The libavcodec Musepack SV8 decoder returned 2 bytes consumed for 1
byte input, which triggered a crash due to negative input packet size
later. Add a sanity check to prevent crashes with this type of minor
decoder overreads. Also add a check to parser consumed data.
Add support for using libavcodec decoders that do not have entries in
codecs.conf. This is currently only used with demux_lavf, and the
codec selection is based on codec_id returned by libavformat. Also
modify codec-related terminal output somewhat to make it use
information from libavcodec and avoid excessively long default output.
The new any-lavc-codec support is implemented with codecs.conf entries
that invoke vd_ffmpeg/ad_ffmpeg without directly specifying any
libavcodec codec name. In this mode, the decoders now instead select
the libavcodec codec based on codec_id previously set by demux_lavf
(if any). These new "generic" codecs.conf entries specify "status
buggy", so that they're tried after any specific entries with
higher-priority status.
Add new directive "anyinput" to codecs.conf syntax. This means the
entry will always match regardless of fourcc. This is used for the
above new codecs.conf entries (so the driver always gets to decide
whether to accept the input, and will fail init() if it can't find a
suitable codec in libavcodec). Remove parsing support for the obsolete
codecs.conf directive "cpuflags". This directive has not had any
effect and has not been used in default codecs.conf since many years
ago.
Shorten codec-related terminal output. When using libavcodec decoders,
show the libavcodec long_name field rather than codecs.conf "info"
field as the name of the codec. Stop showing the codecs.conf entry
name and "vfm/afm" name by default, as these are rarely needed;
they're now in verbose output only. Show "VIDEO:" line at VO
initialization rather than at demuxer open. This didn't really belong
in demuxer code; the new location may show more accurate values (known
after decoder has been opened) and works right if video track is
changed after initial demuxer open.
The vd.c changes (primarily done for terminal output changes) remove
round-to-even behavior from code setting dimensions based on aspect
ratio. I hope nothing depended on this; at least the even values were
not consistently guaranteed anyway, as the rounding code did not run
if the video file did not specify a nonzero aspect value.
Switch libavcodec audio decoding from avcodec_decode_audio3() to
avcodec_decode_audio4(). Instead of decoding directly to the output
buffer, the data is now copied from the libavcodec output packet,
adding an extra memory copy (optimizing this would require some
interface changes).
After libavcodec added avcodec_decode_audio4() earlier, it dropped
support for splitting large audio packets into output chunks of size
AVCODEC_MAX_AUDIO_FRAME_SIZE or less. This caused a regression with
the previous API: audio files with huge packets could fail to decode,
as libavcodec refused to write into the AVCODEC_MAX_AUDIO_FRAME_SIZE
buffer provided by mplayer2. This occurrend mainly with some lossless
audio formats. This commit restores support for those files; there are
now no fixed limits on packet size.
Change various code to use the latest Libav API. The libavcodec
error_recognition setting has been removed and replaced with different
semantics. I removed the "--lavdopts=er=<value>" option accordingly,
as I don't think it's widely enough used to be worth attempting to
emulate the old option semantics using the new API. A new option with
the new semantics can be added later if needed.
Libav dropped APIs that were necessary with all Libav versions
until quite recently (like setting avctx->age), and it would thus not
be possible to keep compatibility with previous Libav versions without
adding workarounds. The new APIs also had some bugs/limitations in the
recent Libav release 0.8, and it would not work fully (at least some
avcodec options would not be set correctly). Because of those issues,
this commit makes no attempt to maintain compatibility with anything
but the latest Libav git head. Hopefully the required fixes and
improvements will be included in a following Libav point release.
Update various code using Libav libraries to remove use of API
features that were deprecated at Libav release 0.7. I think this
removes them all with the exception of URLContext functions still used
in stream_ffmpeg.c (at least other uses that generated deprecation
warnings with libraries from 0.7 are removed).
Avoid calling avcodec_close() in uninit() if avcodec_open() failed.
Calling avcodec_close() on a non-open codec context causes a crash
with recent Libav versions.
ad_ffmpeg init() function did not free resources if opening failed.
Outside code (dec_audio.c) does not automatically call uninit() if
init() returns failure, and the uninit function would have crashed in
some cases had it been called (it did freed lavc_context->extradata,
but lavc_context could have been NULL after early init failure). Add
explicit calls to uninit() after failure and make uninit function safe
to call at any point.
At least the libavcodec WavPack decoder can return output for an audio
frame in multiple parts and return 0 bytes input consumed for the
initial parts. Timing info was not set correctly in this case:
sh_audio->pts and pts_bytes were reset each time when decoding more
from the packet, as if the packet had been new (ds_get_packet_pts()
has a check to return MP_NOPTS_VALUE if the packet has already been
partially read, but that didn't trigger since libavcodec returned
exactly 0 bytes read so the demuxer-visible packet state didn't
change).
Add a field to keep track of whether a packet has already been decoded
from, and don't reset timing info again if so. Adding the field
requires adding a decoder context to store it (there wasn't one
before).
BTW the WavPack decoder behavior and avcodec_decode_audio3()
documentation don't match - the documentation says the return value is
"zero if no frame data was decompressed (used) from the input
AVPacket", while the decoder DOES return some frame data which comes
from the input packet.
Do the global initialization of libavcodec and libavformat
(avcodec_register_all(), av_register_all()) immediately on program
startup and remove the initialization calls from various individual
modules that use libavcodec/libavformat functionality.
The init() method in ad_ffmpeg tries to decode some audio data after
opening the libavcodec decoder; however the method returned success
even if this part failed. Change it to return failure instead,
indicating that the codec could not be successfully opened.
This improves behavior at least with some AAC files, for which the
libavcodec decoder can be successfully initialized but decoding
packets always fails. Before the audio would be decoded with
libavcodec, producing only a constant stream of errors; after this
commit audio decoder initialization falls back to FAAD (if available)
which works for these samples.
Update various code to use newer alternatives instead of deprecated
functions/fields that are being dropped at libav API bump. An
exception is avcodec_thread_init() which is being dropped even though
it's still _necessary_ with fairly recent libav versions, so there's
no good alternative which would work with both those recent versions
and latest libavcodec. I think there are grounds to consider the drop
premature and revert it for now; if that doesn't happen I'll add a
version-test #if check around it later.
One of two alternative code parts passing codec extradata to
libavcodec didn't add the buffer padding that libavcodec requires,
resulting in invalid reads beoynd allocated memory area. Fix.
Use the value of the OutputSamplingFrequency element instead of the
SamplingFrequency element as the "container samplerate". In most cases
this only removes a warning, as those typically differ for SBR AAC
files and there was already a special case detecting this in
ad_ffmpeg.
The implementation adds a new "container_out_samplerate" field to the
sh_audio struct. Reusing the existing "samplerate" field and the
equivalent inside the 'wf' struct and just setting those to the new
value instead would probably work (at least I'm not aware of any codec
that would need the original SamplingFrequency for initialization).
However using a separate field also avoids some ugliness: the 'wf'
struct may not exist (though most demuxers create it), and the
'samplerate' field is overwritten to reflect the final value decided
by codec when decoding is first initialized.
Replace malloc+memset by calloc
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32181 b3059339-0415-0410-9bf9-f77b7e298cf2
Replace malloc+memset by calloc.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32182 b3059339-0415-0410-9bf9-f77b7e298cf2
Replace malloc+memset by calloc.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32183 b3059339-0415-0410-9bf9-f77b7e298cf2
Replace some sizeof(type) by sizeof(*pointer)
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32184 b3059339-0415-0410-9bf9-f77b7e298cf2
Replace malloc+memset by calloc.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32186 b3059339-0415-0410-9bf9-f77b7e298cf2
Replace malloc+memset by calloc.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32187 b3059339-0415-0410-9bf9-f77b7e298cf2
Replace malloc+memset by calloc
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32188 b3059339-0415-0410-9bf9-f77b7e298cf2
Replace sizoef(type) by sizeof(*ptrvar).
Besides being consistent with FFmpeg style,
this reduces the size of a patch to rename these
types to not conflict with the windows.h definitions.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32189 b3059339-0415-0410-9bf9-f77b7e298cf2
Replace malloc+memset by calloc.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32191 b3059339-0415-0410-9bf9-f77b7e298cf2
Replace malloc+memset by calloc.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32192 b3059339-0415-0410-9bf9-f77b7e298cf2
Replace sizeof(type) by sizeof(*ptrvar)
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32193 b3059339-0415-0410-9bf9-f77b7e298cf2
Remove a useless cast.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32194 b3059339-0415-0410-9bf9-f77b7e298cf2
Replace sizeof(type)
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32195 b3059339-0415-0410-9bf9-f77b7e298cf2
Remove a useless cast.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32196 b3059339-0415-0410-9bf9-f77b7e298cf2
Replace several sizeof(WAVEFORMATEX)
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32197 b3059339-0415-0410-9bf9-f77b7e298cf2
Replace one more instance of sizeof(WAVEFORMATEX); fix compilation.
patch by Clément Bœsch, ubitux gmail com
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32199 b3059339-0415-0410-9bf9-f77b7e298cf2
Avoid some pointless uses of sizeof() and one related cast.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32200 b3059339-0415-0410-9bf9-f77b7e298cf2
Merge one malloc() + memset() invocation into calloc().
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32202 b3059339-0415-0410-9bf9-f77b7e298cf2
Replace malloc+memset by calloc
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32203 b3059339-0415-0410-9bf9-f77b7e298cf2
Replace sizeof(WAVEFORMATEX) occurrences.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32205 b3059339-0415-0410-9bf9-f77b7e298cf2
Replace malloc+memset by calloc.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32206 b3059339-0415-0410-9bf9-f77b7e298cf2
Replace sizeof(BITMAPINFOHEADER)
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32207 b3059339-0415-0410-9bf9-f77b7e298cf2
Patch by Vlad Seryakov, vseryakov gmail com
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32043 b3059339-0415-0410-9bf9-f77b7e298cf2
Refactor more instances of avcodec_initialized handling into init_avcodec().
This is a leftover from the previous commit.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32044 b3059339-0415-0410-9bf9-f77b7e298cf2
Add missing #include for vd_ffmpeg.h; fixes the warning:
libmpcodecs/vf_zrmjpeg.c:472: warning: implicit declaration of function 'init_avcodec'
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32176 b3059339-0415-0410-9bf9-f77b7e298cf2
Remove pointless casts of avcodec_find_decoder_by_name() return value.
avcodec_find_decoder_by_name() already returns AVCodec*, so there is
no need to cast the return value to this type.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32007 b3059339-0415-0410-9bf9-f77b7e298cf2
Add support for parameter changes (e.g. channel count) during playback.
This makes decoding AC3 files that switch between 2 and 6 channels
work reasonably well even with -channels 6 and ffac3 decoder.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@31737 b3059339-0415-0410-9bf9-f77b7e298cf2
Fix typo in error message: ACC -> AAC
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32473 b3059339-0415-0410-9bf9-f77b7e298cf2
Avoid printing AAC with SBR warning on every decode call, instead print
it only after every decoder reconfiguration.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32476 b3059339-0415-0410-9bf9-f77b7e298cf2
Note that r30455 is wrong, that commit does not in fact change the
default behavior as claimed in the commit message. It only breaks
"-af-adv force=0", which was already pretty much useless though.
AC3 is still broken due to the libavcodec parser being broken.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@30421 b3059339-0415-0410-9bf9-f77b7e298cf2
a parser, not when just needs_parsing is set.
Fixes playback of e.g. ADPCM in AVI like http://samples.mplayerhq.hu/avi/imaadpcm.avi
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@30314 b3059339-0415-0410-9bf9-f77b7e298cf2
via libavcodec.
Parsing can be done at the demuxer stage (currently disabled) or at the decoder
(ad_ffmpeg, enabled).
Should allow using the libavcodec AAC, DTS, ... decoders independent of container
format.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@30130 b3059339-0415-0410-9bf9-f77b7e298cf2
Patch submitted by Nicolas George, nicolas.george normalesup org
The layout exceptions removed by this patch were rendered unnecessary by
changes in ffmpeg which normalize channel layout for aac (r20067) and vorbis
(r20148).
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@29821 b3059339-0415-0410-9bf9-f77b7e298cf2
As part of merging subtitle-in-terminal changes make
update_subtitles() only clear existing subtitles if called with the
reset argument, and not try to set new ones. Later calls should set
the needed new subtitles, and this change avoids some problems with
trying to set subtitles when mp_property_sub() in command.c gets
called from initialization code before full initialization.
currently requires that.
That probably is an unintended API change and should be fixed/reverted
in lavc but it hurts little to workaround here.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@29709 b3059339-0415-0410-9bf9-f77b7e298cf2
failing to init the decoder completely just because the first packet is broken.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@29553 b3059339-0415-0410-9bf9-f77b7e298cf2
ffdca, ffflac, ffaac, fftruehd). In the process, adds support for 32-bit
samples.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@29533 b3059339-0415-0410-9bf9-f77b7e298cf2
