Autodetected by default. Encode using -codec:v h264_videotoolbox.
Signed-off-by: Rick Kern <kernrj@gmail.com>
Signed-off-by: wm4 <nfxjfg@googlemail.com>
* commit 'f9fbd474676e903e12efe83203697d60a9d28cf9':
msmpeg4data: Move WMV2 data tables to their own file
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
This commit adds a new encoder capable of creating BBC/SMPTE Dirac/VC-2 HQ
profile files.
Dirac is a wavelet based codec created by the BBC a little more than 10
years ago. Since then, wavelets have mostly gone out of style as they
did not provide adequate encoding gains at lower bitrates. Dirac was a
fully featured video codec equipped with perceptual masking, support for
most popular pixel formats, interlacing, overlapped-block motion
compensation, and other features. It found new life after being stripped
of various features and standardized as the VC-2 codec by the SMPTE with
an extra profile, the HQ profile that this encoder supports, added.
The HQ profile was based off of the Low-Delay profile previously
existing in Dirac. The profile forbids DC prediction and arithmetic
coding to focus on high performance and low delay at higher bitrates.
The standard bitrates for this profile vary but generally 1:4
compression is expected (~525 Mbps vs the 2200 Mbps for uncompressed
1080p50). The codec only supports I-frames, hence the high bitrates.
The structure of this encoder is simple: do a DWT transform on the
entire image, split it into multiple slices (specified by the user) and
encode them in parallel. All of the slices are of the same size, making
rate control and threading very trivial. Although only in C, this encoder
is capable of 30 frames per second on an 4 core 8 threads Ivy Bridge.
A lookup table is used to encode most of the coefficients.
No code was used from the GSoC encoder from 2007 except for the 2
transform functions in diracenc_transforms.c. All other code was written
from scratch.
This encoder outperforms any other encoders in quality, usability and in
features. Other existing implementations do not support 4 level
transforms or 64x64 blocks (slices), which greatly increase compression.
As previously said, the codec is meant for broadcasting, hence support
for non-broadcasting image widths, heights, bit depths, aspect ratios,
etc. are limited by the "level". Although this codec supports a few
chroma subsamplings (420, 422, 444), signalling those is generally
outside the specifications of the level used (3) and the reference
decoder will outright refuse to read any image with such a flag
signalled (it only supports 1920x1080 yuv422p10). However, most
implementations will happily read files with alternate dimensions,
framerates and formats signalled.
Therefore, in order to encode files other than 1080p50 yuv422p10le, you
need to provide an "-strict -2" argument to the command line. The FFmpeg
decoder will happily read any files made with non-standard parameters,
dimensions and subsamplings, and so will other implementations. IMO this
should be "-strict -1", but I'll leave that up for discussion.
There are still plenty of stuff to implement, for instance 5 more
wavelet transforms are still in the specs and supported by the decoder.
The encoder can be lossless, given a high enough bitrate.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Decodes YUV 4:2:2 10-bit and RGB 12-bit files.
Older files with more subbands, skips, Bayer, alpha not supported.
Alpha requires addition of GBRAP12 pixel format.
It serves absolutely no purpose other than to confuse potentional
Android developers about how to use hardware acceleration properly
on the the platform. The stagefright "API" is not public, and the
MediaCodec API is the proper way to do this.
Furthermore, stagefright support in avcodec needs a series of
magic incantations and version-specific stuff, such that
using it actually provides downsides compared just using the actual
Android frameworks properly, in that it is a lot more work and confusion
to get it even running. It also leads to a lot of misinformation, like
these sorts of comments (in [1]) that are absolutely incorrect.
[1] http://stackoverflow.com/a/29362353/3115956
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Commit b272c3a5aa has sped up dsd_tablegen, and now table generation takes
~ 40k cycles. Thus, these tables can always be generated at runtime.
Tested with/without --enable-hardcoded-tables.
Signed-off-by: Ganesh Ajjanagadde <gajjanagadde@gmail.com>
* commit 'e02de9df4b218bd6e1e927b67fd4075741545688':
lavc: export Dirac parsing API used by the ogg demuxer as public
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
This gets rid of virtually useless hardcoded tables hackery. The reason
it is useless is that a 320 element lut is anyway placed regardless of
--enable-hardcoded-tables, from which all necessary tables are trivially
derived at runtime at very low cost:
sample benchmark (x86-64, Haswell, GNU/Linux, single run is really
what is relevant here since looping drastically changes the bench). Fluctuations
are on the order of 10% for the single run test:
39400 decicycles in aacsbr_tableinit, 1 runs, 0 skips
25325 decicycles in aacsbr_tableinit, 2 runs, 0 skips
18475 decicycles in aacsbr_tableinit, 4 runs, 0 skips
15008 decicycles in aacsbr_tableinit, 8 runs, 0 skips
13016 decicycles in aacsbr_tableinit, 16 runs, 0 skips
12005 decicycles in aacsbr_tableinit, 32 runs, 0 skips
11546 decicycles in aacsbr_tableinit, 64 runs, 0 skips
11506 decicycles in aacsbr_tableinit, 128 runs, 0 skips
11500 decicycles in aacsbr_tableinit, 256 runs, 0 skips
11183 decicycles in aacsbr_tableinit, 509 runs, 3 skips
Tested with FATE with/without --enable-hardcoded-tables.
Reviewed-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Signed-off-by: Ganesh Ajjanagadde <gajjanagadde@gmail.com>
* commit 'f023d57d355ff3b917f1aad9b03db5c293ec4244':
lavc: G.723.1 encoder
Split existing FFmpeg G.723.1 encoder into a new file.
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
* commit '165cc6fb9defcd79fd71c08167f3e8df26b058ff':
g723_1: Move sharable functions to a separate file
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
* commit 'aac996cc01042194bf621d845bbe684549b5882e':
g723_1: Rename files to better reflect their purpose
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
The commit 932ff70 introducing this header mentions it should be public.
Reviewed-by: Ronald S. Bultje <rsbultje@gmail.com>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Register mmaldec as mpeg2 decoder. Supporting mpeg2 in mmaldec is just a
matter of setting the correct MMAL_ENCODING on the input port. To ease the
addition of further supported mmal codecs a macro is introduced to generate
the decoder and decoder class structs.
Signed-off-by: Julian Scheel <julian@jusst.de>
Signed-off-by: wm4 <nfxjfg@googlemail.com>
Long Term Prediction allows for prediction of spectral coefficients
via the previously decoded time-dependent samples. This feature
works well with harmonic content 2 or more frames long, like speech,
human or non-human, piano music or any constant tones at very low
bitrates.
It should be noted that the current coder is highly efficient and
the rate control system is unable to encode files at extremely
low bitrates (less than 14kbps seems to be impossible) so this
extension isn't capable of optimum operation. Dramatic difference
is observable with some types of audio and speech but for the most
part the audiable differences are subtle. The spectrum looks better
however so the encoder is able to harvest the additional bits that
this feature provies, should the user choose to enable it. So
it's best to enable this feature only if encoding at the absolutely
lowest bitrate that the encoder is capable of.
The bulk of calls to quantize_band_cost are replaced
by a call to a version that memoizes, greatly improving
performance, since during coefficient search there is
a great deal of repeat work.
Memoization cannot always be applied, so do this in a
different function, and leave the original as-is.
It was merged with the iff_ilbm decoder in commit
929a24efff.
Define AV_CODEC_ID_IFF_BYTERUN1 as AV_CODEC_ID_IFF_ILBM for API
compatibility.
Reviewed-by: Ronald S. Bultje <rsbultje@gmail.com>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
* commit 'bb198c4997d5036f3bf91de51e44f807115677d0':
d3d11va: make av_d3d11va_alloc_context() available at all times
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
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.
* commit 'e3d4784eb31b3ea4a97f2d4c698a75fab9bf3d86':
d3d11va: WindowsPhone requires a mutex around ID3D11VideoContext
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
This commit finalizes AAC-Main profile encoding support
by implementing all mandatory and optional tools available
in the specifications and current decoders.
The AAC-Main profile reqires that prediction support be
present (although decoders don't require it to be enabled)
for an encoder to be deemed capable of AAC-Main encoding,
as well as TNS, PNS and IS, all of which were implemented
with previous commits or earlier of this year.
Users are encouraged to test the new functionality using either
-profile:a aac_main or -aac_pred 1, the former of which will enable
the prediction option by default and the latter will change the
profile to AAC-Main. No other options shall be changed by enabling
either, it's currently up to the users to decide what's best.
The current implementation works best using M/S and/or IS,
so users are also welcome to enable both options and any
other options (TNS, PNS) for maximum quality.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This commit implements temporal noise shaping support in the
encoder, along with an -aac_tns option to toggle it on or off
(off by default for now). TNS will increase audio quality
and reduce quantization noise by applying a multitap FIR filter
across allowed coefficients and transmit side information to the
decoder so it could create an inverse filter.
Users are encouraged to test the new functionality by enabling
-aac_tns 1 during encoding.
No major bugs are observable at this time so after a while if no
new problems appear and if the current implementation is deemed
of high enough quality and stability it will be enabled by default,
possibly at the same time the encoder has its experimental flag
removed and becomes the standard aac encoder in ffmpeg.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
