mpv/options/options.h

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#ifndef MPLAYER_OPTIONS_H
#define MPLAYER_OPTIONS_H
#include <stdbool.h>
#include <stdint.h>
#include "m_option.h"
#include "common/common.h"
typedef struct mp_vo_opts {
struct m_obj_settings *video_driver_list;
int taskbar_progress;
int snap_window;
int ontop;
int ontop_level;
int fullscreen;
int border;
int fit_border;
int all_workspaces;
int window_minimized;
int window_maximized;
int screen_id;
int fsscreen_id;
char *winname;
int x11_netwm;
int x11_bypass_compositor;
int native_keyrepeat;
float panscan;
float zoom;
float pan_x, pan_y;
float align_x, align_y;
float margin_x[2];
float margin_y[2];
int unscaled;
struct m_geometry geometry;
struct m_geometry autofit;
struct m_geometry autofit_larger;
struct m_geometry autofit_smaller;
double window_scale;
int cursor_autohide_delay;
int cursor_autohide_fs;
int keepaspect;
int keepaspect_window;
int hidpi_window_scale;
int native_fs;
int64_t WinID;
float force_monitor_aspect;
float monitor_pixel_aspect;
int force_window_position;
char *mmcss_profile;
double override_display_fps;
double timing_offset;
// vo_drm
struct drm_opts *drm_opts;
int swapchain_depth; // max number of images to render ahead
} mp_vo_opts;
// Subtitle options needed by the subtitle decoders/renderers.
struct mp_subtitle_opts {
int sub_visibility;
int sub_pos;
float sub_delay;
float sub_fps;
float sub_speed;
int forced_subs_only;
int stretch_dvd_subs;
int stretch_image_subs;
int image_subs_video_res;
int sub_fix_timing;
int sub_scale_by_window;
int sub_scale_with_window;
int ass_scale_with_window;
struct osd_style_opts *sub_style;
float sub_scale;
float sub_gauss;
int sub_gray;
int sub_filter_SDH;
int sub_filter_SDH_harder;
int ass_enabled;
float ass_line_spacing;
int ass_use_margins;
int sub_use_margins;
int ass_vsfilter_aspect_compat;
int ass_vsfilter_color_compat;
int ass_vsfilter_blur_compat;
int use_embedded_fonts;
char **ass_force_style_list;
char *ass_styles_file;
int ass_style_override;
int ass_hinting;
int ass_shaper;
int ass_justify;
int sub_clear_on_seek;
int teletext_page;
};
struct mp_osd_render_opts {
float osd_bar_align_x;
float osd_bar_align_y;
float osd_bar_w;
float osd_bar_h;
float osd_scale;
int osd_scale_by_window;
struct osd_style_opts *osd_style;
int force_rgba_osd;
};
typedef struct MPOpts {
int property_print_help;
int use_terminal;
char *dump_stats;
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int verbose;
int msg_really_quiet;
char **msg_levels;
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int msg_color;
int msg_module;
int msg_time;
char *log_file;
test: make tests part of the mpv binary Until now, each .c file in test/ was built as separate, self-contained binary. Each binary could be run to execute the tests it contained. Change this and make them part of the normal mpv binary. Now the tests have to be invoked via the --unittest option. Do this for two reasons: - Tests now run within a "properly" initialized mpv instance, so all services are available. - Possibly simplifying the situation for future build systems. The first point is the main motivation. The mpv code is entangled with mp_log and the option system. It feels like a bad idea to duplicate some of the initialization of this just so you can call code using them. I'm also getting rid of cmocka. There wouldn't be any problem to keep it (it's a perfectly sane set of helpers), but NIH calls. I would have had to aggregate all tests into a CMUnitTest list, and I don't see how I'd get different types of entry points easily. Probably easily solvable, but since we made only pretty basic use of this library, NIH-ing this is actually easier (I needed a list of tests with custom metadata anyway, so all what was left was reimplement the assert_* helpers). Unit tests now don't output anything, and if they fail, they'll simply crash and leave a message that typically requires inspecting the test code to figure out what went wrong (and probably editing the test code to get more information). I even merged the various test functions into single ones. Sucks, but here you go. chmap_sel.c is merged into chmap.c, because I didn't see the point of this being separate. json.c drops the print_message() to go along with the new silent-by-default idea, also there's a memory leak fix unrelated to the rest of this commit. The new code is enabled with --enable-tests (--enable-test goes away). Due to waf's option parser, --enable-test still works, because it's a unique prefix to --enable-tests.
2019-11-07 21:42:14 +00:00
char *test_mode;
int operation_mode;
char **reset_options;
char **script_files;
char **script_opts;
int lua_load_osc;
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int lua_load_ytdl;
char *lua_ytdl_format;
char **lua_ytdl_raw_options;
int lua_load_stats;
int auto_load_scripts;
int audio_exclusive;
int ao_null_fallback;
int audio_stream_silence;
float audio_wait_open;
int force_vo;
float softvol_volume;
int rgain_mode;
float rgain_preamp; // Set replaygain pre-amplification
int rgain_clip; // Enable/disable clipping prevention
float rgain_fallback;
int softvol_mute;
float softvol_max;
int gapless_audio;
mp_vo_opts *vo;
struct ao_opts *ao_opts;
char *wintitle;
char *media_title;
struct mp_csp_equalizer_opts *video_equalizer;
int stop_screensaver;
int video_rotate;
core: redo how codecs are mapped, remove codecs.conf Use codec names instead of FourCCs to identify codecs. Rewrite how codecs are selected and initialized. Now each decoder exports a list of decoders (and the codec it supports) via add_decoders(). The order matters, and the first decoder for a given decoder is preferred over the other decoders. E.g. all ad_mpg123 decoders are preferred over ad_lavc, because it comes first in the mpcodecs_ad_drivers array. Likewise, decoders within ad_lavc that are enumerated first by libavcodec (using av_codec_next()) are preferred. (This is actually critical to select h264 software decoding by default instead of vdpau. libavcodec and ffmpeg/avconv use the same method to select decoders by default, so we hope this is sane.) The codec names follow libavcodec's codec names as defined by AVCodecDescriptor.name (see libavcodec/codec_desc.c). Some decoders have names different from the canonical codec name. The AVCodecDescriptor API is relatively new, so we need a compatibility layer for older libavcodec versions for codec names that are referenced internally, and which are different from the decoder name. (Add a configure check for that, because checking versions is getting way too messy.) demux/codec_tags.c is generated from the former codecs.conf (minus "special" decoders like vdpau, and excluding the mappings that are the same as the mappings libavformat's exported RIFF tables). It contains all the mappings from FourCCs to codec name. This is needed for demux_mkv, demux_mpg, demux_avi and demux_asf. demux_lavf will set the codec as determined by libavformat, while the other demuxers have to do this on their own, using the mp_set_audio/video_codec_from_tag() functions. Note that the sh_audio/video->format members don't uniquely identify the codec anymore, and sh->codec takes over this role. Replace the --ac/--vc/--afm/--vfm with new --vd/--ad options, which provide cover the functionality of the removed switched. Note: there's no CODECS_FLAG_FLIP flag anymore. This means some obscure container/video combinations (e.g. the sample Film_200_zygo_pro.mov) are played flipped. ffplay/avplay doesn't handle this properly either, so we don't care and blame ffmeg/libav instead.
2013-02-09 14:15:19 +00:00
char *audio_decoders;
char *video_decoders;
char *audio_spdif;
core: redo how codecs are mapped, remove codecs.conf Use codec names instead of FourCCs to identify codecs. Rewrite how codecs are selected and initialized. Now each decoder exports a list of decoders (and the codec it supports) via add_decoders(). The order matters, and the first decoder for a given decoder is preferred over the other decoders. E.g. all ad_mpg123 decoders are preferred over ad_lavc, because it comes first in the mpcodecs_ad_drivers array. Likewise, decoders within ad_lavc that are enumerated first by libavcodec (using av_codec_next()) are preferred. (This is actually critical to select h264 software decoding by default instead of vdpau. libavcodec and ffmpeg/avconv use the same method to select decoders by default, so we hope this is sane.) The codec names follow libavcodec's codec names as defined by AVCodecDescriptor.name (see libavcodec/codec_desc.c). Some decoders have names different from the canonical codec name. The AVCodecDescriptor API is relatively new, so we need a compatibility layer for older libavcodec versions for codec names that are referenced internally, and which are different from the decoder name. (Add a configure check for that, because checking versions is getting way too messy.) demux/codec_tags.c is generated from the former codecs.conf (minus "special" decoders like vdpau, and excluding the mappings that are the same as the mappings libavformat's exported RIFF tables). It contains all the mappings from FourCCs to codec name. This is needed for demux_mkv, demux_mpg, demux_avi and demux_asf. demux_lavf will set the codec as determined by libavformat, while the other demuxers have to do this on their own, using the mp_set_audio/video_codec_from_tag() functions. Note that the sh_audio/video->format members don't uniquely identify the codec anymore, and sh->codec takes over this role. Replace the --ac/--vc/--afm/--vfm with new --vd/--ad options, which provide cover the functionality of the removed switched. Note: there's no CODECS_FLAG_FLIP flag anymore. This means some obscure container/video combinations (e.g. the sample Film_200_zygo_pro.mov) are played flipped. ffplay/avplay doesn't handle this properly either, so we don't care and blame ffmeg/libav instead.
2013-02-09 14:15:19 +00:00
struct mp_subtitle_opts *subs_rend;
struct mp_osd_render_opts *osd_rend;
int osd_level;
int osd_duration;
int osd_fractions;
int osd_on_seek;
int video_osd;
int untimed;
char *stream_dump;
char *record_file;
int stop_playback_on_init_failure;
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int loop_times;
int loop_file;
int shuffle;
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int ordered_chapters;
char *ordered_chapters_files;
int chapter_merge_threshold;
double chapter_seek_threshold;
char *chapter_file;
int load_unsafe_playlists;
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int merge_files;
int quiet;
int load_config;
char *force_configdir;
int use_filedir_conf;
int hls_bitrate;
int edition_id;
int correct_pts;
int initial_audio_sync;
int video_sync;
double sync_max_video_change;
double sync_max_audio_change;
double sync_audio_drop_size;
int hr_seek;
float hr_seek_demuxer_offset;
int hr_seek_framedrop;
float audio_delay;
float default_max_pts_correction;
int autosync;
int frame_dropping;
int video_latency_hacks;
int term_osd;
int term_osd_bar;
char *term_osd_bar_chars;
char *playing_msg;
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char *osd_playing_msg;
char *status_msg;
char *osd_status_msg;
char *osd_msg[3];
int player_idle_mode;
int consolecontrols;
int playlist_pos;
struct m_rel_time play_start;
struct m_rel_time play_end;
struct m_rel_time play_length;
Implement backwards playback See manpage additions. This is a huge hack. You can bet there are shit tons of bugs. It's literally forcing square pegs into round holes. Hopefully, the manpage wall of text makes it clear enough that the whole shit can easily crash and burn. (Although it shouldn't literally crash. That would be a bug. It possibly _could_ start a fire by entering some sort of endless loop, not a literal one, just something where it tries to do work without making progress.) (Some obvious bugs I simply ignored for this initial version, but there's a number of potential bugs I can't even imagine. Normal playback should remain completely unaffected, though.) How this works is also described in the manpage. Basically, we demux in reverse, then we decode in reverse, then we render in reverse. The decoding part is the simplest: just reorder the decoder output. This weirdly integrates with the timeline/ordered chapter code, which also has special requirements on feeding the packets to the decoder in a non-straightforward way (it doesn't conflict, although a bugmessmass breaks correct slicing of segments, so EDL/ordered chapter playback is broken in backward direction). Backward demuxing is pretty involved. In theory, it could be much easier: simply iterating the usual demuxer output backward. But this just doesn't fit into our code, so there's a cthulhu nightmare of shit. To be specific, each stream (audio, video) is reversed separately. At least this means we can do backward playback within cached content (for example, you could play backwards in a live stream; on that note, it disables prefetching, which would lead to losing new live video, but this could be avoided). The fuckmess also meant that I didn't bother trying to support subtitles. Subtitles are a problem because they're "sparse" streams. They need to be "passively" demuxed: you don't try to read a subtitle packet, you demux audio and video, and then look whether there was a subtitle packet. This means to get subtitles for a time range, you need to know that you demuxed video and audio over this range, which becomes pretty messy when you demux audio and video backwards separately. Backward display is the most weird (and potentially buggy) part. To avoid that we need to touch a LOT of timing code, we negate all timestamps. The basic idea is that due to the navigation, all comparisons and subtractions of timestamps keep working, and you don't need to touch every single of them to "reverse" them. E.g.: bool before = pts_a < pts_b; would need to be: bool before = forward ? pts_a < pts_b : pts_a > pts_b; or: bool before = pts_a * dir < pts_b * dir; or if you, as it's implemented now, just do this after decoding: pts_a *= dir; pts_b *= dir; and then in the normal timing/renderer code: bool before = pts_a < pts_b; Consequently, we don't need many changes in the latter code. But some assumptions inhererently true for forward playback may have been broken anyway. What is mainly needed is fixing places where values are passed between positive and negative "domains". For example, seeking and timestamp user display always uses positive timestamps. The main mess is that it's not obvious which domain a given variable should or does use. Well, in my tests with a single file, it suddenly started to work when I did this. I'm honestly surprised that it did, and that I didn't have to change a single line in the timing code past decoder (just something minor to make external/cached text subtitles display). I committed it immediately while avoiding thinking about it. But there really likely are subtle problems of all sorts. As far as I'm aware, gstreamer also supports backward playback. When I looked at this years ago, I couldn't find a way to actually try this, and I didn't revisit it now. Back then I also read talk slides from the person who implemented it, and I'm not sure if and which ideas I might have taken from it. It's possible that the timestamp reversal is inspired by it, but I didn't check. (I think it claimed that it could avoid large changes by changing a sign?) VapourSynth has some sort of reverse function, which provides a backward view on a video. The function itself is trivial to implement, as VapourSynth aims to provide random access to video by frame numbers (so you just request decreasing frame numbers). From what I remember, it wasn't exactly fluid, but it worked. It's implemented by creating an index, and seeking to the target on demand, and a bunch of caching. mpv could use it, but it would either require using VapourSynth as demuxer and decoder for everything, or replacing the current file every time something is supposed to be played backwards. FFmpeg's libavfilter has reversal filters for audio and video. These require buffering the entire media data of the file, and don't really fit into mpv's architecture. It could be used by playing a libavfilter graph that also demuxes, but that's like VapourSynth but worse.
2019-05-18 00:10:51 +00:00
int play_dir;
int rebase_start_time;
int play_frames;
double ab_loop[2];
double step_sec;
int position_resume;
int position_check_mtime;
int position_save_on_quit;
int write_filename_in_watch_later_config;
int ignore_path_in_watch_later_config;
char *watch_later_directory;
int pause;
int keep_open;
int keep_open_pause;
double image_display_duration;
char *lavfi_complex;
int stream_id[2][STREAM_TYPE_COUNT];
char **stream_lang[STREAM_TYPE_COUNT];
int stream_auto_sel;
int audio_display;
char **display_tags;
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char **audio_files;
char *demuxer_name;
int demuxer_thread;
player: make playback termination asynchronous Until now, stopping playback aborted the demuxer and I/O layer violently by signaling mp_cancel (bound to libavformat's AVIOInterruptCB mechanism). Change it to try closing them gracefully. The main purpose is to silence those libavformat errors that happen when you request termination. Most of libavformat barely cares about the termination mechanism (AVIOInterruptCB), and essentially it's like the network connection is abruptly severed, or file I/O suddenly returns I/O errors. There were issues with dumb TLS warnings, parsers complaining about incomplete data, and some special protocols that require server communication to gracefully disconnect. We still want to abort it forcefully if it refuses to terminate on its own, so a timeout is required. Users can set the timeout to 0, which should give them the old behavior. This also removes the old mechanism that treats certain commands (like "quit") specially, and tries to terminate the demuxers even if the core is currently frozen. This is for situations where the core synchronized to the demuxer or stream layer while network is unresponsive. This in turn can only happen due to the "program" or "cache-size" properties in the current code (see one of the previous commits). Also, the old mechanism doesn't fit particularly well with the new one. We wouldn't want to abort playback immediately on a "quit" command - the new code is all about giving it a chance to end it gracefully. We'd need some sort of watchdog thread or something equally complicated to handle this. So just remove it. The change in osd.c is to prevent that it clears the status line while waiting for termination. The normal status line code doesn't output anything useful at this point, and the code path taken clears it, both of which is an annoying behavior change, so just let it show the old one.
2018-05-19 16:41:13 +00:00
double demux_termination_timeout;
int demuxer_cache_wait;
int prefetch_open;
char *audio_demuxer_name;
char *sub_demuxer_name;
Implement backwards playback See manpage additions. This is a huge hack. You can bet there are shit tons of bugs. It's literally forcing square pegs into round holes. Hopefully, the manpage wall of text makes it clear enough that the whole shit can easily crash and burn. (Although it shouldn't literally crash. That would be a bug. It possibly _could_ start a fire by entering some sort of endless loop, not a literal one, just something where it tries to do work without making progress.) (Some obvious bugs I simply ignored for this initial version, but there's a number of potential bugs I can't even imagine. Normal playback should remain completely unaffected, though.) How this works is also described in the manpage. Basically, we demux in reverse, then we decode in reverse, then we render in reverse. The decoding part is the simplest: just reorder the decoder output. This weirdly integrates with the timeline/ordered chapter code, which also has special requirements on feeding the packets to the decoder in a non-straightforward way (it doesn't conflict, although a bugmessmass breaks correct slicing of segments, so EDL/ordered chapter playback is broken in backward direction). Backward demuxing is pretty involved. In theory, it could be much easier: simply iterating the usual demuxer output backward. But this just doesn't fit into our code, so there's a cthulhu nightmare of shit. To be specific, each stream (audio, video) is reversed separately. At least this means we can do backward playback within cached content (for example, you could play backwards in a live stream; on that note, it disables prefetching, which would lead to losing new live video, but this could be avoided). The fuckmess also meant that I didn't bother trying to support subtitles. Subtitles are a problem because they're "sparse" streams. They need to be "passively" demuxed: you don't try to read a subtitle packet, you demux audio and video, and then look whether there was a subtitle packet. This means to get subtitles for a time range, you need to know that you demuxed video and audio over this range, which becomes pretty messy when you demux audio and video backwards separately. Backward display is the most weird (and potentially buggy) part. To avoid that we need to touch a LOT of timing code, we negate all timestamps. The basic idea is that due to the navigation, all comparisons and subtractions of timestamps keep working, and you don't need to touch every single of them to "reverse" them. E.g.: bool before = pts_a < pts_b; would need to be: bool before = forward ? pts_a < pts_b : pts_a > pts_b; or: bool before = pts_a * dir < pts_b * dir; or if you, as it's implemented now, just do this after decoding: pts_a *= dir; pts_b *= dir; and then in the normal timing/renderer code: bool before = pts_a < pts_b; Consequently, we don't need many changes in the latter code. But some assumptions inhererently true for forward playback may have been broken anyway. What is mainly needed is fixing places where values are passed between positive and negative "domains". For example, seeking and timestamp user display always uses positive timestamps. The main mess is that it's not obvious which domain a given variable should or does use. Well, in my tests with a single file, it suddenly started to work when I did this. I'm honestly surprised that it did, and that I didn't have to change a single line in the timing code past decoder (just something minor to make external/cached text subtitles display). I committed it immediately while avoiding thinking about it. But there really likely are subtle problems of all sorts. As far as I'm aware, gstreamer also supports backward playback. When I looked at this years ago, I couldn't find a way to actually try this, and I didn't revisit it now. Back then I also read talk slides from the person who implemented it, and I'm not sure if and which ideas I might have taken from it. It's possible that the timestamp reversal is inspired by it, but I didn't check. (I think it claimed that it could avoid large changes by changing a sign?) VapourSynth has some sort of reverse function, which provides a backward view on a video. The function itself is trivial to implement, as VapourSynth aims to provide random access to video by frame numbers (so you just request decreasing frame numbers). From what I remember, it wasn't exactly fluid, but it worked. It's implemented by creating an index, and seeking to the target on demand, and a bunch of caching. mpv could use it, but it would either require using VapourSynth as demuxer and decoder for everything, or replacing the current file every time something is supposed to be played backwards. FFmpeg's libavfilter has reversal filters for audio and video. These require buffering the entire media data of the file, and don't really fit into mpv's architecture. It could be used by playing a libavfilter graph that also demuxes, but that's like VapourSynth but worse.
2019-05-18 00:10:51 +00:00
int64_t video_reverse_size;
int64_t audio_reverse_size;
int cache_pause;
int cache_pause_initial;
float cache_pause_wait;
struct image_writer_opts *screenshot_image_opts;
screenshot: make screenshot filenames configurable This adds the --screenshot-template option, which specifies a template for the filename used for a screenshot. The '%' character is parsed as format specifier. These format specifiers insert metadata into the filename. For example, '%f' is replaced with the filename of the currently played file. The following format specifiers are available: %n Insert sequence number (padded with 4 zeros), e.g. "0002". %0Nn Like %n, but pad to N zeros (N = 0 to 9). %n behaves like %04n. %#n Like %n, but reset the sequence counter on every screenshot. (Useful if other parts in the template make the resulting filename already mostly unique.) %#0Nn Use %0Nn and %#n at the same time. %f Insert filename of the currently played video. %F Like %f, but with stripped file extension ("." and rest). %p Insert current playback time, in HH:MM:SS format. %P Like %p, but adds milliseconds: HH:MM:SS.mmmm %tX Insert the current local date/time, using the date format X. X is a single letter and is passed to strftime() as "%X". E.g. "%td" inserts the number of the current day. %{prop} Insert the value of the slave property 'prop'. E.g. %{filename} is the same as %f. If the property doesn't exist or is not available, nothing is inserted, unless a fallback is specified as in %{prop:fallback text}. %% Insert the character '%'. The strings inserted by format specifiers will be checked for characters not allowed in filenames (including '/' and '\'), and replaced with the placeholder '_'. (This doesn't happen for text that was passed with the --screenshot-template option, and allows specifying a screenshot target directory by prefixing the template with a relative or absolute path.)
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char *screenshot_template;
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char *screenshot_directory;
double force_fps;
int index_mode;
struct m_channels audio_output_channels;
int audio_output_format;
int force_srate;
double playback_speed;
int pitch_correction;
struct m_obj_settings *vf_settings, *vf_defs;
struct m_obj_settings *af_settings, *af_defs;
video: rewrite filtering glue code Get rid of the old vf.c code. Replace it with a generic filtering framework, which can potentially handle more than just --vf. At least reimplementing --af with this code is planned. This changes some --vf semantics (including runtime behavior and the "vf" command). The most important ones are listed in interface-changes. vf_convert.c is renamed to f_swscale.c. It is now an internal filter that can not be inserted by the user manually. f_lavfi.c is a refactor of player/lavfi.c. The latter will be removed once --lavfi-complex is reimplemented on top of f_lavfi.c. (which is conceptually easy, but a big mess due to the data flow changes). The existing filters are all changed heavily. The data flow of the new filter framework is different. Especially EOF handling changes - EOF is now a "frame" rather than a state, and must be passed through exactly once. Another major thing is that all filters must support dynamic format changes. The filter reconfig() function goes away. (This sounds complex, but since all filters need to handle EOF draining anyway, they can use the same code, and it removes the mess with reconfig() having to predict the output format, which completely breaks with libavfilter anyway.) In addition, there is no automatic format negotiation or conversion. libavfilter's primitive and insufficient API simply doesn't allow us to do this in a reasonable way. Instead, filters can use f_autoconvert as sub-filter, and tell it which formats they support. This filter will in turn add actual conversion filters, such as f_swscale, to perform necessary format changes. vf_vapoursynth.c uses the same basic principle of operation as before, but with worryingly different details in data flow. Still appears to work. The hardware deint filters (vf_vavpp.c, vf_d3d11vpp.c, vf_vdpaupp.c) are heavily changed. Fortunately, they all used refqueue.c, which is for sharing the data flow logic (especially for managing future/past surfaces and such). It turns out it can be used to factor out most of the data flow. Some of these filters accepted software input. Instead of having ad-hoc upload code in each filter, surface upload is now delegated to f_autoconvert, which can use f_hwupload to perform this. Exporting VO capabilities is still a big mess (mp_stream_info stuff). The D3D11 code drops the redundant image formats, and all code uses the hw_subfmt (sw_format in FFmpeg) instead. Although that too seems to be a big mess for now. f_async_queue is unused.
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struct filter_opts *filter_opts;
2008-04-24 04:36:43 +00:00
float movie_aspect;
int aspect_method;
char **sub_name;
2011-03-03 10:31:12 +00:00
char **sub_paths;
char **audiofile_paths;
char **external_files;
int autoload_files;
int sub_auto;
int audiofile_auto;
int osd_bar_visible;
int w32_priority;
struct cdda_params *stream_cdda_opts;
struct dvb_params *stream_dvb_opts;
struct stream_lavf_params *stream_lavf_opts;
char *cdrom_device;
char *bluray_device;
double mf_fps;
char *mf_type;
struct demux_rawaudio_opts *demux_rawaudio;
struct demux_rawvideo_opts *demux_rawvideo;
struct demux_lavf_opts *demux_lavf;
struct demux_mkv_opts *demux_mkv;
struct demux_cue_opts *demux_cue;
struct demux_opts *demux_opts;
demux: add a on-disk cache Somewhat similar to the old --cache-file, except for the demuxer cache. Instead of keeping packet data in memory, it's written to disk and read back when needed. The idea is to reduce main memory usage, while allowing fast seeking in large cached network streams (especially live streams). Keeping the packet metadata on disk would be rather hard (would use mmap or so, or rewrite the entire demux.c packet queue handling), and since it's relatively small, just keep it in memory. Also for simplicity, the disk cache is append-only. If you're watching really long livestreams, and need pruning, you're probably out of luck. This still could be improved by trying to free unused blocks with fallocate(), but since we're writing multiple streams in an interleaved manner, this is slightly hard. Some rather gross ugliness in packet.h: we want to store the file position of the cached data somewhere, but on 32 bit architectures, we don't have any usable 64 bit members for this, just the buf/len fields, which add up to 64 bit - so the shitty union aliases this memory. Error paths untested. Side data (the complicated part of trying to serialize ffmpeg packets) untested. Stream recording had to be adjusted. Some minor details change due to this, but probably nothing important. The change in attempt_range_joining() is because packets in cache have no valid len field. It was a useful check (heuristically finding broken cases), but not a necessary one. Various other approaches were tried. It would be interesting to list them and to mention the pros and cons, but I don't feel like it.
2019-06-13 17:10:32 +00:00
struct demux_cache_opts *demux_cache_opts;
stream: turn into a ring buffer, make size configurable In some corner cases (see #6802), it can be beneficial to use a larger stream buffer size. Use this as argument to rewrite everything for no reason. Turn stream.c itself into a ring buffer, with configurable size. The latter would have been easily achievable with minimal changes, and the ring buffer is the hard part. There is no reason to have a ring buffer at all, except possibly if ffmpeg don't fix their awful mp4 demuxer, and some subtle issues with demux_mkv.c wanting to seek back by small offsets (the latter was handled with small stream_peek() calls, which are unneeded now). In addition, this turns small forward seeks into reads (where data is simply skipped). Before this commit, only stream_skip() did this (which also mean that stream_skip() simply calls stream_seek() now). Replace all stream_peek() calls with something else (usually stream_read_peek()). The function was a problem, because it returned a pointer to the internal buffer, which is now a ring buffer with wrapping. The new function just copies the data into a buffer, and in some cases requires callers to dynamically allocate memory. (The most common case, demux_lavf.c, required a separate buffer allocation anyway due to FFmpeg "idiosyncrasies".) This is the bulk of the demuxer_* changes. I'm not happy with this. There still isn't a good reason why there should be a ring buffer, that is complex, and most of the time just wastes half of the available memory. Maybe another rewrite soon. It also contains bugs; you're an alpha tester now.
2019-11-06 20:36:02 +00:00
struct stream_opts *stream_opts;
struct vd_lavc_params *vd_lavc_params;
struct ad_lavc_params *ad_lavc_params;
struct input_opts *input_opts;
// may be NULL if encoding is not compiled-in
struct encode_opts *encode_opts;
char *ipc_path;
char *input_file;
int wingl_dwm_flush;
struct mp_resample_opts *resample_opts;
struct gl_video_opts *gl_video_opts;
struct angle_opts *angle_opts;
vo_opengl: refactor into vo_gpu This is done in several steps: 1. refactor MPGLContext -> struct ra_ctx 2. move GL-specific stuff in vo_opengl into opengl/context.c 3. generalize context creation to support other APIs, and add --gpu-api 4. rename all of the --opengl- options that are no longer opengl-specific 5. move all of the stuff from opengl/* that isn't GL-specific into gpu/ (note: opengl/gl_utils.h became opengl/utils.h) 6. rename vo_opengl to vo_gpu 7. to handle window screenshots, the short-term approach was to just add it to ra_swchain_fns. Long term (and for vulkan) this has to be moved to ra itself (and vo_gpu altered to compensate), but this was a stop-gap measure to prevent this commit from getting too big 8. move ra->fns->flush to ra_gl_ctx instead 9. some other minor changes that I've probably already forgotten Note: This is one half of a major refactor, the other half of which is provided by rossy's following commit. This commit enables support for all linux platforms, while his version enables support for all non-linux platforms. Note 2: vo_opengl_cb.c also re-uses ra_gl_ctx so it benefits from the --opengl- options like --opengl-early-flush, --opengl-finish etc. Should be a strict superset of the old functionality. Disclaimer: Since I have no way of compiling mpv on all platforms, some of these ports were done blindly. Specifically, the blind ports included context_mali_fbdev.c and context_rpi.c. Since they're both based on egl_helpers, the port should have gone smoothly without any major changes required. But if somebody complains about a compile error on those platforms (assuming anybody actually uses them), you know where to complain.
2017-09-14 06:04:55 +00:00
struct opengl_opts *opengl_opts;
vo_gpu: vulkan: initial implementation This time based on ra/vo_gpu. 2017 is the year of the vulkan desktop! Current problems / limitations / improvement opportunities: 1. The swapchain/flipping code violates the vulkan spec, by assuming that the presentation queue will be bounded (in cases where rendering is significantly faster than vsync). But apparently, there's simply no better way to do this right now, to the point where even the stupid cube.c examples from LunarG etc. do it wrong. (cf. https://github.com/KhronosGroup/Vulkan-Docs/issues/370) 2. The memory allocator could be improved. (This is a universal constant) 3. Could explore using push descriptors instead of descriptor sets, especially since we expect to switch descriptors semi-often for some passes (like interpolation). Probably won't make a difference, but the synchronization overhead might be a factor. Who knows. 4. Parallelism across frames / async transfer is not well-defined, we either need to use a better semaphore / command buffer strategy or a resource pooling layer to safely handle cross-frame parallelism. (That said, I gave resource pooling a try and was not happy with the result at all - so I'm still exploring the semaphore strategy) 5. We aggressively use pipeline barriers where events would offer a much more fine-grained synchronization mechanism. As a result of this, we might be suffering from GPU bubbles due to too-short dependencies on objects. (That said, I'm also exploring the use of semaphores as a an ordering tactic which would allow cross-frame time slicing in theory) Some minor changes to the vo_gpu and infrastructure, but nothing consequential. NOTE: For safety, all use of asynchronous commands / multiple command pools is currently disabled completely. There are some left-over relics of this in the code (e.g. the distinction between dev_poll and pool_poll), but that is kept in place mostly because this will be re-extended in the future (vulkan rev 2). The queue count is also currently capped to 1, because of the lack of cross-frame semaphores means we need the implicit synchronization from the same-queue semantics to guarantee a correct result.
2016-09-14 18:54:18 +00:00
struct vulkan_opts *vulkan_opts;
struct spirv_opts *spirv_opts;
vo_gpu: d3d11: initial implementation This is a new RA/vo_gpu backend that uses Direct3D 11. The GLSL generated by vo_gpu is cross-compiled to HLSL with SPIRV-Cross. What works: - All of mpv's internal shaders should work, including compute shaders. - Some external shaders have been tested and work, including RAVU and adaptive-sharpen. - Non-dumb mode works, even on very old hardware. Most features work at feature level 9_3 and all features work at feature level 10_0. Some features also work at feature level 9_1 and 9_2, but without high-bit- depth FBOs, it's not very useful. (Hardware this old is probably not fast enough for advanced features anyway.) Note: This is more compatible than ANGLE, which requires 9_3 to work at all (GLES 2.0,) and 10_1 for non-dumb-mode (GLES 3.0.) - Hardware decoding with D3D11VA, including decoding of 10-bit formats without truncation to 8-bit. What doesn't work / can be improved: - PBO upload and direct rendering does not work yet. Direct rendering requires persistent-mapped PBOs because the decoder needs to be able to read data from images that have already been decoded and uploaded. Unfortunately, it seems like persistent-mapped PBOs are fundamentally incompatible with D3D11, which requires all resources to use driver- managed memory and requires memory to be unmapped (and hence pointers to be invalidated) when a resource is used in a draw or copy operation. However it might be possible to use D3D11's limited multithreading capabilities to emulate some features of PBOs, like asynchronous texture uploading. - The blit() and clear() operations don't have equivalents in the D3D11 API that handle all cases, so in most cases, they have to be emulated with a shader. This is currently done inside ra_d3d11, but ideally it would be done in generic code, so it can take advantage of mpv's shader generation utilities. - SPIRV-Cross is used through a NIH C-compatible wrapper library, since it does not expose a C interface itself. The library is available here: https://github.com/rossy/crossc - The D3D11 context could be made to support more modern DXGI features in future. For example, it should be possible to add support for high-bit-depth and HDR output with DXGI 1.5/1.6.
2017-09-07 10:18:06 +00:00
struct d3d11_opts *d3d11_opts;
struct d3d11va_opts *d3d11va_opts;
struct cocoa_opts *cocoa_opts;
struct macos_opts *macos_opts;
struct android_opts *android_opts;
struct wayland_opts *wayland_opts;
struct dvd_opts *dvd_opts;
struct vaapi_opts *vaapi_opts;
struct sws_opts *sws_opts;
struct zimg_opts *zimg_opts;
int cuda_device;
} MPOpts;
struct dvd_opts {
int angle;
int speed;
char *device;
};
video: rewrite filtering glue code Get rid of the old vf.c code. Replace it with a generic filtering framework, which can potentially handle more than just --vf. At least reimplementing --af with this code is planned. This changes some --vf semantics (including runtime behavior and the "vf" command). The most important ones are listed in interface-changes. vf_convert.c is renamed to f_swscale.c. It is now an internal filter that can not be inserted by the user manually. f_lavfi.c is a refactor of player/lavfi.c. The latter will be removed once --lavfi-complex is reimplemented on top of f_lavfi.c. (which is conceptually easy, but a big mess due to the data flow changes). The existing filters are all changed heavily. The data flow of the new filter framework is different. Especially EOF handling changes - EOF is now a "frame" rather than a state, and must be passed through exactly once. Another major thing is that all filters must support dynamic format changes. The filter reconfig() function goes away. (This sounds complex, but since all filters need to handle EOF draining anyway, they can use the same code, and it removes the mess with reconfig() having to predict the output format, which completely breaks with libavfilter anyway.) In addition, there is no automatic format negotiation or conversion. libavfilter's primitive and insufficient API simply doesn't allow us to do this in a reasonable way. Instead, filters can use f_autoconvert as sub-filter, and tell it which formats they support. This filter will in turn add actual conversion filters, such as f_swscale, to perform necessary format changes. vf_vapoursynth.c uses the same basic principle of operation as before, but with worryingly different details in data flow. Still appears to work. The hardware deint filters (vf_vavpp.c, vf_d3d11vpp.c, vf_vdpaupp.c) are heavily changed. Fortunately, they all used refqueue.c, which is for sharing the data flow logic (especially for managing future/past surfaces and such). It turns out it can be used to factor out most of the data flow. Some of these filters accepted software input. Instead of having ad-hoc upload code in each filter, surface upload is now delegated to f_autoconvert, which can use f_hwupload to perform this. Exporting VO capabilities is still a big mess (mp_stream_info stuff). The D3D11 code drops the redundant image formats, and all code uses the hw_subfmt (sw_format in FFmpeg) instead. Although that too seems to be a big mess for now. f_async_queue is unused.
2018-01-16 10:53:44 +00:00
struct filter_opts {
int deinterlace;
};
extern const struct m_sub_options vo_sub_opts;
extern const struct m_sub_options dvd_conf;
extern const struct m_sub_options mp_subtitle_sub_opts;
extern const struct m_sub_options mp_osd_render_sub_opts;
video: rewrite filtering glue code Get rid of the old vf.c code. Replace it with a generic filtering framework, which can potentially handle more than just --vf. At least reimplementing --af with this code is planned. This changes some --vf semantics (including runtime behavior and the "vf" command). The most important ones are listed in interface-changes. vf_convert.c is renamed to f_swscale.c. It is now an internal filter that can not be inserted by the user manually. f_lavfi.c is a refactor of player/lavfi.c. The latter will be removed once --lavfi-complex is reimplemented on top of f_lavfi.c. (which is conceptually easy, but a big mess due to the data flow changes). The existing filters are all changed heavily. The data flow of the new filter framework is different. Especially EOF handling changes - EOF is now a "frame" rather than a state, and must be passed through exactly once. Another major thing is that all filters must support dynamic format changes. The filter reconfig() function goes away. (This sounds complex, but since all filters need to handle EOF draining anyway, they can use the same code, and it removes the mess with reconfig() having to predict the output format, which completely breaks with libavfilter anyway.) In addition, there is no automatic format negotiation or conversion. libavfilter's primitive and insufficient API simply doesn't allow us to do this in a reasonable way. Instead, filters can use f_autoconvert as sub-filter, and tell it which formats they support. This filter will in turn add actual conversion filters, such as f_swscale, to perform necessary format changes. vf_vapoursynth.c uses the same basic principle of operation as before, but with worryingly different details in data flow. Still appears to work. The hardware deint filters (vf_vavpp.c, vf_d3d11vpp.c, vf_vdpaupp.c) are heavily changed. Fortunately, they all used refqueue.c, which is for sharing the data flow logic (especially for managing future/past surfaces and such). It turns out it can be used to factor out most of the data flow. Some of these filters accepted software input. Instead of having ad-hoc upload code in each filter, surface upload is now delegated to f_autoconvert, which can use f_hwupload to perform this. Exporting VO capabilities is still a big mess (mp_stream_info stuff). The D3D11 code drops the redundant image formats, and all code uses the hw_subfmt (sw_format in FFmpeg) instead. Although that too seems to be a big mess for now. f_async_queue is unused.
2018-01-16 10:53:44 +00:00
extern const struct m_sub_options filter_conf;
extern const struct m_sub_options resample_conf;
stream: turn into a ring buffer, make size configurable In some corner cases (see #6802), it can be beneficial to use a larger stream buffer size. Use this as argument to rewrite everything for no reason. Turn stream.c itself into a ring buffer, with configurable size. The latter would have been easily achievable with minimal changes, and the ring buffer is the hard part. There is no reason to have a ring buffer at all, except possibly if ffmpeg don't fix their awful mp4 demuxer, and some subtle issues with demux_mkv.c wanting to seek back by small offsets (the latter was handled with small stream_peek() calls, which are unneeded now). In addition, this turns small forward seeks into reads (where data is simply skipped). Before this commit, only stream_skip() did this (which also mean that stream_skip() simply calls stream_seek() now). Replace all stream_peek() calls with something else (usually stream_read_peek()). The function was a problem, because it returned a pointer to the internal buffer, which is now a ring buffer with wrapping. The new function just copies the data into a buffer, and in some cases requires callers to dynamically allocate memory. (The most common case, demux_lavf.c, required a separate buffer allocation anyway due to FFmpeg "idiosyncrasies".) This is the bulk of the demuxer_* changes. I'm not happy with this. There still isn't a good reason why there should be a ring buffer, that is complex, and most of the time just wastes half of the available memory. Maybe another rewrite soon. It also contains bugs; you're an alpha tester now.
2019-11-06 20:36:02 +00:00
extern const struct m_sub_options stream_conf;
extern const struct m_sub_options mp_opt_root;
#endif