Handle all pending events and exit instead of waiting. When there are lots of
input events (for example, scrolling with trackpad), timeout can add up
to make a huge frame delay. In my tests, if I scroll fast enough, that loop
would never exit.
This function sucks and apparently is not very portable (at least on
mingw, the configure check fails). Also remove the emulation of that
function from osdep/strsep*, and remove the configure check.
mixer_setvolume() accepts float values for volume, but used the
integer function av_clip() to limit range, losing the fractional part
as a side effect. Change the code to use av_clipf() instead. For most
uses this shouldn't make any real difference; actual AO volume
settings may not have that much precision anyway.
af_volnorm can process either int16_t or float audio data. The float
version used 0 to INT_MAX as full value range, when it should be 0 to
1. This effectively disabled the filter (due to all input being
considered to fall in the silence range). Fix.
Reported by Tobias Jacobi <liquid.acid@gmx.net>.
This causes trouble when a hw device is used:
pcm_hw.c:514:(snd_pcm_hw_delay) SNDRV_PCM_IOCTL_DELAY failed (-77): File descriptor in bad state
when running mpv test.mkv --ao=alsa:device=iec958,alsa and pausing
during playback.
Historically, mplayer usually did not call snd_pcm_delay() (which is
called by get_delay()) while paused, so this problem never showed up.
But at least mpv has changes that cause get_delay() to be called when
updating the status line (see commit 3f949cf).
It's possible that calling snd_pcm_delay() is not always legal when the
audio is paused, and at least fails with the error message mentioned
above is the device is a hardware device. Change get_delay() to return
the last delay before the audio was paused. The intention is to get a
continuous playback status display, even when pausing or frame stepping,
otherwise we could just return the audio buffer fill status in
get_delay() or even just 0 when paused.
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.
This mainly serves as a fallback for platforms where nothing better is
available; also as a debugging help. Both the audio and video driver are
not first class - the audio driver lacks delay detection, and the video
driver only supports a single YUV color space.
Configure options: --disable-sdl2 to disable SDL 2.0+ detection,
--disable-sdl to disable SDL 1.2+ detection. Both options need to be
specified to turn off SDL support entirely.
Add `mp_find_config_file` to search different known paths and use that in
ass_mp to look for the fontconfig configuration file.
Some incidental changes spawned by this feature where:
* Buffer allocation for the strings containing the paths is now performed
with talloc. All of the allocations are done on a NULL context, but it still
improves readability of the code.
* Move the OSX function for lookup inside of a bundle: this code path was
currently not used by the bundle generated with `make osxbundle`. The plan
is to use it again in a future commit to get a fontconfig config file.
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).
The option is -no-video. Remove the deprecated "fast" suboption, which
did nothing and instructed the user to use "-novideo" instead.
Fix a reference to -novideo in encoding.rst.
Add a "generic" entry about -no-* to the list of renamed options. The
change is already explicitly mentioned in the text above the table, but
even if it's redundant, it makes it harder to overlook.
This fixes operation with current ffmpeg releases.
Note that this planarization is slow and should be reverted once proper
planar audio support is there in mpv.
PulseAudio allows applications to set volume over 100%. To make this
possible, the PulseAudio daemon raises the global system volume, and
tries to lower other applications volumes. Unfortunately, this doesn't
work out and doesn't manage to keep the effective volume level of these
other applications.
To make it short: this functionality invoked PulseAudio bugs. Disable
it.
This essentially reverts commit 85a64b.
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
Libav 0.8.4 is ridiculously old (in relative terms), so I don't know
how many things are broken silently.
Encoding is disabled, because the required API hasn't been added yet.
(On the other hand, the old API can't be used in newer versions.)
This should improve compatibility with ffmpeg 0.11.2 as well, which
didn't define AV_CODEC_ID_SUBRIP yet.
Lowering volume while muted did not work correctly with audio outputs
that support native mute setting separate from volume (ao_alsa and
ao_pulse), because the AO-level volume was not set while muted but was
still being read back. Fix by setting the AO volume in this case.
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.
Mad Fish
wm4
Uoti Urpala
Rudolf Polzer
Stefano Pigozzi
reimar