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.
Ever since a change in mplayer2 or so, relative seeks were translated to
absolute seeks before sending them to the demuxer in most cases. The
only exception in current mpv is DVD seeking.
Remove the SEEK_ABSOLUTE flag; it's not the implied default. SEEK_FACTOR
is kept, because it's sometimes slightly useful for seeking in things
like transport streams. (And maybe mkv files without duration set?)
DVD seeking is terrible because DVD and libdvdnav are terrible, but
mostly because libdvdnav is terrible. libdvdnav does not expose seeking
with seek tables. (Although I know xbmc/kodi use an undocumented API
that is not declared in the headers by dladdr()ing it - I think the
function is dvdnav_jump_to_sector_by_time().) With the current mpv
policy if not giving a shit about DVD, just revert our half-working seek
hacks and always use dvdnav_time_search(). Relative seeking might get
stuck sometimes; in this case --hr-seek=always is recommended.
This is mainly a refactor. I'm hoping it will make some things easier
in the future due to cleanly separating codec metadata and stream
metadata.
Also, declare that the "codec" field can not be NULL anymore. demux.c
will set it to "" if it's NULL when added. This gets rid of a corner
case everything had to handle, but which rarely happened.
The demuxer infrastructure was originally single-threaded. To make it
suitable for multithreading (specifically, demuxing and decoding on
separate threads), some sort of tripple-buffering was introduced. There
are separate "struct demuxer" allocations. The demuxer thread sets the
state on d_thread. If anything changes, the state is copied to d_buffer
(the copy is protected by a lock), and the decoder thread is notified.
Then the decoder thread copies the state from d_buffer to d_user (again
while holding a lock). This avoids the need for locking in the
demuxer/decoder code itself (only demux.c needs an internal, "invisible"
lock.)
Remove the streams/num_streams fields from this tripple-buffering
schema. Move them to the internal struct, and protect them with the
internal lock. Use accessors for read access outside of demux.c.
Other than replacing all field accesses with accessors, this separates
allocating and adding sh_streams. This is needed to avoid race
conditions. Before this change, this was awkwardly handled by first
initializing the sh_stream, and then sending a stream change event. Now
the stream is allocated, then initialized, and then declared as
immutable and added (at which point it becomes visible to the decoder
thread immediately).
This change is useful for PR #2626. And eventually, we should probably
get entirely of the tripple buffering, and this makes a nice first step.
Remove the old implementation for these properties. It was never very
good, often returned very innaccurate values or just 0, and was static
even if the source was variable bitrate. Replace it with the
implementation of "packet-video-bitrate". Mark the "packet-..."
properties as deprecated. (The effective difference is different
formatting, and returning the raw value in bits instead of kilobits.)
Also extend the documentation a little.
It appears at least some decoders (sipr?) need the
AVCodecContext.bit_rate field set, so this one is still passed through.
Currently, audio packets are always filtered as a whole. Since demux_raw
output a 1 second long packet, this could lead to large delays when
applying softvol volume. It could be fixed by splitting the frames the
decoder outputs before filtering them (like the old filter code used
to), but since this didn't cause any other problems yet, I'm going with
the simpler fix.
Fixes#1558.
stream_cdda's output format is linked to demux_raw's default audio
format, and at least we don't care enough to provide a separate
mechanism to let stream_cdda explicitly set the format, so they must
match.
Judging from the existing code, it looks like CDDA always outputs little
endian. stream_cdda.c changed this back to native endian (what demux_raw
expects). Just make them both little endian. This requires less code,
and also having a raw demuxer's behavior depend on the endianness of the
machine isn't very sane anyway.
For a while, we used this to transfer PCM from demuxer to the filter
chain. We had a special "codec" that mapped what MPlayer used to do
(MPlayer passes the AF sample format over an extra field to ad_pcm,
which specially interprets it).
Do this by providing a mp_set_pcm_codec() function, which describes a
sample format in a generic way, and sets the appropriate demuxer header
fields so that libavcodec interprets it correctly. We use the fact that
libavcodec has separate PCM decoders for each format. These are
systematically named, so we can easily map them.
This has the advantage that we can change the audio filter chain as we
like, without losing features from the "rawaudio" demuxer. In fact, this
commit also gets rid of the audio filter chain formats completely.
Instead have an explicit list of PCM formats. (We could even just have
the user pass libavcodec PCM decoder names directly, but that would be
annoying in other ways.)
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).
Until now, the audio chain could handle both little endian and big
endian formats. This actually doesn't make much sense, since the audio
API and the HW will most likely prefer native formats. Or at the very
least, it should be trivial for audio drivers to do the byte swapping
themselves.
From now on, the audio chain contains native-endian formats only. All
AOs and some filters are adjusted. af_convertsignendian.c is now wrongly
named, but the filter name is adjusted. In some cases, the audio
infrastructure was reused on the demuxer side, but that is relatively
easy to rectify.
This is a quite intrusive and radical change. It's possible that it will
break some things (especially if they're obscure or not Linux), so watch
out for regressions. It's probably still better to do it the bulldozer
way, since slow transition and researching foreign platforms would take
a lot of time and effort.
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.
The i_bps members of the sh_audio and dev_video structs are mostly used
for displaying the average audio and video bitrates. Keeping them in
bits-per-second avoids truncating them to bytes-per-second and changing
them back lateron.
Stop using it in most places, and prefer STREAM_CTRL_GET_SIZE. The
advantage is that always the correct size will be used. There can be no
doubt anymore whether the end_pos value is outdated (as it happens often
with files that are being downloaded).
Some streams still use end_pos. They don't change size, and it's easier
to emulate STREAM_CTRL_GET_SIZE using end_pos, instead of adding a
STREAM_CTRL_GET_SIZE implementation to these streams.
Make sure int64_t is always used for STREAM_CTRL_GET_SIZE (it was
uint64_t before).
Remove the seek flags mess, and replace them with a seekable flag. Every
stream must set it consistently now, and an assertion in stream.c checks
this. Don't distinguish between streams that can only be forward or
backwards seeked, since we have no such stream types.
stream.start_pos was needed for optical media only, and (apparently) not
for very good reasons. Just get rid of it.
For stream_dvd, we don't need to do anything. Byte seeking was already
removed from it earlier.
For stream_cdda and stream_vcd, emulate the start_pos by offsetting the
stream pos as seen by the rest of mpv.
The bits in discnav.c and loadfile.c were for dealing with the code
seeking back to the start in demux.c. Handle this differently by
assuming the demuxer is always initialized with the stream at start
position, and instead seek back if initializing the demuxer fails.
Remove the --sb option, which worked by modifying stream.start_pos. If
someone really wants this option, it could be added back by creating a
"slice" stream (actually ffmpeg already has such a thing).
The TV code pretends to be part of stream/, but it's actually demuxer
code too. The audio_in code is shared between the TV code and
stream_radio.c, so stream_radio.c needs a small hack until stream.c is
converted.
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.
There are 3 users of the image format option type: demux_raw,
vf_format, vf_noformat. Allow the hwaccel formats (like vdpau etc.)
in general, so that the filters can use it. This won't work for
demux_raw, so explicitly reject these formats there.
This used to be needed to access the generic stream header from the
specific headers, which in turn was needed because the decoders had
access only to the specific headers. This is not the case anymore, so
this can finally be removed again.
Also move the "format" field from the specific headers to sh_stream.
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.
Pointless, using stream->start_pos/end_pos instead.
demux_mf was the only place where this was used specially, but we can
rely on timestamps instead for this case.
There are some Microsoft Windows symbols which are traditionally used by
the mplayer core, because it used to be convenient (avi was the big
format, using binary windows decoders made sense...). So these symbols
have the exact same definition as the Windows one, and if mplayer is
compiled on Windows, the symbols from windows.h are used.
This broke recently just because some files were shuffled around, and
the symbols defined in ms_hdr.h collided with windows.h ones. Since we
don't have windows binary decoders anymore, there's not the slightest
reason our symbols should have the same names. Rename them to reduce the
risk for collision, and to fix the recent regression.
Drop WAVEFORMATEXTENSIBLE, because it's mostly unused. ao_dsound defines
its own version if the windows headers don't define it, and ao_wasapi is
not available on systems where this symbol is missing.
Also reindent ms_hdr.h.
The rawaudio demuxer read one frame per packet, basically a few bytes,
which caused insane overhead. (I found this when I couldn't play raw
audio without dropouts when using -v, which printed a line per packet
read.)
Fix this and read 1 second of audio per packet. This is a regression
since cfa5712 (merging of demux_rawaudio and demux_rawvideo).
wm4
Eric Toombs
Marcin Kurczewski
Marcoen Hirschberg
Stefano Pigozzi