Add a CHECK_ALSA_ERROR macro to report ALSA errors. This is similar to
what vo_vdpau does. This removes lots of boiler plate, it almost gives
me the feeling the ao_alsa initialization code is now readable. This
change is squashed with the reformatting, because both changes are
just as noisy and useless.
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).
Like most other AOs, ao_pulse set the channel count only, always using a
default layout. Try to set the exact layout.
For this, we need a big lookup table to map waveex/lavc/mpv speaker
position to PulseAudio's, since PA_CHANNEL_POSITION_ is apparently not
compatible to waveext, and I haven't seen any API functions that would
help mapping them.
Completely untested. (Let's leave that to someone else...)
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.
Schedule mpv's playloop as a high frequency timer inside the main Cocoa event
loop. This has the benefit to allow accessing menus as well as resizing the
window without the playback being blocked and allows to remove countless hacks
from the code that involved manually pumping the event loop as well simulating
manually some of the Cocoa default behaviours.
A huge improvement consists in removing NSApplicationLoad. This is a C function
defined in the Cocoa header and implements a minimal OSX application under ther
hood so that you can use the Cocoa GUI toolkit from C/C++ without having to
respect the Cocoa standards in terms of application initialization. This was
bad because the behaviour implemented by NSApplicationLoad was hard to customize
and had several gotchas especially in the menu department.
mpv was changed to be just a nib-less application. All the Cocoa part is still
generated in code but the event handling is now not dissimilar to what is
present in a stock Mac application.
As a part of reviewing the initialization process, I also removed all of
`osdep/macosx_finder_args`. The useful parts of the code were moved to
`osdep/macosx_appication` which has the broaded responsibility of managing the
full lifecycle of the Cocoa application. By consequence the
`--enable-macosx-finder` configure switch was killed as well, as this feature
is always enabled.
Another change the users will notice is that when using a bundle the `--quiet`
option will be inserted much earlier in the initializaion process. This results
in mpv not spamming mpv.log anymore with all the initialization outputs.
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.
All this option did was deciding whether the resample filter was to be
insert at the beginning or end of the filter chain. Always do what the
option set for accuracy did. I doubt it makes much of a difference.
libavresample does most things in just one go anyway, so it won't
matter.
Dangerous and misleading. If it turns out that this is actually needed
to make certain setups work right, it should be added back in a better
way (in a way it doesn't cause random crashes).
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.
Switch the internal channel order to libavcodec's. If the channel number
mismatches at some point, use libavresample for up- or downmixing.
Remove the old af_pan automatic downmixing.
The libavcodec channel order should be equivalent to WAVEFORMATEX order,
at least nowadays. reorder_ch.h assumes that WAVEFORMATEX and libavcodec
might be different, but all defined channels have the same mappings.
Remove the downmixing with af_pan as well as the channel conversion with
af_channels from af.c, and prefer af_lavrresample for this. The
automatic downmixing behavior should be the same as before (if the
--channels option is set to 2, which is the default, the audio output
is forced to 2 channels, and libavresample does all downmixing).
Note that mpv still can't do channel layouts. It will pick the default
channel layout according to the channel count. This will be fixed later
by passing down the channel layout as well.
af_hrtf depends on the order of the input channels, so reorder to ALSA
(for which this code was written). This is better than changing the
filter code, which is more risky.
ao_pulse can accept waveext order directly, so set that as channel
mapping.
If format negotiation fails, and additional filters are inserted to fix
this, don't try to reinitialize the filter immediately. Instead, correct
the audio format, and let the caller retry.
Add a retry counter to af_reinit() to ensure that misbehaving filters
can't put the format negotiation into an endless loop.
Refactor to remove the duplicated format filter insertion code. Allow
other format converting filters to be inserted on format mismatches.
af_info.test_conversion checks whether conversion between two formats
would work with the given filter; do this to avoid having to insert
multiple conversion filters at once and such things. (Although this
isn't ideal: what if we want to avoid af_format for some conversions?
What if we want to split af_format in endian-swapping filters etc.?)
Prefer af_lavrresample for conversions that it supports natively,
otherwise let af_format handle the full conversion.
Make sure automatically inserted filters are removed on full reinit
(they are re-added later if they are really needed). Automatically
inserted filters were never explicitly removed, instead, it was
expected that redundant conversion filters detach themselves. This
didn't work if there were several chained format conversion filters,
e.g. s16le->floatle->s16le, which could result from repeated filter
insertion and removal. (format filters detach only if input format and
output format are the same.)
Further, the dummy filter (which exists only because af.c can't handle
an empty filter chain for some reason) could introduce bad conversions
due to how the format negotiation works. Change the code so that the
dummy filter never takes part on format negotiation. (It would be better
to fix format negotiation, but that would be much more complicated and
would involving fixing all filters.)
Simplify af_reinit() and remove the start audio filter parameter. This
means format negotiation and filter initialization is run more often,
but should be harmless.
The format was locked to s16. Extend it to accept all other ffmpeg
sample formats, and even allow different in- and output formats. The
generic filter code will still insert af_format on format mismatches,
though.
The change in af_scaletempo actually fixes a memory leak. af->data
contained a pointer to an allocated buffer, which was overwritten
during format negotiation. Set the format explicitly instead.
Consider:
mpv --volume 10 file1.mkv file2.mkv
Before this commit, the volume was reset to 10 when playing file2.mkv.
This was inconsistent to most other options. E.g. --brightness is a
rather similar case.
In general, settings should never be reset when playing the next file,
unless the option was explicitly marked file-local. This commit
corrects the behavior of the --volume and --mute options.
File local --volume still works as expected:
mpv --{ --volume 10 file1.mkv file2.mkv --}
This sets the volume always to 10 on playback start.
Move the m_config_leave_file_local() call down so that the mixer code
in uninit_player() can set the option volume and mute variables without
overwriting the global option values.
Another subtle issue is that we don't want to set volume if there's no
need to, which is why the user_set_volume/mute fields are introduced.
This is important because setting the volume might change the system
volume depending on other options.
Remove `af_resample` and `af_lavcresample`. The former is a mess while the
latter uses an API that was long deprecated in libavcodec and is now removed.
`af_lavrresample` rougly has the same features and structure of
`af_lavcresample`.
libswresample fallback by wm4.
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.
avcodec_encode_audio() was deprecated, and was finally removed from
Libav and FFmpeg git.
This appears to work. I get heavy A/V desync with -ao alsa and -ao pcm,
but this was already so before this change.
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.
The previous name of this filter was misleading, because it doesn’t actually
normalize volume levels. What it does is closer to performing low-quality
dynamic range compression, hence it is now called af_drc.
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.
mpv -ao help and mpv -vo help shouldn't show the encoding outputs (named
"lavc" on both cases). Also make it impossible to select these manually
when not encoding.
On Linux, ao_portaudio has weird freezing issues (possibly specific to
the ALSA backend, though). Also ao_dsound is more likely to get multi-
channel audio output right, and ao_portaudio probably mangles these.
This partially reverts earlier decisions, when I thought it would
always be better to prefer the audio volume filter over the AO's,
because the AO's relies on the underlying audio-API, which could
be broken or exhibit unusual behavior (like it happened with ao_dsound).
However, since the audio buffer can be quite large (500 ms), and we
don't attempt to flush & refilter the audio on volume changes, always
prefer AO volume control (as long as the AO mixer doesn't control the
system mixer).
Also document what the mixer.c related AO fields mean (hopefully not
too brief).
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.
