Make the VF/VO/AO option parser available to audio filters. No audio
filter uses this yet, but it's still a quite intrusive change.
In particular, the commands for manipulating filters at runtime
completely change. We delete the old code, and use the same
infrastructure as for video filters. (This forces complete
reinitialization of the filter chain, which hopefully isn't a problem
for any use cases. The old code forced reinitialization too, but it
could potentially allow a filter to cache things; e.g. consider loaded
ladspa plugins and such.)
Matroska has an output sample rate (OutputSamplingFrequency), which in
theory should be forced instead of whatever the decoder outputs. But it
appears no software (other than mplayer2 and mpv until now) actually
respects this. Even worse, there were broken files around, which played
correctly with (in theory) broken software, but not mplayer2/mpv. Hacks
were added to our code to play these files correctly, but they didn't
catch all cases.
Simplify this by doing what everyone else does, and always use the
decoder's sample rate instead. In particular, we try to handle all
sample rate issues like libavformat's Matroska demuxer does.
It turns out that some code that was removed earlier was still needed.
avcodec_decode_audio4() can decode packets "partially". In that case,
you have to "slice" the packet and call the decode function again.
Codecs which need this are obscure and in low numbers. One sample that
needs it is here:
rsync://fate-suite.ffmpeg.org/fate-suite/lossless-audio/luckynight-partial.shn
(This one decodes in rather small increments.)
The new code is much simpler than what has been removed earlier,
though. The fact that we own the packet returned by the demuxer helps
a lot.
Not sure what should happen if avcodec_decode_audio4() returns 0.
Currently, we throw away the packet in this case. We don't want to be
stuck in an endless loop (could happen if the decoder produces no
output either).
This is not directly related to the handling of format changes itself,
but playing audio normally after the change. This was broken: the output
byte rate was not recalculated, so audio-video sync was simply broken.
Fix this by calculating the byte rate on the fly, instead of storing it
in sh_audio.
Format changes are relatively common (switches between stereo and 5.1
in TV recordings), so this fixes a somewhat critical bug.
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.
We don't need to deal with partial packet reads, manually using an audio
parser, or having to call the libavcodec decoder multiple times per
packet.
Actually, I'm not sure about the last point. ffplay still does this, but
the ffmpeg demuxing.c example doesn't.
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.
Delete demux_avi, demux_asf, demux_mpg, demux_ts. libavformat does
better than them (except in rare corner cases), and the demuxers have
a bad influence on the rest of the code. Often they don't output
proper packets, and require additional audio and video parsing. Most
work only in --no-correct-pts mode.
Remove them to facilitate further cleanups.
Whatever this was supposed to be originally, it doesn't have much value
anymore. It just forced ad_mpg123 to upmix mono to stereo by default
(the audio chain can do that). As an option, it was mostly useless and
misleading, so get rid of it.
Audio and video had their own (very similar) functions to initialize an
AVPacket (ffmpeg's packet struct) from a demux_packet (mplayer's packet
struct). Add a common function for these.
Also use this function for sd_lavc_conv. This is actually a functional
change, as some libavfilter subtitle demuxers add weird out-of-band
stuff as side-data.
Using demux_rawaudio and the --rawaudio-channels option is useful for
testing channel map stuff. The libavcodec PCM decoder normalizes the
channel map to ffmpeg order, though. Prevent this by forcing the
original channel map when using the mp-pcm pseudo decoder entry (used by
demux_rawaudio and stream/tv.c only).
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.
This is done in af_lavrresample now, and as part of format negotiation.
Also remove the remaining reorder_channel calls. They were redundant
and did nothing.
This actually breaks audio for 5/6/8 channels. There's no reordering
done yet. The actual reordering will be done inside of af_lavrresample
and has to be made part of the format negotiation.
mp_audio has some redundant fields. Setters like mp_audio_set_format()
initialize these properly.
Also move the mp_audio struct to a the file audio.c.
We can remove a mysterious line of code from af.c:
in.format |= af_bits2fmt(in.bps * 8);
I'm not sure if this was ever actually needed, or if it was some kind of
"make it work" quick-fix that works against the way things were supposed
to work. All filters etc. now set the format correctly, so if there ever
was a need for this code, it's definitely gone.
Add dummy input and output filters to remove special cases in the format
negotiation code (af_fix_format_conversion() etc.). The output of the
filter chain is now negotiated in exactly the same way as normal
filters.
Negotiate setting the sample rate in the same way as other audio
parameters. As a side effect, the resampler is inserted at the start of
the filter chain instead of the end, but that shouldn't matter much,
especially since conversion and channel mixing are conflated into the
same filter (due to libavresample's API).
Anything this option did has been removed in the preceding 3 commits.
Note that even though these options sounded like a good idea (like
setting accuracy vs. speed tradeoffs), they were not really properly
implemented.
The only thing this option did was changing the behavior of af_volume.
The option decided what sample format af_volume would use, but only if
the sample format was not already float. If the option was set, it would
default to float, otherwise to S16.
Remove use of the option and all associated code, and make af_volume
always use float (unless a af_volume specific sub-option is set).
Silence maximum value tracking. This message is printed when the filter
is destroyed, and it's slightly annoying. Was enabled due to enabling
float by default.
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.
Instead of putting codec header data into WAVEFORMATEX and
BITMAPINFOHEADER, pass it directly via AVCodecContext. To do this, we
add mp_copy_lav_codec_headers(), which copies the codec header data
from one AVCodecContext to another (originally, the plan was to use
avcodec_copy_context() for this, but it looks like this would turn
decoder initialization into an even worse mess).
Get rid of the silly CodecID <-> codec_tag mapping. This was originally
needed for codecs.conf: codec tags were used to identify codecs, but
libavformat didn't always return useful codec tags (different file
formats can have different, overlapping tag numbers). Since we don't
go through WAVEFORMATEX etc. and pass all header data directly via
AVCodecContext, we can be absolutely sure that the codec tag mapping is
not needed anymore.
Note that this also destroys the "standard" MPlayer method of exporting
codec header data. WAVEFORMATEX and BITMAPINFOHEADER made sure that
other non-libavcodec decoders could be initialized. However, all these
decoders have been removed, so this is just cruft full of old hacks that
are not needed anymore. There's still ad_spdif and ad_mpg123, bu neither
of these need codec header data. Should we ever add non-libavcodec
decoders, better data structures without the past hacks could be added
to export the headers.
Rearrange some code to make it easier readable. Remove some dead code,
and stop printing AVI headers in demux_lavf. (These are not actual AVI
headers, just for internal use.)
There should be no functional changes, other than reducing output in
verbose mode.
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.
Uses the same trick as the planarization code to turn per-sample memcpy
calls into mov instructions. Makes decoding a ~25min 48000Hz 2ch floatle
audio file faster from 3.8s to 2.7s.
ad_dvdpcm reads MPEG specific headers directly (passed through codecdata
by demux_mpg), so you couldn't use ffmpeg's "pcm_dvd" with demux_mpg.
Change demux_mpg to set the correct audio parameters directly. The code
for this is taken from ad_dvdpcm.
ad_dvdpcm is evil because it still does partial packet reads (with
demux_read_data()), and it's redundant to libavcodec anyway.
Since libavcodec doesn't have a "generic" PCM decoder, we have to go out
of out way to make it look like ad_lavc provides one: make it provide a
pseudo "pcm" decoder, which maps some format tags manually to the
individual libavcodec PCM decoders.
Format tags which uniquely map to one libavcodec could be mapped via
codecs.conf. Since defining these in tag_map[] is much shorter (one line
vs. a full codec entry in codecs.conf), and since we need tag_map[]
anyway, we don't use codecs.conf for these.
ad_pcm is evil because it still does partial packet reads (with
demux_read_data()), and it's redundant to libavcodec anyway.
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
Reinitialize sh_audio->samplesize and sample_format before falling back
to another audio decoder (some decoders rely on default values). Remove
code setting these fields from demux_mkv and demux_lavf (no decoder
should depend on demuxer-set values for these fields).
Conflicts:
audio/decode/ad_lavc.c
Merged from mplayer2 commit 6b9567. The changes to ad_lavc.c are not
merged, as they are very specific to the mplayer2 libavresample hack;
we deplanarize manually, so we can't get unsupported sample formats
yet (except on raw audio with "pcm_f64le", as we don't support
AV_SAMPLE_FMT_DBL in the audio chain).
When a video filter returned inf as PTS, the player crashed. One
reason for this was that decode_audio() was called with a negative
minlen parameter, which at some point caused it to call a memory
allocation function with a ridiculous value, triggering an out of
memory code path in talloc.c. (talloc.c has been modified to abort()
on out of memory situations.)
Fix this by sanity checking minlen in decode_audio(). (The check
against outbuf->len always succeeded, because it's an unsigned
comparison.)
Make an existing sanity check in mplayer.c more robust: check for NaN
too, which happens if the video PTS is inf.
This happened with "-vf pullup,softpulldown" (but is not triggered when
the following commit is applied).
Most of these are reimar fixing issues found by Coverity static
analyzer, and possibly some more cleanup commits independent from
this.
Since these commits are rather noisy, squash them all together.
Try to make code a bit clearer.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35294 b3059339-0415-0410-9bf9-f77b7e298cf2
Conflicts:
audio/out/ao_alsa.c
Check the correct variable for NULL.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35323 b3059339-0415-0410-9bf9-f77b7e298cf2
Remove pointless unreachable code (the loop condition already checks
the 0xff case).
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35325 b3059339-0415-0410-9bf9-f77b7e298cf2
Fix typo that might have caused reading beyond the string end.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35326 b3059339-0415-0410-9bf9-f77b7e298cf2
Do not needlessly use "long" types.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35331 b3059339-0415-0410-9bf9-f77b7e298cf2
Use AV_RB32 to avoid sign extension issues and validate offset before using it.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35332 b3059339-0415-0410-9bf9-f77b7e298cf2
Remove nonsense casts.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35343 b3059339-0415-0410-9bf9-f77b7e298cf2
Fix crash in case sh_audio allocation failed.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35348 b3059339-0415-0410-9bf9-f77b7e298cf2
Fix potential NULL dereference.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35351 b3059339-0415-0410-9bf9-f77b7e298cf2
Conflicts:
libmpcodecs/ad_ffmpeg.c
Note: Slightly modified.
Fix malloc failure check to check the correct variable.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35353 b3059339-0415-0410-9bf9-f77b7e298cf2
Avoid code duplication and pointless casts.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35363 b3059339-0415-0410-9bf9-f77b7e298cf2
Conflicts:
stream/tv.c
Error out if an invalid channel list name was specified
instead of continuing and reading outside array bounds
all over the place.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35364 b3059339-0415-0410-9bf9-f77b7e298cf2
Conflicts:
stream/tv.c
Make array "static const".
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35365 b3059339-0415-0410-9bf9-f77b7e298cf2
Properly free resources even when encountering many
parse errors.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35367 b3059339-0415-0410-9bf9-f77b7e298cf2
Conflicts:
parser-cfg.c
Avoid leaks in error handling.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35380 b3059339-0415-0410-9bf9-f77b7e298cf2
Do not do sign comparisons on "char" type which can be both signed or unsigned.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35381 b3059339-0415-0410-9bf9-f77b7e298cf2
Free cookies file data after parsing it.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35382 b3059339-0415-0410-9bf9-f77b7e298cf2
http_set_field only makes a copy of the string, so we still need to
free it.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35383 b3059339-0415-0410-9bf9-f77b7e298cf2
check4proxies does not modify input URL, so mark it const.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35390 b3059339-0415-0410-9bf9-f77b7e298cf2
Remove proxy "support" from stream_rtp and stream_upd, trying
to use a http proxy for UDP connections makes no sense.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35394 b3059339-0415-0410-9bf9-f77b7e298cf2
Conflicts:
stream/stream_rtp.c
stream/stream_udp.c
Add url_new_with_proxy function to reduce code duplication and memleaks.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35395 b3059339-0415-0410-9bf9-f77b7e298cf2
Conflicts:
stream/pnm.c
stream/stream_live555.c
stream/stream_nemesi.c
stream/stream_rtsp.c
Fix off-by-one errors in file descriptor validity checks.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35402 b3059339-0415-0410-9bf9-f77b7e298cf2
Remove pointless cast.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35403 b3059339-0415-0410-9bf9-f77b7e298cf2
Abort when opening the file failed instead of calling
"write" with an invalid descriptor.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35404 b3059339-0415-0410-9bf9-f77b7e298cf2
Remove pointless local variable.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35411 b3059339-0415-0410-9bf9-f77b7e298cf2
Conflicts:
stream/http.c
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.
