Until recently, ao_lavc and vo_lavc started encoding whenever the core
happened to send them data. Since audio and video are not initialized at
the same time, and the muxer was not necessarily opened when the first
encoder started to produce data, the resulting packets were put into a
queue. As soon as the muxer was opened, the queue was flushed.
Change this to make the core wait with sending data until all encoders
are initialized. This has the advantage that we don't need to queue up
the packets.
The main change is that we wait with opening the muxer ("writing
headers") until we have data from all streams. This fixes race
conditions at init due to broken assumptions in the old code.
This also changes a lot of other stuff. I found and fixed a few API
violations (often things for which better mechanisms were invented, and
the old ones are not valid anymore). I try to get away from the public
mutex and shared fields in encode_lavc_context. For now it's still
needed for some timestamp-related fields, but most are gone. It also
removes some bad code duplication between audio and video paths.
The af_get_best_sample_formats() function had an argument of
int[AF_FORMAT_COUNT], which is slightly incorrect, because it's 0
terminated and should in theory have AF_FORMAT_COUNT+1 entries. It won't
actually write this many formats (since some formats are fundamentally
incompatible), but it still feels annoying and incorrect. So fix it, and
require that callers pass an AF_FORMAT_COUNT+1 array.
Note that the array size has no meaning in C function arguments (just
another issue with C static arrays being weird and stupid), so get rid
of it completely.
Not changing the af_lavcac3enc use, since that is rewritten in another
branch anyway.
Before this change, AOs could have internal alignment, and play() would
not consume the trailing data if the size passed to it is not aligned.
Change this to require AOs to report their alignment (via period_size),
and make sure to always send aligned data.
The buffer reported by get_space() now always has to be correct and
reliable. If play() does not consume all data provided (which is bounded
by get_space()), an error is printed.
This is preparation for potential further AO changes.
I casually checked alsa/lavc/null/pcm, the other AOs might or might not
work.
This commit adds an --audio-channel=auto-safe mode, and makes it the
default. This mode behaves like "auto" with most AOs, except with
ao_alsa. The intention is to allow multichannel output by default on
sane APIs. ALSA is not sane as in it's so low level that it will e.g.
configure any layout over HDMI, even if the connected A/V receiver does
not support it. The HDMI fuckup is of course not ALSA's fault, but other
audio APIs normally isolate applications from dealing with this and
require the user to globally configure the correct output layout.
This will help with other AOs too. ao_lavc (encoding) is changed to the
new semantics as well, because it used to force stereo (perhaps because
encoding mode is supposed to produce safe files for crap devices?).
Exclusive mode output on Windows might need to be adjusted accordingly,
as it grants the same kind of low level access as ALSA (requires more
research).
In addition to the things mentioned above, the --audio-channels option
is extended to accept a set of channel layouts. This is supposed to be
the correct way to configure mpv ALSA multichannel output. You need to
put a list of channel layouts that your A/V receiver supports.
This is particularly useful for opus which allows only a fairly restrictive set
of samplerates. If the codec doesn't provide a list of samplerates, just
continue to try the requsted one and hope for the best.
fixes#2957
This is just a refactor, which makes it use the previously introduced
function, and allows us to make af_format_conversion_score() private.
(We drop 2 unlikely warning messages too... who cares.)
Replace all the check macros with function calls. Give them all the
same case and naming schema.
Drop af_fmt2bits(). Only af_fmt2bps() survives as af_fmt_to_bytes().
Introduce af_fmt_is_pcm(), and use it in situations that used
!AF_FORMAT_IS_SPECIAL. Nobody really knew what a "special" format
was. It simply meant "not PCM".
The caller set up the "start" pointer array using the number of planes,
the encode() function used the number of channels. This copied
uninitialized values for packed formats, which makes Coverity warn.
When initialization failed, vo_lavc may cause an irrecoverable state in
the ffmpeg-related structs. Therefore, we reject additional
initialization attempts at least until we know a better way to clean up
the mess.
ao_lavc currently cannot be initialized more than once, yet it's good to
do consistent changes there as well.
Also, clean up uninit-after-failure handling to be less spammy.
bstr.c doesn't really deserve its own directory, and compat had just
a few files, most of which may as well be in osdep. There isn't really
any justification for these extra directories, so get rid of them.
The compat/libav.h was empty - just delete it. We changed our approach
to API compatibility, and will likely not need it anymore.
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).
In most places where af_fmt2bits is called to get the bits/sample, the
result is immediately converted to bytes/sample. Avoid this by getting
bytes/sample directly by introducing af_fmt2bps.
Until now, this was always conflated with uninit. This was ugly, and
also many AOs emulated this manually (or just ignored it). Make draining
an explicit operation, so AOs which support it can provide it, and for
all others generic code will emulate it.
For ao_wasapi, we keep it simple and basically disable the internal
draining implementation (maybe it should be restored later).
Tested on Linux only.
Since the AO will run in a thread, and there's lots of shared state with
encoding, we have to add locking.
One case this doesn't handle correctly are the encode_lavc_available()
calls in ao_lavc.c and vo_lavc.c. They don't do much (and usually only
to protect against doing --ao=lavc with normal playback), and changing
it would be a bit messy. So just leave them.
We want to move the AO to its own thread. There's no technical reason
for making the ao struct opaque to do this. But it helps us sleep at
night, because we can control access to shared state better.
This field will be moved out of the ao struct. The encoding code was
basically using an invalid way of accessing this field.
Since the AO will be moved into its own thread too and will do its own
buffering, the AO and the playback core might not even agree which
sample a PTS timestamp belongs to. Add some extrapolation code to handle
this case.
Since m_option.h and options.h are extremely often included, a lot of
files have to be changed.
Moving path.c/h to options/ is a bit questionable, but since this is
mainly about access to config files (which are also handled in
options/), it's probably ok.
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.
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.
This comes with two internal AO API changes:
1. ao_driver.play now can take non-interleaved audio. For this purpose,
the data pointer is changed to void **data, where data[0] corresponds to
the pointer in the old API. Also, the len argument as well as the return
value are now in samples, not bytes. "Sample" in this context means the
unit of the smallest possible audio frame, i.e. sample_size * channels.
2. ao_driver.get_space now returns samples instead of bytes. (Similar to
the play function.)
Change all AOs to use the new API.
The AO API as exposed to the rest of the player still uses the old API.
It's emulated in ao.c. This is purely to split the commits changing all
AOs and the commits adding actual support for outputting N-I audio.
No AO can handle these, so it would be a problem if they get added
later, and non-interleaved formats get accepted erroneously. Let them
gracefully fall back to other formats.
Most AOs actually would fall back, but to an unrelated formats. This is
covered by this commit too, and if possible they should pick the
interleaved variant if a non-interleaved format is requested.
Now to shift audio pts when outputting to e.g. avi, you need an explicit
facility to insert/remove initial samples, to avoid initial regions of
the video to be sped up/slowed down.
One such facility is the delay filter in libavfilter.
ao_lavc.c accesses ao->buffer, which I consider internal. The access was
done in ao_lavc.c/uninit(), which tried to get the left-over audio in
order to write the last (possibly partial) audio frame. The play()
function didn't accept partial frames, because the AOPLAY_FINAL_CHUNK
flag was not correctly set, and handling it otherwise would require an
internal FIFO.
Fix this by making sure that with gapless audio (used with encoding),
the AOPLAY_FINAL_CHUNK is set only once, instead when each file ends.
Basically, move the hack in ao_lavc's uninit to uninit_player.
One thing can not be entirely correctly handled: if gapless audio is
active, we don't know really whether the AO is closed because the file
ended playing (i.e. we want to send the buffered remainder of the audio
to the AO), or whether the user is quitting the player. (The stop_play
flag is overwritten, fixing that is perhaps not worth it.) Handle this
by adding additional code to drain the AO and the buffers when playback
is quit (see play_current_file() change).
Test case: mpv avdevice://lavfi:sine=441 avdevice://lavfi:sine=441 -length 0.2267 -gapless-audio
wm4
Rudolf Polzer
Rudolf Polzer
Kevin Mitchell
Dmitrij D. Czarkoff
Marcin Kurczewski
Marcoen Hirschberg