and 512 at most.
Sizes must be a power of two, and 256 at most.
``alpha=<blend|yes|no>``
Decides what to do if the input has an alpha component (default: blend).
blend
Blend the frame against a black background.
yes
Try to create a framebuffer with alpha component. This only makes sense
if the video contains alpha information (which is extremely rare). May
not be supported on all platforms. If alpha framebuffers are
unavailable, it silently falls back on a normal framebuffer. Note
that when using FBO indirections (such as with ``opengl-hq``), an FBO
format with alpha must be specified with the ``fbo-format`` option.
no
Ignore alpha component.
``alpha``
Try to create a framebuffer with alpha component. This only makes sense
if the video contains alpha information (which is extremely rare). May
not be supported on all platforms. If alpha framebuffers are
unavailable, it silently falls back on a normal framebuffer. Note
that when using FBO indirections (such as with ``opengl-hq``), an FBO
format with alpha must be specified with the ``fbo-format`` option.
``chroma-location=<auto|center|left>``
Set the YUV chroma sample location. auto means use the bitstream
flags (default: auto).
``rectangle-textures``
Force use of rectangle textures (default: no). Normally this shouldn't
have any advantages over normal textures. Note that hardware decoding
overrides this flag.
``opengl-hq``
Same as ``opengl``, but with default settings for high quality rendering.
This is equivalent to::
--vo=opengl:lscale=spline36:dither-depth=auto:fbo-format=rgba16
--vo=opengl:lscale=lanczos2:dither-depth=auto:pbo:fbo-format=rgb16
Note that some cheaper LCDs do dithering that gravely interferes with
``opengl``'s dithering. Disabling dithering with ``dither-depth=no`` helps.
@@ -571,7 +471,7 @@ Available video output drivers are:
can achieve better quality or performance by changing the ``fbo-format``
suboption to ``rgb16f``, ``rgb32f`` or ``rgb``. Known problems include
Mesa/Intel not accepting ``rgb16``, Mesa sometimes not being compiled with
float texture support, and some OS X setups being very slow with ``rgb16``
float texture support, and some OSX setups being very slow with ``rgb16``
but fast with ``rgb32f``.
``opengl-old``
@@ -585,8 +485,6 @@ Available video output drivers are:
implementation, even if you do not get any error message. Use ``glxinfo``
or a similar tool to display the supported OpenGL extensions.
.. note:: This driver is for compatibility with old systems.
``(no-)ati-hack``
ATI drivers may give a corrupted image when PBOs are used (when using
``force-pbo``). This option fixes this, at the expense of using a bit
@@ -695,9 +593,8 @@ Available video output drivers are:
Set the effect strength for the ``lscale``/``cscale`` filters that
support it.
``stereo=<value>``
Select a method for stereo display. You may have to use
``--video-aspect`` to fix the aspect value. Experimental, do not expect
too much from it.
Select a method for stereo display. You may have to use ``--aspect`` to
fix the aspect value. Experimental, do not expect too much from it.
0
Normal 2D display
@@ -706,13 +603,14 @@ Available video output drivers are:
2
Convert side by side input to full-color green-magenta stereo.
3
Convert side by side input to quad buffered stereo. Only supported
Convert side by side input to quadbuffered stereo. Only supported
by very few OpenGL cards.
The following options are only useful if writing your own fragment programs.
``customprog=<filename>``
Load a custom fragment program from ``<filename>``.
Load a custom fragment program from ``<filename>``. See
``TOOLS/edgedect.fp`` for an example.
``customtex=<filename>``
Load a custom "gamma ramp" texture from ``<filename>``. This can be used
in combination with ``yuv=4`` or with the ``customprog`` option.
@@ -748,7 +646,7 @@ Available video output drivers are:
auto
auto-select (default)
cocoa
Cocoa/OS X
Cocoa/OSX
win
Win32/WGL
x11
@@ -761,66 +659,18 @@ Available video output drivers are:
hardware acceleration. Should work on all platforms supported by SDL 2.0.
For tuning, refer to your copy of the file ``SDL_hints.h``.
.. note:: This driver is for compatibility with systems that don't provide
proper graphics drivers, or which support GLES only.
``sw``
Continue even if a software renderer is detected.
``switch-mode``
Instruct SDL to switch the monitor video mode when going fullscreen.
``vaapi``
Intel VA API video output driver with support for hardware decoding. Note
that there is absolutely no reason to use this, other than wanting to use
hardware decoding to save power on laptops, or possibly preventing video
tearing with some setups.
.. note:: This driver is for compatibility with crappy systems. You can
use vaapi hardware decoding with ``--vo=opengl`` too.
``scaling=<algorithm>``
default
Driver default (mpv default as well).
fast
Fast, but low quality.
hq
Unspecified driver dependent high-quality scaling, slow.
nla
``non-linear anamorphic scaling``
``deint-mode=<mode>``
Select deinterlacing algorithm. Note that by default deinterlacing is
initially always off, and needs to be enabled with the ``D`` key
(default key binding for ``cycle deinterlace``).
This option doesn't apply if libva supports video post processing (vpp).
In this case, the default for ``deint-mode`` is ``no``, and enabling
deinterlacing via user interaction using the methods mentioned above
actually inserts the ``vavpp`` video filter. If vpp is not actually
supported with the libva backend in use, you can use this option to
forcibly enable VO based deinterlacing.
no
Don't allow deinterlacing (default for newer libva).
first-field
Show only first field (going by ``--field-dominance``).
bob
bob deinterlacing (default for older libva).
``scaled-osd=<yes|no>``
If enabled, then the OSD is rendered at video resolution and scaled to
display resolution. By default, this is disabled, and the OSD is
rendered at display resolution if the driver supports it.
``null``
Produces no video output. Useful for benchmarking.
``caca``
Color ASCII art video output driver that works on a text console.
.. note:: This driver is a joke.
``image``
Output each frame into an image file in the current directory. Each file
takes the frame number padded with leading zeros as name.
@@ -862,21 +712,3 @@ Available video output drivers are:
JPEG DPI (default: 72)
``outdir=<dirname>``
Specify the directory to save the image files to (default: ``./``).
``wayland`` (Wayland only)
Wayland shared memory video output as fallback for ``opengl``.
.. note:: This driver is for compatibility with systems that don't provide
working OpenGL drivers.
``alpha``
Use a buffer format that supports videos and images with alpha
information
``rgb565``
Use RGB565 as buffer format. This format is implemented on most
platforms, especially on embedded where it is far more efficient then
RGB8888.
``triple-buffering``
Use 3 buffers instead of 2. This can lead to more fluid playback, but
uses more memory.
-1353
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File diff suppressed because it is too large. Load diff
-651
View File
@@ -1,651 +0,0 @@
LUA SCRIPTING
=============
mpv can load Lua scripts. Scripts in ``~/.mpv/lua/`` will be loaded on program
start, or if passed to ``--lua``. mpv provides the built-in module ``mp``, which
provides functions to send commands to the mpv core and to retrieve information
about playback state, user settings, file information, and so on.
These scripts can be used to control mpv in a similar way to slave mode.
Technically, the Lua code uses the client API internally.
Example
-------
A script which leaves fullscreen mode when the player is paused:
::
function on_pause()
mp.set_property("fullscreen", "no")
end
mp.register_event("pause", on_pause)
This script provides a pretty weird feature, but Lua scripting was made to
allow users implement features which are not going to be added to the mpv core.
Mode of operation
-----------------
Your script will be loaded by the player at program start from ``~/.mpv/lua/``,
or ``--lua``, or in some cases, internally (like ``--osc``). Each script runs
in its own thread. Your script is first run "as is", and once that is done,
the event loop is entered. This event loop will dispatch events received by mpv
and call your own event handlers which you have registered with
``mp.register_event``, or timers added with ``mp.add_timeout`` or similar.
When the player quits, all scripts will be asked to terminate. This happens via
a ``shutdown`` event, which by default will make the event loop return. If your
script got into an endless loop, mpv will probably behave fine during playback
(unless the player is suspended, see ``mp.suspend``), but it won't terminate
when quitting, because it's waiting on your script.
Internally, the C code will call the Lua function ``mp_event_loop`` after
loading a Lua script. This function is normally defined by the default prelude
loaded before your script (see ``player/lua/defaults.lua`` in the mpv sources).
The event loop will wait for events and dispatch events registered with
``mp.register_event``. It will also handle timers added with ``mp.add_timeout``
and similar (by waiting with a timeout).
mp functions
------------
The ``mp`` module is preloaded, although it can be loaded manually with
``require 'mp'``. It provides the core client API.
``mp.command(string)``
Run the given command. This is similar to the commands used in input.conf.
See `List of Input Commands`_.
Returns ``true`` on success, or ``nil, error`` on error.
``mp.commandv(arg1, arg2, ...)``
Similar to ``mp.command``, but pass each command argument as separate
parameter. This has the advantage that you don't have to care about
quoting and escaping in some cases.
Example:
::
mp.command("loadfile " .. filename .. " append")
mp.commandv("loadfile", filename, "append")
These two commands are equivalent, except that the first version breaks
if the filename contains spaces or certain special characters.
Note that properties are *not* expanded. You can use either ``mp.command``,
the ``expand-properties`` prefix, or the ``mp.get_property`` family of
functions.
``mp.get_property(name [,def])``
Return the value of the given property as string. These are the same
properties as used in input.conf. See `Properties`_ for a list of
properties. The returned string is formatted similar to ``${=name}``
(see `Property Expansion`_).
Returns the string on success, or ``def, error`` on error. ``def`` is the
second parameter provided to the function, and is nil if it's missing.
``mp.get_property_osd(name [,def])``
Similar to ``mp.get_property``, but return the property value formatted for
OSD. This is the same string as printed with ``${name}`` when used in
input.conf.
Returns the string on success, or ``def, error`` on error. ``def`` is the
second parameter provided to the function, and is an empty string if it's
missing. Unlike ``get_property()``, assigning the return value to a variable
will always result in a string.
``mp.get_property_bool(name [,def])``
Similar to ``mp.get_property``, but return the property value as Boolean.
Returns a Boolean on success, or ``def, error`` on error.
``mp.get_property_number(name [,def])``
Similar to ``mp.get_property``, but return the property value as number.
Note that while Lua does not distinguish between integers and floats,
mpv internals do. This function simply request a double float from mpv,
and mpv will usually convert integer property values to float.
Returns a number on success, or ``def, error`` on error.
``mp.get_property_native(name [,def])``
Similar to ``mp.get_property``, but return the property value using the best
Lua type for the property. Most time, this will return a string, Boolean,
or number. Some properties (for example ``chapter-list``) are returned as
tables.
Returns a value on success, or ``def, error`` on error. Note that ``nil``
might be a possible, valid value too in some corner cases.
``mp.set_property(name, value)``
Set the given property to the given string value. See ``mp.get_property``
and `Properties`_ for more information about properties.
Returns true on success, or ``nil, error`` on error.
``mp.set_property_bool(name, value)``
Similar to ``mp.set_property``, but set the given property to the given
Boolean value.
``mp.set_property_number(name, value)``
Similar to ``mp.set_property``, but set the given property to the given
numeric value.
Note that while Lua does not distinguish between integers and floats,
mpv internals do. This function will test whether the number can be
represented as integer, and if so, it will pass an integer value to mpv,
otherwise a double float.
``mp.set_property_native(name, value)``
Similar to ``mp.set_property``, but set the given property using its native
type.
Since there are several data types which can not represented natively in
Lua, this might not always work as expected. For example, while the Lua
wrapper can do some guesswork to decide whether a Lua table is an array
or a map, this would fail with empty tables. Also, there are not many
properties for which it makes sense to use this, instead of
``set_property``, ``set_property_bool``, ``set_property_number``.
For these reasons, this function should probably be avoided for now, except
for properties that use tables natively.
``mp.get_time()``
Return the current mpv internal time in seconds as a number. This is
basically the system time, with an arbitrary offset.
``mp.add_key_binding(key, name|fn [,fn])``
Register callback to be run on a key binding. The binding will be mapped to
the given ``key``, which is a string describing the physical key. This uses
the same key names as in input.conf, and also allows combinations
(e.g. ``ctrl+a``).
After calling this function, key presses will cause the function ``fn`` to
be called (unless the user remapped the key with another binding).
The ``name`` argument should be a short symbolic string. It allows the user
to remap the key binding via input.conf using the ``script_message``
command, and the name of the key binding (see below for
an example). The name should be unique across other bindings in the same
script - if not, the previous binding with the same name will be
overwritten. You can omit the name, in which case a random name is generated
internally.
Internally, key bindings are dispatched via the ``script_message_to`` input
command and ``mp.register_script_message``.
Trying to map multiple commands to a key will essentially prefer a random
binding, while the other bindings are not called. It is guaranteed that
user defined bindings in the central input.conf are preferred over bindings
added with this function (but see ``mp.add_forced_key_binding``).
Example:
::
function something_handler()
print("the key was pressed")
end
mp.add_key_binding("x", "something", something_handler)
This will print the message ``the key was pressed`` when ``x`` was pressed.
The user can remap these key bindings. Then the user has to put the
following into his input.conf to remap the command to the ``y`` key:
::
y script_message something
This will print the message when the key ``y`` is pressed. (``x`` will
still work, unless the user remaps it.)
You can also explicitly send a message to a named script only. Assume the
above script was using the filename ``fooscript.lua``:
::
y script_message_to fooscript something
``mp.add_forced_key_binding(...)``
This works almost the same as ``mp.add_key_binding``, but registers the
key binding in a way that will overwrite the user's custom bindings in his
input.conf. (``mp.add_key_binding`` overwrites default key bindings only,
but not those by the user's input.conf.)
``mp.remove_key_binding(name)``
Remove a key binding added with ``mp.add_key_binding`` or
``mp.add_forced_key_binding``. Use the same name as you used when adding
the bindings. It's not possible to remove bindings for which you omitted
the name.
``mp.register_event(name, fn)``
Call a specific function when an event happens. The event name is a string,
and the function fn is a Lua function value.
Some events have associated data. This is put into a Lua table and passed
as argument to fn. The Lua table by default contains a ``name`` field,
which is a string containing the event name. If the event has an error
associated, the ``error`` field is set to a string describing the error,
on success it's not set.
If multiple functions are registered for the same event, they are run in
registration order, which the first registered function running before all
the other ones.
Returns true if such an event exists, false otherwise.
See `Events`_ and `List of events`_ for details.
``mp.unregister_event(fn)``
Undo ``mp.register_event(..., fn)``. This removes all event handlers that
are equal to the ``fn`` parameter. This uses normal Lua ``==`` comparison,
so be careful when dealing with closures.
``mp.observe_property(name, type, fn)``
Watch a property for changes. If the property ``name`` is changed, then
the function ``fn(name)`` will be called. ``type`` can be ``nil``, or be
set to one of ``none``, ``native``, ``bool``, ``string``, or ``number``.
``none`` is the same as ``nil``. For all other values, the new value of
the property will be passed as second argument to ``fn``, using
``mp.get_property_<type>`` to retrieve it. This means if ``type`` is for
example ``string``, ``fn`` is roughly called as in
``fn(name, mp.get_property_string(name))``.
If possible, change events are coalesced. If a property is changed a bunch
of times in a row, only the last change triggers the change function. (The
exact behavior depends on timing and other things.)
In some cases the function is not called even if the property changes.
Whether this can happen depends on the property.
If the ``type`` is ``none`` or ``nil``, sporadic property change events are
possible. This means the change function ``fn`` can be called even if the
property doesn't actually change.
``mp.unobserve_property(fn)``
Undo ``mp.observe_property(..., fn)``. This removes all property handlers
that are equal to the ``fn`` parameter. This uses normal Lua ``==``
comparison, so be careful when dealing with closures.
``mp.add_timeout(seconds, fn)``
Call the given function fn when the given number of seconds has elapsed.
Note that the number of seconds can be fractional. For now, the timer's
resolution may be as low as 50 ms, although this will be improved in the
future.
This is a one-shot timer: it will be removed when it's fired.
Returns a timer object. See ``mp.add_periodic_timer`` for details.
``mp.add_periodic_timer(seconds, fn)``
Call the given function periodically. This is like ``mp.add_timeout``, but
the timer is re-added after the function fn is run.
Returns a timer object. The timer object provides the following methods:
``stop()``
Disable the timer. Does nothing if the timer is already disabled.
This will remember the current elapsed time when stopping, so that
``resume()`` essentially unpauses the timer.
``kill()``
Disable the timer. Resets the elapsed time. ``resume()`` will
restart the timer.
``resume()``
Restart the timer. If the timer was disabled with ``stop()``, this
will resume at the time it was stopped. If the timer was disabled
with ``kill()``, or if it's a previously fired one-shot timer (added
with ``add_timeout()``), this starts the timer from the beginning,
using the initially configured timeout.
``timeout`` (RW)
This field contains the current timeout period. This value is not
updated as time progresses. It's only used to calculate when the
timer should fire next when the timer expires.
If you write this, you can call ``t:kill() ; t:resume()`` to reset
the current timeout to the new one. (``t:stop()`` won't use the
new timeout.)
``oneshot`` (RW)
Whether the timer is periodic (``false``) or fires just once
(``true``). This value is used when the timer expires (but before
the timer callback function fn is run).
``mp.get_opt(key)``
Return a setting from the ``--lua-opts`` option. It's up to the user and
the script how this mechanism is used. Currently, all scripts can access
this equally, so you should be careful about collisions.
``mp.get_script_name()``
Return the name of the current script. The name is usually made of the
filename of the script, with directory and file extension removed. If
there are several script which would have the same name, it's made unique
by appending a number.
.. admonition:: Example
The script ``/path/to/fooscript.lua`` becomes ``fooscript``.
``mp.osd_message(text [,duration])``
Show an OSD message on the screen. ``duration`` is in seconds, and is
optional (uses ``--osd-duration`` by default).
Advanced mp functions
---------------------
These also live in the ``mp`` module, but are documented separately as they
are useful only in special situations.
``mp.suspend()``
Suspend the mpv main loop. There is a long-winded explanation of this in
the C API function ``mpv_suspend()``. In short, this prevents the player
from displaying the next video frame, so that you don't get blocked when
trying to access the player.
This is automatically called by the event handler.
``mp.resume()``
Undo one ``mp.suspend()`` call. ``mp.suspend()`` increments an internal
counter, and ``mp.resume()`` decrements it. When 0 is reached, the player
is actually resumed.
``mp.resume_all()``
This resets the internal suspend counter and resumes the player. (It's
like calling ``mp.resume()`` until the player is actually resumed.)
You might want to call this if you're about to do something that takes a
long time, but doesn't really need access to the player (like a network
operation). Note that you still can access the player at any time.
``mp.get_wakeup_pipe()``
Calls ``mpv_get_wakeup_pipe()`` and returns the read end of the wakeup
pipe. (See ``client.h`` for details.)
``mp.get_next_timeout()``
Return the relative time in seconds when the next timer (``mp.add_timeout``
and similar) expires. If there is no timer, return ``nil``.
``mp.dispatch_events([allow_wait])``
This can be used to run custom event loops. If you want to have direct
control what the Lua script does (instead of being called by the default
event loop), you can set the global variable ``mp_event_loop`` to your
own function running the event loop. From your event loop, you should call
``mp.dispatch_events()`` to dequeue and dispatch mpv events.
If the ``allow_wait`` parameter is set to ``true``, the function will block
until the next event is received or the next timer expires. Otherwise (and
this is the default behavior), it returns as soon as the event loop is
emptied. It's strongly recommended to use ``mp.get_next_timeout()`` and
``mp.get_wakeup_pipe()`` if you're interested in properly working
notification of new events and working timers.
This function calls ``mp.suspend()`` and ``mp.resume_all()`` on its own.
``mp.enable_messages(level)``
Set the minimum log level of which mpv message output to receive. These
messages are normally printed to the terminal. By calling this function,
you can set the minimum log level of messages which should be received with
the ``log-message`` event. See the description of this event for details.
The level is a string, see ``msg.log`` for allowed log levels.
``mp.register_script_message(name, fn)``
This is a helper to dispatch ``script_message`` or ``script_message_to``
invocations to Lua functions. ``fn`` is called if ``script_message`` or
``script_message_to`` (with this script as destination) is run
with ``name`` as first parameter. The other parameters are passed to ``fn``.
If a message with the given name is already registered, it's overwritten.
Used by ``mp.add_key_binding``, so be careful about name collisions.
``mp.unregister_script_message(name)``
Undo a previous registration with ``mp.register_script_message``. Does
nothing if the ``name`` wasn't registered.
mp.msg functions
----------------
This module allows outputting messages to the terminal, and can be loaded
with ``require 'mp.msg'``.
``msg.log(level, ...)``
The level parameter is the message priority. It's a string and one of
``fatal``, ``error``, ``warn``, ``info``, ``v``, ``debug``. The user's
settings will determine which of these messages will be visible. Normally,
all messages are visible, except ``v`` and ``debug``.
The parameters after that are all converted to strings. Spaces are inserted
to separate multiple parameters.
You don't need to add newlines.
``msg.fatal(...)``, ``msg.error(...)``, ``msg.warn(...)``, ``msg.info(...)``, ``msg.verbose(...)``, ``msg.debug(...)``
All of these are shortcuts and equivalent to the corresponding
``msg.log(level, ...)`` call.
mp.options functions
--------------------
mpv comes with a built-in module to manage options from config-files and the
command-line. All you have to do is to supply a table with default options to
the read_options function. The function will overwrite the default values
with values found in the config-file and the command-line (in that order).
``options.read_options(table [, identifier])``
A ``table`` with key-value pairs. The type of the default values is
important for converting the values read from the config file or
command-line back. Do not use ``nil`` as a default value!
The ``identifier`` is used to identify the config-file and the command-line
options. These needs to unique to avoid collisions with other scripts.
Defaults to ``mp.get_script_name()``.
Example implementation::
require 'mp.options'
local options = {
optionA = "defaultvalueA",
optionB = -0.5,
optionC = true,
}
options.read_options(options, "myscript")
print(option.optionA)
The config file will be stored in ``lua-settings/identifier.conf`` in mpv's user
folder. Comment lines can be started with # and stray spaces are not removed.
Boolean values will be represented with yes/no.
Example config::
# comment
optionA=Hello World
optionB=9999
optionC=no
Command-line options are read from the ``--lua-opts`` parameter. To avoid
collisions, all keys have to be prefixed with ``identifier-``.
Example command-line::
--lua-opts=myscript-optionA=TEST:myscript-optionB=0:myscript-optionC=yes
mp.utils options
----------------
This built-in module provides generic helper functions for Lua, and have
strictly speaking nothing to do with mpv or video/audio playback. They are
provided for convenience. Most compensate for Lua's scarce standard library.
``utils.readdir(path [, filter])``
Enumerate all entries at the given path on the filesystem, and return them
as array. Each entry is a directory entry (without the path).
The list is unsorted (in whatever order the operating system returns it).
If the ``filter`` argument is given, it must be one of the following
strings:
``files``
List regular files only. This excludes directories, special files
(like UNIX device files or FIFOs), and dead symlinks. It includes
UNIX symlinks to regular files.
``dirs``
List directories only, or symlinks to directories. ``.`` and ``..``
are not included.
``normal``
Include the results of both ``files`` and ``dirs``. (This is the
default.)
``all``
List all entries, even device files, dead symlinks, FIFOs, and the
``.`` and ``..`` entries.
On error, ``nil, error`` is returned.
``utils.split_path(path)``
Split a path into directory component and filename component, and return
them. The first return value is always the directory. The second return
value is the trailing part of the path, the directory entry.
``utils.join_path(p1, p2)``
Return the concatenation of the 2 paths. Tries to be clever. For example,
if ```p2`` is an absolute path, p2 is returned without change.
Events
------
Events are notifications from player core to scripts. You can register an
event handler with ``mp.register_event``.
Note that all scripts (and other parts of the player) receive events equally,
and there's no such thing as blocking other scripts from receiving events.
Example:
::
function my_fn(event)
print("start of playback!")
end
mp.register_event("playback-start", my_fn)
List of events
--------------
``start-file``
Happens right before a new file is loaded. When you receive this, the
player is loading the file (or possibly already done with it).
``end-file``
Happens after a file was unloaded. Typically, the player will load the
next file right away, or quit if this was the last file.
``file-loaded``
Happens after a file was loaded and begins playback.
``seek``
Happens on seeking (including ordered chapter segment changes).
``playback-restart``
Start of playback after seek or after file was loaded.
``tracks-changed``
The list of video/audio/sub tracks was updated. (This happens on playback
start, and very rarely during playback.)
``track-switched``
A video/audio/subtitle track was switched on or off. This usually happens
when the user (or a script) changes the subtitle track and so on.
``idle``
Idle mode is entered. This happens when playback ended, and the player was
started with ``--idle`` or ``--force-window``. This mode is implicitly ended
when the ``start-file`` or ``shutdown`` events happen.
``pause``
Playback was paused. This also happens when for example the player is
paused on low network cache. Then the event type indicates the pause state
(like the property "pause" as opposed to the "core-idle" property), and you
might receive multiple ``pause`` events in a row.
``unpause``
Playback was unpaused. See above for details.
``tick``
Called after a video frame was displayed. This is a hack, and you should
avoid using it. Use timers instead and maybe watch pausing/unpausing events
to avoid wasting CPU when the player is paused.
``shutdown``
Sent when the player quits, and the script should terminate. Normally
handled automatically. See `Mode of operation`_.
``log-message``
Receives messages enabled with ``mp.enable_messages``. The message data
is contained in the table passed as first parameter to the event handler.
The table contains, in addition to the default event fields, the following
fields:
``prefix``
The module prefix, identifies the sender of the message. This is what
the terminal player puts in front of the message text when using the
``--v`` option, and is also what is used for ``--msg-level``.
``level``
The log level as string. See ``msg.log`` for possible log level names.
Note that later versions of mpv might add new levels or remove
(undocumented) existing ones.
``text``
The log message. Note that this is the direct output of a printf()
style output API. The text will contain embedded newlines, and it's
possible that a single message contains multiple lines, or that a
message contains a partial line.
It's safe to display messages only if they end with a newline character,
and to buffer them otherwise.
Keep in mind that these messages are meant to be hints for humans. You
should not parse them, and prefix/level/text of messages might change
any time.
``get-property-reply``
Undocumented (not useful for Lua scripts).
``set-property-reply``
Undocumented (not useful for Lua scripts).
``command-reply``
Undocumented (not useful for Lua scripts).
``script-input-dispatch``
Undocumented (used internally).
``client-message``
Undocumented (used internally).
``video-reconfig``
Happens on video output or filter reconfig.
``audio-reconfig``
Happens on audio output or filter reconfig.
``metadata-update``
Metadata (like file tags) was updated.
``chapter-change``
The current chapter possibly changed.
-252
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ON SCREEN CONTROLLER
====================
The On Screen Controller (short: OSC) is a minimal GUI integrated with mpv to
offer basic mouse-controllability. It is intended to make interaction easier
for new users and to enable precise and direct seeking.
The OSC is enabled by default if mpv was compiled with lua support. It can be
disabled entirely using the ``--osc=no`` option.
Using the OSC
-------------
By default, the OSC will show up whenever the mouse is moved inside the
player window and will hide if the mouse is not moved outside the OSC for
0.5 seconds or if the mouse leaves the window.
The Interface
~~~~~~~~~~~~~
::
+------------------+-----------+--------------------+
| playlist prev | title | playlist next |
+-------+------+---+--+------+-+----+------+--------+
| audio | skip | seek | | seek | skip | full |
+-------+ back | back | play | frwd | frwd | screen |
| sub | | | | | | |
+-------+------+------+------+------+------+--------+
| seekbar |
+----------------+--------------+-------------------+
| time passed | cache status | time remaining |
+----------------+--------------+-------------------+
playlist prev
============= ================================================
left-click play previous file in playlist
shift+L-click show playlist
============= ================================================
title
| Displays current media-title or filename
============= ================================================
left-click show playlist position and length and full title
right-click show filename
============= ================================================
playlist next
============= ================================================
left-click play next file in playlist
shift+L-click show playlist
============= ================================================
audio and sub
| Displays selected track and amount of available tracks
============= ================================================
left-click cycle audio/sub tracks forward
right-click cycle audio/sub tracks backwards
shift+L-click show available audio/sub tracks
============= ================================================
skip back
============= ================================================
left-click go to beginning of chapter / previous chapter
shift+L-click show chapters
============= ================================================
seek back
============= ================================================
left-click skip back 5 seconds
right-click skip back 30 seconds
shift-L-click skip back 1 frame
============= ================================================
play
============= ================================================
left-click toggle play/pause
============= ================================================
seek frwd
============= ================================================
left-click skip forward 10 seconds
right-click skip forward 60 seconds
shift-L-click skip forward 1 frame
============= ================================================
skip frwd
============= ================================================
left-click go to next chapter
shift+L-click show chapters
============= ================================================
fullscreen
============= ================================================
left-click toggle fullscreen
============= ================================================
seekbar
| Indicates current playback position and position of chapters
============= ================================================
left-click seek to position
============= ================================================
time passed
| Shows current playback position timestamp
============= ================================================
left-click toggle displaying timecodes with milliseconds
============= ================================================
cache status
| Shows current cache fill status (only visible when below 45%)
time remaining
| Shows remaining playback time timestamp
============= ================================================
left-click toggle between total and remaining time
============= ================================================
Key Bindings
~~~~~~~~~~~~
These key bindings are active by default if nothing else is already bound to
these keys. In case of collision, the function needs to be bound to a
different key. See `Script Commands`_ section.
============= ================================================
del Hide the OSC permanently until mpv is restarted.
============= ================================================
Configuration
-------------
The OSC offers limited configuration through a config file
``lua-settings/osc.conf`` placed in mpv's user dir and through the
``--lua-opts`` command-line option. Options provided through the command-line
will override those from the config file.
Config Syntax
~~~~~~~~~~~~~
The config file must exactly follow the following syntax::
# this is a comment
optionA=value1
optionB=value2
``#`` can only be used at the beginning of a line and there may be no
spaces around the ``=`` or anywhere else.
Command-line Syntax
~~~~~~~~~~~~~~~~~~~
To avoid collisions with other scripts, all options need to be prefixed with
``osc-``.
Example::
--lua-opts=osc-optionA=value1:osc-optionB=value2
Configurable Options
~~~~~~~~~~~~~~~~~~~~
``showwindowed``
| Default: yes
| Show OSC when windowed?
``showfullscreen``
| Default: yes
| Show OSC when fullscreen?
``scalewindowed``
| Default: 1
| Scaling of the controller when windowed
``scalefullscreen``
| Default: 1
| Scaling of the controller when fullscreen
``scaleforcedwindow``
| Default: 2
| Scaling of the controller when rendered on a forced (dummy) window
``vidscale``
| Default: yes
| Scale the controller with the video?
``valign``
| Default: 0.8
| Vertical alignment, -1 (top) to 1 (bottom)
``halign``
| Default: 0
| Horizontal alignment, -1 (left) to 1 (right)
``boxalpha``
| Default: 80
| Alpha of the background box, 0 (opaque) to 255 (fully transparent)
``hidetimeout``
| Default: 500
| Duration in ms until the OSC hides if no mouse movement, negative value
disables auto-hide
``fadeduration``
| Default: 200
| Duration of fade out in ms, 0 = no fade
``deadzonesize``
| Default: 0
| Size of the deadzone. The deadzone is an area that makes the mouse act
like leaving the window. Movement there won't make the OSC show up and
it will hide immediately if the mouse enters it. The deadzone starts
at the window border opposite to the OSC and the size controls how much
of the window it will span. Values between 0 and 1.
``minmousemove``
| Default: 3
| Minimum amount of pixels the mouse has to move between ticks to make
the OSC show up
``seektooltip``
| Default: yes
| Display a tooltip over the seekbar indicating time at mouse position.
Script Commands
~~~~~~~~~~~~~~~
The OSC script listens to certain script commands. These commands can bound
in ``input.conf``, or sent by other scripts.
``enable-osc``
Undoes ``disable-osc`` or the effect of the ``del`` key.
``disable-osc``
Hide the OSC permanently. This is also what the ``del`` key does.
Example
You could put this into ``input.conf`` to hide the OSC with the ``a`` key and
to unhide it with ``b``::
a script_message disable-osc
b script_message enable-osc
-71
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@@ -1,71 +0,0 @@
Release Policy
==============
Every few months, a new release is cut off of the master branch and is assigned
a 0.X.0 version number.
As part of the maintenance process, minor releases are made, which are assigned
0.X.Y version numbers. Minor releases contain bug fixes only. They never merge
the master branch, and no features are added to it. Only the latest release is
maintained.
The goal of releases is to provide stability and an unchanged base for the sake
of Linux distributions. If you want the newest features, just use the master
branch, which is stable most of the time, except sometimes, when it's not.
Releases other than the latest release are unsupported and unmaintained.
Release procedure
-----------------
- Create branch release/0.X or cherry-pick commits to the relevant branch.
- Create and/or update the ``RELEASE_NOTES`` file.
- Create and/or update the ``VERSION`` file.
- Update ``DOCS/client-api-changes.rst`` (on major releases).
- Create tag v0.X.Y.
- Push branch and tag to GitHub.
- Create a new GitHub release using the content of ``RELEASE_NOTES`` related to
the new version.
Release notes template
----------------------
Here is a template that can be used for writing the ``RELEASE_NOTES`` file.
```markdown
Release 0.X.Y
=============
Changes
-------
- List of changes.
Bug fixes
---------
- List of bug fixes.
New features
------------
- List of new features.
This listing is not complete. There are many more bug fixes and changes. The
complete change log can be viewed by running ``git log <start>..<end>`` in
the git repository.
```
Note that the "Release 0.X.Y" title should be removed when creating a new GitHub
release.
When creating a new point release its changes should be added on top of the
``RELEASE_NOTES`` file (with the appropriate title) so that all the changes in
the current 0.X branch will be included. This way the ``RELEASE_NOTES`` file
can be used by distributors as changelog for point releases too.
+40 -55
View File
@@ -1,11 +1,12 @@
This file intends to give a big picture of how mpv is structured.
NOTE: DOCS/OUTDATED-tech/* may contain more detailed information, but most of it
is possibly or definitely outdated. This file intends to give a big
picture of how mpv is structured.
player/*.c:
Essentially makes up the player applications, including the main() function
and the playback loop.
Generally, it accesses all other subsystems, initializes them, and pushes
data between them during playback.
mpvcore/mplayer.c:
This contains the main play loop, anything related to mpv and playback
related initializations. It also contains the main function. Generally, it
accesses all other subsystems, initializes them, and pushes data between
them during playback.
The structure is as follows (as of commit e13c05366557cb):
* main():
@@ -17,12 +18,12 @@ player/*.c:
* check help options etc. (call handle_help_options()), possibly exit
* call play_files() function that works down the playlist:
* run idle loop (idle_loop()), until there are files in the
playlist or an exit command was given (only if --idle it set)
playlist or an exit command was given (slave mode only)
* actually load and play a file in play_current_file():
* run all the dozens of functions to load the file and
initialize playback
* run a small loop that does normal playback, until the file is
done or a command terminates playback
done or a slave command terminates playback
(on each iteration, run_playloop() is called, which is rather
big and complicated - it decodes some audio and video on
each frame, waits for input, etc.)
@@ -41,7 +42,7 @@ player/*.c:
- the other subsystems rarely call back into the frontend, and the frontend
polls them instead (probably a good thing)
I like to call the player/*.c files the "frontend".
I like to call mplayer.c (and some other files) the "frontend".
talloc.h & talloc.c:
Hierarchical memory manager copied from Samba. It's like a malloc() with
@@ -75,53 +76,37 @@ talloc.h & talloc.c:
replacement. It works on top of system malloc and provides additional
features that are supposed to make memory management easier.
Warning: actually, we're not using talloc anymore. talloc in mpv has been
replaced by a custom re-implementation (TA in ta/). It provides
some talloc emulation (just the parts needed by mpv). We will get
rid of the talloc emulation later and use TA natively.
(See ta/README for details.)
mpvcore/mp_core.h:
Data structures for mplayer.c and command.c. They are usually not accessed
by other parts of mpv for the sake of modularization.
player/command.c:
This contains the implementation for client API commands and properties.
Properties are essentially dynamic variables changed by certain commands.
This is basically responsible for all user commands, like initiating
seeking, switching tracks, etc. It calls into other player/*.c files,
where most of the work is done, but also calls other parts of mpv.
Note that there are lots of global variables floating around everywhere
else. This is an ongoing transition, and eventually there should be no
global variables anymore.
player/core.h:
Data structures and function prototypes for most of player/*.c. They are
usually not accessed by other parts of mpv for the sake of modularization.
options.h contains the global option struct MPOpts, and its default values
are in defaultopts.c for some reason.
player/client.c:
This implements the client API (libmpv/client.h). For the most part, this
just calls into other parts of the player. This also manages a ringbuffer
of events from player to clients.
options/options.h, options/options.c
options.h contains the global option struct MPOpts. The option declarations
(option names, types, and MPOpts offsets for the option parser) are in
options.c. Most default values for options and MPOpts are in
mp_default_opts at the end of options.c.
MPOpts is unfortunarely quite monolithic, and virtually accessed by
everything.But some components (like video outputs and video filters) have
their own sub-option tables separate from MPOpts.
The actual option parser is spread over m_option.c, m_config.c, and
parser-mpcmd.c, and uses the option table in options.c.
input/input.c:
mpvcore/input/input.c:
This translates keyboard input comming from libvo and other sources (such
as remote control devices like Apple IR or client API commands) to the
as remote control devices like Apple IR or slave mode commands) to the
key bindings listed in the user's (or the builtin) input.conf and turns
them into items of type struct mp_cmd. These commands are queued, and read
by playloop.c. They get pushed with run_command() to command.c.
by mplayer.c. They get pushed with run_command() to command.c.
Note that keyboard input and commands used by the client API are the same.
The client API only uses the command parser though, and has its own queue
of input commands somewhere else.
Note that keyboard input and slave mode input are essentially the same
things. Just looking at input.conf should make this clear. (The other
direction of slave mode communication, mpv to application, consists of
random mp_msg() calls all over the code in all parts of the player.)
common/msg.h:
mpvcore/command.c:
This contains the implementation for slave commands and properties.
Properties are essentially dynamic variables changed by certain commands.
This is basically responsible for all user commands, like initiating
seeking, switching tracks, etc. It calls into mplayer.c, where most of the
work is done, but also into other parts of mpv.
mpvcore/mp_msg.h:
All terminal output should go through mp_msg().
stream/*:
@@ -197,8 +182,8 @@ audio/decode/:
audio/filter/:
Audio filter chain. af_lavrresample is inserted if any form of conversion
between audio formats is needed. (af_convert24.c and af_convertsignendian.c
are also used for some formats not directly supported by FFmpeg.)
between audio formats is needed. (af_format.c is also used for some corner
cases.)
audio/out/:
Audio outputs.
@@ -213,7 +198,7 @@ audio/out/:
sub/:
Contains subtitle and OSD rendering.
osd.c/.h is actually the OSD code. It queries dec_sub.c to retrieve
sub.c/.h is actually the OSD code. It queries dec_sub.c to retrieve
decoded/rendered subtitles. osd_libass.c is the actual implementation of
the OSD text renderer (which uses libass, and takes care of all the tricky
fontconfig/freetype API usage and text layouting).
@@ -222,7 +207,7 @@ sub/:
.ass subtitles via libass. demux_lavf.c loads most subtitle types via
FFmpeg. demux_subreader.c is the old MPlayer code. It's used as last
fallback, or to handle some text subtitle types on Libav. (It also can
load UTF-16 encoded subtitles without requiring the use of -sub-codepage.)
load UTF-16 encoded subtitles without requiring the use of -subcp.)
demux_subreader.c should eventually go away (maybe).
The subtitles are passed to dec_sub.c and the subtitle decoders in sd_*.c
@@ -238,8 +223,8 @@ sub/:
detection as well as timing postprocessing work. (Timing postprocessing
removes tiny gaps or overlaps between subtitle events.)
player/timeline/:
A timeline is the abstraction used by loadfile.c to combine several files
mpvcore/timeline/:
A timeline is the abstraction used by mplayer.c to combine several files
into one seemingly linear video. It's mainly used for ordered chapters
playback. The high level code to find and load other files containing the
segments for playing an ordered chapters file is in tl_matroska.c.
-156
View File
@@ -1,156 +0,0 @@
waf build system overview
=========================
mpv's new build system is based on waf and it should completly replace the
custom ./configure + Makefile based system inherited from MPlayer.
Goals and the choice of waf
===========================
The new system comes with some goals, which can be summed up as: be as good as
the old one at what it did well (customizability) and fix some of it's major
shortcomings:
1) The build system must be uniform in how it handles any single feature check.
Repetition and boilerplate have to be avoided.
When adding a new feature using the old configure, one had to add a fair
amount of code to the shell script to do option parsing, detection of the
feature and declaration of variables for the Makefile to pickup. The worst
part is this pieces are spread apart in the configure and copy pasted for
any single case. That brings us to..
2) --enable-feature has to override the user and help him understand that he
has libraries missing and should install them for the feature to be enabled.
3) Must be customizable, hackable, pleasant to the developer eyes and to work
with in general.
4) Must have separate configuration and build steps.
Goal 2 comes as a given on pretty much any build system, since autotools made
this behaviour very popular among users (and rightly so).
Goal 1+3 were somewhat harder to accomplish as it looks like all of the build
systems we evaluated (waf included!) had problems with them. For reference we
had proof of concept build systems with waf, CMake and autotools.
What puts waf apart from CMake and autotools, is that projects using it use
Python to program their build system. Also while the Waf Book shows really
simple API usages, you can write your own build system on top of waf that is
tailored to the project's specific needs.
mpv's custom configure step on top of waf
=========================================
To some extents mpv has a custom build system written on top of waf. This
document will not go over the standard waf behaviour as that is documented in
the `Waf book <http://docs.waf.googlecode.com/git/book_17/single.html>`_.
All of the configuration process is handled with a declarative approach. Lists
of dictionaries define the checks, and some custom Python code traverses these
lists and depending on the check definition it calls into the actual waf API.
A simple example using pkg-config would be::
{
'name': '--vdpau',
'desc': 'VDPAU acceleration',
'deps': [ 'x11' ],
'func': check_pkg_config('vdpau', '>= 0.2'),
}
This defines a feature called ``vdpau`` which can be enabled or disabled by
the users with configure flags (that's the meaning of ``--``). This feature
depends on another feature whose name is ``x11``, and the autodetection check
consists of running ``pkg-config`` and looking for ``vdpau`` with version
``>= 0.2``. If the check succeds a ``#define HAVE_VDPAU 1`` will be added to
``config.h``, if not ``#define HAVE_VDPAU 0`` will be added.
The defines names are automatically prepended with ``HAVE_``, capitalized and
special characters are replaced with underscores. This happens in
``waftools/inflectors.py``.
Mandatory fields:
-----------------
``name``: indicates the unique identifier used by the custom dependency code
to refer to a feature. If the unique identifier is prepended with ``--``
the build system will also generate options for ``./waf configure`` so that
the feature can be enabled and disabled.
``desc``: this is the textual representation of the feature used in the
interactions with the users.
``func``: function that will perform the check. These functions are defined in
``waftools/checks``. The reusable checks are all functions that return
functions. The return functions will then be applied using waf's configuration
context.
The source code for the reusable checks is a bit convoluted, but it should be
easy to pick up their usage from the ``wscript``. Their signature mirrors
the semantics of some of the shell functions used in mplayer.
If someone expresses some interest, I will extend this document with official
documentation for each check function.
Optional fields
---------------
``deps``: list of dependencies of this feature. It is a list of names of
other features as defined in the ``name`` field (minus the eventual leading
``--``). All of the dependencies must be satisfied. If they are not the check
will be skipped without even running ``func``.
``deps_any``: like deps but it is satisfied even if only one of the dependencies
is satisfied. You can think of ``deps`` as a 'and' condition and ``deps_any``
as a 'or' condition.
``deps_neg``: like deps but it is satisfied when none of the dependencies is
satisfied.
``req``: defaults to False. If set to True makes this feature a hard
dependency of mpv (configuration will fail if autodetection fails). If set to
True you must also provide ``fmsg``.
``fmsg``: string with the failure message in case a required dependency is not
satisfied.
``os_specific_checks``: this takes a dictionary that has ``os-`` dependencies
as keys (such as ``os-win32``), and by values has another dictionary that is
merged on top of the current feature definition only for that specific OS.
For example::
{
'name': '--pthreads',
'desc': 'POSIX threads',
'func': check_pthreads,
'os_specific_checks': {
'os-win32': {
'func': check_pthreads_w32_static.
}
}
}
will override the value of ``func`` with ``check_pthreads_w32_static`` only
if the target OS of the build is Windows.
``groups``: groups a dependency with another one. This can be used to disabled
all the grouped dependencies with one ``--disable-``. At the moment this is
only used for OpenGL backends, where you want to disable them when
``--disable-gl`` is passed to the configure.
mpv's custom build step on top of waf
=====================================
Build step is pretty much vanilla waf. The only difference being that the list
of source files can contain both strings or tuples. If a tuple is found,
the second element in the tuple will the used to match the features detected
in the configure step (the ``name`` field described above). If this feature
was not enabled during configure, the source file will not be compiled in.
All of the custom Python for this is inside the function ``filtered_sources``
contained in the file ``waftools/dependencies.py``.
Also ``dependencies_use`` and ``dependencies_includes`` collect cflags and
ldflags that were generated from the features checks in the configure step.
+587 -253
View File
@@ -1,284 +1,625 @@
GNU GENERAL PUBLIC LICENSE
Version 2, June 1991
Version 3, 29 June 2007
Copyright (C) 1989, 1991 Free Software Foundation, Inc.
51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Copyright (C) 2007 Free Software Foundation, Inc. <http://fsf.org/>
Everyone is permitted to copy and distribute verbatim copies
of this license document, but changing it is not allowed.
Preamble
The licenses for most software are designed to take away your
freedom to share and change it. By contrast, the GNU General Public
License is intended to guarantee your freedom to share and change free
software--to make sure the software is free for all its users. This
General Public License applies to most of the Free Software
Foundation's software and to any other program whose authors commit to
using it. (Some other Free Software Foundation software is covered by
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The GNU General Public License is a free, copyleft license for
software and other kinds of works.
The licenses for most software and other practical works are designed
to take away your freedom to share and change the works. By contrast,
the GNU General Public License is intended to guarantee your freedom to
share and change all versions of a program--to make sure it remains free
software for all its users. We, the Free Software Foundation, use the
GNU General Public License for most of our software; it applies also to
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When we speak of free software, we are referring to freedom, not
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NO WARRANTY
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11. BECAUSE THE PROGRAM IS LICENSED FREE OF CHARGE, THERE IS NO WARRANTY
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OTHERWISE STATED IN WRITING THE COPYRIGHT HOLDERS AND/OR OTHER PARTIES
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ALL NECESSARY SERVICING, REPAIR OR CORRECTION.
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IN NO EVENT UNLESS REQUIRED BY APPLICABLE LAW OR AGREED TO IN WRITING
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GENERAL, SPECIAL, INCIDENTAL OR CONSEQUENTIAL DAMAGES ARISING OUT OF THE
USE OR INABILITY TO USE THE PROGRAM (INCLUDING BUT NOT LIMITED TO LOSS OF
DATA OR DATA BEING RENDERED INACCURATE OR LOSSES SUSTAINED BY YOU OR THIRD
PARTIES OR A FAILURE OF THE PROGRAM TO OPERATE WITH ANY OTHER PROGRAMS),
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SUCH DAMAGES.
17. Interpretation of Sections 15 and 16.
If the disclaimer of warranty and limitation of liability provided
above cannot be given local legal effect according to their terms,
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Program, unless a warranty or assumption of liability accompanies a
copy of the Program in return for a fee.
END OF TERMS AND CONDITIONS
How to Apply These Terms to Your New Programs
If you develop a new program, and you want it to be of the greatest
@@ -287,15 +628,15 @@ free software which everyone can redistribute and change under these terms.
To do so, attach the following notices to the program. It is safest
to attach them to the start of each source file to most effectively
convey the exclusion of warranty; and each file should have at least
state the exclusion of warranty; and each file should have at least
the "copyright" line and a pointer to where the full notice is found.
<one line to give the program's name and a brief idea of what it does.>
Copyright (C) <year> <name of author>
This program is free software; you can redistribute it and/or modify
This program is free software: you can redistribute it and/or modify
it under the terms of the GNU General Public License as published by
the Free Software Foundation; either version 2 of the License, or
the Free Software Foundation, either version 3 of the License, or
(at your option) any later version.
This program is distributed in the hope that it will be useful,
@@ -304,37 +645,30 @@ the "copyright" line and a pointer to where the full notice is found.
GNU General Public License for more details.
You should have received a copy of the GNU General Public License
along with this program; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
along with this program. If not, see <http://www.gnu.org/licenses/>.
Also add information on how to contact you by electronic and paper mail.
If the program is interactive, make it output a short notice like this
when it starts in an interactive mode:
If the program does terminal interaction, make it output a short
notice like this when it starts in an interactive mode:
Gnomovision version 69, Copyright (C) year name of author
Gnomovision comes with ABSOLUTELY NO WARRANTY; for details type `show w'.
<program> Copyright (C) <year> <name of author>
This program comes with ABSOLUTELY NO WARRANTY; for details type `show w'.
This is free software, and you are welcome to redistribute it
under certain conditions; type `show c' for details.
The hypothetical commands `show w' and `show c' should show the appropriate
parts of the General Public License. Of course, the commands you use may
be called something other than `show w' and `show c'; they could even be
mouse-clicks or menu items--whatever suits your program.
parts of the General Public License. Of course, your program's commands
might be different; for a GUI interface, you would use an "about box".
You should also get your employer (if you work as a programmer) or your
school, if any, to sign a "copyright disclaimer" for the program, if
necessary. Here is a sample; alter the names:
You should also get your employer (if you work as a programmer) or school,
if any, to sign a "copyright disclaimer" for the program, if necessary.
For more information on this, and how to apply and follow the GNU GPL, see
<http://www.gnu.org/licenses/>.
Yoyodyne, Inc., hereby disclaims all copyright interest in the program
`Gnomovision' (which makes passes at compilers) written by James Hacker.
<signature of Ty Coon>, 1 April 1989
Ty Coon, President of Vice
This General Public License does not permit incorporating your program into
proprietary programs. If your program is a subroutine library, you may
consider it more useful to permit linking proprietary applications with the
library. If this is what you want to do, use the GNU Library General
Public License instead of this License.
The GNU General Public License does not permit incorporating your program
into proprietary programs. If your program is a subroutine library, you
may consider it more useful to permit linking proprietary applications with
the library. If this is what you want to do, use the GNU Lesser General
Public License instead of this License. But first, please read
<http://www.gnu.org/philosophy/why-not-lgpl.html>.
+152 -188
View File
@@ -19,22 +19,21 @@
# with MPlayer; if not, write to the Free Software Foundation, Inc.,
# 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
include old_build/config.mak
include config.mak
###### variable declarations #######
SOURCES_AUDIO_INPUT-$(ALSA) += stream/ai_alsa1x.c
SOURCES_AUDIO_INPUT-$(OSS_AUDIO)+= stream/ai_oss.c
SOURCES_AUDIO_INPUT-$(SNDIO) += stream/ai_sndio.c
SOURCES_AUDIO_INPUT-$(OSS) += stream/ai_oss.c
SOURCES-$(AUDIO_INPUT) += $(SOURCES_AUDIO_INPUT-yes)
SOURCES-$(CDDA) += stream/stream_cdda.c
SOURCES-$(CDDA) += stream/stream_cdda.c \
stream/cdinfo.c
SOURCES-$(DVBIN) += stream/dvb_tune.c \
stream/stream_dvb.c
SOURCES-$(DVDREAD) += stream/stream_dvd.c \
stream/stream_dvd_common.c
SOURCES-$(DVDNAV) += stream/stream_dvdnav.c \
stream/stream_dvd_common.c
SOURCES-$(HAVE_SYS_MMAN_H) += audio/filter/af_export.c
SOURCES-$(LADSPA) += audio/filter/af_ladspa.c
SOURCES-$(LIBASS) += sub/ass_mp.c sub/sd_ass.c \
demux/demux_libass.c
@@ -45,21 +44,43 @@ SOURCES-$(LIBBS2B) += audio/filter/af_bs2b.c
SOURCES-$(LIBPOSTPROC) += video/filter/vf_pp.c
SOURCES-$(LIBSMBCLIENT) += stream/stream_smb.c
SOURCES-$(MACOSX_BUNDLE) += osdep/macosx_bundle.m
SOURCES-$(COCOA) += video/out/cocoa_common.m \
osdep/macosx_application.m \
osdep/macosx_events.m \
osdep/ar/HIDRemote.m
SOURCES-$(MNG) += demux/demux_mng.c
SOURCES-$(MPG123) += audio/decode/ad_mpg123.c
SOURCES-$(NEED_GETTIMEOFDAY) += osdep/gettimeofday.c
SOURCES-$(NEED_GLOB) += osdep/glob-win.c
SOURCES-$(PRIORITY) += osdep/priority.c
SOURCES-$(PVR) += stream/stream_pvr.c
SOURCES-$(RADIO) += stream/stream_radio.c
SOURCES-$(RADIO_CAPTURE) += stream/audio_in.c
SOURCES-$(STREAM_CACHE) += stream/cache.c
SOURCES-$(TV) += stream/stream_tv.c stream/tv.c \
stream/frequencies.c stream/tvi_dummy.c \
demux/demux_tv.c
stream/frequencies.c stream/tvi_dummy.c
SOURCES-$(TV_V4L2) += stream/tvi_v4l2.c stream/audio_in.c
SOURCES-$(VCD) += stream/stream_vcd.c
SOURCES-$(DUMMY_OSD) += sub/osd_dummy.c
SOURCES-$(LIBASS_OSD) += sub/osd_libass.c
SOURCES-$(ALSA) += audio/out/ao_alsa.c
SOURCES-$(CACA) += video/out/vo_caca.c
SOURCES-$(SDL2) += audio/out/ao_sdl.c video/out/vo_sdl.c
SOURCES-$(SDL) += audio/out/ao_sdl.c
SOURCES-$(SDL2) += video/out/vo_sdl.c
SOURCES-$(COREAUDIO) += audio/out/ao_coreaudio.c \
audio/out/ao_coreaudio_utils.c \
audio/out/ao_coreaudio_properties.c
SOURCES-$(COREVIDEO) += video/out/vo_corevideo.c
SOURCES-$(DIRECT3D) += video/out/vo_direct3d.c \
video/out/w32_common.c
SOURCES-$(DSOUND) += audio/out/ao_dsound.c
SOURCES-$(WASAPI) += audio/out/ao_wasapi.c
SOURCES-$(GL) += video/out/gl_common.c video/out/gl_osd.c \
video/out/vo_opengl.c video/out/gl_lcms.c \
video/out/gl_video.c video/out/dither.c \
@@ -67,64 +88,59 @@ SOURCES-$(GL) += video/out/gl_common.c video/out/gl_osd.c \
video/out/pnm_loader.c
SOURCES-$(ENCODING) += video/out/vo_lavc.c audio/out/ao_lavc.c \
common/encode_lavc.c
mpvcore/encode_lavc.c
SOURCES-$(GL_WIN32) += video/out/w32_common.c video/out/gl_w32.c
SOURCES-$(GL_X11) += video/out/x11_common.c video/out/gl_x11.c
SOURCES-$(GL_COCOA) += video/out/gl_cocoa.c
SOURCES-$(GL_WAYLAND) += video/out/wayland_common.c \
video/out/gl_wayland.c
SOURCES-$(JACK) += audio/out/ao_jack.c
SOURCES-$(JOYSTICK) += input/joystick.c
SOURCES-$(LIBQUVI4) += stream/resolve/resolve_quvi.c
SOURCES-$(LIBQUVI9) += stream/resolve/resolve_quvi9.c
SOURCES-$(LIRC) += input/lirc.c
SOURCES-$(JOYSTICK) += mpvcore/input/joystick.c
SOURCES-$(LIBQUVI) += mpvcore/resolve_quvi.c
SOURCES-$(LIBQUVI9) += mpvcore/resolve_quvi9.c
SOURCES-$(LIRC) += mpvcore/input/lirc.c
SOURCES-$(OPENAL) += audio/out/ao_openal.c
SOURCES-$(OSS_AUDIO) += audio/out/ao_oss.c
SOURCES-$(OSS) += audio/out/ao_oss.c
SOURCES-$(PULSE) += audio/out/ao_pulse.c
SOURCES-$(PORTAUDIO) += audio/out/ao_portaudio.c
SOURCES-$(RSOUND) += audio/out/ao_rsound.c
SOURCES-$(SNDIO) += audio/out/ao_sndio.c
SOURCES-$(VDPAU) += video/vdpau.c video/vdpau_mixer.c \
video/out/vo_vdpau.c video/decode/vdpau.c \
video/filter/vf_vdpaupp.c
SOURCES-$(VDPAU_GL_X11) += video/out/gl_hwdec_vdpau.c
SOURCES-$(VAAPI) += video/out/vo_vaapi.c \
video/decode/vaapi.c \
video/vaapi.c
SOURCES-$(VAAPI_VPP) += video/filter/vf_vavpp.c
SOURCES-$(VAAPI_GLX) += video/out/gl_hwdec_vaglx.c
SOURCES-$(VDPAU) += video/vdpau.c video/out/vo_vdpau.c
SOURCES-$(VDPAU_DEC) += video/decode/vdpau.c
SOURCES-$(VDPAU_DEC_OLD) += video/decode/vdpau_old.c
SOURCES-$(X11) += video/out/vo_x11.c video/out/x11_common.c
SOURCES-$(XV) += video/out/vo_xv.c
SOURCES-$(WAYLAND) += video/out/vo_wayland.c video/out/wayland_common.c
SOURCES-$(LIBAVFILTER) += video/filter/vf_lavfi.c \
audio/filter/af_lavfi.c
SOURCES-$(VF_LAVFI) += video/filter/vf_lavfi.c
SOURCES-$(AF_LAVFI) += audio/filter/af_lavfi.c
ifeq ($(HAVE_AVUTIL_REFCOUNTING),no)
SOURCES-yes += video/decode/lavc_dr1.c
endif
SOURCES-$(LUA) += player/lua.c
SOURCES-$(VAPOURSYNTH) += video/filter/vf_vapoursynth.c
SOURCES-$(DLOPEN) += video/filter/vf_dlopen.c
SOURCES = audio/audio.c \
audio/audio_buffer.c \
SOURCES = talloc.c \
audio/audio.c \
audio/chmap.c \
audio/chmap_sel.c \
audio/fmt-conversion.c \
audio/format.c \
audio/mixer.c \
audio/reorder_ch.c \
audio/decode/ad_lavc.c \
audio/decode/ad_spdif.c \
audio/decode/dec_audio.c \
audio/filter/af.c \
audio/filter/af_center.c \
audio/filter/af_channels.c \
audio/filter/af_convert24.c \
audio/filter/af_convertsignendian.c \
audio/filter/af_delay.c \
audio/filter/af_dummy.c \
audio/filter/af_equalizer.c \
audio/filter/af_export.c \
audio/filter/af_extrastereo.c \
audio/filter/af_force.c \
audio/filter/af_format.c \
audio/filter/af_hrtf.c \
audio/filter/af_karaoke.c \
@@ -136,105 +152,77 @@ SOURCES = audio/audio.c \
audio/filter/af_sub.c \
audio/filter/af_surround.c \
audio/filter/af_sweep.c \
audio/filter/af_tools.c \
audio/filter/af_drc.c \
audio/filter/af_volume.c \
audio/filter/filter.c \
audio/filter/tools.c \
audio/filter/window.c \
audio/out/ao.c \
audio/out/ao_null.c \
audio/out/ao_pcm.c \
audio/out/pull.c \
audio/out/push.c \
bstr/bstr.c \
common/av_common.c \
common/av_log.c \
common/codecs.c \
common/common.c \
common/msg.c \
common/playlist.c \
common/tags.c \
common/version.c \
demux/codec_tags.c \
demux/demux.c \
demux/demux_edl.c \
demux/demux_cue.c \
demux/demux_disc.c \
demux/demux_lavf.c \
demux/demux_mf.c \
demux/demux_mkv.c \
demux/demux_playlist.c \
demux/demux_raw.c \
demux/demux_subreader.c \
demux/demux_raw.c \
demux/ebml.c \
demux/mf.c \
demux/packet.c \
input/cmd_list.c \
input/cmd_parse.c \
input/event.c \
input/input.c \
input/keycodes.c \
misc/charset_conv.c \
misc/dispatch.c \
misc/rendezvous.c \
misc/ring.c \
options/m_config.c \
options/m_option.c \
options/m_property.c \
options/options.c \
options/parse_commandline.c \
options/parse_configfile.c \
options/path.c \
mpvcore/asxparser.c \
mpvcore/av_common.c \
mpvcore/av_log.c \
mpvcore/av_opts.c \
mpvcore/bstr.c \
mpvcore/charset_conv.c \
mpvcore/codecs.c \
mpvcore/command.c \
mpvcore/cpudetect.c \
mpvcore/m_config.c \
mpvcore/m_option.c \
mpvcore/m_property.c \
mpvcore/mp_common.c \
mpvcore/mp_msg.c \
mpvcore/mp_ring.c \
mpvcore/mplayer.c \
mpvcore/options.c \
mpvcore/parser-cfg.c \
mpvcore/parser-mpcmd.c \
mpvcore/path.c \
mpvcore/playlist.c \
mpvcore/playlist_parser.c \
mpvcore/screenshot.c \
mpvcore/version.c \
mpvcore/input/input.c \
mpvcore/timeline/tl_edl.c \
mpvcore/timeline/tl_matroska.c \
mpvcore/timeline/tl_cue.c \
osdep/io.c \
osdep/numcores.c \
osdep/terminal-unix.c \
osdep/timer.c \
osdep/timer-linux.c \
osdep/threads.c \
player/audio.c \
player/client.c \
player/configfiles.c \
player/command.c \
player/discnav.c \
player/loadfile.c \
player/main.c \
player/misc.c \
player/osd.c \
player/playloop.c \
player/screenshot.c \
player/scripting.c \
player/sub.c \
player/video.c \
player/timeline/tl_matroska.c \
player/timeline/tl_mpv_edl.c \
player/timeline/tl_cue.c \
stream/cache.c \
stream/cache_file.c \
stream/cookies.c \
stream/rar.c \
stream/stream.c \
stream/stream_avdevice.c \
stream/stream_edl.c \
stream/stream_file.c \
stream/stream_lavf.c \
stream/stream_memory.c \
stream/stream_mf.c \
stream/stream_null.c \
stream/stream_rar.c \
sub/dec_sub.c \
sub/draw_bmp.c \
sub/find_subfiles.c \
sub/img_convert.c \
sub/osd.c \
sub/sd_lavc.c \
sub/sd_lavc_conv.c \
sub/sd_lavf_srt.c \
sub/sd_microdvd.c \
sub/sd_movtext.c \
sub/sd_spu.c \
sub/sd_srt.c \
ta/ta.c \
ta/ta_utils.c \
ta/ta_talloc.c \
sub/spudec.c \
sub/sub.c \
video/csputils.c \
video/fmt-conversion.c \
video/image_writer.c \
@@ -243,13 +231,14 @@ SOURCES = audio/audio.c \
video/mp_image_pool.c \
video/sws_utils.c \
video/decode/dec_video.c \
video/decode/vd.c \
video/decode/vd_lavc.c \
video/filter/vf.c \
video/filter/pullup.c \
video/filter/vf_buffer.c \
video/filter/vf_crop.c \
video/filter/vf_delogo.c \
video/filter/vf_divtc.c \
video/filter/vf_down3dright.c \
video/filter/vf_dsize.c \
video/filter/vf_eq.c \
video/filter/vf_expand.c \
@@ -271,6 +260,7 @@ SOURCES = audio/audio.c \
video/filter/vf_sub.c \
video/filter/vf_swapuv.c \
video/filter/vf_unsharp.c \
video/filter/vf_vo.c \
video/filter/vf_yadif.c \
video/out/bitmap_packer.c \
video/out/aspect.c \
@@ -278,15 +268,17 @@ SOURCES = audio/audio.c \
video/out/vo.c \
video/out/vo_null.c \
video/out/vo_image.c \
video/out/win_state.c \
osdep/$(GETCH) \
osdep/$(TIMER) \
$(SOURCES-yes)
OBJECTS += $(addsuffix .o, $(basename $(SOURCES)))
OBJECTS-$(PE_EXECUTABLE) += osdep/mpv-rc.o
OBJECTS += $(OBJECTS-yes)
DEP_FILES = $(patsubst %.S,%.d,$(patsubst %.cpp,%.d,$(patsubst %.c,%.d,$(SOURCES:.m=.d) $(SOURCES:.m=.d))))
ALL_TARGETS += mpv
ALL_TARGETS += mpv$(EXESUF)
INSTALL_BIN += install-mpv
INSTALL_BIN_STRIP += install-mpv-strip
@@ -294,7 +286,7 @@ INSTALL_MAN =
ifeq ($(BUILD_MAN),yes)
INSTALL_MAN += install-mpv-man
ALL_TARGETS += DOCS/man/mpv.1
ALL_TARGETS += DOCS/man/en/mpv.1
endif
DIRS = . \
@@ -302,21 +294,13 @@ DIRS = . \
audio/decode \
audio/filter \
audio/out \
bstr \
common \
compat \
input \
player/timeline \
core \
mpvcore/input \
mpvcore/timeline \
demux \
misc \
options \
osdep \
osdep/ar \
player \
stream \
stream/resolve \
sub \
ta \
video \
video/decode \
video/filter \
@@ -325,6 +309,7 @@ DIRS = . \
ADDSUFFIXES = $(foreach suf,$(1),$(addsuffix $(suf),$(2)))
ADD_ALL_DIRS = $(call ADDSUFFIXES,$(1),$(DIRS))
ADD_ALL_EXESUFS = $(1) $(call ADDSUFFIXES,$(EXESUFS_ALL),$(1))
###### brief build output #######
@@ -340,14 +325,27 @@ all: $(ALL_TARGETS)
%.1: %.rst
$(RST2MAN) $< $@
%.o: %.S
$(CC) $(DEPFLAGS) $(CFLAGS) -c -o $@ $<
%.o: %.c
$(CC) $(DEPFLAGS) $(CFLAGS) -c -o $@ $<
mpv: $(OBJECTS) player/main_fn.o
%.o: %.cpp
$(CC) $(DEPFLAGS) $(CXXFLAGS) -c -o $@ $<
%.o: %.m
$(CC) $(DEPFLAGS) $(CFLAGS) -c -o $@ $<
%-rc.o: %.rc
$(WINDRES) -I. $< $@
mpv$(EXESUF): $(OBJECTS)
mpv$(EXESUF):
$(CC) -o $@ $^ $(EXTRALIBS)
input/input.c: input/input_conf.h
input/input_conf.h: TOOLS/file2string.pl etc/input.conf
mpvcore/input/input.c: mpvcore/input/input_conf.h
mpvcore/input/input_conf.h: TOOLS/file2string.pl etc/input.conf
./$^ >$@
MKVLIB_DEPS = TOOLS/lib/Parse/Matroska.pm \
@@ -368,25 +366,10 @@ video/out/gl_video.c: video/out/gl_video_shaders.h
video/out/gl_video_shaders.h: TOOLS/file2string.pl video/out/gl_video_shaders.glsl
./$^ >$@
video/out/x11_common.c: video/out/x11_icon.inc
video/out/x11_icon.inc: TOOLS/file2string.pl video/out/x11_icon.bin
./$^ >$@
sub/osd_libass.c: sub/osd_font.h
sub/osd_font.h: TOOLS/file2string.pl sub/osd_font.otf
./$^ >$@
player/lua/%.inc: TOOLS/file2string.pl player/lua/%.lua
./$^ >$@
player/lua.c: player/lua/defaults.inc \
player/lua/assdraw.inc \
player/lua/osc.inc \
player/lua/options.inc
etc/_mpv: TOOLS/zsh.pl ./mpv
./$< > $@
# ./configure must be rerun if it changed
config.mak: configure
@echo "############################################################"
@@ -399,98 +382,79 @@ version.h .version: version.sh
# Force version.sh to run to potentially regenerate version.h
-include .version
%: %.c
%$(EXESUF): %.c
$(CC) $(CFLAGS) -o $@ $^
###### dependency declarations / specific CFLAGS ######
common/version.c: version.h
mpvcore/version.c osdep/mpv-rc.o: version.h
osdep/mpv-rc.o: osdep/mpv.exe.manifest etc/mpv-icon.ico
DOCS/man/en/mpv.1: DOCS/man/en/af.rst \
DOCS/man/en/ao.rst \
DOCS/man/en/changes.rst \
DOCS/man/en/encode.rst \
DOCS/man/en/input.rst \
DOCS/man/en/options.rst \
DOCS/man/en/vf.rst \
DOCS/man/en/vo.rst
DOCS/man/mpv.1: DOCS/man/af.rst \
DOCS/man/ao.rst \
DOCS/man/changes.rst \
DOCS/man/encode.rst \
DOCS/man/input.rst \
DOCS/man/options.rst \
DOCS/man/vf.rst \
DOCS/man/vo.rst
###### installation / clean / generic rules #######
install: $(INSTALL_BIN) install-data $(INSTALL_MAN)
install-no-man: $(INSTALL_BIN) install-data
install-strip: $(INSTALL_BIN_STRIP) install-data $(INSTALL_MAN)
install-strip-no-man: $(INSTALL_BIN_STRIP) install-data
install: $(INSTALL_BIN) $(INSTALL_MAN)
install-no-man: $(INSTALL_BIN)
install-strip: $(INSTALL_BIN_STRIP) $(INSTALL_MAN)
install-strip-no-man: $(INSTALL_BIN_STRIP)
install-dirs:
if test ! -d $(BINDIR) ; then $(INSTALL) -d $(BINDIR) ; fi
install-%: % install-dirs
install-%: %$(EXESUF) install-dirs
$(INSTALL) -m 755 $< $(BINDIR)
install-%-strip: % install-dirs
install-%-strip: %$(EXESUF) install-dirs
$(INSTALL) -m 755 -s $< $(BINDIR)
install-mpv-man: install-mpv-man-en
install-mpv-man-en: DOCS/man/mpv.1
install-mpv-man-en: DOCS/man/en/mpv.1
if test ! -d $(MANDIR)/man1 ; then $(INSTALL) -d $(MANDIR)/man1 ; fi
$(INSTALL) -m 644 DOCS/man/mpv.1 $(MANDIR)/man1/
ICONSIZES = 16x16 32x32 64x64
define ICON_INSTALL_RULE
install-mpv-icon-$(size): etc/mpv-icon-8bit-$(size).png
$(INSTALL) -d $(prefix)/share/icons/hicolor/$(size)/apps
$(INSTALL) -m 644 etc/mpv-icon-8bit-$(size).png $(prefix)/share/icons/hicolor/$(size)/apps/mpv.png
endef
$(foreach size,$(ICONSIZES),$(eval $(ICON_INSTALL_RULE)))
install-mpv-icons: $(foreach size,$(ICONSIZES),install-mpv-icon-$(size))
install-mpv-desktop: etc/mpv.desktop
$(INSTALL) -d $(prefix)/share/applications
$(INSTALL) -m 644 etc/mpv.desktop $(prefix)/share/applications/
install-mpv-config: etc/encoding-profiles.conf
$(INSTALL) -d $(CONFDIR)
$(INSTALL) -m 644 etc/encoding-profiles.conf $(CONFDIR)
install-mpv-zsh: etc/_mpv
$(INSTALL) -d $(prefix)/share/zsh/vendor-completions
$(INSTALL) -m 644 etc/_mpv $(prefix)/share/zsh/vendor-completions/
install-data: install-mpv-icons install-mpv-desktop install-mpv-config install-mpv-zsh
$(INSTALL) -m 644 DOCS/man/en/mpv.1 $(MANDIR)/man1/
uninstall:
$(RM) $(BINDIR)/mpv
$(RM) $(MANDIR)/man1/mpv.1 $(MANDIR)/man1/mpv.1
$(RM) $(prefix)/share/applications/mpv.desktop
$(RM) $(prefix)/share/zsh/vendor-completions/_mpv
$(RM) $(foreach size,$(ICONSIZES),$(prefix)/share/icons/hicolor/$(size)/apps/mpv.png)
$(RM) $(BINDIR)/mpv$(EXESUF)
$(RM) $(MANDIR)/man1/mpv.1
$(RM) $(MANDIR)/en/man1/mpv.1
clean:
-$(RM) $(call ADD_ALL_DIRS,/*.o /*.d /*.a /*.ho /*~)
-$(RM) $(call ADD_ALL_DIRS,/*.o /*.a /*.ho /*~)
-$(RM) mpv
-$(RM) DOCS/man/*/mpv.1
-$(RM) $(call ADD_ALL_EXESUFS,mpv)
-$(RM) DOCS/man/en/mpv.1
-$(RM) version.h
-$(RM) input/input_conf.h
-$(RM) mpvcore/input/input_conf.h
-$(RM) video/out/vdpau_template.c
-$(RM) demux/ebml_types.h demux/ebml_defs.c
-$(RM) video/out/gl_video_shaders.h
-$(RM) video/out/x11_icon.inc
-$(RM) sub/osd_font.h
-$(RM) player/lua/defaults.inc
-$(RM) player/lua/assdraw.inc
-$(RM) player/lua/osc.inc
-$(RM) player/lua/options.inc
distclean: clean
-$(RM) config.log old_build/config.h old_build/config.mak Makefile
-rmdir old_build/
-$(RM) config.log config.mak config.h TAGS tags
TAGS:
$(RM) $@; find . -name '*.[chS]' -o -name '*.asm' | xargs etags -a
tags:
$(RM) $@; find . -name '*.[chS]' -o -name '*.asm' | xargs ctags -a
osxbundle:
@TOOLS/osxbundle.py mpv
osxbundle-skip-deps:
@TOOLS/osxbundle.py --skip-deps mpv
-include $(DEP_FILES)
+26 -82
View File
@@ -16,52 +16,25 @@ Compilation
-----------
Compiling with full features requires development files for several
external libraries. Below is a list of some important requirements.
The mpv build system uses *waf* but we don't store it in your source tree. The
script './bootstrap.py' will download the latest version of waf that was tested
with the build system.
For a list of the available build options use `./waf configure --help`. If
you think you have support for some feature installed but configure fails to
detect it, the file `build/config.log` may contain information about the
reasons for the failure.
NOTE: To avoid cluttering the output with unreadable spam, `--help` only shows
one of the two switches for each option. If the option is autodetected by
default, the `--disable-***` switch is printed; if the option is disabled by
default, the `--enable-***` switch is printed. Either way, you can use
`--enable-***` or `--disable-**` regardless of what is printed by `--help`.
To build the software you can use `./waf build`: the result of the compilation
will be located in `build/mpv`. You can use `./waf install` to install mpv
to the *prefix* after it is compiled.
NOTE: Using the old build system (with `./old-configure`) should still work,
but will be removed in a future version of mpv.
external libraries. Below is a list of some important requirements. For
more information see the output of `./configure --help` for a list of options,
or look at the list of enabled and disabled features printed after running
`./configure`. If you think you have support for some feature installed
but configure fails to detect it, the file `config.log` may contain
information about the reasons for the failure.
Essential dependencies (incomplete list):
- gcc or clang
- X development headers (xlib, X extensions, libvdpau, libGL, libXv, ...)
- Audio output development headers (libasound, pulseaudio)
- FFmpeg libraries (libavutil libavcodec libavformat libswscale libavresample)
At least FFmpeg 2.1.4 or Libav 10 is required.
- libjpeg (for screenshots)
- libquvi if you want to play Youtube videos directly
Libass dependencies:
- gcc or clang, yasm on x86 and x86_64
- fribidi, freetype, fontconfig development headers (for libass)
FFmpeg dependencies:
- gcc or clang, yasm on x86 and x86_64
- OpenSSL (has to be explicitly enabled when compiling ffmpeg)
- libass
- FFmpeg libraries (libavutil libavcodec libavformat libswscale libpostproc)
- libjpeg
- libquvi if you want to play Youtube videos directly
- libx264/libmp3lame/libfdk-aac if you want to use encoding (has to be
explicitly enabled when compiling ffmpeg)
- Libav also works, but some features will not work. (See section below.)
Most of the above libraries are available in suitable versions on normal
Linux distributions. However FFmpeg is an exception (distro versions may be
@@ -74,37 +47,22 @@ If you are running Mac OSX and using homebrew we provide [homebrew-mpv][homebrew
to date formula that compiles mpv with sensible dependencies and defaults for
OSX.
FFmpeg vs. Libav
----------------
configure --enable-* parameters
-------------------------------
Generally, mpv should work with the latest release as well as the git version
of both FFmpeg and Libav. But FFmpeg is preferred, and some mpv features work
with FFmpeg only. See the [wiki article][ffmpeg_vs_libav] about the issue.
Release cycle
-------------
Every few months, a new release is cut off of the master branch and is assigned
a 0.X.0 version number.
As part of the maintenance process, minor releases are made, which are assigned
0.X.Y version numbers. Minor releases contain bug fixes only. They never merge
the master branch, and no features are added to it. Only the latest release is
maintained.
The goal of releases is to provide stability and an unchanged base for the sake
of Linux distributions. If you want the newest features, just use the master
branch, which is stable most of the time, except sometimes, when it's not.
Releases other than the latest release are unsupported and unmaintained.
See the [release policy document][release-policy] for more information.
The --enable-* parameters unconditionally force options on, completely
skipping autodetection. This behavior is unlike what you may be used to from
autoconf-based configure scripts that can decide to override you. This greater
level of control comes at a price. You may have to provide the correct compiler
and linker flags yourself.
If you used one of these options and experience a compilation or
linking failure, make sure you have passed the necessary compiler/linker flags
to configure.
Bug reports
-----------
Please use the [issue tracker][issue-tracker] provided by GitHub to send us bug
Please use the [issue tracker][issue tracker] provided by GitHub to send us bug
reports or feature requests.
Contributing
@@ -114,26 +72,12 @@ For small changes you can just send us pull requests through GitHub. For bigger
changes come and talk to us on IRC before you start working on them. It will
make code review easier for both parties later on.
Contact
-------
Contacts
--------
These forms of contact are meant to ask questions about mpv usage, give
feedback on mpv and discuss it's development.
You can find us on IRC in `#mpv-player` on `irc.freenode.net`
If possible, please avoid posting bugs here and use the [issue tracker][issue-tracker]
instead.
- **Users IRC Channel**: `#mpv-player` on `irc.freenode.net`
- **Users Mailing List**: `mpv-users@googlegroups.com` ([Archive / Subscribe][mpv-users]).
- **Devel Mailing List**: `mpv-devel@googlegroups.com` ([Archive / Subscribe][mpv-devel])
To contact the `mpv` team in private write to `mpv-team@googlegroups.com`.
[changes]: https://github.com/mpv-player/mpv/blob/master/DOCS/man/changes.rst
[changes]: https://github.com/mpv-player/mpv/blob/master/DOCS/man/en/changes.rst
[mpv-build]: https://github.com/mpv-player/mpv-build
[homebrew-mpv]: https://github.com/mpv-player/homebrew-mpv
[issue-tracker]: https://github.com/mpv-player/mpv/issues
[mpv-users]: https://groups.google.com/forum/?hl=en#!forum/mpv-users
[mpv-devel]: https://groups.google.com/forum/?hl=en#!forum/mpv-devel
[ffmpeg_vs_libav]: https://github.com/mpv-player/mpv/wiki/FFmpeg-versus-Libav
[release-policy]: https://github.com/mpv-player/mpv/blob/master/DOCS/release-policy.md
[issue tracker]: https://github.com/mpv-player/mpv/issues
-146
View File
@@ -1,146 +0,0 @@
Release 0.5.2
=============
Changes
-------
- Inherit event flags from parent X11 window when using --wid.
- Properly resume playlists loaded at runtime (this makes mpv try to resume a
playlist when loaded like "mpv playlist.m3u").
- Don't honor $XDG_CONFIG_DIRS anymore for global paths (this fixes a regression
on systems that set the variable by default to somewhere else then /etc,
making mpv skip previously created config files).
- Allow up to 256 MB of extradata to make broken files work.
Bug fixes
---------
- Restore old speed change behavior (this fixes a temporary lack of sound after
restoring normal speed e.g. when pressing Backspace).
- Fix quit handling when seeking past EOF while paused.
- Prevent burning CPU when seeking while paused.
- Don't add the playlist base path twice when using --playlist.
- Fix a crash when the initialization of the EGL context fails on Wayland.
- Make EOF handling properly event-based (this fixes a bug that made mpv hang
when changing from one file to the next with --gapless-audio=no).
- Assume icy title data is terminated with ';' (this fixes handling of streams
with metadata containing apostrophes).
This listing is not complete. There are many more bug fixes and changes. The
complete change log can be viewed by running ``git log 0f1d6c0..e3d3a47`` in
the git repository.
Release 0.5.1
=============
Changes
-------
- Add audio/flac to the MIME type list in mpv.desktop.
- The --slave-broken option has been reintroduced and is equivalent to using
--input-file=/dev/stdin and --input-terminal=no.
Bug fixes
---------
- Fix screenshots with anamorphic videos in the vdpau vo.
- Fix a crash happening when mpv fails to create a vdpau mixer in the vdpau vo.
- Don't quit when the --loop-file option is used.
- Always load the cursor theme in the wayland vo (this fixes a crash caused by
moving the mouse cursor over the video).
- Properly return non-zero from the zsh completion script when no matches are
found (this fixes the use of user-defined zsh completion matchers).
- Fix the description of the opengl-hq vo default options in the manpage.
- Remove duplicated content from the manpage that was accidentally added in the
manpage reorganization from the previous release.
- Avoid clobbering playback position on video end (when both audio and video
streams reached the end, the playback poisition was briefly set at 0).
- Avoid overwriting user input in the coreaudio_device ao.
This listing is not complete. There are many more bug fixes and changes. The
complete change log can be viewed by running ``git log 6729e39..fdd648b`` in
the git repository.
Release 0.5.0
=============
Changes
-------
- Support setting the start time relative to the start PTS with --start.
- Use /dev/sr0 as default cdrom/dvd device on Linux.
- Change "cache" property representation to float.
- Split the ao_coreaudio audio output into ao_coreaudio (which uses the
AudioUnits and the CoreAudio sound server) and ao_coreaudio_device (which uses
direct device access).
- Complete supported URL schemes in the zsh completion script.
- Enable compiler optimizations (-O2) during build.
- Allow precise seeking with percent-based seeks too.
- Do precise seeks when clicking on the OSC seekbar.
- Remove support for older VDA hwaccel versions on Mac OS X (requires recent
versions of ffmpeg/libav).
- Remove the last remains of slave mode (--slave-broken option). Support for
reading input commands from stdin can be emulated using: --input-terminal=no
--input-file=/dev/stdin.
- Some key binding were remapped or removed:
+ "LEFT" and "RIGHT" have been remapped to seek 5 seconds instead of 10.
+ "c" for changing the YUV color space was unmapped.
+ OSD level cycling has been remapped from "o" to "O".
+ "o" is now mapped to the show_progress command (like "P").
+ "PGUP" and "PGDWN" have been remapped to chapter-based seeks (the old
bindings "@" and "!" are still available). The old behvaiour has been
remapped to "Shift+PGUP" and "Shift+PGDWM".
+ Increasing/decreasing audio delay have been remapped from "+" and "-" to
"Ctrl++" and "Ctrl+-".
+ "(" and ")" for adjusting audio balance have been unmapped.
+ "F" for toggling the "sub-forced-only" option has been unmapped.
+ "TAB" for cycling TS programs has been unmapped.
+ "A" for cycling DVD angle has been unmapped.
+ "U" for stopping playback has been unmapped.
+ "l" and "L" for cycling quvi format have been unmapped.
- Make it possible to build the client API (libmpv) on Mac OS X.
- Do necessary checks for nested Wayland compositors which only have limited
interfaces.
- The options in the manpage have been re-organized into sections and ordered by
usage to make it easier to navigate.
- The "hard" framedrop mode has been removed (this behaviour is still available
via the --vd-lavc-framedrop option).
- Show the basename of a subtitle file only (without the directory part).
Bug fixes
---------
- Use a default color space for files with no video (this fixes color managed
OSD messages when no video channel is present).
- Fix key equivalent dispatching for Cocoa on Mac OS X.
- Disable non-interleaved audio formats by default with ao_alsa, since some
ALSA plugins don't cope well with them (non-interleaved formats can still be
enabled using the "non-interleaved" ao_alsa option).
- Fix deadlock in the client API when calling mpv_terminate_destroy() before
mpv_initialize().
- Fix flushing of filter chain (this fixes a bug where the last frames would get
dropped on EOF, if buffered by filters).
- Actually use the channels.conf configuration file for DVB.
- Restore the --dvbin-file option.
- Fix off by one error in the command parser which caused a segfault when using
the mp.commandv() Lua function with 10 or more arguments.
New features
------------
- Search for config files in $XDG_CONFIG_HOME and $XDG_CONFIG_DIRS in addition
to ~/.mpv. The config file has also been renamed to mpv.conf (although the
old file still works and will be loaded even if the new one exists).
- Add "playback-time" property that returns the current playback time always
starting at 0. This is now used to show the playback time in the OSD and the
status line, fixing the shown playback time for transport streams that start
at a completely random timestamp.
- Add cache-related "cache-free", "cache-used" and "cache-idle" properties.
- Add option to list supported protocols (--list-protocols).
- Allow setting libavformat stream options with the --stream-lavf-o option.
- Support running the demuxer in a separate thread and prefetch packets (this
can be enabled using the --demuxer-thread option).
This listing is not complete. There are many more bug fixes and changes. The
complete change log can be viewed by running ``git log cb250d4..52f2f67`` in
the git repository.
-93
View File
@@ -1,93 +0,0 @@
#!/usr/bin/env python
import re
import os
import sys
import shutil
import subprocess
from functools import partial
sys_re = re.compile("^/System")
usr_re = re.compile("^/usr/lib/")
exe_re = re.compile("@executable_path")
def is_user_lib(objfile, libname):
return not sys_re.match(libname) and \
not usr_re.match(libname) and \
not exe_re.match(libname) and \
not "libobjc." in libname and \
not "libSystem." in libname and \
not "libc." in libname and \
not "libgcc." in libname and \
not os.path.basename(objfile) in libname
def otool(objfile):
command = "otool -L %s | grep -e '\t' | awk '{ print $1 }'" % objfile
output = subprocess.check_output(command, shell = True)
return filter(partial(is_user_lib, objfile), output.split())
def install_name_tool_change(old, new, objfile):
subprocess.call(["install_name_tool", "-change", old, new, objfile])
def install_name_tool_id(name, objfile):
subprocess.call(["install_name_tool", "-id", name, objfile])
def libraries(objfile, result = dict()):
libs_list = otool(objfile)
result[objfile] = set(libs_list)
for lib in libs_list:
if lib not in result:
libraries(lib, result)
return result
def leafs(libs_dict, processed = []):
result = []
processed = set(processed)
for objfile, libs in libs_dict.items():
if libs <= processed:
result.append(objfile)
return result
def lib_path(binary):
return os.path.join(os.path.dirname(binary), 'lib')
def lib_name(lib):
return os.path.join("@executable_path", "lib", os.path.basename(lib))
def process_libraries(libs_dict, binary, processed = []):
ls = leafs(libs_dict, processed)
diff = set(ls) - set(processed)
if diff == set([binary]):
return
for src in diff:
name = lib_name(src)
dst = os.path.join(lib_path(binary), os.path.basename(src))
shutil.copy(src, dst)
os.chmod(dst, 0o755)
install_name_tool_id(name, dst)
if src in libs_dict[binary]:
install_name_tool_change(src, name, binary)
for p in processed:
if p in libs_dict[src]:
install_name_tool_change(p, lib_name(p), dst)
process_libraries(libs_dict, binary, ls)
def main():
binary = os.path.abspath(sys.argv[1])
if not os.path.exists(lib_path(binary)):
os.makedirs(lib_path(binary))
libs = libraries(binary)
print(libs)
process_libraries(libs, binary)
if __name__ == "__main__":
main()
-15
View File
@@ -1,15 +0,0 @@
#!/bin/sh
# This script is expected to be called as TOOLS/gen-x11-icon.sh (it will access
# etc/mpv-icon...), and it will write video/out/x11_icon.bin.
conv() {
echo
identify -format "icon: %w %h" $1
convert $1 -depth 8 rgba:-
}
(echo "# File generated by gen-x11-icon.sh" ;
conv etc/mpv-icon-8bit-16x16.png ;
conv etc/mpv-icon-8bit-32x32.png ;
conv etc/mpv-icon-8bit-64x64.png) | gzip -c > video/out/x11_icon.bin
-161
View File
@@ -1,161 +0,0 @@
#!/bin/sh
case "$0" in
*/*)
MYDIR=${0%/*}
;;
*)
MYDIR=.
;;
esac
: ${MPV:=mpv}
: ${ILDETECT_MPV:=$MPV}
: ${ILDETECT_MPV:=$MPV}
: ${ILDETECT_MPVFLAGS:=--start=35% --length=35}
: ${ILDETECT_DRY_RUN:=}
: ${ILDETECT_QUIET:=}
: ${ILDETECT_RUN_INTERLACED_ONLY:=}
: ${ILDETECT_FORCE_RUN:=}
: ${MAKE:=make}
# exit status:
# 0 progressive
# 1 telecine
# 2 interlaced
# 8 unknown
# 16 detect fail
# 17+ mpv's status | 16
testfun()
{
$ILDETECT_MPV "$@" \
--vf-add=lavfi="[idet]" --msg-level ffmpeg=v \
--o= --vo=null --no-audio --untimed \
$ILDETECT_MPVFLAGS \
| { if [ -n "$ILDETECT_QUIET" ]; then cat; else tee /dev/stderr; fi } \
| grep "Parsed_idet_0: Multi frame detection: " | tail -n 1
}
judge()
{
out=`testfun "$@"`
tff=${out##* TFF:}; tff=${tff%% *}
bff=${out##* BFF:}; bff=${bff%% *}
progressive=${out##* Progressive:}; progressive=${progressive%% *}
undetermined=${out##* Undetermined:}; undetermined=${undetermined%% *}
case "$tff$bff$progressive$undetermined" in
*[!0-9]*)
echo >&2 "ERROR: Unrecognized idet output: $out"
exit 16
;;
esac
interlaced=$((bff + tff))
determined=$((interlaced + progressive))
if [ $undetermined -gt $determined ] || [ $determined -lt 250 ]; then
echo >&2 "ERROR: Less than 50% or 250 frames are determined."
[ -n "$ILDETECT_FORCE_RUN" ] || exit 8
echo >&2 "Assuming interlacing."
if [ $tff -gt $((bff * 10)) ]; then
verdict=interlaced-tff
elif [ $bff -gt $((tff * 10)) ]; then
verdict=interlaced-bff
else
verdict=interlaced
fi
elif [ $((interlaced * 20)) -gt $progressive ]; then
# At least 5% of the frames are interlaced!
if [ $tff -gt $((bff * 10)) ]; then
verdict=interlaced-tff
elif [ $bff -gt $((tff * 10)) ]; then
verdict=interlaced-bff
else
echo >&2 "ERROR: Content is interlaced, but can't determine field order."
[ -n "$ILDETECT_FORCE_RUN" ] || exit 8
echo >&2 "Assuming interlacing with default field order."
verdict=interlaced
fi
else
# Likely progrssive.
verdict=progressive
fi
echo "$verdict"
}
judge "$@" --vf-clr
case "$verdict" in
progressive)
[ -n "$ILDETECT_DRY_RUN" ] || \
[ -n "$ILDETECT_RUN_INTERLACED_ONLY" ] || \
$ILDETECT_MPV "$@"
r=$?
[ $r -eq 0 ] || exit $(($r | 16))
exit 0
;;
interlaced-tff)
judge "$@" --vf-clr --vf-pre=pullup --field-dominance=top
case "$verdict" in
progressive)
[ -n "$ILDETECT_DRY_RUN" ] || \
$ILDETECT_MPV "$@" --vf-pre=pullup --field-dominance=top
r=$?
[ $r -eq 0 ] || exit $(($r | 16))
exit 1
;;
*)
[ -n "$ILDETECT_DRY_RUN" ] || \
$ILDETECT_MPV "$@" --vf-pre=yadif --field-dominance=top
r=$?
[ $r -eq 0 ] || exit $(($r | 16))
exit 2
;;
esac
;;
interlaced-bff)
judge "$@" --vf-clr --vf-pre=pullup --field-dominance=bottom
case "$verdict" in
progressive)
[ -n "$ILDETECT_DRY_RUN" ] || \
$ILDETECT_MPV "$@" --vf-pre=pullup --field-dominance=bottom
r=$?
[ $r -eq 0 ] || exit $(($r | 16))
exit 1
;;
*)
[ -n "$ILDETECT_DRY_RUN" ] || \
$ILDETECT_MPV "$@" --vf-pre=yadif --field-dominance=bottom
r=$?
[ $r -eq 0 ] || exit $(($r | 16))
exit 2
;;
esac
;;
interlaced)
judge "$@" --vf-clr --vf-pre=pullup
case "$verdict" in
progressive)
[ -n "$ILDETECT_DRY_RUN" ] || \
$ILDETECT_MPV "$@" --vf-pre=pullup
r=$?
[ $r -eq 0 ] || exit $(($r | 16))
exit 1
;;
*)
[ -n "$ILDETECT_DRY_RUN" ] || \
$ILDETECT_MPV "$@" --vf-pre=yadif
r=$?
[ $r -eq 0 ] || exit $(($r | 16))
exit 2
;;
esac
;;
*)
echo >&2 "ERROR: Internal error."
exit 16
;;
esac
+17 -20
View File
@@ -132,8 +132,7 @@ sub _bin2int($) {
# creates a floating-point number with the given mantissa and exponent
sub _ldexp {
my ($mantissa, $exponent) = @_;
my $r = new Math::BigRat($mantissa);
return $r * Math::BigRat->new(2)**$exponent;
return $mantissa * Math::BigRat->new(2)**$exponent;
}
# NOTE: the read_* functions are hard to read because they're ports
@@ -253,27 +252,25 @@ bytes from the internal filehandle.
Only lengths C<4> and C<8> are supported (C C<float> and C<double>).
=cut
{
my $b1 = new Math::BigInt 1;
sub read_float {
my ($self, $length) = @_;
my $i = $self->read_uint($length);
my $f;
sub read_float {
my ($self, $length) = @_;
my $i = new Math::BigInt $self->read_uint($length)->bstr;
my $f;
use bigrat try => BIGINT_TRY;
# These evil expressions reinterpret an unsigned int as IEEE binary floats
if ($length == 4) {
$f = _ldexp(($i & ((1<<23) - 1)) + (1<<23), ($i>>23 & ((1<<8) - 1)) - 150);
$f = -$f if $i & ($b1<<31);
} elsif ($length == 8) {
$f = _ldexp(($i & (($b1<<52) - 1)) + ($b1<<52), ($i>>52 & ((1<<12) - 1)) - 1075);
$f = -$f if $i & ($b1<<63);
} else {
croak "Matroska Syntax error: unsupported IEEE float byte size $length";
}
return $f;
# These evil expressions reinterpret an unsigned int as IEEE binary floats
if ($length == 4) {
$f = _ldexp(($i & (1<<23 - 1)) + (1<<23), ($i>>23 & (1<<8 - 1)) - 150);
$f = -$f if $i & (1<<31);
} elsif ($length == 8) {
$f = _ldexp(($i & (1<<52 - 1)) + (1<<52), ($i>>52 & (1<<12 - 1)) - 1075);
$f = -$f if $i & (1<<63);
} else {
croak "Matroska Syntax error: unsupported IEEE float byte size $length";
}
return $f;
}
=method read_ebml_id($length)
-19
View File
@@ -1,19 +0,0 @@
mpv lua scripts
===============
The lua scripts in this folder can be loaded on a one-time basis by
adding the option
--lua=/path/to/script.lua
to mpv's command line.
Unless otherwise specified, they are also suitable for inclusion in
the `~/.mpv/lua` directory where they will be loaded every time mpv
starts, obviating the need to load them with the above `--lua=...`
argument. This is acceptable as they do only basic argument parsing
and key-binding registration, until those bound keys are actually
pressed. They should therefore not interfere with normal playback
(unless you have a conflicting user-defined key-binding, in which
case, you may want to modify either the `mp.add_key_binding()` calls
in the scripts, or your keybinding).
-113
View File
@@ -1,113 +0,0 @@
-- This script uses the lavfi cropdetect filter to automatically
-- insert a crop filter with appropriate parameters for the currently
-- playing video.
--
-- It registers the key-binding "C" (shift+c), which when pressed,
-- inserts the filter vf=lavfi=cropdetect. After 1 second, it then
-- inserts the filter vf=crop=w:h:x:y, where w,h,x,y are determined
-- from the vf-metadata gathered by cropdetect. The cropdetect filter
-- is removed immediately after the crop filter is inserted as it is
-- no longer needed.
--
-- If the "C" key is pressed again, the crop filter is removed
-- restoring playback to its original state.
--
-- Since the crop parameters are determined from the 1 second of video
-- between inserting the cropdetect and crop filters, the "C" key
-- should be pressed at a position in the video where the crop region
-- is unambiguous (i.e., not a black frame, black background title
-- card, or dark scene).
--
-- The default delay between insertion of the cropdetect and
-- crop filters may be overridden by adding
--
-- --lua-opts=autocrop.detect_seconds=<number of seconds>
--
-- to mpv's arguments. This may be desirable to allow cropdetect more
-- time to collect data.
require "mp.msg"
script_name = mp.get_script_name()
cropdetect_label = string.format("%s-cropdetect", script_name)
crop_label = string.format("%s-crop", script_name)
-- number of seconds to gather cropdetect data
detect_seconds = tonumber(mp.get_opt(string.format("%s.detect_seconds", script_name)))
if not detect_seconds then
detect_seconds = 1
end
function del_filter_if_present(label)
-- necessary because mp.command('vf del @label:filter') raises an
-- error if the filter doesn't exist
local vfs = mp.get_property_native("vf")
for i,vf in pairs(vfs) do
if vf["label"] == label then
table.remove(vfs, i)
mp.set_property_native("vf", vfs)
return true
end
end
return false
end
function autocrop_start()
-- exit if cropdetection is already in progress
if timer then
mp.msg.warn("already cropdetecting!")
return
end
-- if there's a crop filter, remove it and exit
if del_filter_if_present(crop_label) then
return
end
-- insert the cropdetect filter
ret=mp.command(
string.format(
'vf add @%s:lavfi=graph="cropdetect=limit=24:round=2:reset=0"',
cropdetect_label
)
)
-- wait to gather data
timer=mp.add_timeout(detect_seconds, do_crop)
end
function do_crop()
-- get the metadata
local cropdetect_metadata = mp.get_property_native(
string.format("vf-metadata/%s", cropdetect_label)
)
-- use it to crop if its valid
if cropdetect_metadata then
if cropdetect_metadata["lavfi.cropdetect.w"]
and cropdetect_metadata["lavfi.cropdetect.h"]
and cropdetect_metadata["lavfi.cropdetect.x"]
and cropdetect_metadata["lavfi.cropdetect.y"]
then
mp.command(string.format("vf add @%s:crop=%s:%s:%s:%s",
crop_label,
cropdetect_metadata["lavfi.cropdetect.w"],
cropdetect_metadata["lavfi.cropdetect.h"],
cropdetect_metadata["lavfi.cropdetect.x"],
cropdetect_metadata["lavfi.cropdetect.y"]))
else
mp.msg.error(
"Got empty crop data. You might need to increase detect_seconds."
)
end
else
mp.msg.error(
"No crop data. Was the cropdetect filter successfully inserted?"
)
mp.msg.error(
"Does your version of ffmpeg/libav support AVFrame metadata?"
)
end
-- remove the cropdetect filter
del_filter_if_present(cropdetect_label)
timer=nil
end
mp.add_key_binding("C", "auto_crop", autocrop_start)
-71
View File
@@ -1,71 +0,0 @@
-- This script automatically loads playlist entries before and after the
-- the currently played file. It does so by scanning the directory a file is
-- located in when starting playback. It sorts the directory entries
-- alphabetically, and adds entries before and after the current file to
-- the internal playlist. (It stops if the it would add an already existing
-- playlist entry at the same position - this makes it "stable".)
-- Add at most 5 * 2 files when starting a file (before + after).
MAXENTRIES = 5
mputils = require 'mp.utils'
function add_files_at(index, files)
index = index - 1
local oldcount = mp.get_property_number("playlist-count", 1)
for i = 1, #files do
mp.commandv("loadfile", files[i], "append")
mp.commandv("playlist_move", oldcount + i - 1, index + i - 1)
end
end
function find_and_add_entries()
local path = mp.get_property("path", "")
local dir, filename = mputils.split_path(path)
if #dir == 0 then
return
end
local files = mputils.readdir(dir, "files")
table.sort(files)
local pl = mp.get_property_native("playlist", {})
local pl_current = mp.get_property_number("playlist-pos", 0) + 1
-- Find the current pl entry (dir+"/"+filename) in the sorted dir list
local current
for i = 1, #files do
if files[i] == filename then
current = i
break
end
end
if current == nil then
return
end
local append = {[-1] = {}, [1] = {}}
for dir = -1, 1, 2 do -- 2 iterations, with dir = -1 and +1
for i = 1, MAXENTRIES do
local file = files[current + i * dir]
local pl_e = pl[pl_current + i * dir]
if file == nil or file[1] == "." then
break
end
if pl_e then
-- If there's a playlist entry, and it's the same file, stop.
if pl_e.filename == file then
break
end
end
if dir == -1 then
if pl_current == 1 then -- never add additional entries in the middle
mp.msg.info("Prepending " .. file)
table.insert(append[-1], 1, file)
end
else
mp.msg.info("Adding " .. file)
table.insert(append[1], file)
end
end
end
add_files_at(pl_current + 1, append[1])
add_files_at(pl_current, append[-1])
end
mp.register_event("start-file", find_and_add_entries)
-56
View File
@@ -1,56 +0,0 @@
-- This script cycles between deinterlacing, pullup (inverse
-- telecine), and both filters off. It uses the "deinterlace" property
-- so that a hardware deinterlacer will be used if available.
--
-- It overrides the default deinterlace toggle keybinding "D"
-- (shift+d), so that rather than merely cycling the "deinterlace" property
-- between on and off, it adds a "pullup" step to the cycle.
--
-- It provides OSD feedback as to the actual state of the two filters
-- after each cycle step/keypress.
--
-- Note: if hardware decoding is enabled, pullup filter will likely
-- fail to insert.
--
-- TODO: It might make sense to use hardware assisted vdpaupp=pullup,
-- if available, but I don't have hardware to test it. Patch welcome.
script_name = mp.get_script_name()
pullup_label = string.format("%s-pullup", script_name)
function pullup_on()
for i,vf in pairs(mp.get_property_native('vf')) do
if vf['label'] == pullup_label then
return "yes"
end
end
return "no"
end
function do_cycle()
if pullup_on() == "yes" then
-- if pullup is on remove it
mp.command(string.format("vf del @%s:pullup", pullup_label))
return
elseif mp.get_property("deinterlace") == "yes" then
-- if deinterlace is on, turn it off and insert pullup filter
mp.set_property("deinterlace", "no")
mp.command(string.format("vf add @%s:pullup", pullup_label))
return
else
-- if neither is on, turn on deinterlace
mp.set_property("deinterlace", "yes")
return
end
end
function cycle_deinterlace_pullup_handler()
do_cycle()
-- independently determine current state and give user feedback
mp.osd_message(string.format("deinterlace: %s\n"..
"pullup: %s",
mp.get_property("deinterlace"),
pullup_on()))
end
mp.add_key_binding("D", "cycle-deinterlace-pullup", cycle_deinterlace_pullup_handler)
-99
View File
@@ -1,99 +0,0 @@
-- This script enables live control of the dynamic range compression
-- (drc) audio filter while the video is playing back. This can be
-- useful to avoid having to stop and restart mpv to adjust filter
-- parameters. See the entry for "drc" under the "AUDIO FILTERS"
-- section of the man page for a complete description of the filter.
--
-- This script registers the key-binding "\" to toggle the filter between
--
-- * off
-- * method=1 (single-sample smoothing)
-- * method=2 (multi-sample smoothing)
--
-- It registers the keybindings ctrl+9/ctrl+0 to decrease/increase the
-- target ampltiude. These keys will insert the filter at the default
-- target amplitude of 0.25 if it was not previously present.
--
-- OSD feedback of the current filter state is displayed on pressing
-- each bound key.
script_name = mp.get_script_name()
function print_state(params)
if params then
mp.osd_message(script_name..":\n"
.."method = "..params["method"].."\n"
.."target = "..params["target"])
else
mp.osd_message(script_name..":\noff")
end
end
function get_index_of_drc(afs)
for i,af in pairs(afs) do
if af["label"] == script_name then
return i
end
end
end
function append_drc(afs)
afs[#afs+1] = {
name = "drc",
label = script_name,
params = {
method = "1",
target = "0.25"
}
}
print_state(afs[#afs]["params"])
end
function modify_or_create_af(fun)
afs = mp.get_property_native("af")
i = get_index_of_drc(afs)
if not i then
append_drc(afs)
else
fun(afs, i)
end
mp.set_property_native("af", afs)
end
function drc_toggle_method_handler()
modify_or_create_af(
function (afs, i)
new_method=(afs[i]["params"]["method"]+1)%3
if new_method == 0 then
table.remove(afs, i)
print_state(nil)
else
afs[i]["params"]["method"] = tostring((afs[i]["params"]["method"])%2+1)
print_state(afs[i]["params"])
end
end
)
end
function drc_scale_target(factor)
modify_or_create_af(
function (afs)
afs[i]["params"]["target"] = tostring(afs[i]["params"]["target"]*factor)
print_state(afs[i]["params"])
end
)
end
function drc_louder_handler()
drc_scale_target(2.0)
end
function drc_quieter_handler()
drc_scale_target(0.5)
end
-- toggle between off, method 1 and method 2
mp.add_key_binding("\\", "drc_toggle_method", drc_toggle_method_handler)
-- increase or decrease target volume
mp.add_key_binding("ctrl+9", "drc_quieter", drc_quieter_handler)
mp.add_key_binding("ctrl+0", "drc_louder", drc_louder_handler)
-16
View File
@@ -1,16 +0,0 @@
-- Test script for property change notification mechanism.
function format_property_val(v)
if type(v) == "table" then
return mp.format_table(v) -- undocumented function; might be removed
else
return tostring(v)
end
end
for i,name in ipairs(mp.get_property_native("property-list")) do
mp.observe_property(name, "native", function(name, val)
print("property '" .. name .. "' changed to '" ..
format_property_val(val) .. "'")
end)
end
-69
View File
@@ -1,69 +0,0 @@
-- Rebuild the terminal status line as a lua script
-- Be aware that this will require more cpu power!
-- Add a string to the status line
function atsl(s)
newStatus = newStatus .. s
end
function update_status_line()
-- Reset the status line
newStatus = ""
if mp.get_property_bool("pause") then
atsl("(Paused) ")
elseif mp.get_property_bool("paused-for-cache") then
atsl("(Buffering) ")
end
if mp.get_property("vid") ~= "no" then
atsl("A")
end
if mp.get_property("aid") ~= "no" then
atsl("V")
end
atsl(": ")
atsl(mp.get_property_osd("time-pos"))
atsl(" / ");
atsl(mp.get_property_osd("length"));
atsl(" (")
atsl(mp.get_property_osd("percent-pos", -1))
atsl("%)")
local r = mp.get_property_number("speed", -1)
if r ~= 1 then
atsl(string.format(" x%4.2f", r))
end
r = mp.get_property_number("avsync", nil)
if r ~= nil then
atsl(string.format(" A-V: %7.3f", r))
end
r = mp.get_property("total-avsync-change", 0)
if math.abs(r) > 0.05 then
atsl(string.format(" ct:%7.3f", r))
end
r = mp.get_property_number("drop-frame-count", -1)
if r > 0 then
atsl(" Late: ")
atsl(r)
end
r = mp.get_property_number("cache", 0)
if r > 0 then
atsl(string.format(" Cache: %d%% ", r))
end
-- Set the new status line
mp.set_property("options/term-status-msg", newStatus)
end
-- Register the event
mp.register_event("tick", update_status_line)
+3 -3
View File
@@ -56,7 +56,7 @@ sub generate_c_header {
my $l = max(map { length $_->{valname} } values %{$el->{subelements}});
# Output each variable, with pointers for array (multiple) elements
for my $subel (sort { $a->{definename} cmp $b->{definename} } values %{$el->{subelements}}) {
for my $subel (values %{$el->{subelements}}) {
printf " %-${l}s %s%s;\n",
$subel->{valname}, $subel->{multiple}?'*':' ', $subel->{fieldname};
}
@@ -101,7 +101,7 @@ sub generate_c_definitions {
# define a field for each subelement
# also does lots of macro magic, but doesn't open a scope
for my $subel (sort { $a->{definename} cmp $b->{definename} } values %{$el->{subelements}}) {
for my $subel (values %{$el->{subelements}}) {
print "F($subel->{definename}, $subel->{fieldname}, ".
($subel->{multiple}?'1':'0').")\n";
}
@@ -166,4 +166,4 @@ sub process_elem {
} else {
print "$elem->{type} ". $elem->get_value ."\n";
}
}
}
+2 -2
View File
@@ -86,7 +86,7 @@ __midentify__allprops="
video-bitrate
video-codec
video-format
video-aspect
aspect
fps
width
height
@@ -104,7 +104,7 @@ for __midentify__key in $__midentify__allprops; do
eval unset $__midentify__nextprefix$__midentify__key
done
__midentify__output=`$MPV --term-playing-msg="$__midentify__propstr" --vo=null --ao=null --frames=1 --quiet --no-cache --no-config "$@"`
__midentify__output=`$MPV --playing-msg="$__midentify__propstr" --vo=null --ao=null --frames=1 --quiet "$@"`
__midentify__fileindex=0
__midentify__prefix=
while :; do
+74 -13
View File
@@ -1,44 +1,105 @@
#!/usr/bin/env python
import os
import re
import shutil
import sys
from optparse import OptionParser
from textwrap import dedent
def sh(command):
return os.popen(command).read()
def bundle_path(binary_name):
return "%s.app" % binary_name
def dylib_lst(input_file):
return sh("otool -L %s | grep -e '\t' | awk '{ print $1 }'" % input_file)
sys_re = re.compile("/System")
exe_re = re.compile("@executable_path")
binary_name = sys.argv[1]
def is_user_lib(libname, input_file):
return not sys_re.match(libname) and \
not exe_re.match(libname) and \
not "libobjc" in libname and \
not "libSystem" in libname and \
not "libgcc" in libname and \
not os.path.basename(input_file) in libname and \
not libname == ''
def user_dylib_lst(input_file):
return [lib for lib in dylib_lst(input_file).split("\n") if
is_user_lib(lib, input_file)]
def bundle_name(binary_name):
return os.path.basename(bundle_path(binary_name))
return "%s.app" % binary_name
def target_plist(binary_name):
return os.path.join(bundle_path(binary_name), 'Contents', 'Info.plist')
return os.path.join(bundle_name(binary_name), 'Contents', 'Info.plist')
def target_directory(binary_name):
return os.path.join(bundle_path(binary_name), 'Contents', 'MacOS')
return os.path.join(bundle_name(binary_name), 'Contents', 'MacOS')
def target_binary(binary_name):
return os.path.join(target_directory(binary_name),
os.path.basename(binary_name))
return os.path.join(target_directory(binary_name), binary_name)
def copy_bundle(binary_name):
if os.path.isdir(bundle_path(binary_name)):
shutil.rmtree(bundle_path(binary_name))
if os.path.isdir(bundle_name(binary_name)):
shutil.rmtree(bundle_name(binary_name))
shutil.copytree(
os.path.join('TOOLS', 'osxbundle', bundle_name(binary_name)),
bundle_path(binary_name))
bundle_name(binary_name))
def copy_binary(binary_name):
shutil.copy(binary_name, target_binary(binary_name))
def run_install_name_tool(target_file, dylib_path, dest_dir, root=True):
new_dylib_path = os.path.join("@executable_path", "lib",
os.path.basename(dylib_path))
sh("install_name_tool -change %s %s %s" % \
(dylib_path, new_dylib_path, target_file))
if root:
sh("install_name_tool -id %s %s" % \
(new_dylib_path, os.path.join(dest_dir,
os.path.basename(dylib_path))))
def cp_dylibs(target_file, dest_dir):
for dylib_path in user_dylib_lst(target_file):
dylib_dest_path = os.path.join(dest_dir, os.path.basename(dylib_path))
try:
shutil.copy(dylib_path, dylib_dest_path)
except IOError:
if re.match("dylib$", target_file):
reinstall_what = target_file
else:
reinstall_what = dylib_path
sys.exit(dedent("""\
%s uses library %s which is not available anymore.
This is most likely because you uninstalled %s.
Please reinstall %s to fix it's dependencies.""" % \
(target_file, dylib_path, dylib_path, reinstall_what) ))
os.chmod(dylib_dest_path, 0o755)
cp_dylibs(dylib_dest_path, dest_dir)
def fix_dylibs_paths(target_file, dest_dir, root=True):
for dylib_path in user_dylib_lst(target_file):
dylib_dest_path = os.path.join(dest_dir, os.path.basename(dylib_path))
run_install_name_tool(target_file, dylib_path, dest_dir, root)
fix_dylibs_paths(dylib_dest_path, dest_dir, False)
def apply_plist_template(plist_file, version):
sh("sed -i -e 's/${VERSION}/%s/g' %s" % (version, plist_file))
sh("sed -i -e 's/{{VERSION}}/%s/g' %s" % (version, plist_file))
def bundle_dependencies(binary_name):
lib_bundle_directory = os.path.join(target_directory(binary_name), "lib")
cp_dylibs(binary_name, lib_bundle_directory)
fix_dylibs_paths(target_binary(binary_name), lib_bundle_directory)
def main():
version = sh("./version.sh --print").strip()
version = sh("TOOLS/osxbundle/version.sh").strip()
usage = "usage: %prog [options] arg"
parser = OptionParser(usage)
@@ -63,7 +124,7 @@ def main():
if options.deps:
print("> bundling dependencies")
sh(" ".join(["TOOLS/dylib-unhell.py", target_binary(binary_name)]))
bundle_dependencies(binary_name)
print("done.")
+35 -608
View File
@@ -1,5 +1,5 @@
<?xml version="1.0" encoding="UTF-8"?>
<!DOCTYPE plist PUBLIC "-//Apple//DTD PLIST 1.0//EN" "http://www.apple.com/DTDs/PropertyList-1.0.dtd">
<!DOCTYPE plist PUBLIC "-//Apple Computer//DTD PLIST 1.0//EN" "http://www.apple.com/DTDs/PropertyList-1.0.dtd">
<plist version="1.0">
<dict>
<key>CFBundleDevelopmentRegion</key>
@@ -35,7 +35,7 @@
<string>wma</string>
</array>
<key>CFBundleTypeIconFile</key>
<string>document.icns</string>
<string>audio.icns</string>
<key>CFBundleTypeName</key>
<string>Audio file</string>
<key>CFBundleTypeRole</key>
@@ -86,7 +86,6 @@
<string>VCD</string>
<string>VFW</string>
<string>VOB</string>
<string>WEBM</string>
<string>WMV</string>
<string>asf</string>
<string>avi</string>
@@ -120,11 +119,10 @@
<string>vcd</string>
<string>vfw</string>
<string>vob</string>
<string>webm</string>
<string>wmv</string>
</array>
<key>CFBundleTypeIconFile</key>
<string>document.icns</string>
<string>movie.icns</string>
<key>CFBundleTypeName</key>
<string>Movie file</string>
<key>CFBundleTypeRole</key>
@@ -159,7 +157,7 @@
<string>utf</string>
</array>
<key>CFBundleTypeIconFile</key>
<string>document.icns</string>
<string>subtitles.icns</string>
<key>CFBundleTypeName</key>
<string>Subtitles file</string>
<key>CFBundleTypeRole</key>
@@ -183,7 +181,7 @@
<key>CFBundlePackageType</key>
<string>APPL</string>
<key>CFBundleShortVersionString</key>
<string>${VERSION}</string>
<string>{{VERSION}}</string>
<key>NSHighResolutionCapable</key>
<true/>
<key>CFBundleURLTypes</key>
@@ -192,630 +190,59 @@
<key>CFBundleTypeRole</key>
<string>Viewer</string>
<key>CFBundleURLName</key>
<string>mpv Custom Protocol</string>
<string>Real Time (Streaming) Protocol</string>
<key>CFBundleURLSchemes</key>
<array>
<string>mpv</string>
<string>rtp</string>
<string>rtsp</string>
</array>
</dict>
<dict>
<key>CFBundleTypeRole</key>
<string>Viewer</string>
<key>CFBundleURLName</key>
<string>Streaming Protocol</string>
<string>File over HTTP/FTP/UDP</string>
<key>CFBundleURLSchemes</key>
<array>
<string>icyx</string>
<string>udp</string>
<string>ftp</string>
<string>http_proxy</string>
<string>http</string>
</array>
</dict>
<dict>
<key>CFBundleTypeRole</key>
<string>Viewer</string>
<key>CFBundleURLName</key>
<string>Microsoft Media Services</string>
<key>CFBundleURLSchemes</key>
<array>
<string>mms</string>
<string>mmst</string>
<string>http</string>
<string>httpproxy</string>
<string>rtp</string>
<string>rtsp</string>
<string>ftp</string>
<string>udp</string>
<string>smb</string>
</array>
</dict>
<dict>
<key>CFBundleTypeRole</key>
<string>Viewer</string>
<key>CFBundleURLName</key>
<string>CD/DVD/Bluray Media</string>
<string>Cuesheet</string>
<key>CFBundleURLSchemes</key>
<array>
<string>cdda</string>
<string>cue</string>
</array>
</dict>
<dict>
<key>CFBundleTypeRole</key>
<string>Viewer</string>
<key>CFBundleURLName</key>
<string>CD/DVD Media</string>
<key>CFBundleURLSchemes</key>
<array>
<string>dvdnav</string>
<string>dvd</string>
<string>vcd</string>
<string>bd</string>
</array>
</dict>
</array>
<key>UTImportedTypeDeclarations</key>
<array>
<dict>
<key>UTTypeConformsTo</key>
<array>
<string>public.audio</string>
</array>
<key>UTTypeDescription</key>
<string>AC3 Audio</string>
<key>UTTypeIconFile</key>
<string>document.icns</string>
<key>UTTypeIdentifier</key>
<string>io.mpv.ac3</string>
<key>UTTypeReferenceURL</key>
<string>http://wiki.multimedia.cx/index.php?title=AC3</string>
<key>UTTypeTagSpecification</key>
<dict>
<key>public.filename-extension</key>
<array>
<string>ac3</string>
<string>a52</string>
<string>eac3</string>
</array>
</dict>
</dict>
<dict>
<key>UTTypeConformsTo</key>
<array>
<string>public.audio</string>
</array>
<key>UTTypeDescription</key>
<string>DTS Audio</string>
<key>UTTypeIconFile</key>
<string>document.icns</string>
<key>UTTypeIdentifier</key>
<string>io.mpv.dts</string>
<key>UTTypeReferenceURL</key>
<string>http://wiki.multimedia.cx/index.php?title=DTS</string>
<key>UTTypeTagSpecification</key>
<dict>
<key>public.filename-extension</key>
<array>
<string>dts</string>
</array>
</dict>
</dict>
<dict>
<key>UTTypeConformsTo</key>
<array>
<string>public.audio</string>
</array>
<key>UTTypeDescription</key>
<string>Free Lossless Audio Codec</string>
<key>UTTypeIconFile</key>
<string>document.icns</string>
<key>UTTypeIdentifier</key>
<string>io.mpv.flac</string>
<key>UTTypeReferenceURL</key>
<string>http://flac.sourceforge.net/format.html</string>
<key>UTTypeTagSpecification</key>
<dict>
<key>public.filename-extension</key>
<array>
<string>flac</string>
</array>
</dict>
</dict>
<dict>
<key>UTTypeConformsTo</key>
<array>
<string>public.audio</string>
</array>
<key>UTTypeDescription</key>
<string>Matroska Audio</string>
<key>UTTypeIconFile</key>
<string>document.icns</string>
<key>UTTypeIdentifier</key>
<string>io.mpv.mka</string>
<key>UTTypeReferenceURL</key>
<string>http://www.matroska.org</string>
<key>UTTypeTagSpecification</key>
<dict>
<key>public.filename-extension</key>
<array>
<string>mka</string>
</array>
</dict>
</dict>
<dict>
<key>UTTypeConformsTo</key>
<array>
<string>public.audio</string>
</array>
<key>UTTypeDescription</key>
<string>Ogg Audio</string>
<key>UTTypeIconFile</key>
<string>document.icns</string>
<key>UTTypeIdentifier</key>
<string>io.mpv.ogg-audio</string>
<key>UTTypeReferenceURL</key>
<string>http://xiph.org/ogg</string>
<key>UTTypeTagSpecification</key>
<dict>
<key>public.filename-extension</key>
<array>
<string>ogg</string>
</array>
</dict>
</dict>
<dict>
<key>UTTypeConformsTo</key>
<array>
<string>public.audio</string>
</array>
<key>UTTypeDescription</key>
<string>PCM Audio</string>
<key>UTTypeIconFile</key>
<string>document.icns</string>
<key>UTTypeIdentifier</key>
<string>io.mpv.pcm</string>
<key>UTTypeReferenceURL</key>
<string>http://en.wikipedia.org/wiki/Pulse-code_modulation</string>
<key>UTTypeTagSpecification</key>
<dict>
<key>public.filename-extension</key>
<array>
<string>pcm</string>
</array>
</dict>
</dict>
<dict>
<key>UTTypeConformsTo</key>
<array>
<string>public.audio</string>
</array>
<key>UTTypeDescription</key>
<string>Windows Media Audio</string>
<key>UTTypeIconFile</key>
<string>document.icns</string>
<key>UTTypeIdentifier</key>
<string>io.mpv.wma</string>
<key>UTTypeReferenceURL</key>
<string>http://en.wikipedia.org/wiki/Windows_Media_Audio</string>
<key>UTTypeTagSpecification</key>
<dict>
<key>public.filename-extension</key>
<array>
<string>wma</string>
</array>
</dict>
</dict>
<dict>
<key>UTTypeConformsTo</key>
<array>
<string>public.movie</string>
</array>
<key>UTTypeDescription</key>
<string>Audio Video Interleave</string>
<key>UTTypeIconFile</key>
<string>document.icns</string>
<key>UTTypeIdentifier</key>
<string>io.mpv.avi</string>
<key>UTTypeReferenceURL</key>
<string>http://www.the-labs.com/Video/odmlff2-avidef.pdf</string>
<key>UTTypeTagSpecification</key>
<dict>
<key>public.filename-extension</key>
<array>
<string>avi</string>
</array>
</dict>
</dict>
<dict>
<key>UTTypeConformsTo</key>
<array>
<string>public.movie</string>
</array>
<key>UTTypeDescription</key>
<string>DIVX Video</string>
<key>UTTypeIconFile</key>
<string>document.icns</string>
<key>UTTypeIdentifier</key>
<string>io.mpv.divx</string>
<key>UTTypeReferenceURL</key>
<string>http://www.divx.com</string>
<key>UTTypeTagSpecification</key>
<dict>
<key>public.filename-extension</key>
<array>
<string>divx</string>
</array>
</dict>
</dict>
<dict>
<key>UTTypeConformsTo</key>
<array>
<string>public.movie</string>
</array>
<key>UTTypeDescription</key>
<string>DV Video</string>
<key>UTTypeIconFile</key>
<string>document.icns</string>
<key>UTTypeIdentifier</key>
<string>io.mpv.dv</string>
<key>UTTypeReferenceURL</key>
<string>http://en.wikipedia.org/wiki/DV</string>
<key>UTTypeTagSpecification</key>
<dict>
<key>public.filename-extension</key>
<array>
<string>dv</string>
<string>hdv</string>
</array>
</dict>
</dict>
<dict>
<key>UTTypeConformsTo</key>
<array>
<string>public.movie</string>
</array>
<key>UTTypeDescription</key>
<string>Flash Video</string>
<key>UTTypeIconFile</key>
<string>document.icns</string>
<key>UTTypeIdentifier</key>
<string>io.mpv.flv</string>
<key>UTTypeReferenceURL</key>
<string>http://en.wikipedia.org/wiki/Flash_Video</string>
<key>UTTypeTagSpecification</key>
<dict>
<key>public.filename-extension</key>
<array>
<string>flv</string>
<string>f4v</string>
<string>f4p</string>
<string>swf</string>
</array>
</dict>
</dict>
<dict>
<key>UTTypeConformsTo</key>
<array>
<string>public.movie</string>
</array>
<key>UTTypeDescription</key>
<string>MPEG-2 Transport Stream</string>
<key>UTTypeIconFile</key>
<string>document.icns</string>
<key>UTTypeIdentifier</key>
<string>io.mpv.mts</string>
<key>UTTypeReferenceURL</key>
<string>http://en.wikipedia.org/wiki/.m2ts</string>
<key>UTTypeTagSpecification</key>
<dict>
<key>public.filename-extension</key>
<array>
<string>trp</string>
<string>m2t</string>
<string>m2ts</string>
<string>mts</string>
<string>mtv</string>
<string>ts</string>
</array>
</dict>
</dict>
<dict>
<key>UTTypeConformsTo</key>
<array>
<string>public.movie</string>
</array>
<key>UTTypeDescription</key>
<string>Matroska Video</string>
<key>UTTypeIconFile</key>
<string>document.icns</string>
<key>UTTypeIdentifier</key>
<string>io.mpv.mkv</string>
<key>UTTypeReferenceURL</key>
<string>http://www.matroska.org</string>
<key>UTTypeTagSpecification</key>
<dict>
<key>public.filename-extension</key>
<array>
<string>mkv</string>
<string>webm</string>
</array>
</dict>
</dict>
<dict>
<key>UTTypeConformsTo</key>
<array>
<string>public.movie</string>
</array>
<key>UTTypeDescription</key>
<string>Ogg Video</string>
<key>UTTypeIconFile</key>
<string>document.icns</string>
<key>UTTypeIdentifier</key>
<string>io.mpv.ogv</string>
<key>UTTypeReferenceURL</key>
<string>http://xiph.org/ogg</string>
<key>UTTypeTagSpecification</key>
<dict>
<key>public.filename-extension</key>
<array>
<string>ogm</string>
<string>ogv</string>
</array>
</dict>
</dict>
<dict>
<key>UTTypeConformsTo</key>
<array>
<string>public.movie</string>
</array>
<key>UTTypeDescription</key>
<string>Real Media</string>
<key>UTTypeIconFile</key>
<string>document.icns</string>
<key>UTTypeIdentifier</key>
<string>io.mpv.rmvb</string>
<key>UTTypeReferenceURL</key>
<string>http://www.real.com</string>
<key>UTTypeTagSpecification</key>
<dict>
<key>public.filename-extension</key>
<array>
<string>rmvb</string>
<string>rm</string>
</array>
</dict>
</dict>
<dict>
<key>UTTypeConformsTo</key>
<array>
<string>public.movie</string>
</array>
<key>UTTypeDescription</key>
<string>Video Object</string>
<key>UTTypeIconFile</key>
<string>document.icns</string>
<key>UTTypeIdentifier</key>
<string>io.mpv.vob</string>
<key>UTTypeReferenceURL</key>
<string>http://en.wikipedia.org/wiki/VOB</string>
<key>UTTypeTagSpecification</key>
<dict>
<key>public.filename-extension</key>
<array>
<string>vob</string>
</array>
</dict>
</dict>
<dict>
<key>UTTypeConformsTo</key>
<array>
<string>public.movie</string>
</array>
<key>UTTypeDescription</key>
<string>Windows Media Video</string>
<key>UTTypeIconFile</key>
<string>document.icns</string>
<key>UTTypeIdentifier</key>
<string>io.mpv.wmv</string>
<key>UTTypeReferenceURL</key>
<string>http://en.wikipedia.org/wiki/Windows_Media_Video</string>
<key>UTTypeTagSpecification</key>
<dict>
<key>public.filename-extension</key>
<array>
<string>wmv</string>
</array>
</dict>
</dict>
<dict>
<key>UTTypeConformsTo</key>
<array>
<string>public.movie</string>
</array>
<key>UTTypeDescription</key>
<string>XVID Video</string>
<key>UTTypeIconFile</key>
<string>document.icns</string>
<key>UTTypeIdentifier</key>
<string>io.mpv.xvid</string>
<key>UTTypeReferenceURL</key>
<string>http://www.xvid.org</string>
<key>UTTypeTagSpecification</key>
<dict>
<key>public.filename-extension</key>
<array>
<string>xvid</string>
</array>
</dict>
</dict>
<dict>
<key>UTTypeConformsTo</key>
<array>
<string>public.movie</string>
</array>
<key>UTTypeDescription</key>
<string>AVC raw stream</string>
<key>UTTypeIconFile</key>
<string>document.icns</string>
<key>UTTypeIdentifier</key>
<string>io.mpv.h264</string>
<key>UTTypeReferenceURL</key>
<string>http://www.itu.int/rec/T-REC-H.264</string>
<key>UTTypeTagSpecification</key>
<dict>
<key>public.filename-extension</key>
<array>
<string>264</string>
</array>
</dict>
</dict>
<dict>
<key>UTTypeConformsTo</key>
<array>
<string>public.movie</string>
</array>
<key>UTTypeDescription</key>
<string>HEVC raw stream</string>
<key>UTTypeIconFile</key>
<string>document.icns</string>
<key>UTTypeIdentifier</key>
<string>io.mpv.hevc</string>
<key>UTTypeReferenceURL</key>
<string>http://hevc.info</string>
<key>UTTypeTagSpecification</key>
<dict>
<key>public.filename-extension</key>
<array>
<string>hevc</string>
</array>
</dict>
</dict>
<dict>
<key>UTTypeConformsTo</key>
<array>
<string>public.movie</string>
</array>
<key>UTTypeDescription</key>
<string>YUV stream</string>
<key>UTTypeIconFile</key>
<string>document.icns</string>
<key>UTTypeIdentifier</key>
<string>io.mpv.yuv</string>
<key>UTTypeReferenceURL</key>
<string>http://en.wikipedia.org/wiki/YUV</string>
<key>UTTypeTagSpecification</key>
<dict>
<key>public.filename-extension</key>
<array>
<string>yuv</string>
</array>
</dict>
</dict>
<dict>
<key>UTTypeConformsTo</key>
<array>
<string>public.movie</string>
</array>
<key>UTTypeDescription</key>
<string>YUV4MPEG2 stream</string>
<key>UTTypeIconFile</key>
<string>document.icns</string>
<key>UTTypeIdentifier</key>
<string>io.mpv.y4m</string>
<key>UTTypeReferenceURL</key>
<string>http://wiki.multimedia.cx/index.php?title=YUV4MPEG2</string>
<key>UTTypeTagSpecification</key>
<dict>
<key>public.filename-extension</key>
<array>
<string>y4m</string>
</array>
</dict>
</dict>
<dict>
<key>UTTypeConformsTo</key>
<array>
<string>public.plain-text</string>
</array>
<key>UTTypeDescription</key>
<string>SubRip Subtitle</string>
<key>UTTypeIconFile</key>
<string>document.icns</string>
<key>UTTypeIdentifier</key>
<string>io.mpv.subrip</string>
<key>UTTypeReferenceURL</key>
<string>http://en.wikipedia.org/wiki/SubRip</string>
<key>UTTypeTagSpecification</key>
<dict>
<key>public.filename-extension</key>
<array>
<string>srt</string>
</array>
</dict>
</dict>
<dict>
<key>UTTypeConformsTo</key>
<array>
<string>public.plain-text</string>
</array>
<key>UTTypeDescription</key>
<string>MicroDVD Subtitle</string>
<key>UTTypeIconFile</key>
<string>document.icns</string>
<key>UTTypeIdentifier</key>
<string>io.mpv.sub</string>
<key>UTTypeReferenceURL</key>
<string>http://en.wikipedia.org/wiki/MicroDVD</string>
<key>UTTypeTagSpecification</key>
<dict>
<key>public.filename-extension</key>
<array>
<string>sub</string>
</array>
</dict>
</dict>
<dict>
<key>UTTypeConformsTo</key>
<array>
<string>public.plain-text</string>
</array>
<key>UTTypeDescription</key>
<string>SubStation Alpha Subtitle</string>
<key>UTTypeIconFile</key>
<string>document.icns</string>
<key>UTTypeIdentifier</key>
<string>io.mpv.ass</string>
<key>UTTypeReferenceURL</key>
<string>https://github.com/libass/libass</string>
<key>UTTypeTagSpecification</key>
<dict>
<key>public.filename-extension</key>
<array>
<string>ass</string>
<string>ssa</string>
</array>
</dict>
</dict>
<dict>
<key>UTTypeConformsTo</key>
<array>
<string>public.data</string>
</array>
<key>UTTypeDescription</key>
<string>VobSub Subtitle</string>
<key>UTTypeIconFile</key>
<string>document.icns</string>
<key>UTTypeIdentifier</key>
<string>io.mpv.vobsub</string>
<key>UTTypeReferenceURL</key>
<string>http://en.wikipedia.org/wiki/DirectVobSub</string>
<key>UTTypeTagSpecification</key>
<dict>
<key>public.filename-extension</key>
<array>
<string>idx</string>
<string>sub</string>
</array>
</dict>
</dict>
<dict>
<key>UTTypeConformsTo</key>
<array>
<string>public.plain-text</string>
</array>
<key>UTTypeDescription</key>
<string>SAMI Subtitle</string>
<key>UTTypeIconFile</key>
<string>document.icns</string>
<key>UTTypeIdentifier</key>
<string>io.mpv.smi</string>
<key>UTTypeReferenceURL</key>
<string>http://en.wikipedia.org/wiki/SAMI</string>
<key>UTTypeTagSpecification</key>
<dict>
<key>public.filename-extension</key>
<array>
<string>smi</string>
<string>smil</string>
</array>
</dict>
</dict>
</array>
</dict>
</plist>
@@ -0,0 +1,120 @@
<?xml version="1.0"?>
<!DOCTYPE fontconfig SYSTEM "fonts.dtd">
<fontconfig>
<dir>/System/Library/Fonts</dir>
<dir>/Library/Fonts</dir>
<dir>~/Library/Fonts</dir>
<dir>/Network/Library/Fonts</dir>
<cachedir>~/.fontconfig</cachedir>
<!--
Accept deprecated 'mono' alias, replacing it with 'monospace'
-->
<match target="pattern">
<test qual="any" name="family">
<string>mono</string>
</test>
<edit name="family" mode="assign" binding="same">
<string>monospace</string>
</edit>
</match>
<!--
Accept alternate 'sans serif' spelling, replacing it with 'sans-serif'
-->
<match target="pattern">
<test qual="any" name="family">
<string>sans serif</string>
</test>
<edit name="family" mode="assign" binding="same">
<string>sans-serif</string>
</edit>
</match>
<!--
Accept deprecated 'sans' alias, replacing it with 'sans-serif'
-->
<match target="pattern">
<test qual="any" name="family">
<string>sans</string>
</test>
<edit name="family" mode="assign" binding="same">
<string>sans-serif</string>
</edit>
</match>
<config>
<!--
These are the default Unicode chars that are expected to be blank
in fonts. All other blank chars are assumed to be broken and
won't appear in the resulting charsets
-->
<blank>
<int>0x0020</int> <!-- SPACE -->
<int>0x00A0</int> <!-- NO-BREAK SPACE -->
<int>0x00AD</int> <!-- SOFT HYPHEN -->
<int>0x034F</int> <!-- COMBINING GRAPHEME JOINER -->
<int>0x0600</int> <!-- ARABIC NUMBER SIGN -->
<int>0x0601</int> <!-- ARABIC SIGN SANAH -->
<int>0x0602</int> <!-- ARABIC FOOTNOTE MARKER -->
<int>0x0603</int> <!-- ARABIC SIGN SAFHA -->
<int>0x06DD</int> <!-- ARABIC END OF AYAH -->
<int>0x070F</int> <!-- SYRIAC ABBREVIATION MARK -->
<int>0x115F</int> <!-- HANGUL CHOSEONG FILLER -->
<int>0x1160</int> <!-- HANGUL JUNGSEONG FILLER -->
<int>0x1680</int> <!-- OGHAM SPACE MARK -->
<int>0x17B4</int> <!-- KHMER VOWEL INHERENT AQ -->
<int>0x17B5</int> <!-- KHMER VOWEL INHERENT AA -->
<int>0x180E</int> <!-- MONGOLIAN VOWEL SEPARATOR -->
<int>0x2000</int> <!-- EN QUAD -->
<int>0x2001</int> <!-- EM QUAD -->
<int>0x2002</int> <!-- EN SPACE -->
<int>0x2003</int> <!-- EM SPACE -->
<int>0x2004</int> <!-- THREE-PER-EM SPACE -->
<int>0x2005</int> <!-- FOUR-PER-EM SPACE -->
<int>0x2006</int> <!-- SIX-PER-EM SPACE -->
<int>0x2007</int> <!-- FIGURE SPACE -->
<int>0x2008</int> <!-- PUNCTUATION SPACE -->
<int>0x2009</int> <!-- THIN SPACE -->
<int>0x200A</int> <!-- HAIR SPACE -->
<int>0x200B</int> <!-- ZERO WIDTH SPACE -->
<int>0x200C</int> <!-- ZERO WIDTH NON-JOINER -->
<int>0x200D</int> <!-- ZERO WIDTH JOINER -->
<int>0x200E</int> <!-- LEFT-TO-RIGHT MARK -->
<int>0x200F</int> <!-- RIGHT-TO-LEFT MARK -->
<int>0x2028</int> <!-- LINE SEPARATOR -->
<int>0x2029</int> <!-- PARAGRAPH SEPARATOR -->
<int>0x202A</int> <!-- LEFT-TO-RIGHT EMBEDDING -->
<int>0x202B</int> <!-- RIGHT-TO-LEFT EMBEDDING -->
<int>0x202C</int> <!-- POP DIRECTIONAL FORMATTING -->
<int>0x202D</int> <!-- LEFT-TO-RIGHT OVERRIDE -->
<int>0x202E</int> <!-- RIGHT-TO-LEFT OVERRIDE -->
<int>0x202F</int> <!-- NARROW NO-BREAK SPACE -->
<int>0x205F</int> <!-- MEDIUM MATHEMATICAL SPACE -->
<int>0x2060</int> <!-- WORD JOINER -->
<int>0x2061</int> <!-- FUNCTION APPLICATION -->
<int>0x2062</int> <!-- INVISIBLE TIMES -->
<int>0x2063</int> <!-- INVISIBLE SEPARATOR -->
<int>0x206A</int> <!-- INHIBIT SYMMETRIC SWAPPING -->
<int>0x206B</int> <!-- ACTIVATE SYMMETRIC SWAPPING -->
<int>0x206C</int> <!-- INHIBIT ARABIC FORM SHAPING -->
<int>0x206D</int> <!-- ACTIVATE ARABIC FORM SHAPING -->
<int>0x206E</int> <!-- NATIONAL DIGIT SHAPES -->
<int>0x206F</int> <!-- NOMINAL DIGIT SHAPES -->
<int>0x2800</int> <!-- BRAILLE PATTERN BLANK -->
<int>0x3000</int> <!-- IDEOGRAPHIC SPACE -->
<int>0x3164</int> <!-- HANGUL FILLER -->
<int>0xFEFF</int> <!-- ZERO WIDTH NO-BREAK SPACE -->
<int>0xFFA0</int> <!-- HALFWIDTH HANGUL FILLER -->
<int>0xFFF9</int> <!-- INTERLINEAR ANNOTATION ANCHOR -->
<int>0xFFFA</int> <!-- INTERLINEAR ANNOTATION SEPARATOR -->
<int>0xFFFB</int> <!-- INTERLINEAR ANNOTATION TERMINATOR -->
</blank>
<!--
Rescan configuration every 30 seconds when FcFontSetList is called
-->
<rescan>
<int>30</int>
</rescan>
</config>
</fontconfig>
Binary file not shown.
+15
View File
@@ -0,0 +1,15 @@
#!/bin/sh
# Extract revision number from file used by daily tarball snapshots
# or from "git describe" output
git_revision=$(cat snapshot_version 2> /dev/null)
test $git_revision || test ! -d .git || \
git_revision=`git describe --match "v[0-9]*" --always`
git_revision=$(expr "$git_revision" : v*'\(.*\)')
test $git_revision || git_revision=UNKNOWN
# releases extract the version number from the VERSION file
version=$(cat VERSION 2> /dev/null)
test $version || version=$git_revision
echo $version
-96
View File
@@ -1,96 +0,0 @@
#!/usr/bin/env python3
import matplotlib.pyplot as plot
import sys
filename = sys.argv[1]
"""
This script is meant to display stats written by mpv --dump-stats=filename.
In general, each line in that file is an event of the form:
<timestamp in microseconds> <text> '#' <comment>
e.g.:
10474959 start flip #cplayer
<text> is what MP_STATS(log, "...") writes. The rest is added by msg.c.
Currently, the following event types are supported:
'start' <name> start of the named event
'end' <name> end of the named event
'value' <float> <name> a normal value (as opposed to event)
<event> singular event
"""
class G:
events = {}
sevents = [] # events, deterministically sorted
start = None
class Event:
pass
def get_event(event):
if event not in G.events:
e = Event()
G.events[event] = e
e.name = event
e.vals = []
e.type = "unknown"
G.sevents = list(G.events.values())
G.sevents.sort(key=lambda x: x.name)
return G.events[event]
for line in [line.split("#")[0].strip() for line in open(filename, "r")]:
line = line.strip()
if not line:
continue
ts, event = line.split(" ", 1)
ts = int(ts) / 1000 # milliseconds
if G.start is None:
G.start = ts
ts = ts - G.start
if event.startswith("start "):
e = get_event(event[6:])
e.type = "event"
e.vals.append((ts, 0))
e.vals.append((ts, 1))
elif event.startswith("end "):
e = get_event(event[4:])
e.type = "event"
e.vals.append((ts, 1))
e.vals.append((ts, 0))
elif event.startswith("value "):
_, val, name = event.split(" ", 2)
val = float(val)
e = get_event(name)
e.type = "value"
e.vals.append((ts, val))
else:
e = get_event(event)
e.type = "event-signal"
e.vals.append((ts, 1))
ao_events = ["ao fill", "audio wait"]
for e in G.sevents:
if e.name in ao_events:
e.vals = [(x, y * 0.5) for (x, y) in e.vals]
plot.hold(True)
mainpl = plot.subplot(2, 1, 1)
legend = []
for e in G.sevents:
if e.type == "value":
plot.subplot(2, 1, 2, sharex=mainpl)
else:
plot.subplot(2, 1, 1)
pl, = plot.plot([x for x,y in e.vals], [y for x,y in e.vals], label=e.name)
if e.type == "event-signal":
plot.setp(pl, marker = "o", linestyle = "None")
legend.append(pl)
plot.subplot(2, 1, 1)
plot.legend(legend, [pl.get_label() for pl in legend])
plot.show()
+64
View File
@@ -0,0 +1,64 @@
#!/bin/sh
set -e
if [ $# -ne 2 ]; then
echo >&2 "Usage: $0 input.svg output.ico"
exit 1
fi
# For smooth rendering on high-DPI displays, the standard three app icon sizes
# (16x16, 32x32 and 48x48) have to be scaled to the four Windows DPI settings
# as in the table below:
# Small Regular Tiles
# 96 DPI (100%) 16x16 32x32 48x48
# 120 DPI (125%) 20x20 40x40 60x60
# 144 DPI (150%) 24x24 48x48 72x72
# 196 DPI (200%) 32x32 64x64 96x96
# Also, there should also be an extra large 256x256 icon and some low colour
# 8-bit and 4-bit variants, which are still used in Remote Desktop.
# Note: Windows Vista has a bug in its icon scaling that makes no sense.
# Instead of following the DPI setting, small icons are always a bit larger
# than they should be at 22x22, 26x26 and 36x36 for 120, 144 and 196 DPI. This
# script doesn't generate icons with those sizes, since computers with Vista
# and a high-DPI display are probably fairly rare these days.
temppng=".$(basename "$1" .svg)-temp.png"
inkscape --without-gui --export-png="$temppng" --export-dpi=72 \
--export-background-opacity=0 --export-width=512 --export-height=512 \
"$1" >/dev/null 2>&1
# Old versions of ImageMagick (like the one in Cygwin) use the wrong gamma when
# exporting icon files. To fix, add -gamma 2.2 after the input file.
convert png:"$temppng" -filter lanczos2 \
\( -clone 0 -resize 96x96 \) \
\( -clone 0 -resize 72x72 \) \
\( -clone 0 -resize 64x64 \) \
\( -clone 0 -resize 60x60 \) \
\( -clone 0 -resize 48x48 \) \
\( -clone 0 -resize 40x40 \) \
\( -clone 0 -resize 32x32 \) \
\( -clone 0 -resize 24x24 \) \
\( -clone 0 -resize 20x20 \) \
\( -clone 0 -resize 16x16 \) \
\( -clone 0 -resize 256x256 \) \
\( -clone 0 -resize 32x32 \( -clone 0 -alpha opaque -colors 255 \) \
\( -clone 0 -channel A -threshold 50% \) \
-delete 0 -compose CopyOpacity -composite -colors 256 \) \
\( -clone 0 -resize 16x16 \( -clone 0 -alpha opaque -colors 255 \) \
\( -clone 0 -channel A -threshold 50% \) \
-delete 0 -compose CopyOpacity -composite -colors 256 \) \
\( -clone 0 -resize 32x32 \( -clone 0 -alpha opaque -colors 15 \) \
\( -clone 0 -channel A -threshold 50% \) \
-delete 0 -compose CopyOpacity -composite -colors 16 \) \
\( -clone 0 -resize 16x16 \( -clone 0 -alpha opaque -colors 15 \) \
\( -clone 0 -channel A -threshold 50% \) \
-delete 0 -compose CopyOpacity -composite -colors 16 \) \
-delete 0 \
-define png:compression-level=9 -define png:include-chunk=none ico:"$2"
rm "$temppng"
+17 -48
View File
@@ -3,18 +3,20 @@
"""
This script emulates "unique application" functionality on Linux. When starting
playback with this script, it will try to reuse an already running instance of
mpv (but only if that was started with umpv). Other mpv instances (not started
by umpv) are ignored, and the script doesn't know about them.
mpv (but only if that was started with umpv).
This only takes filenames as arguments. Custom options can't be used; the script
interprets them as filenames. If mpv is already running, the files passed to
umpv are appended to mpv's internal playlist. If a file does not exist or is
otherwise not playable, mpv will skip the playlist entry when attempting to
play it (from the GUI perspective, it's silently ignored).
This script only uses mpv instances started with umpv. Other instances are
ignored, and don't exist for the script.
This only takes filenames as arguments. If mpv is already running, they are
appended to mpv's internal playlist. If the file does not exist or is otherwise
not playable, mpv will skip the playlist entry when attempting to play it (from
the GUI perspective, it's silently ignored).
If mpv isn't running yet, this script will start mpv and let it control the
current terminal. It will not write output to stdout/stderr, because this
will typically just fill ~/.xsession-errors with garbage.
current terminal. If you play an audio-only file, mpv will not open a window
(that is its normal behavior; but it might not be what a user expects if this
script is used in a GUI setting).
mpv will terminate if there are no more files to play, and running the umpv
script after that will start a new mpv instance.
@@ -23,18 +25,10 @@ Note that you can control the mpv instance by writing to the command fifo:
echo "cycle fullscreen" > /tmp/umpv-fifo-*
Note: you can supply custom options with the UMPV_OPTIONS environment variable.
The environment variable will be split on whitespace, and each item is
passed as option to mpv _if_ the script starts mpv. If mpv is not started
by the script (i.e. mpv is already running), the options will be ignored.
Warning:
The script attempts to make sure the FIFO is safely created (i.e. not world-
writable), and checks that it's really a FIFO. This is important for security,
because the FIFO allows anyone with write access to run arbitrary commands
in mpv's context using the "run" input command. If you are worried about
security, you should verify that the code handles these concerns correctly.
readable, not trying to open a bogus
"""
import sys
@@ -43,33 +37,13 @@ import errno
import subprocess
import fcntl
import stat
import string
if len(sys.argv) < 1:
if len(sys.argv) < 2:
print("error: needs at least one file as argument.")
sys.exit(1)
files = sys.argv[1:]
# this is about the same method mpv uses to decide this
def is_url(filename):
parts = filename.split(":", 1)
if len(parts) < 2:
return False
# protocol prefix has no special characters => it's an URL
prefix = parts[0]
for c in prefix:
if string.ascii_letters.find(c) < 0:
return False
return True
# make them absolute; also makes them safe against interpretation as options
def make_abs(filename):
if not is_url(filename):
return os.path.abspath(filename)
return filename
files = [make_abs(f) for f in files]
opts_env = os.getenv("UMPV_OPTIONS")
opts_env = opts_env.split() if opts_env else []
files = [os.path.abspath(f) for f in files]
FIFO = "/tmp/umpv-fifo-" + os.getenv("USER")
@@ -112,11 +86,6 @@ else:
pass
else:
raise e
os.mkfifo(FIFO, int("0600", 8))
os.mkfifo(FIFO, 0600)
opts = ["mpv", "--no-terminal", "--force-window", "--input-file=" + FIFO]
opts.extend(opts_env)
opts.append("--")
opts.extend(files)
subprocess.check_call(opts)
subprocess.check_call(["mpv", "--input-file=" + FIFO] + files)
+2 -2
View File
@@ -32,11 +32,11 @@
/*
* frame stepping filter
*
* usage: --vf=dlopen=/path/to/framestep.so:5
* usage: -vf dlopen=./framestep.so:5
*
* outputs every 5th frame
*
* usage: --vf=dlopen=/path/to/framestep.so:5:3
* usage: -vf dlopen=./framestep.so:5:3
*
* outputs every 5th frame, starting with frame index 3 (default: 0)
*/
+1 -1
View File
@@ -284,7 +284,7 @@ int vf_dlopen_getcontext(struct vf_dlopen_context *ctx, int argc, const char **a
il->tc_decision_threshold = A(10, 0.2);
static struct vf_dlopen_formatpair map[] = {
{ "gray", "gray" },
{ "y8", "y8" },
{ NULL, NULL }
};
ctx->format_mapping = map;
+1 -7
View File
@@ -16,7 +16,6 @@ esac
: ${ILDETECT_DRY_RUN:=}
: ${ILDETECT_QUIET:=}
: ${ILDETECT_RUN_INTERLACED_ONLY:=}
: ${ILDETECT_FORCE_RUN:=}
: ${MAKE:=make}
# exit status:
@@ -77,12 +76,7 @@ case "$out" in
exit 2
;;
*"probably: "*)
[ -n "$ILDETECT_FORCE_RUN" ] || exit 8
[ -n "$ILDETECT_DRY_RUN" ] || \
$ILDETECT_MPV "$@" -vf-pre yadif
r=$?
[ $r -eq 0 ] || exit $(($r | 16))
exit 0
exit 8
;;
*)
exit 16
+2 -2
View File
@@ -35,7 +35,7 @@
/*
* rectangle
*
* usage: --vf=dlopen=/path/to/rectangle.so
* usage: -vf dlopen=./rectangle.so
*
* provides an editable rectangle
* NOTE: unix only, and requires xterm to be installed. Don't ask.
@@ -345,7 +345,7 @@ int vf_dlopen_getcontext(struct vf_dlopen_context *ctx, int argc, const char **a
(void) argc;
(void) argv;
static struct vf_dlopen_formatpair map[] = {
{ "yuv420p", "yuv420p" },
{ "yv12", "yv12" },
{ NULL, NULL }
};
privdata *priv = malloc(sizeof(privdata));
+2 -2
View File
@@ -29,7 +29,7 @@
/*
* qscale visualizer
*
* usage: --vf=dlopen=/path/to/showqscale.so
* usage: -vf dlopen=./showqscale.so
*
* uses reddish colors for high QPs, and greenish colors for low QPs
*/
@@ -111,7 +111,7 @@ int vf_dlopen_getcontext(struct vf_dlopen_context *ctx, int argc, const char **a
(void) argc;
(void) argv;
static struct vf_dlopen_formatpair map[] = {
{ "yuv420p", "yuv420p" },
{ "yv12", "yv12" },
{ NULL, NULL }
};
ctx->format_mapping = map;
+1 -1
View File
@@ -32,7 +32,7 @@
/*
* telecine filter
*
* usage: --vf=dlopen=/path/to/telecine.so:t:23
* usage: -vf dlopen=./telecine.so:t:23
*
* Parameter: first parameter is "t" for top field first, "b" for bottom field first
* then digits (0-9) for how many fields a frame is to be displayed
+5 -5
View File
@@ -30,17 +30,17 @@
/*
* tile filter
*
* usage: --vf=dlopen=/path/to/tile.so:4:3
* usage: -vf dlopen=./tile.so:4:3
*
* only supports rgb24 and yuv420p for now
* only supports rgb24 and yv12 for now
* in theory can support any format where rows are a multiple of bytes, and the
* multiple is known
*/
#define ALLFORMATS \
/* format bytes xmul ymul */ \
FORMAT("rgb24" , 3, 1, 1) \
FORMAT("yuv420p", 1, 2, 2)
/* format bytes xmul ymul */ \
FORMAT("rgb24", 3, 1, 1) \
FORMAT("yv12", 1, 2, 2)
typedef struct {
int rows, cols;
-23
View File
@@ -1,23 +0,0 @@
#!/bin/sh
# Example of script for using mpv with youtube-dl
set -e
cookies_dir="$(mktemp -d /tmp/youtube-dl_mpv.XXXX)"
cookies_file="${cookies_dir}/cookies"
user_agent="$(youtube-dl --dump-user-agent)" # or set whatever you want
video_url="$(youtube-dl \
--user-agent="$user_agent" \
--cookies="$cookies_file" \
--get-url \
"$1")"
shift
mpv \
--cookies \
--cookies-file="$cookies_file" \
--user-agent="$user_agent" \
"$@" -- $video_url
rm -rf "$cookies_dir"
-159
View File
@@ -1,159 +0,0 @@
#!/usr/bin/perl
# Generate ZSH completion
use strict;
use warnings;
my $mpv = $ARGV[0] || 'mpv';
my @opts = parse_opts("$mpv --list-options", '^ (\-\-[^\s\*]*)\*?\s*(.*)', 1);
my @ao = parse_opts("$mpv --ao=help", '^ ([^\s\:]*)\s*: (.*)');
my @vo = parse_opts("$mpv --vo=help", '^ ([^\s\:]*)\s*: (.*)');
my @af = parse_opts("$mpv --af=help", '^ ([^\s\:]*)\s*: (.*)');
my @vf = parse_opts("$mpv --vf=help", '^ ([^\s\:]*)\s*: (.*)');
my @protos = parse_opts("$mpv --list-protocols", '^ ([^\s]*)');
my ($opts_str, $ao_str, $vo_str, $af_str, $vf_str, $protos_str);
$opts_str .= qq{ '$_' \\\n} foreach (@opts);
chomp $opts_str;
$ao_str .= qq{ '$_' \\\n} foreach (@ao);
chomp $ao_str;
$vo_str .= qq{ '$_' \\\n} foreach (@vo);
chomp $vo_str;
$af_str .= qq{ '$_' \\\n} foreach (@af);
chomp $af_str;
$vf_str .= qq{ '$_' \\\n} foreach (@vf);
chomp $vf_str;
$protos_str .= qq{$_ } foreach (@protos);
chomp $protos_str;
my $tmpl = <<"EOS";
#compdef mpv
# mpv zsh completion
local curcontext="\$curcontext" state state_descr line
typeset -A opt_args
local rc=1
_arguments -C -S \\
$opts_str
'*:files:->mfiles' && rc=0
case \$state in
ao)
local -a values
values=(
$ao_str
)
_describe -t values 'audio outputs' values && rc=0
;;
vo)
local -a values
values=(
$vo_str
)
_describe -t values 'video outputs' values && rc=0
;;
af)
local -a values
values=(
$af_str
)
_describe -t values 'audio filters' values && rc=0
;;
vf)
local -a values
values=(
$vf_str
)
_describe -t values 'video filters' values && rc=0
;;
mfiles)
local expl
_tags files urls
while _tags; do
_requested files expl 'media file' _files -g \\
"*.(#i)(asf|asx|avi|flac|flv|m1v|m2p|m2v|m4v|mjpg|mka|mkv|mov|mp3|mp4|mpe|mpeg|mpg|ogg|ogm|ogv|qt|rm|ts|vob|wav|webm|wma|wmv)(-.)" && rc=0
if _requested urls; then
while _next_label urls expl URL; do
_urls "\$expl[@]" && rc=0
compadd -S '' "\$expl[@]" $protos_str && rc=0
done
fi
(( rc )) || return 0
done
;;
esac
return rc
EOS
print $tmpl;
sub parse_opts {
my ($cmd, $regex, $parsing_main_options) = @_;
my @list;
my @lines = split /\n/, `$cmd`;
foreach my $line (@lines) {
if ($line !~ /^$regex/) {
next;
}
my $entry = $1;
if ($parsing_main_options) {
$entry .= '=-';
}
if (defined $2) {
my $desc = $2;
$desc =~ s/\:/\\:/g;
$entry .= ':' . $desc;
}
if ($parsing_main_options) {
$entry .= ':';
$entry .= '->ao' if ($1 eq '--ao');
$entry .= '->vo' if ($1 eq '--vo');
$entry .= '->af' if ($1 eq '--af');
$entry .= '->vf' if ($1 eq '--vf');
}
push @list, $entry
}
if ($parsing_main_options) {
@list = sort {
$a =~ /(.*?)\:/; my $ma = $1;
$b =~ /(.*?)\:/; my $mb = $1;
length($mb) <=> length($ma)
} @list;
}
return @list;
}
+1 -1
View File
@@ -1 +1 @@
0.5.2
0.1.2
+13 -179
View File
@@ -15,65 +15,44 @@
* with mpv. If not, see <http://www.gnu.org/licenses/>.
*/
#include <stdint.h>
#include <limits.h>
#include <stdlib.h>
#include <assert.h>
#include <libavutil/mem.h>
#include "talloc.h"
#include "common/common.h"
#include "mpvcore/mp_talloc.h"
#include "audio.h"
static void update_redundant_info(struct mp_audio *mpa)
{
assert(mp_chmap_is_empty(&mpa->channels) ||
mp_chmap_is_valid(&mpa->channels));
mpa->nch = mpa->channels.num;
mpa->bps = af_fmt2bps(mpa->format);
if (af_fmt_is_planar(mpa->format)) {
mpa->spf = 1;
mpa->num_planes = mpa->nch;
mpa->sstride = mpa->bps;
} else {
mpa->spf = mpa->nch;
mpa->num_planes = 1;
mpa->sstride = mpa->bps * mpa->nch;
}
}
void mp_audio_set_format(struct mp_audio *mpa, int format)
{
mpa->format = format;
update_redundant_info(mpa);
mpa->bps = af_fmt2bits(format) / 8;
}
void mp_audio_set_num_channels(struct mp_audio *mpa, int num_channels)
{
mp_chmap_from_channels(&mpa->channels, num_channels);
update_redundant_info(mpa);
struct mp_chmap map;
mp_chmap_from_channels(&map, num_channels);
mp_audio_set_channels(mpa, &map);
}
// Use old MPlayer/ALSA channel layout.
void mp_audio_set_channels_old(struct mp_audio *mpa, int num_channels)
{
mp_chmap_from_channels_alsa(&mpa->channels, num_channels);
update_redundant_info(mpa);
struct mp_chmap map;
mp_chmap_from_channels_alsa(&map, num_channels);
mp_audio_set_channels(mpa, &map);
}
void mp_audio_set_channels(struct mp_audio *mpa, const struct mp_chmap *chmap)
{
assert(mp_chmap_is_empty(chmap) || mp_chmap_is_valid(chmap));
mpa->channels = *chmap;
update_redundant_info(mpa);
mpa->nch = mpa->channels.num;
}
void mp_audio_copy_config(struct mp_audio *dst, const struct mp_audio *src)
{
dst->format = src->format;
dst->channels = src->channels;
mp_audio_set_format(dst, src->format);
mp_audio_set_channels(dst, &src->channels);
dst->rate = src->rate;
update_redundant_info(dst);
}
bool mp_audio_config_equals(const struct mp_audio *a, const struct mp_audio *b)
@@ -82,17 +61,11 @@ bool mp_audio_config_equals(const struct mp_audio *a, const struct mp_audio *b)
mp_chmap_equals(&a->channels, &b->channels);
}
bool mp_audio_config_valid(const struct mp_audio *mpa)
{
return mp_chmap_is_valid(&mpa->channels) && af_fmt_is_valid(mpa->format)
&& mpa->rate >= 1 && mpa->rate < 10000000;
}
char *mp_audio_fmt_to_str(int srate, const struct mp_chmap *chmap, int format)
{
char *chstr = mp_chmap_to_str(chmap);
char *res = talloc_asprintf(NULL, "%dHz %s %dch %s", srate, chstr,
chmap->num, af_fmt_to_str(format));
chmap->num, af_fmt2str_short(format));
talloc_free(chstr);
return res;
}
@@ -101,142 +74,3 @@ char *mp_audio_config_to_str(struct mp_audio *mpa)
{
return mp_audio_fmt_to_str(mpa->rate, &mpa->channels, mpa->format);
}
void mp_audio_force_interleaved_format(struct mp_audio *mpa)
{
if (af_fmt_is_planar(mpa->format))
mp_audio_set_format(mpa, af_fmt_from_planar(mpa->format));
}
// Return used size of a plane. (The size is the same for all planes.)
int mp_audio_psize(struct mp_audio *mpa)
{
return mpa->samples * mpa->sstride;
}
void mp_audio_set_null_data(struct mp_audio *mpa)
{
for (int n = 0; n < MP_NUM_CHANNELS; n++)
mpa->planes[n] = NULL;
mpa->samples = 0;
}
static void mp_audio_destructor(void *ptr)
{
struct mp_audio *mpa = ptr;
for (int n = 0; n < MP_NUM_CHANNELS; n++) {
// Note: don't free if not allocated by mp_audio_realloc
if (mpa->allocated[n])
av_free(mpa->planes[n]);
}
}
/* Reallocate the data stored in mpa->planes[n] so that enough samples are
* available on every plane. The previous data is kept (for the smallest
* common number of samples before/after resize).
*
* mpa->samples is not set or used.
*
* This function is flexible enough to handle format and channel layout
* changes. In these cases, all planes are reallocated as needed. Unused
* planes are freed.
*
* mp_audio_realloc(mpa, 0) will still yield non-NULL for mpa->data[n].
*
* Allocated data is implicitly freed on talloc_free(mpa).
*/
void mp_audio_realloc(struct mp_audio *mpa, int samples)
{
assert(samples >= 0);
if (samples >= INT_MAX / mpa->sstride)
abort(); // oom
int size = MPMAX(samples * mpa->sstride, 1);
for (int n = 0; n < mpa->num_planes; n++) {
if (size != mpa->allocated[n]) {
// Note: av_realloc() can't be used (see libavutil doxygen)
void *new = av_malloc(size);
if (!new)
abort();
if (mpa->allocated[n])
memcpy(new, mpa->planes[n], MPMIN(mpa->allocated[n], size));
av_free(mpa->planes[n]);
mpa->planes[n] = new;
mpa->allocated[n] = size;
}
}
for (int n = mpa->num_planes; n < MP_NUM_CHANNELS; n++) {
av_free(mpa->planes[n]);
mpa->planes[n] = NULL;
mpa->allocated[n] = 0;
}
talloc_set_destructor(mpa, mp_audio_destructor);
}
// Like mp_audio_realloc(), but only reallocate if the audio grows in size.
// If the buffer is reallocated, also preallocate.
void mp_audio_realloc_min(struct mp_audio *mpa, int samples)
{
if (samples > mp_audio_get_allocated_size(mpa)) {
size_t alloc = ta_calc_prealloc_elems(samples);
if (alloc > INT_MAX)
abort(); // oom
mp_audio_realloc(mpa, alloc);
}
}
/* Get the size allocated for the data, in number of samples. If the allocated
* size isn't on sample boundaries (e.g. after format changes), the returned
* sample number is a rounded down value.
*
* Note that this only works in situations where mp_audio_realloc() also works!
*/
int mp_audio_get_allocated_size(struct mp_audio *mpa)
{
int size = 0;
for (int n = 0; n < mpa->num_planes; n++) {
int s = mpa->allocated[n] / mpa->sstride;
size = n == 0 ? s : MPMIN(size, s);
}
return size;
}
// Clear the samples [start, start + length) with silence.
void mp_audio_fill_silence(struct mp_audio *mpa, int start, int length)
{
assert(start >= 0 && length >= 0 && start + length <= mpa->samples);
int offset = start * mpa->sstride;
int size = length * mpa->sstride;
for (int n = 0; n < mpa->num_planes; n++) {
if (n > 0 && mpa->planes[n] == mpa->planes[0])
continue; // silly optimization for special cases
af_fill_silence((char *)mpa->planes[n] + offset, size, mpa->format);
}
}
// All integer parameters are in samples.
// dst and src can overlap.
void mp_audio_copy(struct mp_audio *dst, int dst_offset,
struct mp_audio *src, int src_offset, int length)
{
assert(mp_audio_config_equals(dst, src));
assert(length >= 0);
assert(dst_offset >= 0 && dst_offset + length <= dst->samples);
assert(src_offset >= 0 && src_offset + length <= src->samples);
for (int n = 0; n < dst->num_planes; n++) {
memmove((char *)dst->planes[n] + dst_offset * dst->sstride,
(char *)src->planes[n] + src_offset * src->sstride,
length * dst->sstride);
}
}
// Set data to the audio after the given number of samples (i.e. slice it).
void mp_audio_skip_samples(struct mp_audio *data, int samples)
{
assert(samples >= 0 && samples <= data->samples);
for (int n = 0; n < data->num_planes; n++)
data->planes[n] = (uint8_t *)data->planes[n] + samples * data->sstride;
data->samples -= samples;
}
+5 -30
View File
@@ -23,22 +23,14 @@
// Audio data chunk
struct mp_audio {
int samples; // number of samples in data (per channel)
void *planes[MP_NUM_CHANNELS]; // data buffer (one per plane)
int rate; // sample rate
void *audio; // data buffer
int len; // buffer length (in bytes)
int rate; // sample rate
struct mp_chmap channels; // channel layout, use mp_audio_set_*() to set
int format; // format (AF_FORMAT_...), use mp_audio_set_format() to set
// Redundant fields, for convenience
int sstride; // distance between 2 samples in bytes on a plane
// interleaved: bps * nch
// planar: bps
int nch; // number of channels (redundant with chmap)
int spf; // sub-samples per sample on each plane
int num_planes; // number of planes
int bps; // size of sub-samples (af_fmt2bps(format))
// private
int allocated[MP_NUM_CHANNELS]; // use mp_audio_get_allocated_size()
int nch; // number of channels (redundant with chmap)
int bps; // bytes per sample (redundant with format)
};
void mp_audio_set_format(struct mp_audio *mpa, int format);
@@ -47,25 +39,8 @@ void mp_audio_set_channels_old(struct mp_audio *mpa, int num_channels);
void mp_audio_set_channels(struct mp_audio *mpa, const struct mp_chmap *chmap);
void mp_audio_copy_config(struct mp_audio *dst, const struct mp_audio *src);
bool mp_audio_config_equals(const struct mp_audio *a, const struct mp_audio *b);
bool mp_audio_config_valid(const struct mp_audio *mpa);
char *mp_audio_fmt_to_str(int srate, const struct mp_chmap *chmap, int format);
char *mp_audio_config_to_str(struct mp_audio *mpa);
void mp_audio_force_interleaved_format(struct mp_audio *mpa);
int mp_audio_psize(struct mp_audio *mpa);
void mp_audio_set_null_data(struct mp_audio *mpa);
void mp_audio_realloc(struct mp_audio *mpa, int samples);
void mp_audio_realloc_min(struct mp_audio *mpa, int samples);
int mp_audio_get_allocated_size(struct mp_audio *mpa);
void mp_audio_fill_silence(struct mp_audio *mpa, int start, int length);
void mp_audio_copy(struct mp_audio *dst, int dst_offset,
struct mp_audio *src, int src_offset, int length);
void mp_audio_skip_samples(struct mp_audio *data, int samples);
#endif
-152
View File
@@ -1,152 +0,0 @@
/*
* This file is part of mpv.
*
* mpv is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* mpv is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License along
* with mpv. If not, see <http://www.gnu.org/licenses/>.
*/
#include <stddef.h>
#include <limits.h>
#include <assert.h>
#include "common/common.h"
#include "audio_buffer.h"
#include "audio.h"
#include "format.h"
struct mp_audio_buffer {
struct mp_audio *buffer;
};
struct mp_audio_buffer *mp_audio_buffer_create(void *talloc_ctx)
{
struct mp_audio_buffer *ab = talloc(talloc_ctx, struct mp_audio_buffer);
*ab = (struct mp_audio_buffer) {
.buffer = talloc_zero(ab, struct mp_audio),
};
return ab;
}
// Reinitialize the buffer, set a new format, drop old data.
// The audio data in fmt is not used, only the format.
void mp_audio_buffer_reinit(struct mp_audio_buffer *ab, struct mp_audio *fmt)
{
mp_audio_copy_config(ab->buffer, fmt);
mp_audio_realloc(ab->buffer, 1);
ab->buffer->samples = 0;
}
void mp_audio_buffer_reinit_fmt(struct mp_audio_buffer *ab, int format,
const struct mp_chmap *channels, int srate)
{
struct mp_audio mpa = {0};
mp_audio_set_format(&mpa, format);
mp_audio_set_channels(&mpa, channels);
mpa.rate = srate;
mp_audio_buffer_reinit(ab, &mpa);
}
void mp_audio_buffer_get_format(struct mp_audio_buffer *ab,
struct mp_audio *out_fmt)
{
*out_fmt = (struct mp_audio){0};
mp_audio_copy_config(out_fmt, ab->buffer);
}
// Make the total size of the internal buffer at least this number of samples.
void mp_audio_buffer_preallocate_min(struct mp_audio_buffer *ab, int samples)
{
mp_audio_realloc_min(ab->buffer, samples);
}
// Get number of samples that can be written without forcing a resize of the
// internal buffer.
int mp_audio_buffer_get_write_available(struct mp_audio_buffer *ab)
{
return mp_audio_get_allocated_size(ab->buffer) - ab->buffer->samples;
}
// Get a pointer to the end of the buffer (where writing would append). If the
// internal buffer is too small for the given number of samples, it's resized.
// After writing to the buffer, mp_audio_buffer_finish_write() has to be used
// to make the written data part of the readable buffer.
void mp_audio_buffer_get_write_buffer(struct mp_audio_buffer *ab, int samples,
struct mp_audio *out_buffer)
{
assert(samples >= 0);
mp_audio_realloc_min(ab->buffer, ab->buffer->samples + samples);
*out_buffer = *ab->buffer;
out_buffer->samples = ab->buffer->samples + samples;
mp_audio_skip_samples(out_buffer, ab->buffer->samples);
}
void mp_audio_buffer_finish_write(struct mp_audio_buffer *ab, int samples)
{
assert(samples >= 0 && samples <= mp_audio_buffer_get_write_available(ab));
ab->buffer->samples += samples;
}
// Append data to the end of the buffer.
// If the buffer is not large enough, it is transparently resized.
// For now always copies the data.
void mp_audio_buffer_append(struct mp_audio_buffer *ab, struct mp_audio *mpa)
{
int offset = ab->buffer->samples;
ab->buffer->samples += mpa->samples;
mp_audio_realloc_min(ab->buffer, ab->buffer->samples);
mp_audio_copy(ab->buffer, offset, mpa, 0, mpa->samples);
}
// Prepend silence to the start of the buffer.
void mp_audio_buffer_prepend_silence(struct mp_audio_buffer *ab, int samples)
{
assert(samples >= 0);
int oldlen = ab->buffer->samples;
ab->buffer->samples += samples;
mp_audio_realloc_min(ab->buffer, ab->buffer->samples);
mp_audio_copy(ab->buffer, samples, ab->buffer, 0, oldlen);
mp_audio_fill_silence(ab->buffer, 0, samples);
}
// Get the start of the current readable buffer.
void mp_audio_buffer_peek(struct mp_audio_buffer *ab, struct mp_audio *out_mpa)
{
*out_mpa = *ab->buffer;
}
// Skip leading samples. (Used with mp_audio_buffer_peek() to read data.)
void mp_audio_buffer_skip(struct mp_audio_buffer *ab, int samples)
{
assert(samples >= 0 && samples <= ab->buffer->samples);
mp_audio_copy(ab->buffer, 0, ab->buffer, samples,
ab->buffer->samples - samples);
ab->buffer->samples -= samples;
}
void mp_audio_buffer_clear(struct mp_audio_buffer *ab)
{
ab->buffer->samples = 0;
}
// Return number of buffered audio samples
int mp_audio_buffer_samples(struct mp_audio_buffer *ab)
{
return ab->buffer->samples;
}
// Return amount of buffered audio in seconds.
double mp_audio_buffer_seconds(struct mp_audio_buffer *ab)
{
return ab->buffer->samples / (double)ab->buffer->rate;
}
-44
View File
@@ -1,44 +0,0 @@
/*
* This file is part of mpv.
*
* mpv is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* mpv is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License along
* with mpv. If not, see <http://www.gnu.org/licenses/>.
*/
#ifndef MP_AUDIO_BUFFER_H
#define MP_AUDIO_BUFFER_H
struct mp_audio_buffer;
struct mp_audio;
struct mp_chmap;
struct mp_audio_buffer *mp_audio_buffer_create(void *talloc_ctx);
void mp_audio_buffer_reinit(struct mp_audio_buffer *ab, struct mp_audio *fmt);
void mp_audio_buffer_reinit_fmt(struct mp_audio_buffer *ab, int format,
const struct mp_chmap *channels, int srate);
void mp_audio_buffer_get_format(struct mp_audio_buffer *ab,
struct mp_audio *out_fmt);
void mp_audio_buffer_preallocate_min(struct mp_audio_buffer *ab, int samples);
int mp_audio_buffer_get_write_available(struct mp_audio_buffer *ab);
void mp_audio_buffer_get_write_buffer(struct mp_audio_buffer *ab, int minsamples,
struct mp_audio *out_buffer);
void mp_audio_buffer_finish_write(struct mp_audio_buffer *ab, int samples);
void mp_audio_buffer_append(struct mp_audio_buffer *ab, struct mp_audio *mpa);
void mp_audio_buffer_prepend_silence(struct mp_audio_buffer *ab, int samples);
void mp_audio_buffer_peek(struct mp_audio_buffer *ab, struct mp_audio *out_mpa);
void mp_audio_buffer_skip(struct mp_audio_buffer *ab, int samples);
void mp_audio_buffer_clear(struct mp_audio_buffer *ab);
int mp_audio_buffer_samples(struct mp_audio_buffer *ab);
double mp_audio_buffer_seconds(struct mp_audio_buffer *ab);
#endif
+12 -12
View File
@@ -18,12 +18,12 @@
#include <stdlib.h>
#include <assert.h>
#include "common/msg.h"
#include "mpvcore/mp_msg.h"
#include "chmap.h"
// Names taken from libavutil/channel_layout.c (Not accessible by API.)
// Use of these names is hard-coded in some places (e.g. ao_alsa.c)
static const char *const speaker_names[MP_SPEAKER_ID_COUNT][2] = {
static const char *speaker_names[MP_SPEAKER_ID_COUNT][2] = {
[MP_SPEAKER_ID_FL] = {"fl", "front left"},
[MP_SPEAKER_ID_FR] = {"fr", "front right"},
[MP_SPEAKER_ID_FC] = {"fc", "front center"},
@@ -53,7 +53,7 @@ static const char *const speaker_names[MP_SPEAKER_ID_COUNT][2] = {
// Names taken from libavutil/channel_layout.c (Not accessible by API.)
// Channel order corresponds to lavc/waveex, except for the alsa entries.
static const char *const std_layout_names[][2] = {
static const char *std_layout_names[][2] = {
{"empty", ""}, // not in lavc
{"mono", "fc"},
{"stereo", "fl-fr"},
@@ -105,7 +105,7 @@ static const struct mp_chmap default_layouts[MP_NUM_CHANNELS + 1] = {
// channels. 3 and 7 channels were likely undefined (no ALSA support).
// I'm not sure about the 4 channel case: ALSA uses "quad", while the ffmpeg
// default layout is "4.0".
static const char *const mplayer_layouts[MP_NUM_CHANNELS + 1] = {
static const char *mplayer_layouts[MP_NUM_CHANNELS + 1] = {
[1] = "mono",
[2] = "stereo",
[4] = "quad",
@@ -471,19 +471,19 @@ bool mp_chmap_from_str(struct mp_chmap *dst, bstr src)
return true;
}
void mp_chmap_print_help(struct mp_log *log)
void mp_chmap_print_help(int msgt, int msgl)
{
mp_info(log, "Speakers:\n");
mp_msg(msgt, msgl, "Speakers:\n");
for (int n = 0; n < MP_SPEAKER_ID_COUNT; n++) {
if (speaker_names[n][0])
mp_info(log, " %-16s (%s)\n",
speaker_names[n][0], speaker_names[n][1]);
mp_msg(msgt, msgl, " %-16s (%s)\n",
speaker_names[n][0], speaker_names[n][1]);
}
mp_info(log, "Standard layouts:\n");
mp_msg(msgt, msgl, "Standard layouts:\n");
for (int n = 0; std_layout_names[n][0]; n++) {
mp_info(log, " %-16s (%s)\n",
std_layout_names[n][0], std_layout_names[n][1]);
mp_msg(msgt, msgl, " %-16s (%s)\n",
std_layout_names[n][0], std_layout_names[n][1]);
}
for (int n = 0; n < MP_NUM_CHANNELS; n++)
mp_info(log, " unknown%d\n", n + 1);
mp_msg(msgt, msgl, " unknown%d\n", n);
}
+3 -5
View File
@@ -20,14 +20,14 @@
#include <inttypes.h>
#include <stdbool.h>
#include "bstr/bstr.h"
#include "mpvcore/bstr.h"
#define MP_NUM_CHANNELS 8
// Speaker a channel can be assigned to.
// This corresponds to WAVEFORMATEXTENSIBLE channel mask bit indexes.
// E.g. channel_mask = (1 << MP_SPEAKER_ID_FL) | ...
enum mp_speaker_id {
enum {
// Official WAVEFORMATEXTENSIBLE (shortened names)
MP_SPEAKER_ID_FL = 0, // FRONT_LEFT
MP_SPEAKER_ID_FR, // FRONT_RIGHT
@@ -121,9 +121,7 @@ void mp_chmap_get_reorder(int dst[MP_NUM_CHANNELS], const struct mp_chmap *from,
char *mp_chmap_to_str(const struct mp_chmap *src);
bool mp_chmap_from_str(struct mp_chmap *dst, bstr src);
struct mp_log;
void mp_chmap_print_help(struct mp_log *log);
void mp_chmap_print_help(int msgt, int msgl);
// Use these to avoid chaos in case lavc's definition should diverge from MS.
#define mp_chmap_to_waveext mp_chmap_to_lavc
+16 -21
View File
@@ -18,21 +18,21 @@
#include <stdlib.h>
#include <assert.h>
#include "common/common.h"
#include "chmap_sel.h"
static const struct mp_chmap speaker_replacements[][2] = {
// 5.1 <-> 5.1 (side)
{ MP_CHMAP2(SL, SR), MP_CHMAP2(BL, BR) },
// 7.1 <-> 7.1 (rear ext)
{ MP_CHMAP2(SL, SR), MP_CHMAP2(SDL, SDR) },
// 5.1 and 5.1(side) are practically the same. It doesn't make much sense to
// reject either of them.
static const int replaceable_speakers[][2] = {
{MP_SPEAKER_ID_SL, MP_SPEAKER_ID_BL},
{MP_SPEAKER_ID_SR, MP_SPEAKER_ID_BR},
{-1},
};
// Try to replace speakers from the left of the list with the ones on the
// right, or the other way around.
static bool replace_speakers(struct mp_chmap *map, struct mp_chmap list[2])
// list[] contains a list of speaker pairs, with each pair indicating how
// a speaker can be swapped for another speaker. Try to replace speakers from
// the left of the list with the ones on the right, or the other way around.
static bool replace_speakers(struct mp_chmap *map, const int list[][2])
{
assert(list[0].num == list[1].num);
if (!mp_chmap_is_valid(map))
return false;
for (int dir = 0; dir < 2; dir++) {
@@ -41,9 +41,9 @@ static bool replace_speakers(struct mp_chmap *map, struct mp_chmap list[2])
bool replaced = false;
struct mp_chmap t = *map;
for (int n = 0; n < t.num; n++) {
for (int i = 0; i < list[0].num; i++) {
if (t.speaker[n] == list[from].speaker[i]) {
t.speaker[n] = list[to].speaker[i];
for (int i = 0; list[i][0] != -1; i++) {
if (t.speaker[n] == list[i][from]) {
t.speaker[n] = list[i][to];
replaced = true;
break;
}
@@ -168,14 +168,9 @@ bool mp_chmap_sel_adjust(const struct mp_chmap_sel *s, struct mp_chmap *map)
return true;
}
}
for (int i = 0; i < MP_ARRAY_SIZE(speaker_replacements); i++) {
struct mp_chmap t = *map;
struct mp_chmap *r = (struct mp_chmap *)speaker_replacements[i];
if (replace_speakers(&t, r) && test_layout(s, &t)) {
*map = t;
return true;
}
}
// 5.1 <-> 5.1(side)
if (replace_speakers(map, replaceable_speakers) && test_layout(s, map))
return true;
// Fallback to mono/stereo as last resort
if (map->num == 1) {
*map = (struct mp_chmap) MP_CHMAP_INIT_MONO;
+8 -10
View File
@@ -19,13 +19,11 @@
#ifndef MPLAYER_AD_H
#define MPLAYER_AD_H
#include "common/codecs.h"
#include "mpvcore/codecs.h"
#include "demux/stheader.h"
#include "demux/demux.h"
#include "audio/format.h"
#include "audio/audio.h"
#include "dec_audio.h"
struct mp_decoder_list;
@@ -33,14 +31,14 @@ struct mp_decoder_list;
struct ad_functions {
const char *name;
void (*add_decoders)(struct mp_decoder_list *list);
int (*init)(struct dec_audio *da, const char *decoder);
void (*uninit)(struct dec_audio *da);
int (*control)(struct dec_audio *da, int cmd, void *arg);
int (*decode_packet)(struct dec_audio *da);
int (*preinit)(sh_audio_t *sh);
int (*init)(sh_audio_t *sh, const char *decoder);
void (*uninit)(sh_audio_t *sh);
int (*control)(sh_audio_t *sh, int cmd, void *arg);
int (*decode_audio)(sh_audio_t *sh, unsigned char *buffer, int minlen,
int maxlen);
};
enum ad_ctrl {
ADCTRL_RESET = 1, // flush and reset state, e.g. after seeking
};
#define ADCTRL_RESYNC_STREAM 1 // resync, called after seeking
#endif /* MPLAYER_AD_H */
+242 -138
View File
@@ -29,57 +29,51 @@
#include "talloc.h"
#include "config.h"
#include "common/av_common.h"
#include "common/codecs.h"
#include "common/msg.h"
#include "options/options.h"
#include "mpvcore/av_common.h"
#include "mpvcore/codecs.h"
#include "mpvcore/mp_msg.h"
#include "mpvcore/options.h"
#include "mpvcore/av_opts.h"
#include "ad.h"
#include "audio/reorder_ch.h"
#include "audio/fmt-conversion.h"
#include "compat/mpbswap.h"
#include "compat/libav.h"
struct priv {
AVCodecContext *avctx;
AVFrame *avframe;
struct mp_audio frame;
uint8_t *output;
uint8_t *output_packed; // used by deplanarize to store packed audio samples
int output_left;
int unitsize;
bool force_channel_map;
struct demux_packet *packet;
};
static void uninit(struct dec_audio *da);
static void uninit(sh_audio_t *sh);
static int decode_audio(sh_audio_t *sh,unsigned char *buffer,int minlen,int maxlen);
#define OPT_BASE_STRUCT struct ad_lavc_params
struct ad_lavc_params {
float ac3drc;
int downmix;
int threads;
char **avopts;
};
#define OPT_BASE_STRUCT struct MPOpts
const struct m_sub_options ad_lavc_conf = {
.opts = (const m_option_t[]) {
OPT_FLOATRANGE("ac3drc", ac3drc, 0, 0, 2),
OPT_FLAG("downmix", downmix, 0),
OPT_INTRANGE("threads", threads, 0, 1, 16),
OPT_KEYVALUELIST("o", avopts, 0),
{0}
},
.size = sizeof(struct ad_lavc_params),
.defaults = &(const struct ad_lavc_params){
.ac3drc = 1.,
.downmix = 1,
.threads = 1,
},
const m_option_t ad_lavc_decode_opts_conf[] = {
OPT_FLOATRANGE("ac3drc", ad_lavc_param.ac3drc, 0, 0, 2),
OPT_FLAG("downmix", ad_lavc_param.downmix, 0),
OPT_STRING("o", ad_lavc_param.avopt, 0),
{0}
};
struct pcm_map
{
int tag;
const char *codecs[6]; // {any, 1byte, 2bytes, 3bytes, 4bytes, 8bytes}
const char *codecs[5]; // {any, 1byte, 2bytes, 3bytes, 4bytes}
};
// NOTE: these are needed to make rawaudio with demux_mkv work.
// NOTE: some of these are needed to make rawaudio with demux_mkv and others
// work. ffmpeg does similar mapping internally, not part of the public
// API. Some of these might be dead leftovers for demux_mov support.
static const struct pcm_map tag_map[] = {
// Microsoft PCM
{0x0, {NULL, "pcm_u8", "pcm_s16le", "pcm_s24le", "pcm_s32le"}},
@@ -87,12 +81,26 @@ static const struct pcm_map tag_map[] = {
// MS PCM, Extended
{0xfffe, {NULL, "pcm_u8", "pcm_s16le", "pcm_s24le", "pcm_s32le"}},
// IEEE float
{0x3, {"pcm_f32le", [5] = "pcm_f64le"}},
{0x3, {"pcm_f32le"}},
// 'raw '
{0x20776172, {"pcm_s16be", [1] = "pcm_u8"}},
// 'twos', used by demux_mkv.c internally
{MKTAG('t', 'w', 'o', 's'),
{NULL, "pcm_s8", "pcm_s16be", "pcm_s24be", "pcm_s32be"}},
// 'twos'/'sowt'
{0x736F7774, {"pcm_s16be", [1] = "pcm_s8"}},
{0x74776F73, {"pcm_s16be", [1] = "pcm_s8"}},
// 'fl32'/'FL32'
{0x32336c66, {"pcm_f32be"}},
{0x32334C46, {"pcm_f32be"}},
// '23lf'/'lpcm'
{0x666c3332, {"pcm_f32le"}},
{0x6D63706C, {"pcm_f32le"}},
// 'in24', bigendian int24
{0x34326e69, {"pcm_s24be"}},
// '42ni', little endian int24, MPlayer internal fourCC
{0x696e3234, {"pcm_s24le"}},
// 'in32', bigendian int32
{0x32336e69, {"pcm_s32be"}},
// '23ni', little endian int32, MPlayer internal fourCC
{0x696e3332, {"pcm_s32le"}},
{-1},
};
@@ -115,8 +123,6 @@ static const struct pcm_map af_map[] = {
{AF_FORMAT_S32_BE, {"pcm_s32be"}},
{AF_FORMAT_FLOAT_LE, {"pcm_f32le"}},
{AF_FORMAT_FLOAT_BE, {"pcm_f32be"}},
{AF_FORMAT_DOUBLE_LE, {"pcm_f64le"}},
{AF_FORMAT_DOUBLE_BE, {"pcm_f64be"}},
{-1},
};
@@ -124,13 +130,11 @@ static const char *find_pcm_decoder(const struct pcm_map *map, int format,
int bits_per_sample)
{
int bytes = (bits_per_sample + 7) / 8;
if (bytes == 8)
bytes = 5; // 64 bit entry
for (int n = 0; map[n].tag != -1; n++) {
const struct pcm_map *entry = &map[n];
if (entry->tag == format) {
const char *dec = NULL;
if (bytes >= 1 && bytes <= 5)
if (bytes >= 1 && bytes <= 4)
dec = entry->codecs[bytes];
if (!dec)
dec = entry->codecs[0];
@@ -141,19 +145,28 @@ static const char *find_pcm_decoder(const struct pcm_map *map, int format,
return NULL;
}
static void set_data_from_avframe(struct dec_audio *da)
static int preinit(sh_audio_t *sh)
{
struct priv *priv = da->priv;
AVCodecContext *lavc_context = priv->avctx;
return 1;
}
// Note: invalid parameters are rejected by dec_audio.c
int fmt = lavc_context->sample_fmt;
mp_audio_set_format(&da->decoded, af_from_avformat(fmt));
if (!da->decoded.format)
MP_FATAL(da, "unsupported lavc format %s", av_get_sample_fmt_name(fmt));
da->decoded.rate = lavc_context->sample_rate;
/* Prefer playing audio with the samplerate given in container data
* if available, but take number the number of channels and sample format
* from the codec, since if the codec isn't using the correct values for
* those everything breaks anyway.
*/
static int setup_format(sh_audio_t *sh_audio,
const AVCodecContext *lavc_context)
{
struct priv *priv = sh_audio->context;
int sample_format =
af_from_avformat(av_get_packed_sample_fmt(lavc_context->sample_fmt));
int samplerate = lavc_context->sample_rate;
// If not set, try container samplerate
if (!samplerate && sh_audio->wf) {
samplerate = sh_audio->wf->nSamplesPerSec;
mp_tmsg(MSGT_DECAUDIO, MSGL_V, "ad_lavc: using container rate.\n");
}
struct mp_chmap lavc_chmap;
mp_chmap_from_lavc(&lavc_chmap, lavc_context->channel_layout);
@@ -161,18 +174,23 @@ static void set_data_from_avframe(struct dec_audio *da)
if (lavc_chmap.num != lavc_context->channels)
mp_chmap_from_channels(&lavc_chmap, lavc_context->channels);
if (priv->force_channel_map) {
struct sh_audio *sh_audio = da->header->audio;
if (lavc_chmap.num == sh_audio->channels.num)
lavc_chmap = sh_audio->channels;
}
mp_audio_set_channels(&da->decoded, &lavc_chmap);
da->decoded.samples = priv->avframe->nb_samples;
for (int n = 0; n < da->decoded.num_planes; n++)
da->decoded.planes[n] = priv->avframe->data[n];
if (!mp_chmap_equals(&lavc_chmap, &sh_audio->channels) ||
samplerate != sh_audio->samplerate ||
sample_format != sh_audio->sample_format) {
sh_audio->channels = lavc_chmap;
sh_audio->samplerate = samplerate;
sh_audio->sample_format = sample_format;
sh_audio->samplesize = af_fmt2bits(sh_audio->sample_format) / 8;
return 1;
}
return 0;
}
static void set_from_wf(AVCodecContext *avctx, MP_WAVEFORMATEX *wf)
static void set_from_wf(AVCodecContext *avctx, WAVEFORMATEX *wf)
{
avctx->channels = wf->nChannels;
avctx->sample_rate = wf->nSamplesPerSec;
@@ -180,45 +198,47 @@ static void set_from_wf(AVCodecContext *avctx, MP_WAVEFORMATEX *wf)
avctx->block_align = wf->nBlockAlign;
avctx->bits_per_coded_sample = wf->wBitsPerSample;
if (wf->cbSize > 0)
mp_lavc_set_extradata(avctx, wf + 1, wf->cbSize);
if (wf->cbSize > 0) {
avctx->extradata = av_mallocz(wf->cbSize + FF_INPUT_BUFFER_PADDING_SIZE);
avctx->extradata_size = wf->cbSize;
memcpy(avctx->extradata, wf + 1, avctx->extradata_size);
}
}
static int init(struct dec_audio *da, const char *decoder)
static int init(sh_audio_t *sh_audio, const char *decoder)
{
struct MPOpts *mpopts = da->opts;
struct ad_lavc_params *opts = mpopts->ad_lavc_params;
struct MPOpts *mpopts = sh_audio->opts;
struct ad_lavc_param *opts = &mpopts->ad_lavc_param;
AVCodecContext *lavc_context;
AVCodec *lavc_codec;
struct sh_stream *sh = da->header;
struct sh_audio *sh_audio = sh->audio;
struct priv *ctx = talloc_zero(NULL, struct priv);
da->priv = ctx;
sh_audio->context = ctx;
if (sh_audio->wf && strcmp(decoder, "pcm") == 0) {
decoder = find_pcm_decoder(tag_map, sh->format,
decoder = find_pcm_decoder(tag_map, sh_audio->format,
sh_audio->wf->wBitsPerSample);
} else if (sh_audio->wf && strcmp(decoder, "mp-pcm") == 0) {
decoder = find_pcm_decoder(af_map, sh->format, 0);
decoder = find_pcm_decoder(af_map, sh_audio->format, 0);
ctx->force_channel_map = true;
}
lavc_codec = avcodec_find_decoder_by_name(decoder);
if (!lavc_codec) {
MP_ERR(da, "Cannot find codec '%s' in libavcodec...\n", decoder);
uninit(da);
mp_tmsg(MSGT_DECAUDIO, MSGL_ERR,
"Cannot find codec '%s' in libavcodec...\n", decoder);
uninit(sh_audio);
return 0;
}
lavc_context = avcodec_alloc_context3(lavc_codec);
ctx->avctx = lavc_context;
ctx->avframe = av_frame_alloc();
lavc_context->refcounted_frames = 1;
ctx->avframe = avcodec_alloc_frame();
lavc_context->codec_type = AVMEDIA_TYPE_AUDIO;
lavc_context->codec_id = lavc_codec->id;
if (opts->downmix) {
lavc_context->request_channels = mpopts->audio_output_channels.num;
lavc_context->request_channel_layout =
mp_chmap_to_lavc(&mpopts->audio_output_channels);
}
@@ -227,11 +247,18 @@ static int init(struct dec_audio *da, const char *decoder)
av_opt_set_double(lavc_context, "drc_scale", opts->ac3drc,
AV_OPT_SEARCH_CHILDREN);
mp_set_avopts(da->log, lavc_context, opts->avopts);
if (opts->avopt) {
if (parse_avopts(lavc_context, opts->avopt) < 0) {
mp_msg(MSGT_DECVIDEO, MSGL_ERR,
"ad_lavc: setting AVOptions '%s' failed.\n", opts->avopt);
uninit(sh_audio);
return 0;
}
}
lavc_context->codec_tag = sh->format;
lavc_context->codec_tag = sh_audio->format;
lavc_context->sample_rate = sh_audio->samplerate;
lavc_context->bit_rate = sh_audio->bitrate;
lavc_context->bit_rate = sh_audio->i_bps * 8;
lavc_context->channel_layout = mp_chmap_to_lavc(&sh_audio->channels);
if (sh_audio->wf)
@@ -240,53 +267,79 @@ static int init(struct dec_audio *da, const char *decoder)
// demux_mkv, demux_mpg
if (sh_audio->codecdata_len && sh_audio->codecdata &&
!lavc_context->extradata) {
mp_lavc_set_extradata(lavc_context, sh_audio->codecdata,
sh_audio->codecdata_len);
lavc_context->extradata = av_malloc(sh_audio->codecdata_len +
FF_INPUT_BUFFER_PADDING_SIZE);
lavc_context->extradata_size = sh_audio->codecdata_len;
memcpy(lavc_context->extradata, (char *)sh_audio->codecdata,
lavc_context->extradata_size);
}
if (sh->lav_headers)
mp_copy_lav_codec_headers(lavc_context, sh->lav_headers);
mp_set_avcodec_threads(lavc_context, opts->threads);
if (sh_audio->gsh->lav_headers)
mp_copy_lav_codec_headers(lavc_context, sh_audio->gsh->lav_headers);
/* open it */
if (avcodec_open2(lavc_context, lavc_codec, NULL) < 0) {
MP_ERR(da, "Could not open codec.\n");
uninit(da);
mp_tmsg(MSGT_DECAUDIO, MSGL_ERR, "Could not open codec.\n");
uninit(sh_audio);
return 0;
}
mp_msg(MSGT_DECAUDIO, MSGL_V, "INFO: libavcodec \"%s\" init OK!\n",
lavc_codec->name);
if (lavc_context->bit_rate != 0)
da->bitrate = lavc_context->bit_rate;
else if (sh_audio->wf && sh_audio->wf->nAvgBytesPerSec)
da->bitrate = sh_audio->wf->nAvgBytesPerSec * 8;
// Decode at least 1 byte: (to get header filled)
for (int tries = 0;;) {
int x = decode_audio(sh_audio, sh_audio->a_buffer, 1,
sh_audio->a_buffer_size);
if (x > 0) {
sh_audio->a_buffer_len = x;
break;
}
if (++tries >= 5) {
mp_msg(MSGT_DECAUDIO, MSGL_ERR,
"ad_lavc: initial decode failed\n");
uninit(sh_audio);
return 0;
}
}
sh_audio->i_bps = lavc_context->bit_rate / 8;
if (sh_audio->wf && sh_audio->wf->nAvgBytesPerSec)
sh_audio->i_bps = sh_audio->wf->nAvgBytesPerSec;
int af_sample_fmt =
af_from_avformat(av_get_packed_sample_fmt(lavc_context->sample_fmt));
if (af_sample_fmt == AF_FORMAT_UNKNOWN) {
uninit(sh_audio);
return 0;
}
return 1;
}
static void uninit(struct dec_audio *da)
static void uninit(sh_audio_t *sh)
{
struct priv *ctx = da->priv;
struct priv *ctx = sh->context;
if (!ctx)
return;
AVCodecContext *lavc_context = ctx->avctx;
if (lavc_context) {
if (avcodec_close(lavc_context) < 0)
MP_ERR(da, "Could not close codec.\n");
mp_tmsg(MSGT_DECVIDEO, MSGL_ERR, "Could not close codec.\n");
av_freep(&lavc_context->extradata);
av_freep(&lavc_context);
}
av_frame_free(&ctx->avframe);
avcodec_free_frame(&ctx->avframe);
talloc_free(ctx);
sh->context = NULL;
}
static int control(struct dec_audio *da, int cmd, void *arg)
static int control(sh_audio_t *sh, int cmd, void *arg)
{
struct priv *ctx = da->priv;
struct priv *ctx = sh->context;
switch (cmd) {
case ADCTRL_RESET:
case ADCTRL_RESYNC_STREAM:
avcodec_flush_buffers(ctx->avctx);
mp_audio_set_null_data(&da->decoded);
ctx->output_left = 0;
talloc_free(ctx->packet);
ctx->packet = NULL;
return CONTROL_TRUE;
@@ -294,71 +347,121 @@ static int control(struct dec_audio *da, int cmd, void *arg)
return CONTROL_UNKNOWN;
}
static int decode_packet(struct dec_audio *da)
static av_always_inline void deplanarize(struct sh_audio *sh)
{
struct priv *priv = da->priv;
struct priv *priv = sh->context;
uint8_t **planes = priv->avframe->extended_data;
size_t bps = av_get_bytes_per_sample(priv->avctx->sample_fmt);
size_t nb_samples = priv->avframe->nb_samples;
size_t channels = priv->avctx->channels;
size_t size = bps * nb_samples * channels;
if (talloc_get_size(priv->output_packed) != size)
priv->output_packed =
talloc_realloc_size(priv, priv->output_packed, size);
reorder_to_packed(priv->output_packed, planes, bps, channels, nb_samples);
priv->output = priv->output_packed;
}
static int decode_new_packet(struct sh_audio *sh)
{
struct priv *priv = sh->context;
AVCodecContext *avctx = priv->avctx;
mp_audio_set_null_data(&da->decoded);
struct demux_packet *mpkt = priv->packet;
if (!mpkt) {
if (demux_read_packet_async(da->header, &mpkt) == 0)
return AD_WAIT;
}
if (!mpkt)
mpkt = demux_read_packet(sh->gsh);
if (!mpkt)
return -1; // error or EOF
priv->packet = talloc_steal(priv, mpkt);
int in_len = mpkt ? mpkt->len : 0;
int in_len = mpkt->len;
AVPacket pkt;
mp_set_av_packet(&pkt, mpkt, NULL);
mp_set_av_packet(&pkt, mpkt);
// If we don't have a PTS yet, use the first packet PTS we can get.
if (da->pts == MP_NOPTS_VALUE && mpkt && mpkt->pts != MP_NOPTS_VALUE) {
da->pts = mpkt->pts;
da->pts_offset = 0;
if (mpkt->pts != MP_NOPTS_VALUE) {
sh->pts = mpkt->pts;
sh->pts_bytes = 0;
}
int got_frame = 0;
av_frame_unref(priv->avframe);
int ret = avcodec_decode_audio4(avctx, priv->avframe, &got_frame, &pkt);
if (mpkt) {
// At least "shorten" decodes sub-frames, instead of the whole packet.
// At least "mpc8" can return 0 and wants the packet again next time.
if (ret >= 0) {
ret = FFMIN(ret, mpkt->len); // sanity check against decoder overreads
mpkt->buffer += ret;
mpkt->len -= ret;
mpkt->pts = MP_NOPTS_VALUE; // don't reset PTS next time
}
if (mpkt->len == 0 || ret < 0) {
talloc_free(mpkt);
priv->packet = NULL;
}
// LATM may need many packets to find mux info
if (ret == AVERROR(EAGAIN))
return AD_OK;
if (ret > 0) {
ret = FFMIN(ret, mpkt->len); // sanity check against decoder overreads
mpkt->buffer += ret;
mpkt->len -= ret;
mpkt->pts = MP_NOPTS_VALUE; // don't reset PTS next time
}
if (mpkt->len == 0 || ret <= 0) {
talloc_free(mpkt);
priv->packet = NULL;
}
// LATM may need many packets to find mux info
if (ret == AVERROR(EAGAIN))
return 0;
if (ret < 0) {
MP_ERR(da, "Error decoding audio.\n");
return AD_ERR;
mp_msg(MSGT_DECAUDIO, MSGL_V, "lavc_audio: error\n");
return -1;
}
if (!got_frame)
return mpkt ? AD_OK : AD_EOF;
set_data_from_avframe(da);
double out_pts = mp_pts_from_av(priv->avframe->pkt_pts, NULL);
if (out_pts != MP_NOPTS_VALUE) {
da->pts = out_pts;
da->pts_offset = 0;
return 0;
uint64_t unitsize = (uint64_t)av_get_bytes_per_sample(avctx->sample_fmt) *
avctx->channels;
if (unitsize > 100000)
abort();
priv->unitsize = unitsize;
uint64_t output_left = unitsize * priv->avframe->nb_samples;
if (output_left > 500000000)
abort();
priv->output_left = output_left;
if (av_sample_fmt_is_planar(avctx->sample_fmt) && avctx->channels > 1) {
deplanarize(sh);
} else {
priv->output = priv->avframe->data[0];
}
MP_DBG(da, "Decoded %d -> %d samples\n", in_len, da->decoded.samples);
mp_dbg(MSGT_DECAUDIO, MSGL_DBG2, "Decoded %d -> %d \n", in_len,
priv->output_left);
return 0;
}
static int decode_audio(sh_audio_t *sh_audio, unsigned char *buf, int minlen,
int maxlen)
{
struct priv *priv = sh_audio->context;
AVCodecContext *avctx = priv->avctx;
int len = -1;
while (len < minlen) {
if (!priv->output_left) {
if (decode_new_packet(sh_audio) < 0)
break;
continue;
}
if (setup_format(sh_audio, avctx))
return len;
int size = (minlen - len + priv->unitsize - 1);
size -= size % priv->unitsize;
size = FFMIN(size, priv->output_left);
if (size > maxlen)
abort();
memcpy(buf, priv->output, size);
priv->output += size;
priv->output_left -= size;
if (len < 0)
len = size;
else
len += size;
buf += size;
maxlen -= size;
sh_audio->pts_bytes += size;
}
return len;
}
static void add_decoders(struct mp_decoder_list *list)
{
mp_add_lavc_decoders(list, AVMEDIA_TYPE_AUDIO);
@@ -369,8 +472,9 @@ static void add_decoders(struct mp_decoder_list *list)
const struct ad_functions ad_lavc = {
.name = "lavc",
.add_decoders = add_decoders,
.preinit = preinit,
.init = init,
.uninit = uninit,
.control = control,
.decode_packet = decode_packet,
.decode_audio = decode_audio,
};
+341 -161
View File
@@ -1,7 +1,7 @@
/*
* MPEG 1.0/2.0/2.5 audio layer I, II, III decoding with libmpg123
*
* Copyright (C) 2010-2013 Thomas Orgis <thomas@orgis.org>
* Copyright (C) 2010-2012 Thomas Orgis <thomas@orgis.org>
*
* MPlayer is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
@@ -25,38 +25,42 @@
#include "config.h"
#include "ad.h"
#include "common/msg.h"
#include "mpvcore/mp_msg.h"
/* Reducing the ifdeffery to two main variants:
* 1. most compatible to any libmpg123 version
* 2. fastest variant with recent libmpg123 (>=1.14)
* Running variant 2 on older libmpg123 versions may work in
* principle, but is not supported.
* So, please leave the check for MPG123_API_VERSION there, m-kay?
*/
#include <mpg123.h>
#if (defined MPG123_API_VERSION) && (MPG123_API_VERSION < 33)
#error "This should not happen"
/* Enable faster mode of operation with newer libmpg123, avoiding
* unnecessary memcpy() calls. */
#if (defined MPG123_API_VERSION) && (MPG123_API_VERSION >= 33)
#define AD_MPG123_FRAMEWISE
#endif
/* Switch for updating bitrate info of VBR files. Not essential. */
#define AD_MPG123_MEAN_BITRATE
struct ad_mpg123_context {
mpg123_handle *handle;
int sample_size;
#ifdef AD_MPG123_MEAN_BITRATE
/* Running mean for bit rate, stream length estimation. */
float mean_rate;
unsigned int mean_count;
/* Time delay for updates. */
short delay;
#endif
/* If the stream is actually VBR. */
char vbr;
};
static void uninit(struct dec_audio *da)
{
struct ad_mpg123_context *con = da->priv;
mpg123_close(con->handle);
mpg123_delete(con->handle);
mpg123_exit();
}
/* This initializes libmpg123 and prepares the handle, including funky
* parameters. */
static int init(struct dec_audio *da, const char *decoder)
static int preinit(sh_audio_t *sh)
{
int err;
struct ad_mpg123_context *con;
@@ -67,8 +71,8 @@ static int init(struct dec_audio *da, const char *decoder)
if (mpg123_init() != MPG123_OK)
return 0;
da->priv = talloc_zero(NULL, struct ad_mpg123_context);
con = da->priv;
sh->context = malloc(sizeof(struct ad_mpg123_context));
con = sh->context;
/* Auto-choice of optimized decoder (first argument NULL). */
con->handle = mpg123_new(NULL, &err);
if (!con->handle)
@@ -91,53 +95,39 @@ static int init(struct dec_audio *da, const char *decoder)
* Let's try to run with the default for now. */
/* That would produce floating point output.
* You can get 32 and 24 bit ints, even 8 bit via format matrix.
* If wanting a specific encoding here, configure format matrix and
* make sure it is in set_format(). */
* You can get 32 and 24 bit ints, even 8 bit via format matrix. */
/* mpg123_param(con->handle, MPG123_ADD_FLAGS, MPG123_FORCE_FLOAT, 0.); */
/* Example for RVA choice (available since libmpg123 1.0.0):
mpg123_param(con->handle, MPG123_RVA, MPG123_RVA_MIX, 0.0) */
#ifdef AD_MPG123_FRAMEWISE
/* Prevent funky automatic resampling.
* This way, we can be sure that one frame will never produce
* more than 1152 stereo samples.
* Background:
* Going to decode directly to the output buffer. It is important to have
* MPG123_AUTO_RESAMPLE disabled for the buffer size being an all-time
* limit.
* We need at least 1152 samples. dec_audio.c normally guarantees this. */
* more than 1152 stereo samples. */
mpg123_param(con->handle, MPG123_REMOVE_FLAGS, MPG123_AUTO_RESAMPLE, 0.);
err = mpg123_open_feed(con->handle);
if (err != MPG123_OK)
goto bad_end;
#else
/* Older mpg123 is vulnerable to concatenated streams when gapless cutting
* is enabled (will only play the jingle of a badly constructed radio
* stream). The versions using framewise decoding are fine with that. */
mpg123_param(con->handle, MPG123_REMOVE_FLAGS, MPG123_GAPLESS, 0.);
#endif
return 1;
bad_end:
if (!con->handle) {
MP_ERR(da, "mpg123 preinit error: %s\n", mpg123_plain_strerror(err));
} else {
MP_ERR(da, "mpg123 preinit error: %s\n", mpg123_strerror(con->handle));
}
if (!con->handle)
mp_msg(MSGT_DECAUDIO, MSGL_ERR, "mpg123 preinit error: %s\n",
mpg123_plain_strerror(err));
else
mp_msg(MSGT_DECAUDIO, MSGL_ERR, "mpg123 preinit error: %s\n",
mpg123_strerror(con->handle));
uninit(da);
return 0;
}
static int mpg123_format_to_af(int mpg123_encoding)
{
/* Without external force, mpg123 will always choose signed encoding,
* and non-16-bit only on builds that don't support it.
* Be reminded that it doesn't matter to the MPEG file what encoding
* is produced from it. */
switch (mpg123_encoding) {
case MPG123_ENC_SIGNED_8: return AF_FORMAT_S8;
case MPG123_ENC_SIGNED_16: return AF_FORMAT_S16;
case MPG123_ENC_SIGNED_32: return AF_FORMAT_S32;
case MPG123_ENC_FLOAT_32: return AF_FORMAT_FLOAT;
}
if (con->handle)
mpg123_delete(con->handle);
mpg123_exit();
free(sh->context);
sh->context = NULL;
return 0;
}
@@ -150,139 +140,328 @@ static int compute_bitrate(struct mpg123_frameinfo *i)
{-1, 384, 1152, 576}, /* MPEG 2.5 */
{-1, -1, -1, -1}, /* Unknown */
};
return (int) ((i->framesize + 4) * 8 * i->rate /
return (int) ((i->framesize + 4) * 8 * i->rate * 0.001 /
samples_per_frame[i->version][i->layer] + 0.5);
}
/* Opted against the header printout from old mp3lib, too much
* irrelevant info. This is modelled after the mpg123 app's
* standard output line.
* If more verbosity is demanded, one can add more detail and
* also throw in ID3v2 info which libmpg123 collects anyway. */
static void print_header_compact(struct mpg123_frameinfo *i)
{
static const char *smodes[5] = {
"stereo", "joint-stereo", "dual-channel", "mono", "invalid"
};
static const char *layers[4] = {
"Unknown", "I", "II", "III"
};
static const char *versions[4] = {
"1.0", "2.0", "2.5", "x.x"
};
mp_msg(MSGT_DECAUDIO, MSGL_V, "MPEG %s layer %s, ",
versions[i->version], layers[i->layer]);
switch (i->vbr) {
case MPG123_CBR:
if (i->bitrate)
mp_msg(MSGT_DECAUDIO, MSGL_V, "%d kbit/s", i->bitrate);
else
mp_msg(MSGT_DECAUDIO, MSGL_V, "%d kbit/s (free format)",
compute_bitrate(i));
break;
case MPG123_VBR:
mp_msg(MSGT_DECAUDIO, MSGL_V, "VBR");
break;
case MPG123_ABR:
mp_msg(MSGT_DECAUDIO, MSGL_V, "%d kbit/s ABR", i->abr_rate);
break;
default:
mp_msg(MSGT_DECAUDIO, MSGL_V, "???");
}
mp_msg(MSGT_DECAUDIO, MSGL_V, ", %ld Hz %s\n", i->rate,
smodes[i->mode]);
}
/* This tries to extract a requested amount of decoded data.
* Even when you request 0 bytes, it will feed enough input so that
* the decoder _could_ have delivered something.
* Returns byte count >= 0, -1 on error.
*
* Thoughts on exact pts keeping:
* We have to assume that MPEG frames are cut in pieces by packet boundaries.
* Also, it might be possible that the first packet does not contain enough
* data to ensure initial stream sync... or re-sync on erroneous streams.
* So we need something robust to relate the decoded byte count to the correct
* time stamp. This is tricky, though. From the outside, you cannot tell if,
* after having fed two packets until the first output arrives, one should
* start counting from the first packet's pts or the second packet's.
* So, let's just count from the last fed package's pts. If the packets are
* exactly cut to MPEG frames, this will cause one frame mismatch in the
* beginning (when mpg123 peeks ahead for the following header), but will
* be corrected with the third frame already. One might add special code to
* not increment the base pts past the first packet's after a resync before
* the first decoded bytes arrived. */
static int decode_a_bit(sh_audio_t *sh, unsigned char *buf, int count)
{
int ret = MPG123_OK;
int got = 0;
struct ad_mpg123_context *con = sh->context;
/* There will be one MPG123_NEW_FORMAT message on first open.
* This will be handled in init(). */
do {
size_t got_now = 0;
/* Feed the decoder. This will only fire from the second round on. */
if (ret == MPG123_NEED_MORE) {
/* Feed more input data. */
struct demux_packet *pkt = demux_read_packet(sh->gsh);
if (!pkt)
break; /* Apparently that's it. EOF. */
/* Next bytes from that presentation time. */
if (pkt->pts != MP_NOPTS_VALUE) {
sh->pts = pkt->pts;
sh->pts_bytes = 0;
}
#ifdef AD_MPG123_FRAMEWISE
/* Have to use mpg123_feed() to avoid decoding here. */
ret = mpg123_feed(con->handle, pkt->buffer, pkt->len);
#else
/* Do not use mpg123_feed(), added in later libmpg123 versions. */
ret = mpg123_decode(con->handle, pkt->buffer, pkt->len, NULL, 0, NULL);
#endif
talloc_free(pkt);
if (ret == MPG123_ERR)
break;
}
/* Theoretically, mpg123 could return MPG123_DONE, so be prepared.
* Should not happen in our usage, but it is a valid return code. */
else if (ret == MPG123_ERR || ret == MPG123_DONE)
break;
/* Try to decode a bit. This is the return value that counts
* for the loop condition. */
#ifdef AD_MPG123_FRAMEWISE
if (!buf) { /* fake call just for feeding to get format */
ret = mpg123_getformat(con->handle, NULL, NULL, NULL);
} else { /* This is the decoding. One frame at a time. */
ret = mpg123_replace_buffer(con->handle, buf, count);
if (ret == MPG123_OK)
ret = mpg123_decode_frame(con->handle, NULL, NULL, &got_now);
}
#else
ret = mpg123_decode(con->handle, NULL, 0, buf + got, count - got,
&got_now);
#endif
got += got_now;
sh->pts_bytes += got_now;
#ifdef AD_MPG123_FRAMEWISE
} while (ret == MPG123_NEED_MORE || (got == 0 && count != 0));
#else
} while (ret == MPG123_NEED_MORE || got < count);
#endif
if (ret == MPG123_ERR) {
mp_msg(MSGT_DECAUDIO, MSGL_ERR, "mpg123 decoding failed: %s\n",
mpg123_strerror(con->handle));
mpg123_close(con->handle);
return -1;
}
return got;
}
/* Close, reopen stream. Feed data until we know the format of the stream.
* 1 on success, 0 on error */
static int reopen_stream(sh_audio_t *sh)
{
struct ad_mpg123_context *con = (struct ad_mpg123_context*) sh->context;
mpg123_close(con->handle);
/* No resetting of the context:
* We do not want to loose the mean bitrate data. */
/* Open and make sure we have fed enough data to get stream properties. */
if (MPG123_OK == mpg123_open_feed(con->handle) &&
/* Feed data until mpg123 is ready (has found stream beginning). */
!decode_a_bit(sh, NULL, 0)) {
return 1;
} else {
mp_msg(MSGT_DECAUDIO, MSGL_ERR,
"mpg123 failed to reopen stream: %s\n",
mpg123_strerror(con->handle));
mpg123_close(con->handle);
return 0;
}
}
/* Now we really start accessing some data and determining file format.
* Paranoia note: The mpg123_close() on errors is not really necessary,
* But it ensures that we don't accidentally continue decoding with a
* bad state (possibly interpreting the format badly or whatnot). */
static int init(sh_audio_t *sh, const char *decoder)
{
long rate = 0;
int channels = 0;
int encoding = 0;
mpg123_id3v2 *v2;
struct mpg123_frameinfo finfo;
struct ad_mpg123_context *con = sh->context;
/* We're open about any output format that libmpg123 will suggest.
* Note that a standard build will always default to 16 bit signed and
* the native sample rate of the file. */
if (MPG123_OK == mpg123_format_all(con->handle) &&
reopen_stream(sh) &&
MPG123_OK == mpg123_getformat(con->handle, &rate, &channels, &encoding) &&
/* Forbid the format to change later on. */
MPG123_OK == mpg123_format_none(con->handle) &&
MPG123_OK == mpg123_format(con->handle, rate, channels, encoding) &&
/* Get MPEG header info. */
MPG123_OK == mpg123_info(con->handle, &finfo) &&
/* Since we queried format, mpg123 should have read past ID3v2 tags.
* We need to decide if printing of UTF-8 encoded text info is wanted. */
MPG123_OK == mpg123_id3(con->handle, NULL, &v2)) {
/* If we are here, we passed all hurdles. Yay! Extract the info. */
print_header_compact(&finfo);
/* Do we want to print out the UTF-8 Id3v2 info?
if (v2)
print_id3v2(v2); */
/* Have kb/s, want B/s
* For VBR, the first frame will be a bad estimate. */
sh->i_bps = (finfo.bitrate ? finfo.bitrate : compute_bitrate(&finfo))
* 1000 / 8;
#ifdef AD_MPG123_MEAN_BITRATE
con->delay = 1;
con->mean_rate = 0.;
con->mean_count = 0;
#endif
con->vbr = (finfo.vbr != MPG123_CBR);
mp_chmap_from_channels(&sh->channels, channels);
sh->samplerate = rate;
/* Without external force, mpg123 will always choose signed encoding,
* and non-16-bit only on builds that don't support it.
* Be reminded that it doesn't matter to the MPEG file what encoding
* is produced from it. */
switch (encoding) {
case MPG123_ENC_SIGNED_8:
sh->sample_format = AF_FORMAT_S8;
sh->samplesize = 1;
break;
case MPG123_ENC_SIGNED_16:
sh->sample_format = AF_FORMAT_S16_NE;
sh->samplesize = 2;
break;
/* To stay compatible with the oldest libmpg123 headers, do not rely
* on float and 32 bit encoding symbols being defined.
* Those formats came later */
case 0x1180: /* MPG123_ENC_SIGNED_32 */
sh->sample_format = AF_FORMAT_S32_NE;
sh->samplesize = 4;
break;
case 0x200: /* MPG123_ENC_FLOAT_32 */
sh->sample_format = AF_FORMAT_FLOAT_NE;
sh->samplesize = 4;
break;
default:
mp_msg(MSGT_DECAUDIO, MSGL_ERR,
"Bad encoding from mpg123: %i.\n", encoding);
mpg123_close(con->handle);
return 0;
}
#ifdef AD_MPG123_FRAMEWISE
/* Going to decode directly to MPlayer's memory. It is important
* to have MPG123_AUTO_RESAMPLE disabled for the buffer size
* being an all-time limit. */
sh->audio_out_minsize = 1152 * 2 * sh->samplesize;
#endif
return 1;
} else {
mp_msg(MSGT_DECAUDIO, MSGL_ERR, "mpg123 init error: %s\n",
mpg123_strerror(con->handle));
mpg123_close(con->handle);
return 0;
}
}
static void uninit(sh_audio_t *sh)
{
struct ad_mpg123_context *con = (struct ad_mpg123_context*) sh->context;
mpg123_close(con->handle);
mpg123_delete(con->handle);
free(sh->context);
sh->context = NULL;
mpg123_exit();
}
#ifdef AD_MPG123_MEAN_BITRATE
/* Update mean bitrate. This could be dropped if accurate time display
* on audio file playback is not desired. */
static void update_info(struct dec_audio *da)
static void update_info(sh_audio_t *sh)
{
struct ad_mpg123_context *con = da->priv;
struct mpg123_frameinfo finfo;
if (mpg123_info(con->handle, &finfo) != MPG123_OK)
return;
/* finfo.bitrate is expressed in kilobits */
const int bitrate = finfo.bitrate * 1000;
if (finfo.vbr != MPG123_CBR) {
if (--con->delay < 1) {
struct ad_mpg123_context *con = sh->context;
if (con->vbr && --con->delay < 1) {
struct mpg123_frameinfo finfo;
if (MPG123_OK == mpg123_info(con->handle, &finfo)) {
if (++con->mean_count > ((unsigned int) -1) / 2)
con->mean_count = ((unsigned int) -1) / 4;
/* Might not be numerically optimal, but works fine enough. */
con->mean_rate = ((con->mean_count - 1) * con->mean_rate +
bitrate) / con->mean_count;
da->bitrate = (int) (con->mean_rate + 0.5);
finfo.bitrate) / con->mean_count;
sh->i_bps = (int) (con->mean_rate * 1000 / 8);
con->delay = 10;
}
} else {
da->bitrate = bitrate ? bitrate : compute_bitrate(&finfo);
con->delay = 1;
con->mean_rate = 0.;
con->mean_count = 0;
}
}
#endif
/* libmpg123 has a new format ready; query and store, return return value
of mpg123_getformat() */
static int set_format(struct dec_audio *da)
static int decode_audio(sh_audio_t *sh, unsigned char *buf, int minlen,
int maxlen)
{
struct ad_mpg123_context *con = da->priv;
int ret;
long rate;
int channels;
int encoding;
ret = mpg123_getformat(con->handle, &rate, &channels, &encoding);
if (ret == MPG123_OK) {
mp_audio_set_num_channels(&da->decoded, channels);
da->decoded.rate = rate;
int af = mpg123_format_to_af(encoding);
if (!af) {
/* This means we got a funny custom build of libmpg123 that only supports an unknown format. */
MP_ERR(da, "Bad encoding from mpg123: %i.\n", encoding);
return MPG123_ERR;
}
mp_audio_set_format(&da->decoded, af);
con->sample_size = channels * af_fmt2bps(af);
}
return ret;
int bytes;
bytes = decode_a_bit(sh, buf, maxlen);
if (bytes == 0)
return -1; /* EOF */
#ifdef AD_MPG123_MEAN_BITRATE
update_info(sh);
#endif
return bytes;
}
static int decode_packet(struct dec_audio *da)
static int control(sh_audio_t *sh, int cmd, void *arg)
{
struct ad_mpg123_context *con = da->priv;
int ret;
mp_audio_set_null_data(&da->decoded);
struct demux_packet *pkt;
if (demux_read_packet_async(da->header, &pkt) == 0)
return AD_WAIT;
if (!pkt)
return AD_EOF;
/* Next bytes from that presentation time. */
if (pkt->pts != MP_NOPTS_VALUE) {
da->pts = pkt->pts;
da->pts_offset = 0;
}
/* Have to use mpg123_feed() to avoid decoding here. */
ret = mpg123_feed(con->handle, pkt->buffer, pkt->len);
talloc_free(pkt);
if (ret != MPG123_OK)
goto mpg123_fail;
unsigned char *audio = NULL;
size_t bytes = 0;
ret = mpg123_decode_frame(con->handle, NULL, &audio, &bytes);
if (ret == MPG123_NEED_MORE)
return 0;
if (ret != MPG123_OK && ret != MPG123_DONE && ret != MPG123_NEW_FORMAT)
goto mpg123_fail;
ret = set_format(da);
if (ret != MPG123_OK)
goto mpg123_fail;
if (con->sample_size < 1) {
MP_ERR(da, "no sample size\n");
return AD_ERR;
}
int got_samples = bytes / con->sample_size;
da->decoded.planes[0] = audio;
da->decoded.samples = got_samples;
update_info(da);
return 0;
mpg123_fail:
MP_ERR(da, "mpg123 decoding error: %s\n", mpg123_strerror(con->handle));
return AD_ERR;
}
static int control(struct dec_audio *da, int cmd, void *arg)
{
struct ad_mpg123_context *con = da->priv;
switch (cmd) {
case ADCTRL_RESET:
mp_audio_set_null_data(&da->decoded);
mpg123_close(con->handle);
if (mpg123_open_feed(con->handle) != MPG123_OK) {
MP_ERR(da, "mpg123 failed to reopen stream: %s\n",
mpg123_strerror(con->handle));
case ADCTRL_RESYNC_STREAM:
/* Close/reopen the stream for mpg123 to make sure it doesn't
* think that it still knows the exact stream position.
* Otherwise, we would have funny effects from the gapless code.
* Oh, and it helps to minimize artifacts from jumping in the stream. */
if (reopen_stream(sh)) {
#ifdef AD_MPG123_MEAN_BITRATE
update_info(sh);
#endif
return CONTROL_TRUE;
} else {
/* MPlayer ignores this case! It just keeps on decoding.
* So we have to make sure resync never fails ... */
mp_msg(MSGT_DECAUDIO, MSGL_ERR,
"mpg123 cannot reopen stream for resync.\n");
return CONTROL_FALSE;
}
return CONTROL_TRUE;
break;
}
return CONTROL_UNKNOWN;
}
@@ -296,8 +475,9 @@ static void add_decoders(struct mp_decoder_list *list)
const struct ad_functions ad_mpg123 = {
.name = "mpg123",
.add_decoders = add_decoders,
.preinit = preinit,
.init = init,
.uninit = uninit,
.control = control,
.decode_packet = decode_packet,
.decode_audio = decode_audio,
};
+203 -178
View File
@@ -19,211 +19,60 @@
*/
#include <string.h>
#include <assert.h>
#include <libavformat/avformat.h>
#include <libavcodec/avcodec.h>
#include <libavutil/opt.h>
#include "config.h"
#include "common/msg.h"
#include "common/av_common.h"
#include "options/options.h"
#include "mpvcore/mp_msg.h"
#include "mpvcore/av_common.h"
#include "mpvcore/options.h"
#include "ad.h"
#define FILENAME_SPDIFENC "spdif"
#define OUTBUF_SIZE 65536
struct spdifContext {
struct mp_log *log;
AVFormatContext *lavf_ctx;
int iec61937_packet_size;
int out_buffer_len;
uint8_t out_buffer[OUTBUF_SIZE];
bool need_close;
int out_buffer_size;
uint8_t *out_buffer;
uint8_t pb_buffer[OUTBUF_SIZE];
};
static void uninit(sh_audio_t *sh);
static int read_packet(void *p, uint8_t *buf, int buf_size)
{
// spdifenc does not use read callback.
return 0;
}
static int write_packet(void *p, uint8_t *buf, int buf_size)
{
int len;
struct spdifContext *ctx = p;
int buffer_left = OUTBUF_SIZE - ctx->out_buffer_len;
if (buf_size > buffer_left) {
MP_ERR(ctx, "spdif packet too large.\n");
buf_size = buffer_left;
}
memcpy(&ctx->out_buffer[ctx->out_buffer_len], buf, buf_size);
ctx->out_buffer_len += buf_size;
return buf_size;
len = FFMIN(buf_size, ctx->out_buffer_size -ctx->out_buffer_len);
memcpy(&ctx->out_buffer[ctx->out_buffer_len], buf, len);
ctx->out_buffer_len += len;
return len;
}
static void uninit(struct dec_audio *da)
static int64_t seek(void *p, int64_t offset, int whence)
{
struct spdifContext *spdif_ctx = da->priv;
AVFormatContext *lavf_ctx = spdif_ctx->lavf_ctx;
if (lavf_ctx) {
if (spdif_ctx->need_close)
av_write_trailer(lavf_ctx);
if (lavf_ctx->pb)
av_freep(&lavf_ctx->pb->buffer);
av_freep(&lavf_ctx->pb);
avformat_free_context(lavf_ctx);
}
// spdifenc does not use seek callback.
return 0;
}
static int init(struct dec_audio *da, const char *decoder)
static int preinit(sh_audio_t *sh)
{
struct spdifContext *spdif_ctx = talloc_zero(NULL, struct spdifContext);
da->priv = spdif_ctx;
spdif_ctx->log = da->log;
AVFormatContext *lavf_ctx = avformat_alloc_context();
if (!lavf_ctx)
goto fail;
lavf_ctx->oformat = av_guess_format("spdif", NULL, NULL);
if (!lavf_ctx->oformat)
goto fail;
spdif_ctx->lavf_ctx = lavf_ctx;
void *buffer = av_mallocz(OUTBUF_SIZE);
if (!buffer)
abort();
lavf_ctx->pb = avio_alloc_context(buffer, OUTBUF_SIZE, 1, spdif_ctx, NULL,
write_packet, NULL);
if (!lavf_ctx->pb) {
av_free(buffer);
goto fail;
}
// Request minimal buffering (not available on Libav)
#if LIBAVFORMAT_VERSION_MICRO >= 100
lavf_ctx->pb->direct = 1;
#endif
AVStream *stream = avformat_new_stream(lavf_ctx, 0);
if (!stream)
goto fail;
stream->codec->codec_id = mp_codec_to_av_codec_id(decoder);
AVDictionary *format_opts = NULL;
int num_channels = 0;
int sample_format = 0;
int samplerate = 0;
switch (stream->codec->codec_id) {
case AV_CODEC_ID_AAC:
spdif_ctx->iec61937_packet_size = 16384;
sample_format = AF_FORMAT_IEC61937_LE;
samplerate = 48000;
num_channels = 2;
break;
case AV_CODEC_ID_AC3:
spdif_ctx->iec61937_packet_size = 6144;
sample_format = AF_FORMAT_AC3_LE;
samplerate = 48000;
num_channels = 2;
break;
case AV_CODEC_ID_DTS:
if (da->opts->dtshd) {
av_dict_set(&format_opts, "dtshd_rate", "768000", 0); // 4*192000
spdif_ctx->iec61937_packet_size = 32768;
sample_format = AF_FORMAT_IEC61937_LE;
samplerate = 192000;
num_channels = 2*4;
} else {
spdif_ctx->iec61937_packet_size = 32768;
sample_format = AF_FORMAT_AC3_LE;
samplerate = 48000;
num_channels = 2;
}
break;
case AV_CODEC_ID_EAC3:
spdif_ctx->iec61937_packet_size = 24576;
sample_format = AF_FORMAT_IEC61937_LE;
samplerate = 192000;
num_channels = 2;
break;
case AV_CODEC_ID_MP3:
spdif_ctx->iec61937_packet_size = 4608;
sample_format = AF_FORMAT_MPEG2;
samplerate = 48000;
num_channels = 2;
break;
case AV_CODEC_ID_TRUEHD:
spdif_ctx->iec61937_packet_size = 61440;
sample_format = AF_FORMAT_IEC61937_LE;
samplerate = 192000;
num_channels = 8;
break;
default:
abort();
}
mp_audio_set_num_channels(&da->decoded, num_channels);
mp_audio_set_format(&da->decoded, sample_format);
da->decoded.rate = samplerate;
if (avformat_write_header(lavf_ctx, &format_opts) < 0) {
MP_FATAL(da, "libavformat spdif initialization failed.\n");
av_dict_free(&format_opts);
goto fail;
}
av_dict_free(&format_opts);
spdif_ctx->need_close = true;
sh->samplesize = 2;
return 1;
fail:
uninit(da);
return 0;
}
static int decode_packet(struct dec_audio *da)
{
struct spdifContext *spdif_ctx = da->priv;
AVFormatContext *lavf_ctx = spdif_ctx->lavf_ctx;
mp_audio_set_null_data(&da->decoded);
spdif_ctx->out_buffer_len = 0;
struct demux_packet *mpkt;
if (demux_read_packet_async(da->header, &mpkt) == 0)
return AD_WAIT;
if (!mpkt)
return AD_EOF;
AVPacket pkt;
mp_set_av_packet(&pkt, mpkt, NULL);
pkt.pts = pkt.dts = 0;
MP_VERBOSE(da, "spdif packet, size=%d\n", pkt.size);
if (mpkt->pts != MP_NOPTS_VALUE) {
da->pts = mpkt->pts;
da->pts_offset = 0;
}
int ret = av_write_frame(lavf_ctx, &pkt);
talloc_free(mpkt);
avio_flush(lavf_ctx->pb);
if (ret < 0)
return AD_ERR;
da->decoded.planes[0] = spdif_ctx->out_buffer;
da->decoded.samples = spdif_ctx->out_buffer_len / da->decoded.sstride;
return 0;
}
static int control(struct dec_audio *da, int cmd, void *arg)
{
return CONTROL_UNKNOWN;
}
static const int codecs[] = {
static int codecs[] = {
AV_CODEC_ID_AAC,
AV_CODEC_ID_AC3,
AV_CODEC_ID_DTS,
@@ -233,6 +82,181 @@ static const int codecs[] = {
AV_CODEC_ID_NONE
};
static int init(sh_audio_t *sh, const char *decoder)
{
int srate, bps, *dtshd_rate;
AVFormatContext *lavf_ctx = NULL;
AVStream *stream = NULL;
const AVOption *opt = NULL;
struct spdifContext *spdif_ctx = NULL;
spdif_ctx = av_mallocz(sizeof(*spdif_ctx));
if (!spdif_ctx)
goto fail;
spdif_ctx->lavf_ctx = avformat_alloc_context();
if (!spdif_ctx->lavf_ctx)
goto fail;
sh->context = spdif_ctx;
lavf_ctx = spdif_ctx->lavf_ctx;
lavf_ctx->oformat = av_guess_format(FILENAME_SPDIFENC, NULL, NULL);
if (!lavf_ctx->oformat)
goto fail;
lavf_ctx->priv_data = av_mallocz(lavf_ctx->oformat->priv_data_size);
if (!lavf_ctx->priv_data)
goto fail;
lavf_ctx->pb = avio_alloc_context(spdif_ctx->pb_buffer, OUTBUF_SIZE, 1, spdif_ctx,
read_packet, write_packet, seek);
if (!lavf_ctx->pb)
goto fail;
stream = avformat_new_stream(lavf_ctx, 0);
if (!stream)
goto fail;
lavf_ctx->duration = AV_NOPTS_VALUE;
lavf_ctx->start_time = AV_NOPTS_VALUE;
lavf_ctx->streams[0]->codec->codec_id = mp_codec_to_av_codec_id(decoder);
lavf_ctx->raw_packet_buffer_remaining_size = RAW_PACKET_BUFFER_SIZE;
if (AVERROR_PATCHWELCOME == lavf_ctx->oformat->write_header(lavf_ctx)) {
mp_msg(MSGT_DECAUDIO,MSGL_INFO,
"This codec is not supported by spdifenc.\n");
goto fail;
}
srate = 48000; //fake value
bps = 768000/8; //fake value
int num_channels = 0;
switch (lavf_ctx->streams[0]->codec->codec_id) {
case AV_CODEC_ID_AAC:
spdif_ctx->iec61937_packet_size = 16384;
sh->sample_format = AF_FORMAT_IEC61937_LE;
sh->samplerate = srate;
num_channels = 2;
sh->i_bps = bps;
break;
case AV_CODEC_ID_AC3:
spdif_ctx->iec61937_packet_size = 6144;
sh->sample_format = AF_FORMAT_AC3_LE;
sh->samplerate = srate;
num_channels = 2;
sh->i_bps = bps;
break;
case AV_CODEC_ID_DTS:
if(sh->opts->dtshd) {
opt = av_opt_find(&lavf_ctx->oformat->priv_class,
"dtshd_rate", NULL, 0, 0);
if (!opt)
goto fail;
dtshd_rate = (int*)(((uint8_t*)lavf_ctx->priv_data) +
opt->offset);
*dtshd_rate = 192000*4;
spdif_ctx->iec61937_packet_size = 32768;
sh->sample_format = AF_FORMAT_IEC61937_LE;
sh->samplerate = 192000; // DTS core require 48000
num_channels = 2*4;
sh->i_bps = bps;
} else {
spdif_ctx->iec61937_packet_size = 32768;
sh->sample_format = AF_FORMAT_AC3_LE;
sh->samplerate = srate;
num_channels = 2;
sh->i_bps = bps;
}
break;
case AV_CODEC_ID_EAC3:
spdif_ctx->iec61937_packet_size = 24576;
sh->sample_format = AF_FORMAT_IEC61937_LE;
sh->samplerate = 192000;
num_channels = 2;
sh->i_bps = bps;
break;
case AV_CODEC_ID_MP3:
spdif_ctx->iec61937_packet_size = 4608;
sh->sample_format = AF_FORMAT_MPEG2;
sh->samplerate = srate;
num_channels = 2;
sh->i_bps = bps;
break;
case AV_CODEC_ID_TRUEHD:
spdif_ctx->iec61937_packet_size = 61440;
sh->sample_format = AF_FORMAT_IEC61937_LE;
sh->samplerate = 192000;
num_channels = 8;
sh->i_bps = bps;
break;
default:
break;
}
if (num_channels)
mp_chmap_from_channels(&sh->channels, num_channels);
return 1;
fail:
uninit(sh);
return 0;
}
static int decode_audio(sh_audio_t *sh, unsigned char *buf,
int minlen, int maxlen)
{
struct spdifContext *spdif_ctx = sh->context;
AVFormatContext *lavf_ctx = spdif_ctx->lavf_ctx;
AVPacket pkt;
spdif_ctx->out_buffer_len = 0;
spdif_ctx->out_buffer_size = maxlen;
spdif_ctx->out_buffer = buf;
while (spdif_ctx->out_buffer_len + spdif_ctx->iec61937_packet_size < maxlen
&& spdif_ctx->out_buffer_len < minlen) {
struct demux_packet *mpkt = demux_read_packet(sh->gsh);
if (!mpkt)
break;
mp_set_av_packet(&pkt, mpkt);
mp_msg(MSGT_DECAUDIO,MSGL_V, "pkt.data[%p] pkt.size[%d]\n",
pkt.data, pkt.size);
if (mpkt->pts != MP_NOPTS_VALUE) {
sh->pts = mpkt->pts;
sh->pts_bytes = 0;
}
int out_len = spdif_ctx->out_buffer_len;
int ret = lavf_ctx->oformat->write_packet(lavf_ctx, &pkt);
avio_flush(lavf_ctx->pb);
sh->pts_bytes += spdif_ctx->out_buffer_len - out_len;
talloc_free(mpkt);
if (ret < 0)
break;
}
return spdif_ctx->out_buffer_len;
}
static int control(sh_audio_t *sh, int cmd, void *arg)
{
return CONTROL_UNKNOWN;
}
static void uninit(sh_audio_t *sh)
{
struct spdifContext *spdif_ctx = sh->context;
AVFormatContext *lavf_ctx = spdif_ctx->lavf_ctx;
if (lavf_ctx) {
if (lavf_ctx->oformat)
lavf_ctx->oformat->write_trailer(lavf_ctx);
av_freep(&lavf_ctx->pb);
if (lavf_ctx->streams) {
av_freep(&lavf_ctx->streams[0]->codec);
av_freep(&lavf_ctx->streams[0]->info);
av_freep(&lavf_ctx->streams[0]);
}
av_freep(&lavf_ctx->streams);
av_freep(&lavf_ctx->priv_data);
}
av_freep(&lavf_ctx);
av_freep(&spdif_ctx);
}
static void add_decoders(struct mp_decoder_list *list)
{
for (int n = 0; codecs[n] != AV_CODEC_ID_NONE; n++) {
@@ -247,8 +271,9 @@ static void add_decoders(struct mp_decoder_list *list)
const struct ad_functions ad_spdif = {
.name = "spdif",
.add_decoders = add_decoders,
.preinit = preinit,
.init = init,
.uninit = uninit,
.control = control,
.decode_packet = decode_packet,
.decode_audio = decode_audio,
};
+184 -180
View File
@@ -26,9 +26,9 @@
#include "demux/codec_tags.h"
#include "config.h"
#include "common/codecs.h"
#include "common/msg.h"
#include "bstr/bstr.h"
#include "mpvcore/codecs.h"
#include "mpvcore/mp_msg.h"
#include "mpvcore/bstr.h"
#include "stream/stream.h"
#include "demux/demux.h"
@@ -38,8 +38,6 @@
#include "dec_audio.h"
#include "ad.h"
#include "audio/format.h"
#include "audio/audio.h"
#include "audio/audio_buffer.h"
#include "audio/filter/af.h"
@@ -48,7 +46,7 @@ extern const struct ad_functions ad_lavc;
extern const struct ad_functions ad_spdif;
static const struct ad_functions * const ad_drivers[] = {
#if HAVE_MPG123
#ifdef CONFIG_MPG123
&ad_mpg123,
#endif
&ad_lavc,
@@ -56,44 +54,51 @@ static const struct ad_functions * const ad_drivers[] = {
NULL
};
// Drop audio buffer and reinit it (after format change)
// Returns whether the format was valid at all.
static bool reinit_audio_buffer(struct dec_audio *da)
static int init_audio_codec(sh_audio_t *sh_audio, const char *decoder)
{
if (!mp_audio_config_valid(&da->decoded)) {
MP_ERR(da, "Audio decoder did not specify audio "
"format, or requested an unsupported configuration!\n");
return false;
assert(!sh_audio->initialized);
resync_audio_stream(sh_audio);
sh_audio->samplesize = 4;
sh_audio->sample_format = AF_FORMAT_FLOAT_NE;
sh_audio->audio_out_minsize = 8192; // default, preinit() may change it
if (!sh_audio->ad_driver->preinit(sh_audio)) {
mp_tmsg(MSGT_DECAUDIO, MSGL_ERR, "Audio decoder preinit failed.\n");
return 0;
}
mp_audio_buffer_reinit(da->decode_buffer, &da->decoded);
return true;
}
static void uninit_decoder(struct dec_audio *d_audio)
{
if (d_audio->ad_driver) {
MP_VERBOSE(d_audio, "Uninit audio decoder.\n");
d_audio->ad_driver->uninit(d_audio);
const int base_size = 65536;
// At least 64 KiB plus rounding up to next decodable unit size
sh_audio->a_buffer_size = base_size + sh_audio->audio_out_minsize;
mp_tmsg(MSGT_DECAUDIO, MSGL_V,
"dec_audio: Allocating %d + %d = %d bytes for output buffer.\n",
sh_audio->audio_out_minsize, base_size,
sh_audio->a_buffer_size);
sh_audio->a_buffer = av_mallocz(sh_audio->a_buffer_size);
if (!sh_audio->a_buffer)
abort();
sh_audio->a_buffer_len = 0;
if (!sh_audio->ad_driver->init(sh_audio, decoder)) {
mp_tmsg(MSGT_DECAUDIO, MSGL_V, "Audio decoder init failed.\n");
uninit_audio(sh_audio); // free buffers
return 0;
}
d_audio->ad_driver = NULL;
talloc_free(d_audio->priv);
d_audio->priv = NULL;
}
sh_audio->initialized = 1;
static int init_audio_codec(struct dec_audio *d_audio, const char *decoder)
{
if (!d_audio->ad_driver->init(d_audio, decoder)) {
MP_VERBOSE(d_audio, "Audio decoder init failed.\n");
d_audio->ad_driver = NULL;
uninit_decoder(d_audio);
if (mp_chmap_is_empty(&sh_audio->channels) || !sh_audio->samplerate) {
mp_tmsg(MSGT_DECAUDIO, MSGL_ERR, "Audio decoder did not specify "
"audio format!\n");
uninit_audio(sh_audio); // free buffers
return 0;
}
d_audio->decode_buffer = mp_audio_buffer_create(NULL);
return 1;
}
struct mp_decoder_list *audio_decoder_list(void)
struct mp_decoder_list *mp_audio_decoder_list(void)
{
struct mp_decoder_list *list = talloc_zero(NULL, struct mp_decoder_list);
for (int i = 0; ad_drivers[i] != NULL; i++)
@@ -101,10 +106,10 @@ struct mp_decoder_list *audio_decoder_list(void)
return list;
}
static struct mp_decoder_list *audio_select_decoders(const char *codec,
char *selection)
static struct mp_decoder_list *mp_select_audio_decoders(const char *codec,
char *selection)
{
struct mp_decoder_list *list = audio_decoder_list();
struct mp_decoder_list *list = mp_audio_decoder_list();
struct mp_decoder_list *new = mp_select_decoders(list, codec, selection);
talloc_free(list);
return new;
@@ -119,92 +124,103 @@ static const struct ad_functions *find_driver(const char *name)
return NULL;
}
int audio_init_best_codec(struct dec_audio *d_audio, char *audio_decoders)
int init_best_audio_codec(sh_audio_t *sh_audio, char *audio_decoders)
{
assert(!d_audio->ad_driver);
audio_reset_decoding(d_audio);
assert(!sh_audio->initialized);
struct mp_decoder_entry *decoder = NULL;
struct mp_decoder_list *list =
audio_select_decoders(d_audio->header->codec, audio_decoders);
mp_select_audio_decoders(sh_audio->gsh->codec, audio_decoders);
mp_print_decoders(d_audio->log, MSGL_V, "Codec list:", list);
mp_print_decoders(MSGT_DECAUDIO, MSGL_V, "Codec list:", list);
for (int n = 0; n < list->num_entries; n++) {
struct mp_decoder_entry *sel = &list->entries[n];
const struct ad_functions *driver = find_driver(sel->family);
if (!driver)
continue;
MP_VERBOSE(d_audio, "Opening audio decoder %s:%s\n",
sel->family, sel->decoder);
d_audio->ad_driver = driver;
if (init_audio_codec(d_audio, sel->decoder)) {
mp_tmsg(MSGT_DECAUDIO, MSGL_V, "Opening audio decoder %s:%s\n",
sel->family, sel->decoder);
sh_audio->ad_driver = driver;
if (init_audio_codec(sh_audio, sel->decoder)) {
decoder = sel;
break;
}
MP_WARN(d_audio, "Audio decoder init failed for "
sh_audio->ad_driver = NULL;
mp_tmsg(MSGT_DECAUDIO, MSGL_WARN, "Audio decoder init failed for "
"%s:%s\n", sel->family, sel->decoder);
}
if (d_audio->ad_driver) {
d_audio->decoder_desc =
talloc_asprintf(d_audio, "%s [%s:%s]", decoder->desc, decoder->family,
if (sh_audio->initialized) {
sh_audio->gsh->decoder_desc =
talloc_asprintf(NULL, "%s [%s:%s]", decoder->desc, decoder->family,
decoder->decoder);
MP_VERBOSE(d_audio, "Selected audio codec: %s\n", d_audio->decoder_desc);
mp_msg(MSGT_DECAUDIO, MSGL_INFO, "Selected audio codec: %s\n",
sh_audio->gsh->decoder_desc);
mp_msg(MSGT_DECAUDIO, MSGL_V,
"AUDIO: %d Hz, %d ch, %s\n",
sh_audio->samplerate, sh_audio->channels.num,
af_fmt2str_short(sh_audio->sample_format));
mp_msg(MSGT_IDENTIFY, MSGL_INFO,
"ID_AUDIO_BITRATE=%d\nID_AUDIO_RATE=%d\n" "ID_AUDIO_NCH=%d\n",
sh_audio->i_bps * 8, sh_audio->samplerate, sh_audio->channels.num);
} else {
MP_ERR(d_audio, "Failed to initialize an audio decoder for codec '%s'.\n",
d_audio->header->codec ? d_audio->header->codec : "<unknown>");
mp_msg(MSGT_DECAUDIO, MSGL_ERR,
"Failed to initialize an audio decoder for codec '%s'.\n",
sh_audio->gsh->codec ? sh_audio->gsh->codec : "<unknown>");
}
talloc_free(list);
return !!d_audio->ad_driver;
return sh_audio->initialized;
}
void audio_uninit(struct dec_audio *d_audio)
void uninit_audio(sh_audio_t *sh_audio)
{
if (!d_audio)
return;
if (d_audio->afilter) {
MP_VERBOSE(d_audio, "Uninit audio filters...\n");
af_destroy(d_audio->afilter);
d_audio->afilter = NULL;
if (sh_audio->afilter) {
mp_msg(MSGT_DECAUDIO, MSGL_V, "Uninit audio filters...\n");
af_destroy(sh_audio->afilter);
sh_audio->afilter = NULL;
}
uninit_decoder(d_audio);
talloc_free(d_audio->decode_buffer);
talloc_free(d_audio);
if (sh_audio->initialized) {
mp_tmsg(MSGT_DECAUDIO, MSGL_V, "Uninit audio.\n");
sh_audio->ad_driver->uninit(sh_audio);
sh_audio->initialized = 0;
}
talloc_free(sh_audio->gsh->decoder_desc);
sh_audio->gsh->decoder_desc = NULL;
av_freep(&sh_audio->a_buffer);
}
int audio_init_filters(struct dec_audio *d_audio, int in_samplerate,
int init_audio_filters(sh_audio_t *sh_audio, int in_samplerate,
int *out_samplerate, struct mp_chmap *out_channels,
int *out_format)
{
if (!d_audio->afilter)
d_audio->afilter = af_new(d_audio->global);
struct af_stream *afs = d_audio->afilter;
if (!sh_audio->afilter)
sh_audio->afilter = af_new(sh_audio->opts);
struct af_stream *afs = sh_audio->afilter;
// input format: same as codec's output format:
mp_audio_buffer_get_format(d_audio->decode_buffer, &afs->input);
// Sample rate can be different when adjusting playback speed
afs->input.rate = in_samplerate;
mp_audio_set_channels(&afs->input, &sh_audio->channels);
mp_audio_set_format(&afs->input, sh_audio->sample_format);
// output format: same as ao driver's input format (if missing, fallback to input)
afs->output.rate = *out_samplerate;
mp_audio_set_channels(&afs->output, out_channels);
mp_audio_set_format(&afs->output, *out_format);
afs->replaygain_data = d_audio->replaygain_data;
char *s_from = mp_audio_config_to_str(&afs->input);
char *s_to = mp_audio_config_to_str(&afs->output);
MP_VERBOSE(d_audio, "Building audio filter chain for %s -> %s...\n", s_from, s_to);
mp_tmsg(MSGT_DECAUDIO, MSGL_V,
"Building audio filter chain for %s -> %s...\n", s_from, s_to);
talloc_free(s_from);
talloc_free(s_to);
// let's autoprobe it!
if (af_init(afs) != 0) {
af_destroy(afs);
d_audio->afilter = NULL;
sh_audio->afilter = NULL;
return 0; // failed :(
}
@@ -215,149 +231,137 @@ int audio_init_filters(struct dec_audio *d_audio, int in_samplerate,
return 1;
}
/* Decode packets until we know the audio format. Then reinit the buffer.
* Returns AD_OK on success, negative AD_* code otherwise.
* Also returns AD_OK if already initialized (and does nothing).
*/
int initial_audio_decode(struct dec_audio *da)
static void set_min_out_buffer_size(struct bstr *outbuf, int len)
{
while (!mp_audio_config_valid(&da->decoded)) {
if (da->decoded.samples > 0)
return AD_ERR; // invalid format, rather than uninitialized
int ret = da->ad_driver->decode_packet(da);
if (ret < 0)
return ret;
size_t oldlen = talloc_get_size(outbuf->start);
if (oldlen < len) {
assert(outbuf->start); // talloc context should be already set
mp_msg(MSGT_DECAUDIO, MSGL_V, "Increasing filtered audio buffer size "
"from %zd to %d\n", oldlen, len);
outbuf->start = talloc_realloc_size(NULL, outbuf->start, len);
}
if (mp_audio_buffer_samples(da->decode_buffer) > 0) // avoid accidental flush
return AD_OK;
return reinit_audio_buffer(da) ? AD_OK : AD_ERR;
}
// Filter len bytes of input, put result into outbuf.
static int filter_n_bytes(struct dec_audio *da, struct mp_audio_buffer *outbuf,
int len)
static int filter_n_bytes(sh_audio_t *sh, struct bstr *outbuf, int len)
{
bool format_change = false;
assert(len - 1 + sh->audio_out_minsize <= sh->a_buffer_size);
int error = 0;
assert(len > 0); // would break EOF logic below
while (mp_audio_buffer_samples(da->decode_buffer) < len) {
// Check for a format change
struct mp_audio config;
mp_audio_buffer_get_format(da->decode_buffer, &config);
format_change = !mp_audio_config_equals(&da->decoded, &config);
if (format_change) {
error = AD_EOF; // drain remaining data left in the current buffer
// Decode more bytes if needed
int old_samplerate = sh->samplerate;
struct mp_chmap old_channels = sh->channels;
int old_sample_format = sh->sample_format;
while (sh->a_buffer_len < len) {
unsigned char *buf = sh->a_buffer + sh->a_buffer_len;
int minlen = len - sh->a_buffer_len;
int maxlen = sh->a_buffer_size - sh->a_buffer_len;
int ret = sh->ad_driver->decode_audio(sh, buf, minlen, maxlen);
int format_change = sh->samplerate != old_samplerate
|| !mp_chmap_equals(&sh->channels, &old_channels)
|| sh->sample_format != old_sample_format;
if (ret <= 0 || format_change) {
error = format_change ? -2 : -1;
// samples from format-changing call get discarded too
len = sh->a_buffer_len;
break;
}
if (da->decoded.samples > 0) {
int copy = MPMIN(da->decoded.samples, len);
struct mp_audio append = da->decoded;
append.samples = copy;
mp_audio_buffer_append(da->decode_buffer, &append);
mp_audio_skip_samples(&da->decoded, copy);
da->pts_offset += copy;
continue;
}
error = da->ad_driver->decode_packet(da);
if (error < 0)
break;
sh->a_buffer_len += ret;
}
if (error == AD_WAIT)
return error;
// Filter
struct mp_audio filter_data;
mp_audio_buffer_peek(da->decode_buffer, &filter_data);
filter_data.rate = da->afilter->input.rate; // due to playback speed change
len = MPMIN(filter_data.samples, len);
filter_data.samples = len;
bool eof = error == AD_EOF && filter_data.samples == 0;
struct mp_audio filter_input = {
.audio = sh->a_buffer,
.len = len,
.rate = sh->samplerate,
};
mp_audio_set_format(&filter_input, sh->sample_format);
mp_audio_set_channels(&filter_input, &sh->channels);
if (af_filter(da->afilter, &filter_data, eof ? AF_FILTER_FLAG_EOF : 0) < 0)
return AD_ERR;
mp_audio_buffer_append(outbuf, &filter_data);
if (error == AD_EOF && filter_data.samples > 0)
error = 0; // don't end playback yet
struct mp_audio *filter_output = af_play(sh->afilter, &filter_input);
if (!filter_output)
return -1;
set_min_out_buffer_size(outbuf, outbuf->len + filter_output->len);
memcpy(outbuf->start + outbuf->len, filter_output->audio,
filter_output->len);
outbuf->len += filter_output->len;
// remove processed data from decoder buffer:
mp_audio_buffer_skip(da->decode_buffer, len);
// if format was changed, and all data was drained, execute the format change
if (format_change && eof) {
error = AD_NEW_FMT;
if (!reinit_audio_buffer(da))
error = AD_ERR; // switch to invalid format
}
sh->a_buffer_len -= len;
memmove(sh->a_buffer, sh->a_buffer + len, sh->a_buffer_len);
return error;
}
/* Try to get at least minsamples decoded+filtered samples in outbuf
/* Try to get at least minlen decoded+filtered bytes in outbuf
* (total length including possible existing data).
* Return 0 on success, or negative AD_* error code.
* In the former case outbuf has at least minsamples buffered on return.
* In case of EOF/error it might or might not be. */
int audio_decode(struct dec_audio *d_audio, struct mp_audio_buffer *outbuf,
int minsamples)
* Return 0 on success, -1 on error/EOF (not distinguished).
* In the former case outbuf->len is always >= minlen on return.
* In case of EOF/error it might or might not be.
* Outbuf.start must be talloc-allocated, and will be reallocated
* if needed to fit all filter output. */
int decode_audio(sh_audio_t *sh_audio, struct bstr *outbuf, int minlen)
{
if (!d_audio->afilter)
return AD_ERR;
// Indicates that a filter seems to be buffering large amounts of data
int huge_filter_buffer = 0;
// Decoded audio must be cut at boundaries of this many bytes
int unitsize = sh_audio->channels.num * sh_audio->samplesize * 16;
/* Filter output size will be about filter_multiplier times input size.
* If some filter buffers audio in big blocks this might only hold
* as average over time. */
double filter_multiplier = af_calc_filter_multiplier(d_audio->afilter);
double filter_multiplier = af_calc_filter_multiplier(sh_audio->afilter);
int prev_buffered = -1;
int res = 0;
MP_STATS(d_audio, "start audio");
while (res >= 0 && minsamples >= 0) {
int buffered = mp_audio_buffer_samples(outbuf);
if (minsamples < buffered || buffered == prev_buffered)
break;
prev_buffered = buffered;
/* If the decoder set audio_out_minsize then it can do the equivalent of
* "while (output_len < target_len) output_len += audio_out_minsize;",
* so we must guarantee there is at least audio_out_minsize-1 bytes
* more space in the output buffer than the minimum length we try to
* decode. */
int max_decode_len = sh_audio->a_buffer_size - sh_audio->audio_out_minsize;
if (!unitsize)
return -1;
max_decode_len -= max_decode_len % unitsize;
int decsamples = (minsamples - buffered) / filter_multiplier;
// + some extra for possible filter buffering, and avoid 0
decsamples += 512;
if (huge_filter_buffer) {
while (minlen >= 0 && outbuf->len < minlen) {
// + some extra for possible filter buffering
int declen = (minlen - outbuf->len) / filter_multiplier + (unitsize << 5);
if (huge_filter_buffer)
/* Some filter must be doing significant buffering if the estimated
* input length didn't produce enough output from filters.
* Feed the filters 250 samples at a time until we have enough
* output. Very small amounts could make filtering inefficient while
* large amounts can make mpv demux the file unnecessarily far ahead
* Feed the filters 2k bytes at a time until we have enough output.
* Very small amounts could make filtering inefficient while large
* amounts can make MPlayer demux the file unnecessarily far ahead
* to get audio data and buffer video frames in memory while doing
* so. However the performance impact of either is probably not too
* significant as long as the value is not completely insane. */
decsamples = 250;
}
/* if this iteration does not fill buffer, we must have lots
* of buffering in filters */
huge_filter_buffer = 1;
res = filter_n_bytes(d_audio, outbuf, decsamples);
declen = 2000;
declen -= declen % unitsize;
if (declen > max_decode_len)
declen = max_decode_len;
else
/* if this iteration does not fill buffer, we must have lots
* of buffering in filters */
huge_filter_buffer = 1;
int res = filter_n_bytes(sh_audio, outbuf, declen);
if (res < 0)
return res;
}
MP_STATS(d_audio, "end audio");
return res;
return 0;
}
void audio_reset_decoding(struct dec_audio *d_audio)
void decode_audio_prepend_bytes(struct bstr *outbuf, int count, int byte)
{
if (d_audio->ad_driver)
d_audio->ad_driver->control(d_audio, ADCTRL_RESET, NULL);
if (d_audio->afilter)
af_control_all(d_audio->afilter, AF_CONTROL_RESET, NULL);
d_audio->pts = MP_NOPTS_VALUE;
d_audio->pts_offset = 0;
if (d_audio->decode_buffer)
mp_audio_buffer_clear(d_audio->decode_buffer);
set_min_out_buffer_size(outbuf, outbuf->len + count);
memmove(outbuf->start + count, outbuf->start, outbuf->len);
memset(outbuf->start, byte, count);
outbuf->len += count;
}
void resync_audio_stream(sh_audio_t *sh_audio)
{
sh_audio->pts = MP_NOPTS_VALUE;
if (!sh_audio->initialized)
return;
sh_audio->ad_driver->control(sh_audio, ADCTRL_RESYNC_STREAM, NULL);
}
+9 -41
View File
@@ -20,52 +20,20 @@
#define MPLAYER_DEC_AUDIO_H
#include "audio/chmap.h"
#include "audio/audio.h"
#include "demux/demux.h"
#include "demux/stheader.h"
struct mp_audio_buffer;
struct bstr;
struct mp_decoder_list;
struct dec_audio {
struct mp_log *log;
struct MPOpts *opts;
struct mpv_global *global;
const struct ad_functions *ad_driver;
struct sh_stream *header;
struct mp_audio_buffer *decode_buffer;
struct af_stream *afilter;
char *decoder_desc;
struct replaygain_data *replaygain_data;
int init_retries;
// set by decoder
struct mp_audio decoded; // decoded audio set by last decode_packet() call
int bitrate; // input bitrate, can change with VBR sources
// last known pts value in output from decoder
double pts;
// number of samples output by decoder after last known pts
int pts_offset;
// For free use by the ad_driver
void *priv;
};
struct mp_decoder_list *mp_audio_decoder_list(void);
int init_best_audio_codec(sh_audio_t *sh_audio, char *audio_decoders);
int decode_audio(sh_audio_t *sh_audio, struct bstr *outbuf, int minlen);
void decode_audio_prepend_bytes(struct bstr *outbuf, int count, int byte);
void resync_audio_stream(sh_audio_t *sh_audio);
void skip_audio_frame(sh_audio_t *sh_audio);
void uninit_audio(sh_audio_t *sh_audio);
enum {
AD_OK = 0,
AD_ERR = -1,
AD_EOF = -2,
AD_NEW_FMT = -3,
AD_WAIT = -4,
};
struct mp_decoder_list *audio_decoder_list(void);
int audio_init_best_codec(struct dec_audio *d_audio, char *audio_decoders);
int audio_decode(struct dec_audio *d_audio, struct mp_audio_buffer *outbuf,
int minsamples);
int initial_audio_decode(struct dec_audio *d_audio);
void audio_reset_decoding(struct dec_audio *d_audio);
void audio_uninit(struct dec_audio *d_audio);
int audio_init_filters(struct dec_audio *d_audio, int in_samplerate,
int init_audio_filters(sh_audio_t *sh_audio, int in_samplerate,
int *out_samplerate, struct mp_chmap *out_channels,
int *out_format);
+178 -211
View File
@@ -22,53 +22,48 @@
#include <string.h>
#include <assert.h>
#include "common/common.h"
#include "common/global.h"
#include "options/m_option.h"
#include "options/m_config.h"
#include "mpvcore/m_option.h"
#include "mpvcore/m_config.h"
#include "af.h"
// Static list of filters
extern const struct af_info af_info_dummy;
extern const struct af_info af_info_delay;
extern const struct af_info af_info_channels;
extern const struct af_info af_info_format;
extern const struct af_info af_info_force;
extern const struct af_info af_info_volume;
extern const struct af_info af_info_equalizer;
extern const struct af_info af_info_pan;
extern const struct af_info af_info_surround;
extern const struct af_info af_info_sub;
extern const struct af_info af_info_export;
extern const struct af_info af_info_drc;
extern const struct af_info af_info_extrastereo;
extern const struct af_info af_info_lavcac3enc;
extern const struct af_info af_info_lavrresample;
extern const struct af_info af_info_sweep;
extern const struct af_info af_info_hrtf;
extern const struct af_info af_info_ladspa;
extern const struct af_info af_info_center;
extern const struct af_info af_info_sinesuppress;
extern const struct af_info af_info_karaoke;
extern const struct af_info af_info_scaletempo;
extern const struct af_info af_info_bs2b;
extern const struct af_info af_info_lavfi;
extern const struct af_info af_info_convert24;
extern const struct af_info af_info_convertsignendian;
extern struct af_info af_info_dummy;
extern struct af_info af_info_delay;
extern struct af_info af_info_channels;
extern struct af_info af_info_format;
extern struct af_info af_info_force;
extern struct af_info af_info_volume;
extern struct af_info af_info_equalizer;
extern struct af_info af_info_pan;
extern struct af_info af_info_surround;
extern struct af_info af_info_sub;
extern struct af_info af_info_export;
extern struct af_info af_info_drc;
extern struct af_info af_info_extrastereo;
extern struct af_info af_info_lavcac3enc;
extern struct af_info af_info_lavrresample;
extern struct af_info af_info_sweep;
extern struct af_info af_info_hrtf;
extern struct af_info af_info_ladspa;
extern struct af_info af_info_center;
extern struct af_info af_info_sinesuppress;
extern struct af_info af_info_karaoke;
extern struct af_info af_info_scaletempo;
extern struct af_info af_info_bs2b;
extern struct af_info af_info_lavfi;
static const struct af_info *const filter_list[] = {
static struct af_info* filter_list[] = {
&af_info_dummy,
&af_info_delay,
&af_info_channels,
&af_info_format,
&af_info_force,
&af_info_volume,
&af_info_equalizer,
&af_info_pan,
&af_info_surround,
&af_info_sub,
#if HAVE_SYS_MMAN_H
#ifdef HAVE_SYS_MMAN_H
&af_info_export,
#endif
&af_info_drc,
@@ -77,22 +72,21 @@ static const struct af_info *const filter_list[] = {
&af_info_lavrresample,
&af_info_sweep,
&af_info_hrtf,
#if HAVE_LADSPA
#ifdef CONFIG_LADSPA
&af_info_ladspa,
#endif
&af_info_center,
&af_info_sinesuppress,
&af_info_karaoke,
&af_info_scaletempo,
#if HAVE_LIBBS2B
#ifdef CONFIG_LIBBS2B
&af_info_bs2b,
#endif
#if HAVE_LIBAVFILTER
#ifdef CONFIG_AF_LAVFI
&af_info_lavfi,
#endif
// Must come last, because they're fallback format conversion filter
&af_info_convert24,
&af_info_convertsignendian,
// Must come last, because it's the fallback format conversion filter
&af_info_format,
NULL
};
@@ -115,10 +109,7 @@ static bool get_desc(struct m_obj_desc *dst, int index)
const struct m_obj_list af_obj_list = {
.get_desc = get_desc,
.description = "audio filters",
.aliases = {
{"force", "format"},
{0}
},
.legacy_hacks = true, // many filters have custom option parsing
};
static bool af_config_equals(struct mp_audio *a, struct mp_audio *b)
@@ -142,7 +133,7 @@ static int input_control(struct af_instance* af, int cmd, void* arg)
{
switch (cmd) {
case AF_CONTROL_REINIT:
assert(arg == &((struct af_stream *)af->priv)->input);
assert(arg == &((struct af_stream *)af->setup)->input);
return AF_OK;
}
return AF_UNKNOWN;
@@ -150,7 +141,7 @@ static int input_control(struct af_instance* af, int cmd, void* arg)
static int output_control(struct af_instance* af, int cmd, void* arg)
{
struct af_stream *s = af->priv;
struct af_stream *s = af->setup;
struct mp_audio *output = &s->output;
struct mp_audio *filter_output = &s->filter_output;
@@ -168,9 +159,9 @@ static int output_control(struct af_instance* af, int cmd, void* arg)
return AF_UNKNOWN;
}
static int dummy_filter(struct af_instance* af, struct mp_audio* data, int f)
static struct mp_audio *dummy_play(struct af_instance* af, struct mp_audio* data)
{
return 0;
return data;
}
/* Function for creating a new filter of type name.The name may
@@ -180,7 +171,8 @@ static struct af_instance *af_create(struct af_stream *s, char *name,
{
struct m_obj_desc desc;
if (!m_obj_list_find(&desc, &af_obj_list, bstr0(name))) {
MP_ERR(s, "Couldn't find audio filter '%s'.\n", name);
mp_tmsg(MSGT_VFILTER, MSGL_ERR,
"Couldn't find audio filter '%s'.\n", name);
return NULL;
}
const struct af_info *info = desc.p;
@@ -189,38 +181,40 @@ static struct af_instance *af_create(struct af_stream *s, char *name,
if (info->flags & AF_FLAGS_NOT_REENTRANT) {
for (struct af_instance *cur = s->first; cur; cur = cur->next) {
if (cur->info == info) {
MP_ERR(s, "There can only be one "
mp_msg(MSGT_AFILTER, MSGL_ERR, "[libaf] There can only be one "
"instance of the filter '%s' in each stream\n", name);
return NULL;
}
}
}
MP_VERBOSE(s, "Adding filter %s \n", name);
mp_msg(MSGT_AFILTER, MSGL_V, "[libaf] Adding filter %s \n", name);
struct af_instance *af = talloc_zero(NULL, struct af_instance);
*af = (struct af_instance) {
.info = info,
.mul = 1,
.data = talloc_zero(af, struct mp_audio),
.log = mp_log_new(af, s->log, name),
.replaygain_data = s->replaygain_data,
};
struct m_config *config = m_config_from_obj_desc(af, s->log, &desc);
if (m_config_apply_defaults(config, name, s->opts->af_defs) < 0)
struct m_config *config = m_config_from_obj_desc(af, &desc);
if (m_config_initialize_obj(config, &desc, &af->priv, &args) < 0)
goto error;
if (m_config_set_obj_params(config, args) < 0)
goto error;
af->priv = config->optstruct;
// Initialize the new filter
if (af->info->open(af) != AF_OK)
goto error;
if (args && af->control) {
// Single option string for old filters
char *opts = (char *)args; // m_config_initialize_obj did this
assert(!af->priv);
if (af->control(af, AF_CONTROL_COMMAND_LINE, opts) <= AF_ERROR)
goto error;
}
return af;
error:
MP_ERR(s, "Couldn't create or open audio filter '%s'\n", name);
mp_msg(MSGT_AFILTER, MSGL_ERR,
"[libaf] Couldn't create or open audio filter '%s'\n", name);
talloc_free(af);
return NULL;
}
@@ -281,14 +275,17 @@ static void af_remove(struct af_stream *s, struct af_instance *af)
return;
// Print friendly message
MP_VERBOSE(s, "Removing filter %s \n", af->info->name);
mp_msg(MSGT_AFILTER, MSGL_V, "[libaf] Removing filter %s \n",
af->info->name);
// Notify filter before changing anything
af->control(af, AF_CONTROL_PRE_DESTROY, 0);
// Detach pointers
af->prev->next = af->next;
af->next->prev = af->prev;
if (af->uninit)
af->uninit(af);
af->uninit(af);
talloc_free(af);
}
@@ -306,26 +303,26 @@ repeat:
static void af_print_filter_chain(struct af_stream *s, struct af_instance *at,
int msg_level)
{
MP_MSG(s, msg_level, "Audio filter chain:\n");
mp_msg(MSGT_AFILTER, msg_level, "Audio filter chain:\n");
struct af_instance *af = s->first;
while (af) {
char b[128] = {0};
mp_snprintf_cat(b, sizeof(b), " [%s] ", af->info->name);
mp_msg(MSGT_AFILTER, msg_level, " [%s] ", af->info->name);
if (af->data) {
char *info = mp_audio_config_to_str(af->data);
mp_snprintf_cat(b, sizeof(b), "%s", info);
mp_msg(MSGT_AFILTER, msg_level, "%s", info);
talloc_free(info);
}
if (af == at)
mp_snprintf_cat(b, sizeof(b), " <-");
MP_MSG(s, msg_level, "%s\n", b);
mp_msg(MSGT_AFILTER, msg_level, " <-");
mp_msg(MSGT_AFILTER, msg_level, "\n");
af = af->next;
}
mp_msg(MSGT_AFILTER, msg_level, " [ao] ");
char *info = mp_audio_config_to_str(&s->output);
MP_MSG(s, msg_level, " [ao] %s\n", info);
mp_msg(MSGT_AFILTER, msg_level, "%s\n", info);
talloc_free(info);
}
@@ -337,76 +334,14 @@ static int af_count_filters(struct af_stream *s)
return count;
}
// Finds the first conversion filter on the way from srcfmt to dstfmt.
// Conversions form a DAG: each node is a format/filter pair, and possible
// conversions are edges. We search the DAG for the shortest path.
// Some cases visit the same filter multiple times, but with different formats
// (like u24le->s8), so one node per format or filter separate is not enough.
// Returns the filter and dest. format for the first conversion step.
// (So we know what conversion filter with what format to insert next.)
static char *af_find_conversion_filter(int srcfmt, int *dstfmt)
static char *af_find_conversion_filter(int srcfmt, int dstfmt)
{
#define MAX_NODES (64 * 32)
int num_fmt = 0, num_filt = 0;
for (int n = 0; filter_list[n]; n++)
num_filt = n + 1;
for (int n = 0; af_fmtstr_table[n].format; n++)
num_fmt = n + 1;
int num_nodes = num_fmt * num_filt;
assert(num_nodes < MAX_NODES);
bool visited[MAX_NODES] = {0};
unsigned char distance[MAX_NODES];
short previous[MAX_NODES] = {0};
for (int n = 0; n < num_nodes; n++) {
distance[n] = 255;
if (af_fmtstr_table[n % num_fmt].format == srcfmt)
distance[n] = 0;
for (int n = 0; filter_list[n]; n++) {
struct af_info *af = filter_list[n];
if (af->test_conversion && af->test_conversion(srcfmt, dstfmt))
return (char *)af->name;
}
while (1) {
int next = -1;
for (int n = 0; n < num_nodes; n++) {
if (!visited[n] && (next < 0 || (distance[n] < distance[next])))
next = n;
}
if (next < 0 || distance[next] == 255)
return NULL;
visited[next] = true;
int fmt = next % num_fmt;
if (af_fmtstr_table[fmt].format == *dstfmt) {
// Best match found
for (int cur = next; cur >= 0; cur = previous[cur] - 1) {
if (distance[cur] == 1) {
*dstfmt = af_fmtstr_table[cur % num_fmt].format;
return (char *)filter_list[cur / num_fmt]->name;
}
}
return NULL;
}
for (int n = 0; filter_list[n]; n++) {
const struct af_info *af = filter_list[n];
if (!af->test_conversion)
continue;
for (int i = 0; af_fmtstr_table[i].format; i++) {
if (i != fmt && af->test_conversion(af_fmtstr_table[fmt].format,
af_fmtstr_table[i].format))
{
int other = n * num_fmt + i;
int ndist = distance[next] + 1;
if (ndist < distance[other]) {
distance[other] = ndist;
previous[other] = next + 1;
}
}
}
}
}
assert(0);
#undef MAX_NODES
return NULL;
}
static bool af_is_conversion_filter(struct af_instance *af)
@@ -417,7 +352,7 @@ static bool af_is_conversion_filter(struct af_instance *af)
// in is what af can take as input - insert a conversion filter if the actual
// input format doesn't match what af expects.
// Returns:
// AF_OK: must call af_reinit() or equivalent, format matches (or is closer)
// AF_OK: must call af_reinit() or equivalent, format matches
// AF_FALSE: nothing was changed, format matches
// else: error
static int af_fix_format_conversion(struct af_stream *s,
@@ -430,21 +365,18 @@ static int af_fix_format_conversion(struct af_stream *s,
struct mp_audio actual = *prev->data;
if (actual.format == in.format)
return AF_FALSE;
int dstfmt = in.format;
char *filter = af_find_conversion_filter(actual.format, &dstfmt);
if (prev->control(prev, AF_CONTROL_FORMAT_FMT, &in.format) == AF_OK) {
*p_af = prev;
return AF_OK;
}
char *filter = af_find_conversion_filter(actual.format, in.format);
if (!filter)
return AF_ERROR;
if (strcmp(filter, prev->info->name) == 0) {
if (prev->control(prev, AF_CONTROL_SET_FORMAT, &dstfmt) == AF_OK) {
*p_af = prev;
return AF_OK;
}
}
struct af_instance *new = af_prepend(s, af, filter, NULL);
if (new == NULL)
return AF_ERROR;
new->auto_inserted = true;
if (AF_OK != (rv = new->control(new, AF_CONTROL_SET_FORMAT, &dstfmt)))
if (AF_OK != (rv = new->control(new, AF_CONTROL_FORMAT_FMT, &in.format)))
return rv;
*p_af = new;
return AF_OK;
@@ -460,7 +392,7 @@ static int af_fix_channels(struct af_stream *s, struct af_instance **p_af,
struct mp_audio actual = *prev->data;
if (mp_chmap_equals(&actual.channels, &in.channels))
return AF_FALSE;
if (prev->control(prev, AF_CONTROL_SET_CHANNELS, &in.channels) == AF_OK) {
if (prev->control(prev, AF_CONTROL_CHANNELS, &in.channels) == AF_OK) {
*p_af = prev;
return AF_OK;
}
@@ -469,7 +401,7 @@ static int af_fix_channels(struct af_stream *s, struct af_instance **p_af,
if (new == NULL)
return AF_ERROR;
new->auto_inserted = true;
if (AF_OK != (rv = new->control(new, AF_CONTROL_SET_CHANNELS, &in.channels)))
if (AF_OK != (rv = new->control(new, AF_CONTROL_CHANNELS, &in.channels)))
return rv;
*p_af = new;
return AF_OK;
@@ -484,7 +416,7 @@ static int af_fix_rate(struct af_stream *s, struct af_instance **p_af,
struct mp_audio actual = *prev->data;
if (actual.rate == in.rate)
return AF_FALSE;
if (prev->control(prev, AF_CONTROL_SET_RESAMPLE_RATE, &in.rate) == AF_OK) {
if (prev->control(prev, AF_CONTROL_RESAMPLE_RATE, &in.rate) == AF_OK) {
*p_af = prev;
return AF_OK;
}
@@ -493,7 +425,7 @@ static int af_fix_rate(struct af_stream *s, struct af_instance **p_af,
if (new == NULL)
return AF_ERROR;
new->auto_inserted = true;
if (AF_OK != (rv = new->control(new, AF_CONTROL_SET_RESAMPLE_RATE, &in.rate)))
if (AF_OK != (rv = new->control(new, AF_CONTROL_RESAMPLE_RATE, &in.rate)))
return rv;
*p_af = new;
return AF_OK;
@@ -505,14 +437,14 @@ static int af_fix_rate(struct af_stream *s, struct af_instance **p_af,
// state (for example, format filters that were tentatively inserted stay
// inserted).
// In that case, you should always rebuild the filter chain, or abort.
// Also, note that for complete reinit, fixup_output_format() may have to be
// called after this function.
static int af_reinit(struct af_stream *s)
{
remove_auto_inserted_filters(s);
// Start with the second filter, as the first filter is the special input
// filter which needs no initialization.
struct af_instance *af = s->first->next;
// Up to 7 retries per filter (channel, rate, 5x format conversions)
int max_retry = af_count_filters(s) * 7;
int max_retry = af_count_filters(s) * 4; // up to 4 retries per filter
int retry = 0;
while (af) {
if (retry >= max_retry)
@@ -521,11 +453,10 @@ static int af_reinit(struct af_stream *s)
// Check if this is the first filter
struct mp_audio in = *af->prev->data;
// Reset just in case...
mp_audio_set_null_data(&in);
in.audio = NULL;
in.len = 0;
int rv = af->control(af, AF_CONTROL_REINIT, &in);
if (rv == AF_OK && !mp_audio_config_equals(&in, af->prev->data))
rv = AF_FALSE; // conversion filter needed
switch (rv) {
case AF_OK:
af = af->next;
@@ -555,8 +486,8 @@ static int af_reinit(struct af_stream *s)
break;
}
default:
MP_ERR(s, "Reinitialization did not work, "
"audio filter '%s' returned error code %i\n",
mp_msg(MSGT_AFILTER, MSGL_ERR, "[libaf] Reinitialization did not "
"work, audio filter '%s' returned error code %i\n",
af->info->name, rv);
af_print_filter_chain(s, af, MSGL_ERR);
return AF_ERROR;
@@ -565,15 +496,11 @@ static int af_reinit(struct af_stream *s)
af_print_filter_chain(s, NULL, MSGL_V);
/* Set previously unset fields in s->output to those of the filter chain
* output. This is used to make the output format fixed, and even if you
* insert new filters or change the input format, the output format won't
* change. (Audio outputs generally can't change format at runtime.) */
af_copy_unset_fields(&s->output, &s->filter_output);
return af_config_equals(&s->output, &s->filter_output) ? AF_OK : AF_ERROR;
return AF_OK;
negotiate_error:
MP_ERR(s, "Unable to convert audio input format to output format.\n");
mp_msg(MSGT_AFILTER, MSGL_ERR, "[libaf] Unable to correct audio format. "
"This error should never occur, please send a bug report.\n");
af_print_filter_chain(s, af, MSGL_ERR);
return AF_ERROR;
}
@@ -585,33 +512,32 @@ void af_uninit(struct af_stream *s)
af_remove(s, s->first->next);
}
struct af_stream *af_new(struct mpv_global *global)
struct af_stream *af_new(struct MPOpts *opts)
{
struct af_stream *s = talloc_zero(NULL, struct af_stream);
static const struct af_info in = { .name = "in" };
static struct af_info in = { .name = "in" };
s->first = talloc(s, struct af_instance);
*s->first = (struct af_instance) {
.info = &in,
.control = input_control,
.filter = dummy_filter,
.priv = s,
.play = dummy_play,
.setup = s,
.data = &s->input,
.mul = 1.0,
};
static const struct af_info out = { .name = "out" };
static struct af_info out = { .name = "out" };
s->last = talloc(s, struct af_instance);
*s->last = (struct af_instance) {
.info = &out,
.control = output_control,
.filter = dummy_filter,
.priv = s,
.play = dummy_play,
.setup = s,
.data = &s->filter_output,
.mul = 1.0,
};
s->first->next = s->last;
s->last->prev = s->first;
s->opts = global->opts;
s->log = mp_log_new(s, global->log, "!af");
s->opts = opts;
return s;
}
@@ -621,10 +547,25 @@ void af_destroy(struct af_stream *s)
talloc_free(s);
}
/*
* Set previously unset fields in s->output to those of the filter chain
* output. This is used to make the output format fixed, and even if you insert
* new filters or change the input format, the output format won't change.
* \return AF_ERROR on error, AF_OK if successful.
*/
static int fixup_output_format(struct af_stream *s)
{
if (AF_OK != af_reinit(s))
return AF_ERROR;
af_copy_unset_fields(&s->output, &s->filter_output);
return af_config_equals(&s->output, &s->filter_output) ? AF_OK : AF_ERROR;
}
/* Initialize the stream "s". This function creates a new filter list
if necessary according to the values set in input and output. Input
and output should contain the format of the current movie and the
format of the preferred output respectively. The function is
formate of the preferred output respectively. The function is
reentrant i.e. if called with an already initialized stream the
stream will be reinitialized.
If one of the prefered output parameters is 0 the one that needs
@@ -637,24 +578,31 @@ int af_init(struct af_stream *s)
return -1;
// Precaution in case caller is misbehaving
mp_audio_set_null_data(&s->input);
mp_audio_set_null_data(&s->output);
s->input.audio = s->output.audio = NULL;
s->input.len = s->output.len = 0;
// Check if this is the first call
if (s->first->next == s->last) {
// Add all filters in the list (if there are any)
struct m_obj_settings *list = s->opts->af_settings;
for (int i = 0; list && list[i].name; i++) {
if (!af_prepend(s, s->last, list[i].name, list[i].attribs)) {
af_uninit(s);
if (!af_prepend(s, s->last, list[i].name, list[i].attribs))
return -1;
}
}
}
if (af_reinit(s) != AF_OK) {
remove_auto_inserted_filters(s);
// Init filters
if (AF_OK != af_reinit(s))
return -1;
if (AF_OK != fixup_output_format(s)) {
// Something is stuffed audio out will not work
MP_ERR(s, "Could not create audio filter chain.\n");
mp_msg(
MSGT_AFILTER, MSGL_ERR,
"[libaf] Unable to setup filter system can not"
" meet sound-card demands, please send a bug report. \n");
af_uninit(s);
return -1;
}
@@ -680,7 +628,7 @@ struct af_instance *af_add(struct af_stream *s, char *name, char **args)
return NULL;
// Reinitalize the filter list
if (af_reinit(s) != AF_OK) {
if (af_reinit(s) != AF_OK || fixup_output_format(s) != AF_OK) {
af_uninit(s);
af_init(s);
return NULL;
@@ -688,31 +636,36 @@ struct af_instance *af_add(struct af_stream *s, char *name, char **args)
return new;
}
/* Filter data chunk through the filters in the list.
* On success, *data is set to the filtered data/format.
* Warning: input audio data will be overwritten.
*/
int af_filter(struct af_stream *s, struct mp_audio *data, int flags)
// Filter data chunk through the filters in the list
struct mp_audio *af_play(struct af_stream *s, struct mp_audio *data)
{
struct af_instance *af = s->first;
assert(mp_audio_config_equals(af->data, data));
// Iterate through all filters
while (af) {
int r = af->filter(af, data, flags);
if (r < 0)
return r;
assert(mp_audio_config_equals(af->data, data));
do {
if (data->len <= 0)
break;
data = af->play(af, data);
af = af->next;
}
return 0;
} while (af && data);
return data;
}
// Calculate average ratio of filter output samples to input samples.
// e.g: num_output_samples = mul * num_input_samples
/* Calculate the minimum output buffer size for given input data d
* when using the RESIZE_LOCAL_BUFFER macro. The +t+1 part ensures the
* value is >= len*mul rounded upwards to whole samples even if the
* double 'mul' is inexact. */
int af_lencalc(double mul, struct mp_audio *d)
{
int t = d->bps * d->nch;
return d->len * mul + t + 1;
}
// Calculate average ratio of filter output size to input size
double af_calc_filter_multiplier(struct af_stream *s)
{
struct af_instance *af = s->first;
double mul = 1;
// Iterate through all filters and calculate total multiplication factor
do {
mul *= af->mul;
af = af->next;
@@ -721,20 +674,41 @@ double af_calc_filter_multiplier(struct af_stream *s)
return mul;
}
/* Calculate the total delay [seconds of output] caused by the filters */
/* Calculate the total delay [bytes output] caused by the filters */
double af_calc_delay(struct af_stream *s)
{
struct af_instance *af = s->first;
double delay = 0.0;
register double delay = 0.0;
// Iterate through all filters
while (af) {
delay += af->delay;
delay *= af->mul;
af = af->next;
}
return delay;
}
/* Send control to all filters, starting with the last until one accepts the
* command with AF_OK. Return the accepting filter. */
/* Helper function called by the macro with the same name this
function should not be called directly */
int af_resize_local_buffer(struct af_instance *af, struct mp_audio *data)
{
// Calculate new length
register int len = af_lencalc(af->mul, data);
mp_msg(MSGT_AFILTER, MSGL_V, "[libaf] Reallocating memory in module %s, "
"old len = %i, new len = %i\n", af->info->name, af->data->len, len);
// If there is a buffer free it
free(af->data->audio);
// Create new buffer and check that it is OK
af->data->audio = malloc(len);
if (!af->data->audio) {
mp_msg(MSGT_AFILTER, MSGL_FATAL, "[libaf] Could not allocate memory \n");
return AF_ERROR;
}
af->data->len = len;
return AF_OK;
}
// documentation in af.h
struct af_instance *af_control_any_rev(struct af_stream *s, int cmd, void *arg)
{
int res = AF_UNKNOWN;
@@ -747,10 +721,3 @@ struct af_instance *af_control_any_rev(struct af_stream *s, int cmd, void *arg)
}
return NULL;
}
/* Send control to all filters. Never stop, even if a filter returns AF_OK. */
void af_control_all(struct af_stream *s, int cmd, void *arg)
{
for (struct af_instance *af = s->first; af; af = af->next)
af->control(af, cmd, arg);
}
+193 -52
View File
@@ -21,16 +21,17 @@
#include <stdio.h>
#include <stdbool.h>
#include <sys/types.h>
#include "options/options.h"
#include "config.h"
#include "mpvcore/options.h"
#include "audio/format.h"
#include "audio/chmap.h"
#include "audio/audio.h"
#include "common/msg.h"
#include "control.h"
#include "mpvcore/mp_msg.h"
struct af_instance;
struct mpv_global;
// Number of channels
#define AF_NCH MP_NUM_CHANNELS
@@ -39,14 +40,13 @@ struct mpv_global;
#define AF_FLAGS_REENTRANT 0x00000000
#define AF_FLAGS_NOT_REENTRANT 0x00000001
// Flags for af->filter()
#define AF_FILTER_FLAG_EOF 1
/* Audio filter information not specific for current instance, but for
a specific filter */
struct af_info {
const char *info;
const char *name;
const char *author;
const char *comment;
const int flags;
int (*open)(struct af_instance *vf);
bool (*test_conversion)(int src_format, int dst_format);
@@ -58,23 +58,18 @@ struct af_info {
// Linked list of audio filters
struct af_instance {
const struct af_info *info;
struct mp_log *log;
struct replaygain_data *replaygain_data;
int (*control)(struct af_instance *af, int cmd, void *arg);
void (*uninit)(struct af_instance *af);
/* flags is a bit mask of AF_FILTER_FLAG_* values
* returns 0 on success, negative value on error
*/
int (*filter)(struct af_instance *af, struct mp_audio *data, int flags);
struct mp_audio * (*play)(struct af_instance *af, struct mp_audio *data);
void *setup; // old field for priv structs
void *priv;
struct mp_audio *data; // configuration and buffer for outgoing data stream
struct mp_audio *data; // configuration for outgoing data stream
struct af_instance *next;
struct af_instance *prev;
double delay; /* Delay caused by the filter, in seconds of audio consumed
* without corresponding output */
double delay; /* Delay caused by the filter, in units of bytes read without
* corresponding output */
double mul; /* length multiplier: how much does this instance change
* the number of samples passed though. (Ratio of input
* and output, e.g. mul=4 => 1 sample becomes 4 samples) .*/
the length of the buffer. */
bool auto_inserted; // inserted by af.c, such as conversion filters
};
@@ -85,16 +80,18 @@ struct af_stream {
struct af_instance *last;
// The user sets the input format (what the decoder outputs), and sets some
// or all fields in output to the output format the AO accepts.
// See fixup_output_format().
struct mp_audio input;
struct mp_audio output;
struct mp_audio filter_output;
struct mp_log *log;
struct MPOpts *opts;
struct replaygain_data *replaygain_data;
};
// Return values
/*********************************************
// Return values
*/
#define AF_DETACH 2
#define AF_OK 1
#define AF_TRUE 1
@@ -103,46 +100,190 @@ struct af_stream {
#define AF_ERROR -2
#define AF_FATAL -3
// Parameters for af_control_*
enum af_control {
AF_CONTROL_REINIT = 1,
AF_CONTROL_RESET,
AF_CONTROL_SET_RESAMPLE_RATE,
AF_CONTROL_SET_FORMAT,
AF_CONTROL_SET_CHANNELS,
AF_CONTROL_SET_VOLUME,
AF_CONTROL_GET_VOLUME,
AF_CONTROL_SET_PAN_LEVEL,
AF_CONTROL_SET_PAN_NOUT,
AF_CONTROL_SET_PAN_BALANCE,
AF_CONTROL_GET_PAN_BALANCE,
AF_CONTROL_SET_PLAYBACK_SPEED,
};
// Argument for AF_CONTROL_SET_PAN_LEVEL
typedef struct af_control_ext_s {
void* arg; // Argument
int ch; // Chanel number
} af_control_ext_t;
struct af_stream *af_new(struct mpv_global *global);
/*********************************************
// Export functions
*/
/**
* \defgroup af_chain Audio filter chain functions
* \{
* \param s filter chain
*/
struct af_stream *af_new(struct MPOpts *opts);
void af_destroy(struct af_stream *s);
int af_init(struct af_stream *s);
void af_uninit(struct af_stream *s);
struct af_instance *af_add(struct af_stream *s, char *name, char **args);
int af_filter(struct af_stream *s, struct mp_audio *data, int flags);
struct af_instance *af_control_any_rev(struct af_stream *s, int cmd, void *arg);
void af_control_all(struct af_stream *s, int cmd, void *arg);
/**
* \brief Initialize the stream "s".
* \return 0 on success, -1 on failure
*
* This function creates a new filter list if necessary, according
* to the values set in input and output. Input and output should contain
* the format of the current movie and the format of the preferred output
* respectively.
* Filters to convert to the preferred output format are inserted
* automatically, except when they are set to 0.
* The function is reentrant i.e. if called with an already initialized
* stream the stream will be reinitialized.
*/
int af_init(struct af_stream *s);
/**
* \brief Uninit and remove all filters from audio filter chain
*/
void af_uninit(struct af_stream *s);
/**
* \brief This function adds the filter "name" to the stream s.
* \param name name of filter to add
* \return pointer to the new filter, NULL if insert failed
*
* The filter will be inserted somewhere nice in the
* list of filters (i.e. at the beginning unless the
* first filter is the format filter (why??).
*/
struct af_instance *af_add(struct af_stream *s, char *name, char **args);
/**
* \brief filter data chunk through the filters in the list
* \param data data to play
* \return resulting data
* \ingroup af_chain
*/
struct mp_audio *af_play(struct af_stream *s, struct mp_audio *data);
/**
* \brief send control to all filters, starting with the last until
* one accepts the command with AF_OK.
* \param cmd filter control command
* \param arg argument for filter command
* \return the accepting filter or NULL if none was found
*/
struct af_instance *af_control_any_rev(struct af_stream *s, int cmd, void *arg);
/**
* \brief calculate average ratio of filter output lenth to input length
* \return the ratio
*/
double af_calc_filter_multiplier(struct af_stream *s);
/**
* \brief Calculate the total delay caused by the filters
* \return delay in bytes of "missing" output
*/
double af_calc_delay(struct af_stream *s);
/** \} */ // end of af_chain group
// Helper functions and macros used inside the audio filters
/**
* \defgroup af_filter Audio filter helper functions
* \{
*/
/* Helper function called by the macro with the same name only to be
called from inside filters */
int af_resize_local_buffer(struct af_instance *af, struct mp_audio *data);
/* Helper function used to calculate the exact buffer length needed
when buffers are resized. The returned length is >= than what is
needed */
int af_lencalc(double mul, struct mp_audio *data);
/**
* \brief convert dB to gain value
* \param n number of values to convert
* \param in [in] values in dB, <= -200 will become 0 gain
* \param out [out] gain values
* \param k input values are divided by this
* \param mi minimum dB value, input will be clamped to this
* \param ma maximum dB value, input will be clamped to this
* \return AF_ERROR on error, AF_OK otherwise
*/
int af_from_dB(int n, float *in, float *out, float k, float mi, float ma);
/**
* \brief convert gain value to dB
* \param n number of values to convert
* \param in [in] gain values, 0 wil become -200 dB
* \param out [out] values in dB
* \param k output values will be multiplied by this
* \return AF_ERROR on error, AF_OK otherwise
*/
int af_to_dB(int n, float *in, float *out, float k);
/**
* \brief convert milliseconds to sample time
* \param n number of values to convert
* \param in [in] values in milliseconds
* \param out [out] sample time values
* \param rate sample rate
* \param mi minimum ms value, input will be clamped to this
* \param ma maximum ms value, input will be clamped to this
* \return AF_ERROR on error, AF_OK otherwise
*/
int af_from_ms(int n, float *in, int *out, int rate, float mi, float ma);
/**
* \brief convert sample time to milliseconds
* \param n number of values to convert
* \param in [in] sample time values
* \param out [out] values in milliseconds
* \param rate sample rate
* \return AF_ERROR on error, AF_OK otherwise
*/
int af_to_ms(int n, int *in, float *out, int rate);
/**
* \brief test if output format matches
* \param af audio filter
* \param out needed format, will be overwritten by available
* format if they do not match
* \return AF_FALSE if formats do not match, AF_OK if they match
*
* compares the format, bps, rate and nch values of af->data with out
*/
int af_test_output(struct af_instance *af, struct mp_audio *out);
int af_from_dB(int n, float *in, float *out, float k, float mi, float ma);
int af_to_dB(int n, float *in, float *out, float k);
int af_from_ms(int n, float *in, int *out, int rate, float mi, float ma);
int af_to_ms(int n, int *in, float *out, int rate);
/**
* \brief soft clipping function using sin()
* \param a input value
* \return clipped value
*/
float af_softclip(float a);
/** \} */ // end of af_filter group, but more functions of this group below
/** Memory reallocation macro: if a local buffer is used (i.e. if the
filter doesn't operate on the incoming buffer this macro must be
called to ensure the buffer is big enough.
* \ingroup af_filter
*/
#define RESIZE_LOCAL_BUFFER(a, d) \
((a->data->len < af_lencalc(a->mul, d)) ? af_resize_local_buffer(a, d) : AF_OK)
/* Some other useful macro definitions*/
#ifndef min
#define min(a, b)(((a) > (b)) ? (b) : (a))
#endif
#ifndef max
#define max(a, b)(((a) > (b)) ? (a) : (b))
#endif
#ifndef clamp
#define clamp(a, min, max) (((a) > (max)) ? (max) : (((a) < (min)) ? (min) : (a)))
#endif
#ifndef sign
#define sign(a) (((a) > 0) ? (1) : (-1))
#endif
#ifndef lrnd
#define lrnd(a, b) ((b)((a) >= 0.0 ? (a) + 0.5 : (a) - 0.5))
#endif
#endif /* MPLAYER_AF_H */
+58 -48
View File
@@ -27,7 +27,7 @@
#include <string.h>
#include "af.h"
#include "options/m_option.h"
#include "mpvcore/m_option.h"
/// Internal specific data of the filter
struct af_bs2b {
@@ -37,33 +37,33 @@ struct af_bs2b {
t_bs2bdp filter; ///< instance of a library filter
};
#define FILTER(name, type) \
static int filter_##name(struct af_instance *af, struct mp_audio *data, int f) \
#define PLAY(name, type) \
static struct mp_audio *play_##name(struct af_instance *af, struct mp_audio *data) \
{ \
/* filter is called for all pairs of samples available in the buffer */ \
bs2b_cross_feed_##name(((struct af_bs2b*)(af->priv))->filter, \
(type*)(data->planes[0]), data->samples); \
(type*)(data->audio), data->len/data->bps/2); \
\
return 0; \
return data; \
}
FILTER(f, float)
FILTER(fbe, float)
FILTER(fle, float)
FILTER(s32be, int32_t)
FILTER(u32be, uint32_t)
FILTER(s32le, int32_t)
FILTER(u32le, uint32_t)
FILTER(s24be, bs2b_int24_t)
FILTER(u24be, bs2b_uint24_t)
FILTER(s24le, bs2b_int24_t)
FILTER(u24le, bs2b_uint24_t)
FILTER(s16be, int16_t)
FILTER(u16be, uint16_t)
FILTER(s16le, int16_t)
FILTER(u16le, uint16_t)
FILTER(s8, int8_t)
FILTER(u8, uint8_t)
PLAY(f, float)
PLAY(fbe, float)
PLAY(fle, float)
PLAY(s32be, int32_t)
PLAY(u32be, uint32_t)
PLAY(s32le, int32_t)
PLAY(u32le, uint32_t)
PLAY(s24be, bs2b_int24_t)
PLAY(u24be, bs2b_uint24_t)
PLAY(s24le, bs2b_int24_t)
PLAY(u24le, bs2b_uint24_t)
PLAY(s16be, int16_t)
PLAY(u16be, uint16_t)
PLAY(s16le, int16_t)
PLAY(u16le, uint16_t)
PLAY(s8, int8_t)
PLAY(u8, uint8_t)
/// Initialization and runtime control
@@ -74,6 +74,7 @@ static int control(struct af_instance *af, int cmd, void *arg)
switch (cmd) {
case AF_CONTROL_REINIT: {
int format;
char buf[256];
// Sanity check
if (!arg) return AF_ERROR;
@@ -86,69 +87,70 @@ static int control(struct af_instance *af, int cmd, void *arg)
and assign corresponding handlers */
switch (format) {
case AF_FORMAT_FLOAT_BE:
af->filter = filter_fbe;
af->play = play_fbe;
break;
case AF_FORMAT_FLOAT_LE:
af->filter = filter_fle;
af->play = play_fle;
break;
case AF_FORMAT_S32_BE:
af->filter = filter_s32be;
af->play = play_s32be;
break;
case AF_FORMAT_U32_BE:
af->filter = filter_u32be;
af->play = play_u32be;
break;
case AF_FORMAT_S32_LE:
af->filter = filter_s32le;
af->play = play_s32le;
break;
case AF_FORMAT_U32_LE:
af->filter = filter_u32le;
af->play = play_u32le;
break;
case AF_FORMAT_S24_BE:
af->filter = filter_s24be;
af->play = play_s24be;
break;
case AF_FORMAT_U24_BE:
af->filter = filter_u24be;
af->play = play_u24be;
break;
case AF_FORMAT_S24_LE:
af->filter = filter_s24le;
af->play = play_s24le;
break;
case AF_FORMAT_U24_LE:
af->filter = filter_u24le;
af->play = play_u24le;
break;
case AF_FORMAT_S16_BE:
af->filter = filter_s16be;
af->play = play_s16be;
break;
case AF_FORMAT_U16_BE:
af->filter = filter_u16be;
af->play = play_u16be;
break;
case AF_FORMAT_S16_LE:
af->filter = filter_s16le;
af->play = play_s16le;
break;
case AF_FORMAT_U16_LE:
af->filter = filter_u16le;
af->play = play_u16le;
break;
case AF_FORMAT_S8:
af->filter = filter_s8;
af->play = play_s8;
break;
case AF_FORMAT_U8:
af->filter = filter_u8;
af->play = play_u8;
break;
default:
af->filter = filter_f;
mp_audio_set_format(af->data, AF_FORMAT_FLOAT);
af->play = play_f;
mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE);
break;
}
// bs2b have srate limits, try to resample if needed
if (af->data->rate > BS2B_MAXSRATE || af->data->rate < BS2B_MINSRATE) {
af->data->rate = BS2B_DEFAULT_SRATE;
MP_WARN(af, "Requested sample rate %d Hz is out of bounds [%d..%d] Hz.\n"
"Trying to resample to %d Hz.\n",
mp_msg(MSGT_AFILTER, MSGL_WARN,
"[bs2b] Requested sample rate %d Hz is out of bounds [%d..%d] Hz.\n"
"[bs2b] Trying to resample to %d Hz.\n",
af->data->rate, BS2B_MINSRATE, BS2B_MAXSRATE, BS2B_DEFAULT_SRATE);
}
bs2b_set_srate(s->filter, (long)af->data->rate);
MP_VERBOSE(af, "using format %s\n",
af_fmt_to_str(af->data->format));
mp_msg(MSGT_AFILTER, MSGL_V, "[bs2b] using format %s\n",
af_fmt2str(af->data->format,buf,256));
return af_test_output(af,(struct mp_audio*)arg);
}
@@ -160,6 +162,7 @@ static int control(struct af_instance *af, int cmd, void *arg)
static void uninit(struct af_instance *af)
{
struct af_bs2b *s = af->priv;
free(af->data);
if (s->filter)
bs2b_close(s->filter);
}
@@ -170,9 +173,13 @@ static int af_open(struct af_instance *af)
struct af_bs2b *s = af->priv;
af->control = control;
af->uninit = uninit;
af->mul = 1;
if (!(af->data = calloc(1, sizeof(struct mp_audio))))
return AF_ERROR;
// NULL means failed initialization
if (!(s->filter = bs2b_open())) {
free(af->data);
return AF_ERROR;
}
@@ -189,10 +196,13 @@ static int af_open(struct af_instance *af)
#define OPT_BASE_STRUCT struct af_bs2b
/// Description of this filter
const struct af_info af_info_bs2b = {
.info = "Bauer stereophonic-to-binaural audio filter",
.name = "bs2b",
.open = af_open,
struct af_info af_info_bs2b = {
"Bauer stereophonic-to-binaural audio filter",
"bs2b",
"Andrew Savchenko",
"",
AF_FLAGS_REENTRANT,
af_open,
.priv_size = sizeof(struct af_bs2b),
.options = (const struct m_option[]) {
OPT_INTRANGE("fcut", fcut, 0, BS2B_MINFCUT, BS2B_MAXFCUT),
+55 -26
View File
@@ -29,19 +29,18 @@
#include <stdlib.h>
#include <string.h>
#include "common/common.h"
#include "af.h"
// Data for specific instances of this filter
typedef struct af_center_s
{
int ch; // Channel number which to insert the filtered data
int ch; // Channel number which to insert the filtered data
}af_center_t;
// Initialization and runtime control
static int control(struct af_instance* af, int cmd, void* arg)
{
af_center_t* s = af->priv;
af_center_t* s = af->setup;
switch(cmd){
case AF_CONTROL_REINIT:{
@@ -49,24 +48,48 @@ static int control(struct af_instance* af, int cmd, void* arg)
if(!arg) return AF_ERROR;
af->data->rate = ((struct mp_audio*)arg)->rate;
mp_audio_set_channels_old(af->data, MPMAX(s->ch+1,((struct mp_audio*)arg)->nch));
mp_audio_set_format(af->data, AF_FORMAT_FLOAT);
mp_audio_set_channels_old(af->data, max(s->ch+1,((struct mp_audio*)arg)->nch));
mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE);
return af_test_output(af,(struct mp_audio*)arg);
}
case AF_CONTROL_COMMAND_LINE:{
int ch=1;
sscanf(arg,"%i", &ch);
return control(af,AF_CONTROL_CENTER_CH | AF_CONTROL_SET, &ch);
}
case AF_CONTROL_CENTER_CH | AF_CONTROL_SET: // Requires reinit
// Sanity check
if((*(int*)arg >= AF_NCH) || (*(int*)arg < 0)){
mp_msg(MSGT_AFILTER, MSGL_ERR, "[sub] Center channel number must be between "
" 0 and %i current value is %i\n", AF_NCH-1, *(int*)arg);
return AF_ERROR;
}
s->ch = *(int*)arg;
return AF_OK;
case AF_CONTROL_CENTER_CH | AF_CONTROL_GET:
*(int*)arg = s->ch;
return AF_OK;
}
return AF_UNKNOWN;
}
// Filter data through filter
static int filter(struct af_instance* af, struct mp_audio* data, int flags)
// Deallocate memory
static void uninit(struct af_instance* af)
{
struct mp_audio* c = data; // Current working data
af_center_t* s = af->priv; // Setup for this instance
float* a = c->planes[0]; // Audio data
int nch = c->nch; // Number of channels
int len = c->samples*c->nch; // Number of samples in current audio block
int ch = s->ch; // Channel in which to insert the center audio
free(af->data);
free(af->setup);
}
// Filter data through filter
static struct mp_audio* play(struct af_instance* af, struct mp_audio* data)
{
struct mp_audio* c = data; // Current working data
af_center_t* s = af->setup; // Setup for this instance
float* a = c->audio; // Audio data
int len = c->len/4; // Number of samples in current audio block
int nch = c->nch; // Number of channels
int ch = s->ch; // Channel in which to insert the center audio
register int i;
// Run filter
@@ -75,25 +98,31 @@ static int filter(struct af_instance* af, struct mp_audio* data, int flags)
a[i+ch] = (a[i]/2) + (a[i+1]/2);
}
return 0;
return c;
}
// Allocate memory and set function pointers
static int af_open(struct af_instance* af){
af_center_t* s;
af->control=control;
af->filter=filter;
af->uninit=uninit;
af->play=play;
af->mul=1;
af->data=calloc(1,sizeof(struct mp_audio));
af->setup=s=calloc(1,sizeof(af_center_t));
if(af->data == NULL || af->setup == NULL)
return AF_ERROR;
// Set default values
s->ch = 1; // Channel nr 2
return AF_OK;
}
#define OPT_BASE_STRUCT af_center_t
const struct af_info af_info_center = {
.info = "Audio filter for adding a center channel",
.name = "center",
.flags = AF_FLAGS_NOT_REENTRANT,
.open = af_open,
.priv_size = sizeof(af_center_t),
.options = (const struct m_option[]) {
OPT_INTRANGE("channel", ch, 0, 0, AF_NCH - 1, OPTDEF_INT(1)),
{0}
},
// Description of this filter
struct af_info af_info_center = {
"Audio filter for adding a center channel",
"center",
"Alex Beregszaszi",
"",
AF_FLAGS_NOT_REENTRANT,
af_open
};
+104 -84
View File
@@ -3,8 +3,6 @@
* command line parameter channels. It is stupid and can only add
* silence or copy channels, not mix or filter.
*
* Original author: Anders
*
* This file is part of MPlayer.
*
* MPlayer is free software; you can redistribute it and/or modify
@@ -27,7 +25,6 @@
#include <string.h>
#include <inttypes.h>
#include "common/common.h"
#include "af.h"
#define FR 0
@@ -35,14 +32,12 @@
typedef struct af_channels_s{
int route[AF_NCH][2];
int nch, nr;
int nr;
int router;
char *routes;
}af_channels_t;
// Local function for copying data
static void copy(struct af_instance *af, void* in, void* out,
int ins, int inos,int outs, int outos, int len, int bps)
static void copy(void* in, void* out, int ins, int inos,int outs, int outos, int len, int bps)
{
switch(bps){
case 1:{
@@ -113,25 +108,24 @@ static void copy(struct af_instance *af, void* in, void* out,
break;
}
default:
MP_ERR(af, "Unsupported number of bytes/sample: %i"
" please report this error on the MPlayer mailing list. \n",bps);
mp_msg(MSGT_AFILTER, MSGL_ERR, "[channels] Unsupported number of bytes/sample: %i"
" please report this error on the MPlayer mailing list. \n",bps);
}
}
// Make sure the routes are sane
static int check_routes(struct af_instance *af, int nin, int nout)
static int check_routes(af_channels_t* s, int nin, int nout)
{
af_channels_t* s = af->priv;
int i;
if((s->nr < 1) || (s->nr > AF_NCH)){
MP_ERR(af, "The number of routing pairs must be"
" between 1 and %i. Current value is %i\n",AF_NCH,s->nr);
mp_msg(MSGT_AFILTER, MSGL_ERR, "[channels] The number of routing pairs must be"
" between 1 and %i. Current value is %i\n",AF_NCH,s->nr);
return AF_ERROR;
}
for(i=0;i<s->nr;i++){
if((s->route[i][FR] >= nin) || (s->route[i][TO] >= nout)){
MP_ERR(af, "Invalid routing in pair nr. %i.\n", i);
mp_msg(MSGT_AFILTER, MSGL_ERR, "[channels] Invalid routing in pair nr. %i.\n", i);
return AF_ERROR;
}
}
@@ -141,111 +135,137 @@ static int check_routes(struct af_instance *af, int nin, int nout)
// Initialization and runtime control
static int control(struct af_instance* af, int cmd, void* arg)
{
af_channels_t* s = af->priv;
af_channels_t* s = af->setup;
switch(cmd){
case AF_CONTROL_REINIT: ;
struct mp_chmap chmap;
mp_chmap_set_unknown(&chmap, s->nch);
mp_audio_set_channels(af->data, &chmap);
case AF_CONTROL_REINIT:
// Set default channel assignment
if(!s->router){
int i;
// Make sure this filter isn't redundant
if(af->data->nch == ((struct mp_audio*)arg)->nch)
return AF_DETACH;
return AF_DETACH;
// If mono: fake stereo
if(((struct mp_audio*)arg)->nch == 1){
s->nr = MPMIN(af->data->nch,2);
for(i=0;i<s->nr;i++){
s->route[i][FR] = 0;
s->route[i][TO] = i;
}
s->nr = min(af->data->nch,2);
for(i=0;i<s->nr;i++){
s->route[i][FR] = 0;
s->route[i][TO] = i;
}
}
else{
s->nr = MPMIN(af->data->nch, ((struct mp_audio*)arg)->nch);
for(i=0;i<s->nr;i++){
s->route[i][FR] = i;
s->route[i][TO] = i;
}
s->nr = min(af->data->nch, ((struct mp_audio*)arg)->nch);
for(i=0;i<s->nr;i++){
s->route[i][FR] = i;
s->route[i][TO] = i;
}
}
}
af->data->rate = ((struct mp_audio*)arg)->rate;
mp_audio_force_interleaved_format((struct mp_audio*)arg);
mp_audio_set_format(af->data, ((struct mp_audio*)arg)->format);
return check_routes(af,((struct mp_audio*)arg)->nch,af->data->nch);
af->mul = (double)af->data->nch / ((struct mp_audio*)arg)->nch;
return check_routes(s,((struct mp_audio*)arg)->nch,af->data->nch);
case AF_CONTROL_COMMAND_LINE:{
int nch = 0;
int n = 0;
// Check number of channels and number of routing pairs
sscanf(arg, "%i:%i%n", &nch, &s->nr, &n);
// If router scan commandline for routing pairs
if(s->nr){
char* cp = &((char*)arg)[n];
int ch = 0;
// Sanity check
if((s->nr < 1) || (s->nr > AF_NCH)){
mp_msg(MSGT_AFILTER, MSGL_ERR, "[channels] The number of routing pairs must be"
" between 1 and %i. Current value is %i\n",AF_NCH,s->nr);
}
s->router = 1;
// Scan for pairs on commandline
while((*cp == ':') && (ch < s->nr)){
sscanf(cp, ":%i:%i%n" ,&s->route[ch][FR], &s->route[ch][TO], &n);
mp_msg(MSGT_AFILTER, MSGL_V, "[channels] Routing from channel %i to"
" channel %i\n",s->route[ch][FR],s->route[ch][TO]);
cp = &cp[n];
ch++;
}
}
struct mp_chmap chmap;
mp_chmap_from_channels(&chmap, nch);
if (AF_OK != af->control(af, AF_CONTROL_CHANNELS | AF_CONTROL_SET, &chmap))
return AF_ERROR;
return AF_OK;
}
case AF_CONTROL_CHANNELS | AF_CONTROL_SET:
// Reinit must be called after this function has been called
mp_audio_set_channels(af->data, (struct mp_chmap *)arg);
if(!s->router)
mp_msg(MSGT_AFILTER, MSGL_V, "[channels] Changing number of channels"
" to %i\n",af->data->nch);
return AF_OK;
}
return AF_UNKNOWN;
}
// Filter data through filter
static int filter(struct af_instance* af, struct mp_audio* data, int flags)
// Deallocate memory
static void uninit(struct af_instance* af)
{
struct mp_audio* c = data; // Current working data
struct mp_audio* l = af->data; // Local data
af_channels_t* s = af->priv;
int i;
free(af->setup);
if (af->data)
free(af->data->audio);
free(af->data);
}
mp_audio_realloc_min(af->data, data->samples);
// Filter data through filter
static struct mp_audio* play(struct af_instance* af, struct mp_audio* data)
{
struct mp_audio* c = data; // Current working data
struct mp_audio* l = af->data; // Local data
af_channels_t* s = af->setup;
int i;
if(AF_OK != RESIZE_LOCAL_BUFFER(af,data))
return NULL;
// Reset unused channels
memset(l->planes[0],0,mp_audio_psize(c) / c->nch * l->nch);
memset(l->audio,0,c->len / c->nch * l->nch);
if(AF_OK == check_routes(af,c->nch,l->nch))
if(AF_OK == check_routes(s,c->nch,l->nch))
for(i=0;i<s->nr;i++)
copy(af, c->planes[0],l->planes[0],c->nch,s->route[i][FR],
l->nch,s->route[i][TO],mp_audio_psize(c),c->bps);
copy(c->audio,l->audio,c->nch,s->route[i][FR],
l->nch,s->route[i][TO],c->len,c->bps);
// Set output data
c->planes[0] = l->planes[0];
c->audio = l->audio;
c->len = c->len / c->nch * l->nch;
mp_audio_set_channels(c, &l->channels);
return 0;
return c;
}
// Allocate memory and set function pointers
static int af_open(struct af_instance* af){
af->control=control;
af->filter=filter;
af_channels_t *s = af->priv;
// If router scan commandline for routing pairs
if(s->routes && s->routes[0]){
char* cp = s->routes;
int ch = 0;
// Scan for pairs on commandline
do {
int n = 0;
if (ch >= AF_NCH) {
MP_FATAL(af, "Can't have more than %d routes.\n", AF_NCH);
return AF_ERROR;
}
sscanf(cp, "%i-%i%n" ,&s->route[ch][FR], &s->route[ch][TO], &n);
MP_VERBOSE(af, "Routing from channel %i to"
" channel %i\n",s->route[ch][FR],s->route[ch][TO]);
cp = &cp[n];
ch++;
} while(*cp == ',' && *(cp++));
s->nr = ch;
if (s->nr > 0)
s->router = 1;
}
return AF_OK;
af->control=control;
af->uninit=uninit;
af->play=play;
af->mul=1;
af->data=calloc(1,sizeof(struct mp_audio));
af->setup=calloc(1,sizeof(af_channels_t));
if((af->data == NULL) || (af->setup == NULL))
return AF_ERROR;
return AF_OK;
}
#define OPT_BASE_STRUCT af_channels_t
const struct af_info af_info_channels = {
.info = "Insert or remove channels",
.name = "channels",
.open = af_open,
.priv_size = sizeof(af_channels_t),
.options = (const struct m_option[]) {
OPT_INTRANGE("nch", nch, 0, 1, AF_NCH, OPTDEF_INT(2)),
OPT_STRING("routes", routes, 0),
{0}
},
// Description of this filter
struct af_info af_info_channels = {
"Insert or remove channels",
"channels",
"Anders",
"",
AF_FLAGS_REENTRANT,
af_open
};
-116
View File
@@ -1,116 +0,0 @@
/*
* This file is part of mpv.
*
* mpv is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* mpv is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License along
* with mpv. If not, see <http://www.gnu.org/licenses/>.
*/
#include <stdlib.h>
#include <assert.h>
#include "audio/format.h"
#include "af.h"
static bool test_conversion(int src_format, int dst_format)
{
return (src_format == AF_FORMAT_U24 && dst_format == AF_FORMAT_U32) ||
(src_format == AF_FORMAT_S24 && dst_format == AF_FORMAT_S32) ||
(src_format == AF_FORMAT_U32 && dst_format == AF_FORMAT_U24) ||
(src_format == AF_FORMAT_S32 && dst_format == AF_FORMAT_S24);
}
static int control(struct af_instance *af, int cmd, void *arg)
{
switch (cmd) {
case AF_CONTROL_REINIT: {
struct mp_audio *in = arg;
struct mp_audio orig_in = *in;
struct mp_audio *out = af->data;
if (!test_conversion(in->format, out->format))
return AF_DETACH;
if ((in->format & AF_FORMAT_BITS_MASK) == AF_FORMAT_24BIT) {
mp_audio_set_format(out, af_fmt_change_bits(in->format, 32));
} else if ((in->format & AF_FORMAT_BITS_MASK) == AF_FORMAT_32BIT) {
mp_audio_set_format(out, af_fmt_change_bits(in->format, 24));
} else {
abort();
}
out->rate = in->rate;
mp_audio_set_channels(out, &in->channels);
assert(test_conversion(in->format, out->format));
return mp_audio_config_equals(in, &orig_in) ? AF_OK : AF_FALSE;
}
case AF_CONTROL_SET_FORMAT: {
mp_audio_set_format(af->data, *(int*)arg);
return AF_OK;
}
}
return AF_UNKNOWN;
}
// The LSB is always ignored.
#if BYTE_ORDER == BIG_ENDIAN
#define SHIFT(x) ((3-(x))*8)
#else
#define SHIFT(x) (((x)+1)*8)
#endif
static int filter(struct af_instance *af, struct mp_audio *data, int flags)
{
mp_audio_realloc_min(af->data, data->samples);
struct mp_audio *out = af->data;
size_t len = mp_audio_psize(data) / data->bps;
if (data->bps == 4) {
for (int s = 0; s < len; s++) {
uint32_t val = *((uint32_t *)data->planes[0] + s);
uint8_t *ptr = (uint8_t *)out->planes[0] + s * 3;
ptr[0] = val >> SHIFT(0);
ptr[1] = val >> SHIFT(1);
ptr[2] = val >> SHIFT(2);
}
mp_audio_set_format(data, af_fmt_change_bits(data->format, 24));
} else {
for (int s = 0; s < len; s++) {
uint8_t *ptr = (uint8_t *)data->planes[0] + s * 3;
uint32_t val = ptr[0] << SHIFT(0)
| ptr[1] << SHIFT(1)
| ptr[2] << SHIFT(2);
*((uint32_t *)out->planes[0] + s) = val;
}
mp_audio_set_format(data, af_fmt_change_bits(data->format, 32));
}
data->planes[0] = out->planes[0];
return 0;
}
static int af_open(struct af_instance *af)
{
af->control = control;
af->filter = filter;
return AF_OK;
}
const struct af_info af_info_convert24 = {
.info = "Convert between 24 and 32 bit sample format",
.name = "convert24",
.open = af_open,
.test_conversion = test_conversion,
};
-131
View File
@@ -1,131 +0,0 @@
/*
* This file is part of mpv.
*
* mpv is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* mpv is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License along
* with mpv. If not, see <http://www.gnu.org/licenses/>.
*/
#include <stdlib.h>
#include <assert.h>
#include "af.h"
#include "audio/format.h"
#include "compat/mpbswap.h"
static bool test_conversion(int src_format, int dst_format)
{
if ((src_format & AF_FORMAT_PLANAR) ||
(dst_format & AF_FORMAT_PLANAR))
return false;
int src_noend = src_format & ~AF_FORMAT_END_MASK;
int dst_noend = dst_format & ~AF_FORMAT_END_MASK;
// We can swap endian for all formats, but sign only for integer formats.
if (src_noend == dst_noend)
return true;
if (((src_noend & ~AF_FORMAT_SIGN_MASK) ==
(dst_noend & ~AF_FORMAT_SIGN_MASK)) &&
((src_noend & AF_FORMAT_POINT_MASK) == AF_FORMAT_I))
return true;
return false;
}
static int control(struct af_instance *af, int cmd, void *arg)
{
switch (cmd) {
case AF_CONTROL_REINIT: {
struct mp_audio *in = arg;
struct mp_audio orig_in = *in;
struct mp_audio *out = af->data;
if (!test_conversion(in->format, out->format))
return AF_DETACH;
out->rate = in->rate;
mp_audio_set_channels(out, &in->channels);
return mp_audio_config_equals(in, &orig_in) ? AF_OK : AF_FALSE;
}
case AF_CONTROL_SET_FORMAT: {
mp_audio_set_format(af->data, *(int*)arg);
return AF_OK;
}
}
return AF_UNKNOWN;
}
static void endian(void *data, int len, int bps)
{
switch (bps) {
case 2:
for (int i = 0; i < len; i++) {
((uint16_t*)data)[i] = bswap_16(((uint16_t *)data)[i]);
}
break;
case 3:
for(int i = 0; i < len; i++) {
uint8_t s = ((uint8_t *)data)[3 * i];
((uint8_t *)data)[3 * i] = ((uint8_t *)data)[3 * i + 2];
((uint8_t *)data)[3 * i + 2] = s;
}
break;
case 4:
for(int i = 0; i < len; i++) {
((uint32_t*)data)[i] = bswap_32(((uint32_t *)data)[i]);
}
break;
}
}
static void si2us(void *data, int len, int bps, bool le)
{
ptrdiff_t i = -(len * bps);
uint8_t *p = &((uint8_t *)data)[len * bps];
if (le && bps > 1)
p += bps - 1;
if (len <= 0)
return;
do {
p[i] ^= 0x80;
} while (i += bps);
}
static int filter(struct af_instance *af, struct mp_audio *data, int flags)
{
int infmt = data->format;
int outfmt = af->data->format;
size_t len = data->samples * data->nch;
if ((infmt & AF_FORMAT_END_MASK) != (outfmt & AF_FORMAT_END_MASK))
endian(data->planes[0], len, data->bps);
if ((infmt & AF_FORMAT_SIGN_MASK) != (outfmt & AF_FORMAT_SIGN_MASK))
si2us(data->planes[0], len, data->bps,
(outfmt & AF_FORMAT_END_MASK) == AF_FORMAT_LE);
mp_audio_set_format(data, outfmt);
return 0;
}
static int af_open(struct af_instance *af)
{
af->control = control;
af->filter = filter;
return AF_OK;
}
const struct af_info af_info_convertsignendian = {
.info = "Convert between sample format sign/endian",
.name = "convertsignendian",
.open = af_open,
.test_conversion = test_conversion,
};
+82 -75
View File
@@ -3,8 +3,6 @@
* channels and can be used for simple position panning.
* An extension for this filter would be a reverb.
*
* Original author: Anders
*
* This file is part of MPlayer.
*
* MPlayer is free software; you can redistribute it and/or modify
@@ -27,7 +25,6 @@
#include <string.h>
#include <inttypes.h>
#include "common/common.h"
#include "af.h"
#define L 65536
@@ -37,52 +34,71 @@
// Data for specific instances of this filter
typedef struct af_delay_s
{
void* q[AF_NCH]; // Circular queues used for delaying audio signal
int wi[AF_NCH]; // Write index
int ri; // Read index
float d[AF_NCH]; // Delay [ms]
char *delaystr;
void* q[AF_NCH]; // Circular queues used for delaying audio signal
int wi[AF_NCH]; // Write index
int ri; // Read index
float d[AF_NCH]; // Delay [ms]
}af_delay_t;
// Initialization and runtime control
static int control(struct af_instance* af, int cmd, void* arg)
{
af_delay_t* s = af->priv;
af_delay_t* s = af->setup;
switch(cmd){
case AF_CONTROL_REINIT:{
int i;
struct mp_audio *in = arg;
if (in->bps != 1 && in->bps != 2 && in->bps != 4) {
MP_FATAL(af, "Sample format not supported\n");
return AF_ERROR;
}
// Free prevous delay queues
for(i=0;i<af->data->nch;i++)
free(s->q[i]);
mp_audio_force_interleaved_format(in);
mp_audio_copy_config(af->data, in);
mp_audio_copy_config(af->data, (struct mp_audio*)arg);
// Allocate new delay queues
for(i=0;i<af->data->nch;i++){
s->q[i] = calloc(L,af->data->bps);
if(NULL == s->q[i])
MP_FATAL(af, "Out of memory\n");
mp_msg(MSGT_AFILTER, MSGL_FATAL, "[delay] Out of memory\n");
}
if(AF_OK != af_from_ms(AF_NCH, s->d, s->wi, af->data->rate, 0.0, 1000.0))
return control(af,AF_CONTROL_DELAY_LEN | AF_CONTROL_SET,s->d);
}
case AF_CONTROL_COMMAND_LINE:{
int n = 1;
int i = 0;
char* cl = arg;
while(n && i < AF_NCH ){
sscanf(cl,"%f:%n",&s->d[i],&n);
if(n==0 || cl[n-1] == '\0')
break;
cl=&cl[n];
i++;
}
return AF_OK;
}
case AF_CONTROL_DELAY_LEN | AF_CONTROL_SET:{
int i;
if(AF_OK != af_from_ms(AF_NCH, arg, s->wi, af->data->rate, 0.0, 1000.0))
return AF_ERROR;
s->ri = 0;
for(i=0;i<AF_NCH;i++){
MP_DBG(af, "Channel %i delayed by %0.3fms\n",
i,MPCLAMP(s->d[i],0.0,1000.0));
MP_TRACE(af, "Channel %i delayed by %i samples\n",
i,s->wi[i]);
mp_msg(MSGT_AFILTER, MSGL_DBG2, "[delay] Channel %i delayed by %0.3fms\n",
i,clamp(s->d[i],0.0,1000.0));
mp_msg(MSGT_AFILTER, MSGL_DBG3, "[delay] Channel %i delayed by %i samples\n",
i,s->wi[i]);
}
return AF_OK;
}
case AF_CONTROL_DELAY_LEN | AF_CONTROL_GET:{
int i;
for(i=0;i<AF_NCH;i++){
if(s->ri > s->wi[i])
s->wi[i] = L - (s->ri - s->wi[i]);
else
s->wi[i] = s->wi[i] - s->ri;
}
return af_to_ms(AF_NCH, s->wi, arg, af->data->rate);
}
}
return AF_UNKNOWN;
}
@@ -92,59 +108,61 @@ static void uninit(struct af_instance* af)
{
int i;
free(af->data);
for(i=0;i<AF_NCH;i++)
free(((af_delay_t*)(af->priv))->q[i]);
free(((af_delay_t*)(af->setup))->q[i]);
free(af->setup);
}
// Filter data through filter
static int filter(struct af_instance* af, struct mp_audio* data, int flags)
static struct mp_audio* play(struct af_instance* af, struct mp_audio* data)
{
struct mp_audio* c = data; // Current working data
af_delay_t* s = af->priv; // Setup for this instance
int nch = c->nch; // Number of channels
int len = mp_audio_psize(c)/c->bps; // Number of sample in data chunk
int ri = 0;
int ch,i;
struct mp_audio* c = data; // Current working data
af_delay_t* s = af->setup; // Setup for this instance
int nch = c->nch; // Number of channels
int len = c->len/c->bps; // Number of sample in data chunk
int ri = 0;
int ch,i;
for(ch=0;ch<nch;ch++){
switch(c->bps){
case 1:{
int8_t* a = c->planes[0];
int8_t* a = c->audio;
int8_t* q = s->q[ch];
int wi = s->wi[ch];
ri = s->ri;
for(i=ch;i<len;i+=nch){
q[wi] = a[i];
a[i] = q[ri];
UPDATEQI(wi);
UPDATEQI(ri);
q[wi] = a[i];
a[i] = q[ri];
UPDATEQI(wi);
UPDATEQI(ri);
}
s->wi[ch] = wi;
break;
}
case 2:{
int16_t* a = c->planes[0];
int16_t* a = c->audio;
int16_t* q = s->q[ch];
int wi = s->wi[ch];
ri = s->ri;
for(i=ch;i<len;i+=nch){
q[wi] = a[i];
a[i] = q[ri];
UPDATEQI(wi);
UPDATEQI(ri);
q[wi] = a[i];
a[i] = q[ri];
UPDATEQI(wi);
UPDATEQI(ri);
}
s->wi[ch] = wi;
break;
}
case 4:{
int32_t* a = c->planes[0];
int32_t* a = c->audio;
int32_t* q = s->q[ch];
int wi = s->wi[ch];
ri = s->ri;
for(i=ch;i<len;i+=nch){
q[wi] = a[i];
a[i] = q[ri];
UPDATEQI(wi);
UPDATEQI(ri);
q[wi] = a[i];
a[i] = q[ri];
UPDATEQI(wi);
UPDATEQI(ri);
}
s->wi[ch] = wi;
break;
@@ -152,39 +170,28 @@ static int filter(struct af_instance* af, struct mp_audio* data, int flags)
}
}
s->ri = ri;
return 0;
return c;
}
// Allocate memory and set function pointers
static int af_open(struct af_instance* af){
af->control=control;
af->uninit=uninit;
af->filter=filter;
af_delay_t *s = af->priv;
int n = 1;
int i = 0;
char* cl = s->delaystr;
while(cl && n && i < AF_NCH ){
sscanf(cl,"%f%n",&s->d[i],&n);
if(n==0 || cl[n-1] == '\0')
break;
cl=&cl[n];
if (*cl != ',')
break;
cl++;
i++;
}
return AF_OK;
af->control=control;
af->uninit=uninit;
af->play=play;
af->mul=1;
af->data=calloc(1,sizeof(struct mp_audio));
af->setup=calloc(1,sizeof(af_delay_t));
if(af->data == NULL || af->setup == NULL)
return AF_ERROR;
return AF_OK;
}
#define OPT_BASE_STRUCT af_delay_t
const struct af_info af_info_delay = {
.info = "Delay audio filter",
.name = "delay",
.open = af_open,
.priv_size = sizeof(af_delay_t),
.options = (const struct m_option[]) {
OPT_STRING("delays", delaystr, 0),
{0}
},
// Description of this filter
struct af_info af_info_delay = {
"Delay audio filter",
"delay",
"Anders",
"",
AF_FLAGS_REENTRANT,
af_open
};
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