| Commit message (Collapse) | Author | Age |
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See #356
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In the cocoa backend you can use cmd+0/1/2 to scale the window. This commit
makes it use the new window-scale functionality.
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This commit adds a new build system based on waf. configure and Makefile
are deprecated effective immediately and someday in the future they will be
removed (they are still available by running ./old-configure).
You can find how the choice for waf came to be in `DOCS/waf-buildsystem.rst`.
TL;DR: we couldn't get the same level of abstraction and customization with
other build systems we tried (CMake and autotools).
For guidance on how to build the software now, take a look at README.md
and the cross compilation guide.
CREDITS:
This is a squash of ~250 commits. Some of them are not by me, so here is the
deserved attribution:
- @wm4 contributed some Windows fixes, renamed configure to old-configure
and contributed to the bootstrap script. Also, GNU/Linux testing.
- @lachs0r contributed some Windows fixes and the bootstrap script.
- @Nikoli contributed a lot of testing and discovered many bugs.
- @CrimsonVoid contributed changes to the bootstrap script.
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This is Better(tm).
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Also add an explanation why this special code path for rtsp is needed.
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This was forgotten when the parser for mplayer2 EDL files was removed.
Change the header of the mpv EDL format to include a '#', so a naive
parser could skip the header as comment. (Maybe this is questionable;
on the other hand, if it can be simpler, why not.)
Also, strip the header in demux_edl.c before passing on the data, so the
header check doesn't need to be duplicated in tl_mpv_edl.c.
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This possibly enables code that has never been tested before
(accidentally), so let's hope this works out ok.
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It was a bit too complicated and inconvenient, and I doubt anyone
actively used it. The mpv EDL format should cover all use cases.
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Questionable change from user perspective, but internally needed to
implement the next commit. Also useful for testing timeline stuff.
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Edit Decision Lists (EDL) allow combining parts from multiple source
files into one virtual file. MPlayer had an EDL format (which sucked),
which mplayer2 tried to improve with its own format (which sucked). As
logic demands, mpv introduces its very own format (which sucks).
The new format should actually be much simpler and easier to use, and
its implementation is simpler and smaller too.
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The intention of the existing code was trying to match demuxer-reported
stream IDs, instead of using possibly arbitrary ordering of the frontend
track list. But EDL files can consist of quite different files, for
which trying to match the stream IDs doesn't always make sense.
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This didn't have any consequences, other than suddenly reinitializing
video when it works again (such as with EDL timeline mixing video and
audio-only files).
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OpenGL interop was essentially disabled, because the decoder didn't
request vdpau device creation from vo_opengl.
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The size accidentally defaulted to 200 seconds instead of 200
milliseconds, which had fatal consequences when trying to use it.
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This always bothered me.
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This always bothered me.
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This could lead to (barely) audible artifacts with --af=scaletempo and
modified playback speed.
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The AF control commands used an elaborate and unnecessary organization
for the command constants. Get rid of all that and convert the
definitions to a simple enum. Also remove the control commands that
were not really needed, because they were not used outside of the
filters that implemented them.
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And by "cleanup", I mean "remove". Actually, only remove the parts that
are redundant and doxygen noise. Move useful parts to the comment above
the function's implementation in the C source file.
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Some decoders used to read packets and decode data when calling
resync_audio_stream(). This required a special case in mp_seek() for
audio. (A comment mentions liba52, which is long gone; but until
recently ad_mpg123.c actually exposed this behavior.)
No decoder does this anymore, and resync_audio_stream() works similar
as resync_video_stream(). Remove the special case.
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The priv struct is now allocated by talloc in stream.c. It doesn't need
to be manually freed, and using free() instead of talloc_free() probably
crashes.
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Same as MPlayer svn commit r36515 "Chose cheaper alpha blend equation."
No idea if this is actually faster, but can't hurt.
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When the decoder detects a format change, it overwrites the values
stored in sh_audio (this affects the members sample_format, samplerate,
channels). In the case when the old audio data still needs to be
played/filtered, the audio format as identified by sh_audio and the
format used for the decoder buffer can mismatch. In particular, they
will mismatch in the very unlikely but possible case the audio chain is
reinitialized while old data is draining during a format change.
Or in other words, sh_audio might contain the new format, while the
audio chain is still configured to use the old format.
Currently, the audio code (player/audio.c and init_audio_filters) access
sh_audio to get the current format. This is in theory incorrect for the
reasons mentioned above. Use the decoder buffer's format instead, which
should be correct at any point.
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Commit 22b3f522 not only redid major aspects of audio decoding, but also
attempted to fix audio format change handling. Before that commit, data
that was already decoded but not yet filtered was thrown away on a
format change. After that commit, data was supposed to finish playing
before rebuilding filters and so on.
It was still buggy, though: the decoder buffer was initialized to the
new format too early, triggering an assertion failure. Move the reinit
call below filtering to fix this.
ad_mpg123.c needs to be adjusted so that it doesn't decode new data
before the format change is actually executed.
Add some more assertions to af_play() (audio filtering) to make sure
input data and configured format don't mismatch. This will also catch
filters which don't set the format on their output data correctly.
Regression due to planar_audio branch.
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Also split the function itself into 3.
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Regression introduced with commit a89549e8.
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Don't call ao_play() if there's nothing left. Of course this still asks
the AO to play internally buffered audio by setting drain=true.
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Simulate proper handling of AOPLAY_FINAL_CHUNK. Print when underruns
occur (i.e. running out of data). Add some options that control
simulated buffer and outburst sizes.
All this is useful for debugging and self-documentation. (Note that
ao_null always was supposed to simulate an ideal AO, which is the reason
why it fools people who try to use it for benchmarking video.)
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Also break that line a bit.
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Might be helpful later.
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These were replaced with DEMUXER_CTRL_SWITCHED_TRACKS a while ago.
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demuxer->filepos contains the byte offset of the last read packet. This
is so that the player can estimate the current playback position, if no
proper timestamps are available. Simplify it to use demux_packet->pos in
the generic demuxer code, instead of bothering every demuxer
implementation about it.
(Note that this is still a bit incorrect: it relfects the position of
the last packet read by the demuxer, not that returned to the user. But
that was already broken, and is not that trivial to fix.)
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This was originally added for better seeking where libavformat's seek
function won't work well: files with timestamp resets. In these cases,
the code tried to calculate an average bitrate, and then do byte based
seeks by multiplying the seek target time with the bitrate.
Apparently this was unreliable enough that the code was just commented
(and other parts became inactive). Get rid of it.
Note that the player still does byte based seeks in these cases when
doing percent-seeks.
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This should be almost equivalent, but is slightly better because the
EOF flag is reset earlier.
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This should allow it to select better fallback formats, instead of
picking the first encoder sample format if ao->format is not equal to
any of the encoder sample formats.
Not sure what is supposed to happen if the encoder provides no
compatible sample format (or no sample format list at all), but in this
case ao_lavc.c still fails gracefully.
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The added function af_format_conversion_score() can be used to select
the best sample format to convert to in order to reduce loss and extra
conversion work.
It calculates a "loss" score when going from one format to another, and
for each conversion that needs to be done a certain score is subtracted.
Thus, if you have to convert from one format to a set of other formats,
you can calculate the score for each conversion, and pick the one with
the highest score.
Conversion between int and float is considered the worst case. One odd
consequence is that when converting from s32 to u8 or float, u8 will be
picked.
Test program used to develop this follows:
#define MAX_FMT 200
struct entry {
const char *name;
int score;
};
static int compentry(const void *px1, const void *px2)
{
const struct entry *x1 = px1;
const struct entry *x2 = px2;
if (x1->score > x2->score)
return 1;
if (x1->score < x2->score)
return -1;
return 0;
}
int main(int argc, char *argv[])
{
for (int n = 0; af_fmtstr_table[n].name; n++) {
struct entry entry[MAX_FMT];
int entries = 0;
for (int i = 0; af_fmtstr_table[i].name; i++) {
assert(i < MAX_FMT);
entry[entries].name = af_fmtstr_table[i].name;
entry[entries].score =
af_format_conversion_score(af_fmtstr_table[i].format,
af_fmtstr_table[n].format);
entries++;
}
qsort(&entry[0], entries, sizeof(entry[0]), compentry);
for (int i = 0; i < entries; i++) {
printf("%s -> %s: %d \n", af_fmtstr_table[n].name,
entry[i].name, entry[i].score);
}
}
}
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This was accidentally equivalent to floatp.
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These must be written even if there was no "final frame", e.g. due to
the player being exited with "q".
Although the issue is mostly of theoretical nature, as most audio codecs
don't need the final encoding calls with NULL data. Maybe will be more
relevant in the future.
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You get the native format by not appending any suffix to the format.
This change includes user-facing names, e.g. for the --format option.
|
enkore
wm4
Josh Driver
Stefano Pigozzi
Rudolf Polzer