If the PVQ search picked a place to increment/decrement on the y[]
vector which had no pulse then it would cause a desync since it would
change the sum in the wrong direction. Fix this by not considering
places without pulses as viable.
This makes the PVQ search slightly worse at K < 5 which isn't all that
common. Still, this is a workaround to prevent making broken files until
I can think of a better way of fixing it.
Also add an assertion, which can be removed or moved to assert1/2 once
the PVQ search is stable.
Fixes: runtime error: shift exponent 132 is too large for 32-bit type 'int' Fixes: 609/clusterfuzz-testcase-4825202619842560
See 11.2.2 IHDR Image header
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/targets/ffmpeg Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
avformat/utils: Also fill dts==RELATIVE_TS_BASE packets in update_initial_durations()
This dts value can end up in the list in the absence of durations and is in that
case semantically identical to AV_NOPTS_VALUE. We can alternatively prevent
storing RELATIVE_TS_BASE if there is no duration.
Fixes Ticket3640
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Steven Liu [Sat, 18 Feb 2017 01:42:51 +0000 (09:42 +0800)]
avformat/hlsenc: set default http method to PUT when method is null
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>
Felicia [Mon, 6 Feb 2017 23:49:36 +0000 (15:49 -0800)]
libopus: decode ambisonics with non-diegetic sources
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>
Thomas Stephens [Tue, 7 Feb 2017 20:20:32 +0000 (14:20 -0600)]
avformat/dashenc: Only use temporary files when outputting to file protocol
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>
Joel Cunningham [Mon, 30 Jan 2017 16:00:44 +0000 (10:00 -0600)]
HTTP: improve performance by reducing forward seeks
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.
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).
opus_celt: rename structures to better names and reorganize them
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).
opus_celt: move quantization and band decoding to opus_pvq.c
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.
avformat/http: Check for truncated buffers in http_connect()
Reported-by: SleepProgger <security@gnutp.com> Reviewed-by: Steven Liu <lingjiujianke@gmail.com> Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Alex Converse [Thu, 9 Feb 2017 16:58:47 +0000 (08:58 -0800)]
aac_latm: Align inband PCE to the start of the payload
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.
wm4 [Fri, 10 Feb 2017 11:17:24 +0000 (12:17 +0100)]
hwcontext_dxva2: support D3D9Ex
D3D9Ex uses different driver paths. This helps with "headless"
configurations when no user logs in. Plain D3D9 device creation will
fail if no user is logged in, while it works with D3D9Ex.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Merges Libav commit c2f97f0508708.
wm4 [Thu, 2 Feb 2017 10:27:54 +0000 (11:27 +0100)]
AVFrame: add an opaque_ref field
This is an extended version of the AVFrame.opaque field, which can be
used to attach arbitrary user information to an AVFrame.
The usefulness of the opaque field is rather limited, because it can
store only up to 32 bits of information (or 64 bit on 64 bit systems).
It's not possible to set this field to a memory allocation, because
there is no way to deallocate it correctly.
The opaque_ref field circumvents this by letting the user set an
AVBuffer, which makes the user data refcounted.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Merges Libav commit 04f3bd349651.
Jan Ekström [Fri, 10 Feb 2017 23:21:14 +0000 (01:21 +0200)]
movenc: add support for track names in ISML manifests
This enables having multiple tracks of the same type which would
be treated as different things by the media server (as opposed to
different bit rate versions of the same track). According to the
smooth streaming specification, just setting the systemLanguage
tag is not enough to note that a track with the same attributes
differs from another one.
Reviewed-by: Martin Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Steven Liu [Sat, 11 Feb 2017 04:32:31 +0000 (12:32 +0800)]
avformat/hlsenc: deprecate hls_wrap option
When user use the hls_wrap, there have many problem:
1. some platform refersh the old but usefull segment
2. CDN(Content Delivery Network) Deliver HLS not friendly
The hls_wrap is used to wrap segments for use little space,
now user can use hls_list_size and hls_flags delete_segments
instead it.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc> Reviewed-by: Carl Eugen Hoyos <ceffmpeg@gmail.com> Signed-off-by: Steven Liu <lq@chinaffmpeg.org>