Convert them to zigzag order, as the rest of them are.
When I was adding support for 10-bit DNxHD, I just copy-pasted the
missing quant matrices from the spec. Now it turns out the existing
matrices in dnxhddata.c were in zigzag order. This resulted in wrong
quantization for 10-bit DNxHD. The attached patch fixes the problem by
converting 10-bit quant matrices to zigzag order.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
It is only (mis-)used to set the dsp fucntions clear_block(s). But
these functions always work on 16bits-wide elements, which make
the parameter useless and actually harmful, as it causes all content
on more than 8-bits to not use accelerated functions.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
When forwarding the frame type information, by default x264 can
decide which kind of keyframe output, add an option to force it
to output IDR frames in to support use-cases such as preparing
the content for segmented streams formats.
x264 build 147 adds the native support for NV21.
Useful to avoid additional pixel format conversion when encoding
from a wide range of capture devices, Android among those.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Fixes deadlock with threads
Found-by: Paul B Mahol
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The variable is not a constant and can lead to race conditions
Fixes: repro.webm (not reproducable with FFmpeg alone)
Found-by: Dale Curtis <dalecurtis@google.com>
Tested-by: Dale Curtis <dalecurtis@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
These field are difficult to interpret, and are provided by a single
encoder (mpegvideoenc). In general they do not belong to a structure
containing raw data only, so remove them from AVFrame.
Mpegvideoenc now uses a private field in Picture for its internal
computations.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
Commits 43bc5cf9 and c5371f77 add code for skipping initial zeros in mp3
packets. This code forgot to report to the user that data was skipped at
all.
Since audio codecs allow partial packet decoding, the user application
has to rely on the return value. It will remove the data reported as
consumed by the decoder, and feed it to the decoder again. This resulted
in the mp3 frame after the zero region to be decoded over and over
again, until the zero region was finally skipped by the application.
Fix this by including the amount of skipped bytes to the number of
consumed bytes returned by the decode call.
Fixes trac ticket #4890.
CID 1256 is specified as using the same table for luma and chroma,
which is the same as CID 1235 luma table. This is consistent with
the format supposedly being RGB, although most sequences seem to
actually be YCbCr-encoded.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
VideoToolbox also implements a software decoder for h264, and will fallback to
using it if the file cannot be decoded on the GPU. In these cases though,
we want the hwaccel to fail so that we can use the libavcodec software decoder
instead of the Apple one.
Signed-off-by: wm4 <nfxjfg@googlemail.com>
* commit '4885bde3187a2bb0cae85b67796e07db233bf77f':
motion_est_template: Fix undefined left shift of negative number
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
* commit '948f3c19a8bd069768ca411212aaf8c1ed96b10d':
lavc: Make AVPacket.duration int64, and deprecate convergence_duration
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
This bit is 1 in some samples, and seems to coincide with interlaced
mbs and CID1260. 2008 specs do not know about it, and maintain qscale
is 11 bits. This looks oversized, but may help larger bitdepths.
Currently, it leads to an obviously incorrect qscale value, meaning
its syntax is shifted by 1. However, reading 11 bits also leads to
obviously incorrect decoding: qscale seems to be 10 bits.
However, as most profiles still have 11bits qscale, the feature is
restricted to the CID1260 profile (this flag is dependent on
a higher-level flag located in the header).
The encoder writes 12 bits of syntax, last and first bits always 0,
which is now somewhat inconsistent with the decoder, but ends up with
the same effect (progressive + reserved bit).
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
Note that convergence_duration had another meaning, one which was in
practice never used. The only real use for it was a 64 bit replacement
for the duration field. It's better just to make duration 64 bits, and
to get rid of it.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
* commit '83847cc8fa97e0fc637a0962bafb837acdb6eacc':
qsvenc: do not try to close the encoder if the session is NULL
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
File libopenh264enc.c has been modified so that the encoder uses av_log()
to log messages (error, warning, info, etc.) instead of logging them
directly to stderr. At the time the encoder is created, the current
libav log level is mapped to an equivalent libopenh264 log level. This
log level, and a message logging function that invokes av_log() to
actually log messages, are then set on the encoder.
This contains further changes and simplifications by Michael Niedermayer
and Martin Storsjö.
Signed-off-by: Martin Storsjö <martin@martin.st>
pix_fmt was declared presumably to shorten the argument passed to the function.
However, it is currently not being used for such a purpose.
This patch simply removes it instead.
This fixes -Wunused-but-set-variable reported at e.g:
http://fate.ffmpeg.org/log.cgi?time=20150919194249&log=compile&slot=x86_64-darwin-gcc-4.9.
Signed-off-by: Ganesh Ajjanagadde <gajjanagadde@gmail.com>
Reviewed-by: wm4 <nfxjfg@googlemail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
It appears vdpau drivers can return constrained baseline as unsupported,
even if libvdpau knows about the symbol, and the main profile is
supported.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This function can intrinsically not deal with codec profile fallback
(for H.264 Constrained Baseline especially), and was made redundant
by av_vdpau_bind_context().
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Each line is padded by the format, and errors are now reported and
stop the decoding. Around 5% speedup.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Default field order to top field first when interlaced frame is detected and pic_struct_present_flag is not set.
Since bottom field first comes from the old NTSC standard and is not used with HD anymore I think it´s straight forward to favor the majority.
Signed-off-by: Thomas Mundt <loudmax@yahoo.de>
Reviewed-by: Kieran Kunhya <kierank@obe.tv>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Around 3x speedup with 4 threads. Maybe more mb lines should be
batched per thread, but that's good enough for a first try.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Indicates a YCbCr->RGB transform at the block level. Although nothing
explicitly states it, this would assume the actual content is planar
RGB.
Currently unsupported, but the one sequence I found using it flagged
every mb that way, actually meaning the content was YCbCr, and thus
best left to the output format to decide what to do of it.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
It currently only applies to CID 1260, but this flag is dependent on
a higher-level flag located in the header.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* commit 'bbf71d46db3417b43bcbd745cbf235e8e2ff69ae':
dxv: Print texture information after header parsing
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
CID 1260 (as evidenced by incorrect decoding of a sample from ticket
4876) seems to use incorrect weight tables. It appears those tables
were not zigzag-scanned.
Apply zigzag on weight tables for new CIDs 1258, 1259, and 1260, and
fix an incorrect chroma table for CID 1256.
Fixes last issue from ticket #4876.
Found-by: Christophe Gisquet <christophe.gisquet@gmail.com>
Signed-off-by: Christophe Gisquet <christophe.gisquet@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This bit is 1 in some samples, and seems to coincide with interlaced
mbs and CID1260. 2008 specs do not know about it, and maintain qscale
is 11 bits. This looks oversized, but may help larger bitdepths.
Currently, it leads to an obviously incorrect qscale value, meaning
its syntax is shifted by 1. However, reading 11 bits also leads to
obviously incorrect decoding: qscale seems to be 10 bits.
However, as most profiles still have 11bits qscale, the feature is
restricted to the CID1260 profile.
The encoder writes 12 bits of syntax, last and first bits always 0,
which is now somewhat inconsistent with the decoder, but ends up with
the same effect (progressive + reserved bit).
Partially fixes ticket #4876.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This patch tweaks search_for_pns to be both more
aggressive and more careful when applying PNS. On
the one side, it will again try to use PNS on zero
(or effectively zero) bands. For this, both zeroes
and band_type have to be checked (some ZERO bands
aren't marked in zeroes). On the other side, a more
accurate rate-distortion measure avoids using PNS
where it would cause audible distortion.
Also fixed a small bug in the computation of freq
that caused PNS usage on low-frequency bands during
8-short windows. This allows re-enabling PNS during
8-short.
AVBufferRef.data and AVPacket.data don't need to have the same value.
AVPacket could point anywhere into the buffer. Likewise, the sizes
don't need to be the same.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
This works only for extradata sizes up to 128 bytes. Additionally, I
could never actually see it doing anything. The new code using
MMAL_BUFFER_HEADER_FLAG_CONFIG now takes care of this.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
We can send mp4-style data directly. But for some reason, this requires
sending the extradata as buffer with MMAL_BUFFER_HEADER_FLAG_CONFIG
set. Reuse the infrastructure for sending AVPackets to do this.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Commit 7404f3bdb switched bitrate to 64 bits.
This triggers -Wabsolute-value on clang, e.g
http://fate.ffmpeg.org/log.cgi?time=20150917122742&log=compile&slot=x86_64-darwin-clang-3.7-O3.
Therefore, usage of abs is changed to llabs, which is available on all of the platforms.
Unfortunately, LLONG_MAX is not always available, so INT64_MAX is used instead.
Signed-off-by: Ganesh Ajjanagadde <gajjanagadde@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This patch modifies the encode frame function to
retry encoding the frame when the resulting bit count
is too far off target, but only adjusting lambda
in small, incremental step. It also makes the logic
more conservative - otherwise it will contend with
bit reservoir-related variations in bit allocation,
and result in artifacts when frame have to be truncated
(usually at high bit rates transitioning from low
complexity to high complexity).
Reduces the number of times the vbv retry code is used and should have no
effect on quality
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
If cmd_pos is broken, this would just keep accumulating packets in the
reassembly buffer, until it fails and flushes the buffer on overflow.
Since packets are usually rather small, this will take a lot of subtitle
packets. The perceived effect is that subtitles are not displayed
anymore after the faulty packet was passed to the decoder.
I'm not terribly sure about this, but on the other hand this code is
active only when fragmented packets need to be reassembled.
Fixes sample file in trac issue #4872.
Assuming the first and second packets are partial, this would append the
reassembly buffer (ctx->buf) to itself with the second
append_to_cached_buf() call, because buf is set to ctx->buf.
I do not know a valid sample file which triggers this, and do not know
if packets can be split into more than 2 sub-packets, but it triggered
with a (differently) broken sample file in trac issue #4872.
With the move of some functions into templates
in aaccoder_twoloop.h and aaccoder_trellis.h,
make checkheaders started failing. Add them to
SKIPHEADERS as should be.
This resolves implementation defined behavior, and also silences -Wabsolute-value in clang 3.5+.
Moreover, the generated asm is identical to before modulo nop padding.
Signed-off-by: Ganesh Ajjanagadde <gajjanagadde@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* commit 'e3d4784eb31b3ea4a97f2d4c698a75fab9bf3d86':
d3d11va: WindowsPhone requires a mutex around ID3D11VideoContext
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
* commit '777885983533235ccda5145f96317fc8cd0a18ab':
dcadec: set channel layout in a separate function
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
* commit '971177f751a6e2931232accceab438bce277bde8':
dcadec: scan for extensions in a separate function
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
This patch refactors the AAC coders to reuse code
between the MIPS port and the regular, portable C code.
There were two main functions that had to use
hand-optimized versions of quantization code:
- search_for_quantizers_twoloop
- codebook_trellis_rate
Those two were split into their own template header
files so they can be inlined inside both the MIPS port
and the generic code. In each context, they'll link
to their specialized implementations, and thus be
optimized by the compiler.
This approach I believe is better than maintaining
several copies of each function. As past experience has
proven, having to keep those in sync was error prone.
In this way, they will remain in sync by default.
Also, an implementation of the dequantized output
argument for the optimized quantize_and_encode
functions is included in the patch. While the current
implementation of search_for_pred still isn't using
it, future iterations of main prediction probably will.
It should not imply any measurable performance hit while
not being used.
Failing earlier causes the context to be insufficiently initialized which
can break decoding future frames with threads
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* commit '95a41311ac3a44773cc4dc407408aca35b1f8e26':
jpeg2000: Factor out band stepsize initialization
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
* commit '41bcc3d15204f290400ba02e4e8f87fc07bcc00e':
jpeg2000: Split codeblock decoding from the main tile decoding
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
* commit '5d14cf199990cd378904a2618b5c72c4b02290f6':
mpegvideo: Make sure mpegutils.h is included where needed
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
* commit '525f58977c93e189fda49a5c4928feaf4d89fac6':
mpegvideo: Move macros to more appropriate headers
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
This fixes cases where the shifted number is 64, but we shifted non-
zero numbers away in the shift. The change makes behaviour consistent
with libvpx.
Code in aaccoder_mips.c was not synced with changes in aaccoder.c for
some time.
That was cause for some fate-aac tests failing.
This patch fixes the problems.
Optimizations disabled in 933309a are enabled again.
Signed-off-by: Nedeljko Babic <nedeljko.babic@imgtec.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
"int" is useful in testing because provides accurate results across
different plaftforms, so remove it from the scheduled FF_API_UNUSED_MEMBERS
deprecation.
Deprecate the now unused option, but temporarily retain the capability
to disable the now default behaviour.
Mention this change in the AVPacket documentation.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
This also drops setting the frame->pts field. This is usually not set by
decoders, so this would be an inconsistency that's at worst a danger to
the API user.
It appears the buffer->dts field is normally not set by the MMAL
decoder, so don't use it. If it's ever going to be set by MMAL, we
don't know whether the value will be what we want.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
The generic code in utils.c sets the AVFrame.pkt_dts field from the
packet it was supposedly decoded. This does not have to be true for a
fully asynchronous decoder like mmaldec. It could be overwritten with an
incorrect value. Even if the decoder doesn't determine the DTS (but sets
it to AV_NOPTS_VALUE), it's impossible to determine a correct value in
utils.c.
Decoders can now be marked with FF_CODEC_CAP_SETS_PKT_DTS, in which case
utils.c won't overwrite the field. The decoders are expected to set this
field (even if they only set it to AV_NOPTS_VALUE).
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
This MMAL feature fills in missing timestamps from the framerate set on
the input port. This is generally unwanted, since libavcodec decoders
merely pass through timestamps without ever "fixing" them. The framerate
is also unknown, and even the timebase doesn't have to be set.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Don't try to do a blocking wait for MMAL output if we haven't even sent
a single real packet, but only flush packets. Obviously we can't expect
to get anything back.
Additionally, don't send a flush packet to MMAL in the same case. It
appears the MMAL decoder will sometimes hang in mmal_vc_port_disable()
(called from ffmmal_close_decoder()), waiting for a reply from the GPU
which never arrives. Either MMAL disallows sending flush packets without
preceding real data, or it's a MMAL bug.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
I can't come up with a nice way to handle this. It's hard to keep the
lock-stepped input/output in this case. You can't predict whether the
MMAL decoder will output a picture (because it's asynchronous), so
you have to assume in general that any packet could produce 0 or 1
frames. You can't continue to write input packets to the decoder,
because then you might get too many output frames, which you can't
get rid of because the lavc decoding API does not allow the decoder
to return an output frame without consuming an input frame (except
when flushing).
The ideal fix is a M:N decoding API (preferably asynchronous), which
would make this code potentially much cleaner. For now, this hack
will do.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Since TNS was fixed with the recent commits retweak the values
so it's more frequently used.
Still not enabled by default yet, though it's possible that it
will be made enabled by default in the near future.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This commit was made possible with the earlier commits since the
new quantization method basically means we're working always with
unsigned values. The specifications mention to use compression when
the first 2 bits are identical but they didn't mention if this should
happen before or after the conversion to signed values. Actually
they said nothing about conversion to signed values.
With this commit, coefficient compression usually always happens
which saves a lot of space, especially at extremely low bitrates
and doesn't change the quality at all.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This finally (and again) gets rid of basically everything the
specifications say about how TNS should be done. The main
problem used to be that a single filter was used for all
coefficients which despite being explicitly recommended by
the specifications usually sounds wrong, therefore it's
a corner case in the current TNS implementation.
This commit also changes the coefficient bit size, as apparently
it's better to use lower precision in case the windows are eight
short. This is apparently what fdk_aac uses, looking at the bit
stream and makes sense. Also the order when 8 SHORT windows happen
is important as 7 was too much and according to PSNR was worse
while 5 is just about correct.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
