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Commit Graph

64 Commits

Author SHA1 Message Date
wm4
65db3291b3 client API: better error reporting
Give somewhat more information on playback failure.
2014-10-28 20:30:12 +01:00
wm4
c9234d769d player: fix exiting if both audio and video fail initializing
The player was supposed to exit playback if both video and audio failed
to initialize (or if one of the streams was not selected when the other
stream failed). This didn't work; for one this check was missing from
one of the failure paths. And more importantly, both checked the
current_track array incorrectly.

Fix these issues, and move the failure handling code into a common
function.

CC: @mpv-player/stable
2014-10-23 18:31:43 +02:00
wm4
ca038d9a22 audio: don't go to sleep after audio reinit
It possibly goes to sleep without actually starting to decode audio.
Possibly fixes a problem with --no-osc --no-video reported on IRC.

CC: @mpv-player/stable
2014-10-17 01:10:49 +02:00
wm4
0e0dc5c5be player: fix crash on early audio uninit
Could crash when exiting playback in very early stages of
initialization.

CC: @mpv-player/stable
2014-10-16 01:03:02 +02:00
wm4
19d21103e5 player: exit if audio init fails and there's no video
Seems logical. For some reason, the player allows deselecting both audio
and video stream without quitting (a deliberate feature of which I have
no idea why it was added years ago), so this is needed.
2014-10-10 15:15:58 +02:00
wm4
9d5d031b6d player: remove central uninit_player() function and flags mess
Each subsystem (or similar thing) had an INITIALIZED_ flag assigned. The
main use of this was that you could pass a bitmask of these flags to
uninit_player(). Except in some situations where you wanted to
uninitialize nearly everything, this wasn't really useful. Moreover, it
was quite annoying that subsystems had most of the code in a specific
file, but the uninit code in loadfile.c (because that's where
uninit_player() was implemented).

Simplify all this. Remove the flags; e.g. instead of testing for the
INITIALIZED_AO flag, test whether mpctx->ao is set. Move uninit code
to separate functions, e.g. uninit_audio_out().
2014-10-03 23:05:09 +02:00
wm4
f62f984404 player: don't print audio/video init failure message twice
The messages "Audio: no audio" and "Video: no video" could be printed
twice each if initializing them failed. Prevent his silliness.

CC: @mpv-player/stable
2014-10-02 03:12:45 +02:00
wm4
ae2e2b9740 audio: enable pitch correction by default when playing fast
Apparently this is what users want. When playing with normal speed,
nothing is done. When playing slower than normal, resampling is used
instead, because scaletempo (which does the pitch correction) adds
too many artifacts.
2014-10-02 02:58:52 +02:00
wm4
c3e2a1febc command: move setting playback speed to a separate function 2014-10-02 02:49:05 +02:00
wm4
7dd3822d09 audio: refactor some aspects of filter chain setup
There's no real reason why audio_init_filter() should exist. Just use
af_init or af_reinit directly. (We lose a useless message; the same
information is printed in a quite close place with more details.)

Requires less code, and the way the filter chain is marked as having
failed to initialize allows just switching off audio instead of
crashing if trying to insert a volume filter in mixer.c fails, and
recreating the old filter chain fails too.
2014-10-02 02:42:23 +02:00
wm4
4ea05577bd audio: remove --audiodrop
This would play some silence in case video was slower than audio. If
framedropping is already enabled, there's no other way to keep A/V
sync, short of changing audio playback speed (which would give worse
results). The --audiodrop option inserted silence if there was more
than 500ms desync.

This worked somewhat, but I think it was a silly idea after all. Whether
the playback experience is really bad or slightly worse doesn't really
matter. There also was a subtle bug with PTS handling, that apparently
caused A/V desync anyway at ridiculous playback speeds.

Just remove this feature; nobody is going to use it anyway.
2014-09-30 18:05:55 +02:00
wm4
81bf9a1963 audio: cleanup spdif format definitions
Before this commit, there was AF_FORMAT_AC3 (the original spdif format,
used for AC3 and DTS core), and AF_FORMAT_IEC61937 (used for AC3, DTS
and DTS-HD), which was handled as some sort of superset for
AF_FORMAT_AC3. There also was AF_FORMAT_MPEG2, which used
IEC61937-framing, but still was handled as something "separate".

Technically, all of them are pretty similar, but may use different
bitrates. Since digital passthrough pretends to be PCM (just with
special headers that wrap digital packets), this is easily detectable by
the higher samplerate or higher number of channels, so I don't know why
you'd need a separate "class" of sample formats (AF_FORMAT_AC3 vs.
AF_FORMAT_IEC61937) to distinguish them. Actually, this whole thing is
just a mess.

Simplify this by handling all these formats the same way.
AF_FORMAT_IS_IEC61937() now returns 1 for all spdif formats (even MP3).
All AOs just accept all spdif formats now - whether that works or not is
not really clear (seems inconsistent due to earlier attempts to make
DTS-HD work). But on the other hand, enabling spdif requires manual user
interaction, so it doesn't matter much if initialization fails in
slightly less graceful ways if it can't work at all.

At a later point, we will support passthrough with ao_pulse. It seems
the PulseAudio API wants to know the codec type (or maybe not - feeding
it DTS while telling it it's AC3 works), add separate formats for each
codecs. While this reminds of the earlier chaos, it's stricter, and most
code just uses AF_FORMAT_IS_IEC61937().

Also, modify AF_FORMAT_TYPE_MASK (renamed from AF_FORMAT_POINT_MASK) to
include special formats, so that it always describes the fundamental
sample format type. This also ensures valid AF formats are never 0 (this
was probably broken in one of the earlier commits from today).
2014-09-23 23:11:54 +02:00
wm4
70f4721574 player: reset last_av_difference if not applicable
Don't let stale values linger around.

Also fix a slightly related case in audio.c.
2014-09-20 00:44:37 +02:00
wm4
7791e25fa0 audio: fix initial sync with huge AO buffer
With e.g --start=-3 --audio-buffer=10 the decoder entered EOF state
before the initial sync was finished, entered STATUS_EOF, and just
started playing audio from a random position.

This doesn't handle seeking outside of the file, which is a different
case. E.g. --start=30:00 with audio and video enabled in a file shorter
than 30:00 will play a random last part of audio. This could perhaps be
fixed by using the hr-seek target for cutting audio, instead of the
video PTS, but that would be kind of intrusive, so don't do it for now.
The simpler solution, assuming audio EOF on video EOF, wouldn't work,
because we allow audio to start before video, or to last after video.
2014-09-06 13:33:29 +02:00
wm4
eaa1f16564 audio: correctly initialize output buffer
Just like the previous commit, this takes care of fallout from commit
7ab228, which fixed a bug, but introduced some new ones.

CC: @mpv-player/stable
2014-09-05 17:51:45 +02:00
wm4
7ab228629e audio: fix obscure audio resync failure with timelines
Somehow, there was a larger misunderstanding in the code: ao_buffer
does not need to be preserved over audio reinit for proper support of
gapless audio. The actual AO internal buffer takes care of this.

In fact, preserving ao_buffer just breaks audio resync. In the ordered
chapter case, end_pts is used, which means not all audio data in the
buffer is played, thus some data is left over when audio decoding
resumes on the next segment. This triggers some code that aborts resync
if there's "audio decoded" (ao_buffer contains something), but no PTS
is known (nothing was actually decoded yet).

Simplify, and always bind the output buffer to the decoder.

CC: @mpv-player/stable (maybe)
2014-09-05 01:53:10 +02:00
wm4
787839e8ec cosmetics: remove a stray ';' 2014-09-05 01:53:10 +02:00
wm4
fc0fa9a221 audio: go to draining state instead of EOF if audio starts later
Probably no observable effect, but it's more correct. Setting audio to
EOF could have bad effects otherwise (anywhere the player logic for
example decides whether EOF was reached, and such).
2014-08-31 14:48:58 +02:00
wm4
f9f436a490 audio: restore old speed change behavior
Don't attempt to resync after speed changes. Note that most other cases
of audio reinit (like switching tracks etc.) still resync, but other
code paths take care of setting the audio_status accordingly.

This restores the old behavior of not trying to fix audio desync, which
was probably changed with commit 261506e3.

Note that the code as of now wasn't even entirely correct, since the A/V
sync values are slightly shifted. The dsync depends on the audio buffer
size, so a larger buffer size will show more extreme desync. Also see
mplayer2 commit 213a224e, which should fixed this - it was not merged
into mpv, because it disabled audio for too long, resulting in a worse
user experience. This is similar to the issue this commit attempts to
fix.

Fixes: #1042 (probably)
CC: @mpv-player-stable
2014-08-28 14:26:38 +02:00
wm4
f104ef1282 player: minor changes
This shouldn't change anything functionally.

Change the A/V desync message. --framedrop is enabled by default now, so
the text must be changed a little. I've never heard of audio outputs
messing up A/V sync recently, so remove that part.

Remove the unused ao_pts field.

Reorder 2 A/V sync related expressions so that they look the same.
2014-08-25 21:39:24 +02:00
wm4
21f52aeeba audio: minor improvements to timeline switching
In theory, timestamps can be negative, so we shouldn't just return -1
as special value.

Remove the separate code for clearing decode buffers; use the same code
that is used for normal seek reset.
2014-08-23 11:39:07 +02:00
wm4
4c25b000b5 player: fix recent speed change regression
Commit 5afc025c broke this. The reason is that mpctx->delay is updated
when a new video frame is added. This value is also needed to resync
audio, but it will be for the wrong PTS. They must be consistent with
each other, and if they aren't, initial sync will be off by N video
frames, which results at least in worse user experience.

This can be reproduced by for example heavily switching between normal
and 2x speed, or similar.

Fix by readding the video_next_pts field (keeping its use minimal,
instead of reverting the commit that removed it).
2014-08-22 15:36:48 +02:00
wm4
5afc025cc9 video: get rid of video_next_pts field
Not really needed anymore. Code should be mostly equivalent.

Also get rid of some other now-unused or outdated things.
2014-08-22 14:22:06 +02:00
wm4
07aba86b37 audio: add a mode to insert silence on severe A/V desync
This is probably a stupid idea, but it can't be denied that this
actually allows playing video without larger desync, even if video is
too slow.
2014-08-15 23:52:42 +02:00
wm4
fa7c421588 player: use virtual time for --audio-file with ordered chapters
Apparently users prefer this behavior.

It was used for subtitles too, so move the code to calculate the video
offset into a separate function. Seeking also needs to be fixed.

Fixes #1018.
2014-08-15 23:32:37 +02:00
wm4
be64535a4e audio: fix inverted condition
Recent regression. Could perhaps make gapless audio fail to work
correctly.
2014-08-06 20:30:46 +02:00
wm4
4cf3f3ca2c audio: simplify condition
The expression added with the previous commit (0cce8fe6) looked slightly
more complicated than it has to be. The code is equivalent.
2014-07-31 21:11:49 +02:00
wm4
0cce8fe64f audio: fix A/V sync in encoding mode
In encoding mode, the AO pretends to be infinitely fast (it will take
whatever we write, without ever rejecting input). Commit 261506e3 broke
this somehow. It turns out an old hack dealing with this was accidentally
dropped.

This is the hunk of code whose semantics were (partially) dropped:

    if (mpctx->d_audio && (mpctx->restart_playback ? !video_left :
                           ao_untimed(mpctx->ao) && (mpctx->delay <= 0 ||
                                                     !video_left)))
    {
        int status = fill_audio_out_buffers(mpctx, endpts);
        // Not at audio stream EOF yet
        audio_left = status > -2;
    }

This if condition is pretty wild, and it looked like it was pretty much
for audio-only mode, rather than subtle handling for encoding mode.
2014-07-31 04:49:44 +02:00
wm4
1423bd0bfd player: move video display code out of the playloop
Basically move the code from playloop.c to video.c. The new function
write_video() now contains the code that was part of run_playloop().

There are no functional changes, except handling "new_frame_shown"
slightly differently. This is done so that we don't need new a new
MPContext field or a return value for write_video() to signal this
condition. Instead, it's handled indirectly.
2014-07-30 23:29:00 +02:00
wm4
8f2e9f1d61 player: split seek_reset()
This also reduces some code duplication with other parts of the code.

The changfe is mostly cosmetic, although there are also some subtle
changes in behavior. At least one change is that the big desync message
is now printed after every seek.
2014-07-30 23:29:00 +02:00
wm4
8d990408d7 audio: better sync behavior on bogus EOF
In situations when the demuxer reports EOF, but immediately "recovers"
after that and returns new data, it could happen that audio sync was
skipped. Deal with this by actually entering the EOF state, instead of
assuming this will happen later.
2014-07-30 23:29:00 +02:00
wm4
da780309d7 audio: better initial sync for files where audio starts later
Some files have the first audio much later into the video (for whatever
reasons). Instead of appending large amounts of silence to the audio
buffer (and refusing to sync if the audio to append is "too large"),
just wait until enough video has played.
2014-07-30 00:40:45 +02:00
wm4
1cd2b5976d audio: cosmetics: remove unused return value 2014-07-30 00:24:57 +02:00
wm4
63d1d53d2f audio: ignore (some) decoding errors on initialization
It probably happens relatively often that the first packet (or even the
first N packets) of a stream will fail to decode, but decoding will
eventually succeed at a later point. Before commit 261506e3, this was
handled by an explicit retry loop (although this was also for other
purposes), but with then was changed to abort on the first error. This
makes it impossible to decode some audio streams.

Change this so that errors are ignored for the first 50 packets, which
should make it equivalent to the old code.
2014-07-29 18:05:55 +02:00
wm4
862d7d8a1a player: fix desync when seeking and switching external tracks
If you for example use --audio-file, disable the external track, seek,
and enable the external track again, the playback position of the
external file was off, and you would get major A/V desync. This was
actually supposed to work, but broke at some time ago (probably commit
2b87415f). It didn't work, because it attempted to seek the stream if it
was already selected, which was always true due to
reselect_demux_streams() being called before that.

Fix by putting the initial selection and the seek together.
2014-07-29 17:55:28 +02:00
wm4
261506e36e audio: change playback restart and resyncing
This commit makes audio decoding non-blocking. If e.g. the network is
too slow the playloop will just go to sleep, instead of blocking until
enough data is available.

For video, this was already done with commit 7083f88c. For audio, it's
unfortunately much more complicated, because the audio decoder was used
in a blocking manner. Large changes are required to get around this.
The whole playback restart mechanism must be turned into a statemachine,
especially since it has close interactions with video restart. Lots of
video code is thus also changed.

(For the record, I don't think switching this code to threads would
make this conceptually easier: the code would still have to deal with
external input while blocked, so these in-between states do get visible
[and thus need to be handled] anyway. On the other hand, it certainly
should be possible to modularize this code a bit better.)

This will probably cause a bunch of regressions.
2014-07-28 21:20:37 +02:00
wm4
623eac2b1b audio: cosmetics: collapse a function
There's no need for build_afilter_chain() to be a separate function
anymore.
2014-07-24 15:27:40 +02:00
wm4
986099d323 audio: fix race condition in EOF code
Don't return an EOF code if there's still buffered data.

Also, don't call demux_stream_eof() in the playloop. There's probably
nothing wrong with it, but it's cleaner not to use it.

Also give AD_EOF its own value, so that a decoding error doesn't drain
audio by causing an EOF condition.
2014-07-24 15:26:07 +02:00
wm4
967add9f0f audio: remove unused metadata field
This was used for replaygain at some point, until replaygain info was
passed through explicitly.
2014-07-21 19:29:58 +02:00
wm4
9736f3309a audio: use symbolic constants instead of magic integers
Similar to commit 26468743.
2014-07-20 20:42:03 +02:00
Rudolf Polzer
073b2becfe ao_lavc: Fix design of audio pts handling.
There was confusion about what should go into audio pts calculation and
what not (mainly due to the audio push thread). This has been fixed by
using the playing - not written - audio pts (which properly takes into
account the ao's buffer), and incrementing the samples count only by the
amount of samples actually taken from the buffer (unfortunately this
now forces us to keep the lock too long for my taste).
2014-07-16 16:18:34 +02:00
wm4
58880c00ee demux: make replaygain per-track
It's unlikely that files with multiple audio tracks and with replaygain
actually happen, but this change might help avoid minor corner cases
with later changes.
2014-07-05 17:07:14 +02:00
wm4
3e2f16a89e audio: add a "weak" gapless mode, and make it default
Basically, this allows gapless playback with similar files (including
the ordered chapter case), while still being robust in general.

The implementation is quite simplistic on purpose, in order to avoid
all the weird corner cases that can occur when creating the filter
chain. The consequence is that it might do not-gapless playback in
more cases when needed, but if that bothers you, you still can use
the normal gapless mode.

Just using "--gapless-audio" or "--gapless-audio=yes" selects the old
mode.
2014-06-09 01:20:53 +02:00
wm4
c79689206c audio: change handling of an EOF corner case
This code handles buggy AOs (even if all AOs are bug-free, it's good for
robustness). Move handling of it to the AO feed thread. Now this check
doesn't require magic numbers and does exactly what's it supposed to do.
2014-05-30 02:16:43 +02:00
Alessandro Ghedini
e7977ec875 af: add replaygain_data field to af_stream and af_instance
Closes #664
2014-04-04 18:35:29 +02:00
wm4
239dc2851a command: allow changing filters before video chain initialization
Apparently this is more intuitive.

Somewhat tricky, because of the odd state after loading a file but
before initializing the VO.
2014-03-30 19:59:26 +02:00
wm4
ae448e198f audio: remove sample rate limit checks
This played the file at a wrong sample rate if the rate was out of
certain bounds.

A comment says this was for the sake of libaf/af_resample.c. This
resampler has been long removed. Our current resampler
(libav/swresample) checks supported sample rates on reconfiguration, and
will error out if a sample rate is not supported. And I think that is
the correct behavior.
2014-03-30 07:34:43 +02:00
Alessandro Ghedini
04e14ec8f6 af: add metadata field to af_stream and af_instance
This allows to propagate metadata information to audio filters.

Closes #632
2014-03-13 14:36:20 +01:00
wm4
b0b0e69570 audio: don't downmix when doing digital passthrough
This obviously doesn't work. It wasn't much of a problem in the past
because most passthrough formats use 2 channels, which is also the
default for downmix.
2014-03-10 02:14:51 +01:00
wm4
249789c256 audio: make --channels option always force the output layout
Use the --channels value directly on the AO, instead of doing it only in
the --channels=stereo (default) case and if the decoder output is not
stereo.
2014-03-10 02:09:18 +01:00