If there is progressive input it will disable deinterlacing in cuvid for
all future frames even those interlaced.
Signed-off-by: Timo Rothenpieler <timo@rothenpieler.org>
Fixes Ticket 6018
This fixes a regression, and allows playback of files containing mpeg4video that are otherwise
not supported
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
When the http method is not set, the method will use POST for ts,
PUT for m3u8, it is not unify, now set it unify.
This ticket id: #5315
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Reviewed-by: Moritz Barsnick <barsnick@gmx.net>
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
Channel mapping 2 additionally supports a non-diegetic stereo track
appended to the end of a full-order ambisonics signal, such that the
total channel count is either
(n + 1) ^ 2, or
(n + 1) ^ 2 + 2
where n is the ambisonics order
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Skips using temporary files when outputting to a protocol other than
"file", which enables dash to output content over network
protocols. The logic has been copied from the HLS format.
Reviewed-by: Steven Liu <lingjiujianke@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This commit optimizes HTTP performance by reducing forward seeks, instead
favoring a read-ahead and discard on the current connection (referred to
as a short seek) for seeks that are within a TCP window's worth of data.
This improves performance because with TCP flow control, a window's worth
of data will be in the local socket buffer already or in-flight from the
sender once congestion control on the sender is fully utilizing the window.
Note: this approach doesn't attempt to differentiate from a newly opened
connection which may not be fully utilizing the window due to congestion
control vs one that is. The receiver can't get at this information, so we
assume worst case; that full window is in use (we did advertise it after all)
and that data could be in-flight
The previous behavior of closing the connection, then opening a new
with a new HTTP range value results in a massive amounts of discarded
and re-sent data when large TCP windows are used. This has been observed
on MacOS/iOS which starts with an initial window of 256KB and grows up to
1MB depending on the bandwidth-product delay.
When seeking within a window's worth of data and we close the connection,
then open a new one within the same window's worth of data, we discard
from the current offset till the end of the window. Then on the new
connection the server ends up re-sending the previous data from new
offset till the end of old window.
Example (assumes full window utilization):
TCP window size: 64KB
Position: 32KB
Forward seek position: 40KB
* (Next window)
32KB |--------------| 96KB |---------------| 160KB
*
40KB |---------------| 104KB
Re-sent amount: 96KB - 40KB = 56KB
For a real world test example, I have MP4 file of ~25MB, which ffplay
only reads ~16MB and performs 177 seeks. With current ffmpeg, this results
in 177 HTTP GETs and ~73MB worth of TCP data communication. With this
patch, ffmpeg issues 4 HTTP GETs and 3 seeks for a total of ~22MB of TCP data
communication.
To support this feature, the short seek logic in avio_seek() has been
extended to call a function to get the short seek threshold value. This
callback has been plumbed to the URLProtocol structure, which now has
infrastructure in HTTP and TCP to get the underlying receiver window size
via SO_RCVBUF. If the underlying URL and protocol don't support returning
a short seek threshold, the default s->short_seek_threshold is used
This feature has been tested on Windows 7 and MacOS/iOS. Windows support
is slightly complicated by the fact that when TCP window auto-tuning is
enabled, SO_RCVBUF doesn't report the real window size, but it does if
SO_RCVBUF was manually set (disabling auto-tuning). So we can only use
this optimization on Windows in the later case
Signed-off-by: Joel Cunningham <joel.cunningham@me.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This marks the first time anyone has written an Opus encoder without
using any libopus code. The aim of the encoder is to prove how far
the format can go by writing the craziest encoder for it.
Right now the encoder's basic, it only supports CBR encoding, however
internally every single feature the CELT layer has is implemented
(except the pitch pre-filter which needs to work well with the rest of
whatever gets implemented). Psychoacoustic and rate control systems are
under development.
The encoder takes in frames of 120 samples and depending on the value of
opus_delay the plan is to use the extra buffered frames as lookahead.
Right now the encoder will pick the nearest largest legal frame size and
won't use the lookahead, but that'll change once there's a
psychoacoustic system.
Even though its a pretty basic encoder its already outperforming
any other native encoder FFmpeg has by a huge amount.
The PVQ search algorithm is faster and more accurate than libopus's
algorithm so the encoder's performance is close to that of libopus
at zero complexity (libopus has more SIMD).
The algorithm might be ported to libopus or other codecs using PVQ in
the future.
The encoder still has a few minor bugs, like desyncs at ultra low
bitrates (below 9kbps with 20ms frames).
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This is meant to be applied on top of my previous patch which
split PVQ into celt_pvq.c and made opus_celt.h
Essentially nothing has been changed other than renaming CeltFrame
to CeltBlock (CeltFrame had absolutely nothing at all to do with
a frame) and CeltContext to CeltFrame.
3 variables have been put in CeltFrame as they make more sense
there rather than being passed around as arguments.
The coefficients have been moved to the CeltBlock structure
(why the hell were they in CeltContext and not in CeltFrame??).
Now the encoder would be able to use the exact context the decoder
uses (plus a couple of extra fields in there).
FATE passes, no slowdowns, etc.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
A huge amount can be reused by the encoder, as the only thing
which needs to be done would be to add a 10 line celt_icwrsi,
a wrapper around it (celt_alg_quant) and templating the
ff_celt_decode_band to replace entropy decoding functions
with entropy encoding.
There is no performance loss but in fact a performance gain of
around 6% which is caused by the compiler being able to optimize
the decoding more efficiently.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Handles strides (needed for Opus transients), does pre-reindexing and folding
without needing a copy.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Mostly used the RFC document, the decoding functions and
the reference encoder's implmenentation as a reference.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Reported-by: SleepProgger <security@gnutp.com>
Reviewed-by: Steven Liu <lingjiujianke@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
A strict reading of the spec seems to imply that it should be aligned to
the start of the element instance tag, but that would break all of the
samples with PCEs.
It seems like a well formed LATM stream should have its PCE in the ASC
rather than inband.
Fixes ticket 4544
Miroslav Slugeň
Michael Niedermayer
Steven Liu
Rostislav Pehlivanov
Thierry Foucu
Felicia
Paul B Mahol
Carl Eugen Hoyos
Andreas Cadhalpun
Mark Thompson
Mulvya
Pavel Koshevoy
Matthieu Bouron
James Almer
Thomas Stephens
Joel Cunningham
Timo Rothenpieler
Lou Logan
Alex Converse