This deprecates the --ad-spdif-dtshd option, and replaces it with a
pseudo decoder. This means ad_spdif will report two decoders, "dts" and
"dts-hd", of which the second simply enables what the option did.
The --ad-spdif-dtshd option will actually be deprecated in the next
commit.
Apparently some A/V receivers do not behave well if "normal" DTS is
passed through using the high bitrate spdif format normally used for
DTS-HD (other receivers are fine with it).
Parse the first packet passed to ad_spdif by decoding it with libavcodec
in order to get the profile. Ignore the --ad-spdif-dtshd if it's not
DTS-HD. (If the codec profile changes midstream, the user is out of
luck. But this is probably an insignificant corner case.)
I thought about parsing the bitstream, but let's not. While it probably
wouldn't be that much effort, we are trying to keep it down on codec
details here - otherwise we could just do our own spdif framing instead
of using libavformat's spdif pseudo-muxer.
Another possibility, using the codec parameters signalled by
libavformat, is disregarded. Our builtin Matroska decoder doesn't do
this, and also we do not want on the demuxer having to decode some
packets in order to retrieve codec params (as libavformat does).
Fixes#1949.
In commit 5f8b060e I blindly assumed that the packet sizes were in
pseudo-samples, but they were actually in bytes. Oops.
(The effect was that cutting the audio was a bit less precise than it
can be.)
Also remove the packet size from ad_spdif.c; it didn't actually use it,
and simply takes what the spdif "muxer" returns.
Needed for the next commit. This commit should probably be reverted as
soon as we're working with full audio frames internally, instead of
"flat" FIFOs.
This rewrites the audio decode loop to some degree. Audio filters don't
do refcounted frames yet, so af.c contains a hacky "emulation".
Remove some of the weird heuristic-heavy code in dec_audio.c. Instead of
estimating how much audio we need to filter, we always filter full
frames. Maybe this should be adjusted later: in case filtering increases
the volume of the audio data, we should try not to buffer too much
filter output by reducing the input that is fed at once.
For ad_spdif.c and ad_mpg123.c, we don't avoid extra copying yet - it
doesn't seem worth the trouble.
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).
IEC 61937 frames should always be little endian (little endian 16 bit
words). I don't see any apparent need why the audio chain should handle
swapped-endian formats.
It could be that some audio outputs might want them (especially on big
endian architectures). On the other hand, it's not clear how that works
on these architectures, and it's not even known whether the current code
works on big endian at all. If something should break, and it should
turn out that swapped-endian spdif is needed on any platform/AO,
swapping still could be done in-place within the affected AO, and
there's no need for the additional complexity in the rest of the player.
Note that af_lavcac3enc outputs big endian spdif frames for unknown
reasons. Normally, the resulting data is just pulled through an auto-
inserted conversion filter and turned into little endian. Maybe this was
done as a trick so that the code didn't have to byte-swap the actual
audio frame. In any case, just make it output little endian frames.
All of this is untested, because I have no receiver hardware.
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.
Accidentally broken in b6af44d3. For ad_lavc (and in general), the PTS
was not updated correctly when filtering only parts of audio frames,
and for ad_mpg123 and ad_spdif the PTS was additionally offset by the
frame size.
This could lead to incorrect time display, and possibly broken A/V sync.
This commit mainly moves the initial decoding of data (done to probe the
audio format) to generic code. This will make it easier to make audio
decoding non-blocking in a later commit.
This commit also changes how decoders return data: instead of having
them write the data into a prepared buffer, they return a reference to
an internal buffer (by setting dec_audio.decoded). This makes it
significantly easier to handle audio format changes, since the decoders
don't really need to care anymore.
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.
If the timebase is set, it's used for converting the packet timestamps.
Otherwise, the previous method of reinterpret-casting the mpv style
double timestamps to libavcodec style int64_t timestamps is used.
Also replace the kind of awkward mp_get_av_frame_pkt_ts() function by
mp_pts_from_av(), which simply converts timestamps in a way the old
function did. (Plus it takes a timebase parameter, similar to the
addition to mp_set_av_packet().)
Note that this should not change anything yet. The code in ad_lavc.c and
vd_lavc.c passes NULL for the timebase parameters. We could set
AVCodecContext.pkt_timebase and use that if we want to give libavcodec
"proper" timestamps.
This could be important for ad_lavc.c: some codecs (opus, probably mp3
and aac too) have weird requirements about doing decoding preroll on the
container level, and thus require adjusting the audio start timestamps
in some cases. libavcodec doesn't tell us how much was skipped, so we
either get shifted timestamps (by the length of the skipped data), or we
give it proper timestamps. (Note: libavcodec interprets or changes
timestamps only if pkt_timebase is set, which by default it is not.)
This would require selecting a timebase though, so I feel uncomfortable
with the idea. At least this change paves the way, and will allow some
testing.
sh_audio is supposed to contain file headers, not whatever was decoded.
Fix this, and write the decoded format to separate fields in the decoder
context, the dec_audio.decoded field. (Note that this field is really
only needed to communicate the audio format from decoder driver to the
generic code, so no other code accesses it.)
Move all state that basically changes during decoding or is needed in
order to manage decoding itself into a new struct (dec_audio).
sh_audio (defined in stheader.h) is supposed to be the audio stream
header. This should reflect the file headers for the stream. Putting the
decoder context there is strange design, to say the least.
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.
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.
Apparently we were using FFmpeg-specific APIs. I have no idea whether
this code is correct on both FFmpeg and Libav (no examples, bad
doxygen... why do they even complaint aht people are using their APIs
incorrectly?), but it appears to work on FFmpeg. That was also the case
before commit ebc4ccb though, where it used internal libavformat
symbols.
Untested on Libav, Travis will tell us.
This accessed tons of private libavformat symbols all over the place.
Don't do this and convert all code to proper public APIs. As a
consequence, the code becomes shorter and cleaner (many things the code
tried are done by libavformat APIs).
pts_bytes can't just be changed at the end. It must be offset to the pts
value, which is reset with each packet read from the demuxer. Make sure
the pts_byte field is always reset after receiving a new PTS, i.e.
increment it after actually writing to the output buffer.
Flush the AVFormatContext's write buffer, because otherwise the audio
PTS will jump around too much: the calculation doesn't use the exact
output buffer size if there's still data in the avio buffer.
Partial packet reads were needed because the video/audio parsers were
working on top of them. So it could happen that a parser read a part of
a packet, and returned that to the decoder. With libavformat/libavcodec,
packets are already parsed, and everything is much simpler.
Most of the simplifications in ad_spdif could have been done earlier.
Remove some other stuff as well, like the questionable slave mode start
time reporting (could be replaced by proper code, but we don't bother).
Remove the unused skip_audio_frame() functionality as well (it was used
by old demuxers). Some functions become private to demux.c, like
demux_fill_buffer(). Introduce new packet read functions, which have
simpler semantics. Packets returned from them are owned by the caller,
and all packets in the demux.c packet queue are considered unread.
Remove special code that dropped subtitle packets with size 0. This
used to be needed because it caused special cases in the old code.
The audio parser was needed only by the "old" demuxers, and
demux_rawaudio. All other demuxers output already parsed packets.
demux_rawaudio is usually for raw audio, so using a parser with it
doesn't usually make sense. But you can also force it to read
compressed formats with fixed packet sizes, in which case the parser
would have been used. This use case is probably broken now, but you
will be able to do the same thing with libavformat demuxers.
This helps passing the channel layout correctly from decoder to audio
filter chain. (Because that part "reuses" the demuxer level codec
parameters, which is very disgusting.)
Note that ffmpeg stuff already passed the channel layout via
mp_copy_lav_codec_headers(). So other than easier dealing with the
demuxer/decoder parameters mess, there's no real advantage to doing
this.
Make the --channels option accept a channel map. Since simple numbers
map to standard layouts with the given number of channels, this is
downwards compatible. Likewise for demux_rawaudio.
The old names have been deprecated a while ago, but were needed for
supporting older ffmpeg/libav versions. The deprecated identifiers
have been removed from recent Libav and FFmpeg git.
This change breaks compatibility with Libav 0.8.x and equivalent
FFmpeg releases.
The spdif decoder was hardcoded to assume that the spdif output is
capable of accepting high (>1.5Mbps) bitrates. While this is true
for modern HDMI spdif interfaces, the original coax/toslink system
cannot deal with this and will fail to work.
This patch adds an option --dtshd which can be enabled if you use
a DTS-capable receiver behind a HDMI link.
Use codec names instead of FourCCs to identify codecs. Rewrite how
codecs are selected and initialized. Now each decoder exports a list
of decoders (and the codec it supports) via add_decoders(). The order
matters, and the first decoder for a given decoder is preferred over
the other decoders. E.g. all ad_mpg123 decoders are preferred over
ad_lavc, because it comes first in the mpcodecs_ad_drivers array.
Likewise, decoders within ad_lavc that are enumerated first by
libavcodec (using av_codec_next()) are preferred. (This is actually
critical to select h264 software decoding by default instead of vdpau.
libavcodec and ffmpeg/avconv use the same method to select decoders by
default, so we hope this is sane.)
The codec names follow libavcodec's codec names as defined by
AVCodecDescriptor.name (see libavcodec/codec_desc.c). Some decoders
have names different from the canonical codec name. The AVCodecDescriptor
API is relatively new, so we need a compatibility layer for older
libavcodec versions for codec names that are referenced internally,
and which are different from the decoder name. (Add a configure check
for that, because checking versions is getting way too messy.)
demux/codec_tags.c is generated from the former codecs.conf (minus
"special" decoders like vdpau, and excluding the mappings that are the
same as the mappings libavformat's exported RIFF tables). It contains
all the mappings from FourCCs to codec name. This is needed for
demux_mkv, demux_mpg, demux_avi and demux_asf. demux_lavf will set the
codec as determined by libavformat, while the other demuxers have to do
this on their own, using the mp_set_audio/video_codec_from_tag()
functions. Note that the sh_audio/video->format members don't uniquely
identify the codec anymore, and sh->codec takes over this role.
Replace the --ac/--vc/--afm/--vfm with new --vd/--ad options, which
provide cover the functionality of the removed switched.
Note: there's no CODECS_FLAG_FLIP flag anymore. This means some obscure
container/video combinations (e.g. the sample Film_200_zygo_pro.mov)
are played flipped. ffplay/avplay doesn't handle this properly either,
so we don't care and blame ffmeg/libav instead.
Do not fall back to 0 for samplerate when parser is not initialized.
Might fix some issues with using -ac spdifenc with audio in MKV
or MP4.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35517 b3059339-0415-0410-9bf9-f77b7e298cf2
Replace outdated list of unsupported formats by list of supported formats.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35534 b3059339-0415-0410-9bf9-f77b7e298cf2
Do not call af_fmt2str on the same data over and over.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35535 b3059339-0415-0410-9bf9-f77b7e298cf2
ad_spdif: use the more specific AF_FORMAT_AC3_LE when
we handle AC3.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35536 b3059339-0415-0410-9bf9-f77b7e298cf2
Make AF_FORMAT_IS_IEC61937 include AF_FORMAT_IS_AC3.
Our AC3 "sample format" is also iec61937.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35537 b3059339-0415-0410-9bf9-f77b7e298cf2
af_format: support endianness conversion also for iec61937
formats in general, not just AC3.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35538 b3059339-0415-0410-9bf9-f77b7e298cf2
Conflicts:
audio/filter/af_format.c
af_format: Fix check_format, non-special formats are of course supported.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35545 b3059339-0415-0410-9bf9-f77b7e298cf2
Note: see mplayer bug #2110
Finish renaming directories and moving files. Adjust all include
statements to make the previous commit compile.
The two commits are separate, because git is bad at tracking renames
and content changes at the same time.
Also take this as an opportunity to remove the separation between
"common" and "mplayer" sources in the Makefile. ("common" used to be
shared between mplayer and mencoder.)
Tis drops the silly lib prefixes, and attempts to organize the tree in
a more logical way. Make the top-level directory less cluttered as
well.
Renames the following directories:
libaf -> audio/filter
libao2 -> audio/out
libvo -> video/out
libmpdemux -> demux
Split libmpcodecs:
vf* -> video/filter
vd*, dec_video.* -> video/decode
mp_image*, img_format*, ... -> video/
ad*, dec_audio.* -> audio/decode
libaf/format.* is moved to audio/ - this is similar to how mp_image.*
is located in video/.
Move most top-level .c/.h files to core. (talloc.c/.h is left on top-
level, because it's external.) Park some of the more annoying files
in compat/. Some of these are relicts from the time mplayer used
ffmpeg internals.
sub/ is not split, because it's too much of a mess (subtitle code is
mixed with OSD display and rendering).
Maybe the organization of core is not ideal: it mixes playback core
(like mplayer.c) and utility helpers (like bstr.c/h). Should the need
arise, the playback core will be moved somewhere else, while core
contains all helper and common code.
wm4
Marcin Kurczewski
Stefano Pigozzi
Wessel Dankers
reimar