| Commit message (Collapse) | Author | Age | Files | Lines |
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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.
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This used to be in bytes, now it's in samples. Divide the value by 8
(assuming a typical audio format, float samples with 2 channels).
Fix some editing mistake or non-sense about the extra buffering added
(1<<x instead of x<<5).
Also sneak in a s/MPlayer/mpv/.
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This was needed for ao_jack.c., but not anymore.
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This changes option parsing as well as filter defaults slightly. The
default is now to encode to spdif (this is way more useful than writing
raw AC3 - what was this even useful for, other than writing broken ac3
-in-wav files?). The bitrate parameter is now always in kbps.
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Apparently this was completely broken after commit 22b3f522. Basically,
this locked up immediately completely while decoding the first packet.
The reason was that the buffer calculations confused bytes and number of
samples. Also, EOF reporting was broken (wrong return code).
The special-casing of ad_mpg123 and ad_spdif (with DECODE_MAX_UNIT) is a
bit annoying, but will eventually be solved in a better way.
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This could cause the bundle to recache stuff because of differences with
configuration of other software using fonconfig. The defaults OS X directories
should be added to fontconfig at build time (through configure).
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I thought this would always work... how disappointing.
Revert to interleaved format if requesting non-interleaved fails.
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Slightly simplifies memory management. This might make adding a demuxer
cache wrapper easier at a later point, because you can just copy the
complete stream header, without worrying that the wrapper will free the
individual stream header fields.
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Practically all audio decoding and filtering code leaked sample data
memory after uninitialization due to a simple logic bug (or typo).
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rejected it
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The existing code tried to remove the "extra" profile flags for h264.
FF_PROFILE_H264_INTRA doesn't matter for us at all, because it's set
only for profiles the vdpau/vaapi APIs don't support.
The FF_PROFILE_H264_CONSTRAINED flag on the other hand is added to
H264_BASELINE, except that it makes the file a real subset of H264_MAIN
and H264_HIGH. Removing that flag would select the BASELINE profile,
which appears to be rarely supported by hardware decoders. This means we
accidentally rejected perfectly hardware decodable files. Use MAIN for
it instead.
(vaapi has explicit support for CONSTRAINED_BASELINE, but it seems to be
a new thing, and is not reported as supported where I tried. So don't
bother to check it, and do the same as on vdpau.)
See github issue #204.
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We got rid of this some time ago, but apparently not completely.
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This used to be needed for teletext support. Teletext commit has been
removed (see commit ebaaa41f), and it appears this code is inactive.
It was just forgotten with the removal. Get rid of it completely.
Untested. (Like all changes to the TV code.)
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Signed-off-by: wm4 <wm4@nowhere>
Significant modifications over the original patch by not overriding
syscalls with macros ("#define open v4l2open") for fallback, but the
other way around ("#define v4l2open open"). As consequence, the calls
have to be replaced throughout the file.
Untested, although the original patch probably was tested.
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Apparently this is not portable to FreeBSD. It turns out that we
(probably) don't use any symbols defined by this header directly, so
the includes are not needed.
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These use the _oldargs_ hack, which failed in combination with playback
resume. Make it work.
It would be better to port all filters to new option parsing, but that's
obviously too much work, and most filters will probably be deleted and
replaced by libavfilter in the long run.
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ao_null should simulate a "perfect" AO, but framestepping behaved quite
badly with it. Framstepping usually exposes problems with AOs dropping
their buffers on pause, and that's what happened here.
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Fixes a problem where the passed size doesn't match the actuall string.
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Conflicts:
audio/out/ao_lavc.c
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Most libavcodec decoders output non-interleaved audio. Add direct
support for this, and remove the hack that repacked non-interleaved
audio back to packed audio.
Remove the minlen argument from the decoder callback. Instead of
forcing every decoder to have its own decode loop to fill the buffer
until minlen is reached, leave this to the caller. So if a decoder
doesn't return enough data, it's simply called again. (In future, I
even want to change it so that decoders don't read packets directly,
but instead the caller has to pass packets to the decoders. This fits
well with this change, because now the decoder callback typically
decodes at most one packet.)
ad_mpg123.c receives some heavy refactoring. The main problem is that
it wanted to handle format changes when there was no data in the decode
output buffer yet. This sounds reasonable, but actually it would write
data into a buffer prepared for old data, since the caller doesn't know
about the format change yet. (I.e. the best place for a format change
would be _after_ writing the last sample to the output buffer.) It's
possible that this code was not perfectly sane before this commit,
and perhaps lost one frame of data after a format change, but I didn't
confirm this. Trying to fix this, I ended up rewriting the decoding
and also the probing.
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Drop support for anything before 1.14.0.
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