| Commit message (Collapse) | Author | Age | Files | Lines |
|---|
| |
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| |
AVCodec.channel_layouts is deprecated and Clang (unlike GCC)
warns when setting this field in a codec definition.
Fortunately, Clang (unlike GCC) allows to use
FF_DISABLE_DEPRECATION_WARNINGS inside a definition (of an FFCodec),
so that one can create simple macros to set AVCodec.channel_layouts
that also suppress deprecation warnings for Clang.
(Notice that some of the codec definitions were already
inside FF_DISABLE/ENABLE_DEPRECATION_WARNINGS (that were not
guarded by FF_API_OLD_CHANNEL_LAYOUT); these have been removed.
Also notice that setting AVCodec.channel_layouts was not guarded
by FF_API_OLD_CHANNEL_LAYOUT either, so testing disabling it
it without removing all the codeblocks would not have worked.)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
|
| |
|
|
|
|
|
|
| |
It reduces typing: Before this patch, there were 105 codecs
whose long_name-definition exceeded the 80 char line length
limit. Now there are only nine of them.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
|
| |
|
|
|
|
|
|
| |
Just return AVERROR_EXTERNAL immediately when encode error.
The other logic should keep the old behavior before commit 7c05b7951.
Suggested-By: Zhao Zhili <zhilizhao@tencent.com>
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
|
| |
|
|
|
|
|
|
|
|
|
| |
and remove FF_CODEC_CAP_INIT_THREADSAFE
All our native codecs are already init-threadsafe
(only wrappers for external libraries and hwaccels
are typically not marked as init-threadsafe yet),
so it is only natural for this to also be the default state.
Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
|
| |
|
|
|
|
|
|
|
|
|
|
| |
because the AudioConverterFillComplexBuffer can return 0 or 1 if
success.
so set the ret to 0 it AudioConverterFillComplexBuffer success and
return ret value for success or return AVERROR_EXTERNAL when
AudioConverterFillComplexBuffer failed.
BTW change the error message log level from warning to error.
Reviewed-by: Zhao Zhili <zhilizhao@tencent.com>
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
|
| |
|
|
|
|
|
|
|
|
|
| |
This is possible, because every given FFCodec has to implement
exactly one of these. Doing so decreases sizeof(FFCodec) and
therefore decreases the size of the binary.
Notice that in case of position-independent code the decrease
is in .data.rel.ro, so that this translates to decreased
memory consumption.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
|
| |
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| |
Up until now, codec.h contains both public and private parts
of AVCodec. This exposes the internals of AVCodec to users
and leads them into the temptation of actually using them
and forces us to forward-declare structures and types that
users can't use at all.
This commit changes this by adding a new structure FFCodec to
codec_internal.h that extends AVCodec, i.e. contains the public
AVCodec as first member; the private fields of AVCodec are moved
to this structure, leaving codec.h clean.
Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
|
| |
|
|
|
|
|
|
|
|
| |
Also move FF_CODEC_TAGS_END as well as struct AVCodecDefault.
This reduces the amount of files that have to include internal.h
(which comes with quite a lot of indirect inclusions), as e.g.
most encoders don't need it. It is furthemore in preparation
for moving the private part of AVCodec out of the public codec.h.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
|
| |
|
|
| |
Signed-off-by: James Almer <jamrial@gmail.com>
|
| |
|
|
| |
Signed-off-by: James Almer <jamrial@gmail.com>
|
| |
|
|
| |
Signed-off-by: Martin Storsjö <martin@martin.st>
|
| |
|
|
| |
Signed-off-by: James Almer <jamrial@gmail.com>
|
| |
|
|
| |
Signed-off-by: James Almer <jamrial@gmail.com>
|
| |
|
|
|
|
| |
Also include channel_layout.h directly wherever used.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
|
| |
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| |
Up until now, ff_alloc_packet2() has a min_size parameter:
It is supposed to be a lower bound on the final size of the packet
to allocate. If it is not too far from the upper bound (namely,
if it is at least half the upper bound), then ff_alloc_packet2()
already allocates the final, already refcounted packet; if it is
not, then the packet is not refcounted and its data only points to
a buffer owned by the AVCodecContext (in this case, the packet will
be made refcounted in encode_simple_internal() in libavcodec/encode.c).
The goal of this was to avoid data copies and intermediate buffers
if one has a precise lower bound.
Yet those encoders for which precise lower bounds exist have recently
been switched to ff_get_encode_buffer() (which automatically allocates
final buffers), leaving only two encoders to actually set the min_size
to something else than zero (namely aliaspixenc and hapenc). Both of
these encoders use a very low lower bound that is not helpful in any
nontrivial case.
This commit therefore removes the min_size parameter as well as the
codepath in ff_alloc_packet2() for the allocation of final buffers.
Furthermore, the function has been renamed to ff_alloc_packet() and
moved to encode.h alongside ff_get_encode_buffer().
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
|
| |
|
|
|
|
|
|
|
| |
It has been added in 2016 when this flag made no sense for encoders at
all; now that it makes sense, audiotoolboxenc doesn't support it,
despite claiming to do so.
Reviewed-by: James Almer <jamrial@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
|
| |
|
|
|
|
|
|
|
|
| |
Given that the AVCodec.next pointer has now been removed, most of the
AVCodecs are not modified at all any more and can therefore be made
const (as this patch does); the only exceptions are the very few codecs
for external libraries that have a init_static_data callback.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
|
| | |
|
| |
|
|
|
|
|
| |
Both are codec properties and not encoder capabilities. The relevant
AVCodecDescriptor.props flags exist for this purpose.
Signed-off-by: James Almer <jamrial@gmail.com>
|
| |
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| |
Previously, there was no way to flush an encoder such that after
draining, the encoder could be used again. We generally suggested
that clients teardown and replace the encoder instance in these
situations. However, for at least some hardware encoders, the cost of
this tear down/replace cycle is very high, which can get in the way of
some use-cases - for example: segmented encoding with nvenc.
To help address that use case, we added support for calling
avcodec_flush_buffers() to nvenc and things worked in practice,
although it was not clearly documented as to whether this should work
or not. There was only one previous example of an encoder implementing
the flush callback (audiotoolboxenc) and it's unclear if that was
intentional or not. However, it was clear that calling
avocdec_flush_buffers() on any other encoder would leave the encoder in
an undefined state, and that's not great.
As part of cleaning this up, this change introduces a formal capability
flag for encoders that support flushing and ensures a flush call is a
no-op for any other encoder. This allows client code to check if it is
meaningful to call flush on an encoder before actually doing it.
I have not attempted to separate the steps taken inside
avcodec_flush_buffers() because it's not doing anything that's wrong
for an encoder. But I did add a sanity check to reject attempts to
flush a frame threaded encoder because I couldn't wrap my head around
whether that code path was actually safe or not. As this combination
doesn't exist today, we'll deal with it if it ever comes up.
|
| |
|
|
|
|
|
|
|
|
|
|
|
|
| |
This fixes #6940
Although undocumented, AudioToolbox seems to require the data supplied
by the callback (i.e. ffat_encode_callback) being unchanged until the
next time the callback is called. In the old implementation, the
AVBuffer backing the frame is recycled after the frame is freed, and
somebody else (maybe the decoder) will write into the AVBuffer and
change the data. AudioToolbox then encodes some wrong data and noise
is produced. Retaining a frame reference solves this problem.
Signed-off-by: James Almer <jamrial@gmail.com>
|
| |
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| |
Explicitly identify decoder/encoder wrappers with a common name. This
saves API users from guessing by the name suffix. For example, they
don't have to guess that "h264_qsv" is the h264 QSV implementation, and
instead they can just check the AVCodec .codec and .wrapper_name fields.
Explicitly mark AVCodec entries that are hardware decoders or most
likely hardware decoders with new AV_CODEC_CAPs. The purpose is allowing
API users listing hardware decoders in a more generic way. The proposed
AVCodecHWConfig does not provide this information fully, because it's
concerned with decoder configuration, not information about the fact
whether the hardware is used or not.
AV_CODEC_CAP_HYBRID exists specifically for QSV, which can have software
implementations in case the hardware is not capable.
Based on a patch by Philip Langdale <philipl@overt.org>.
Merges Libav commit 47687a2f8aca3f65b6fdd117b1cb66a7409a7fd1.
|
| |
|
|
|
|
|
| |
AudioConverterFillComplexBuffer() doesn't always call its callback. A frame
queue is used to prevent skipped audio samples.
Signed-off-by: Rick Kern <kernrj@gmail.com>
|
| |
|
|
|
|
|
| |
The build failure here is caused by the enum value not being defined, but
as long as we're on a newer SDK that has it, it's safe to use it even
when our deployment target is older. Setting the property will error, but
we're not failing on errors there.
|
| | |
|
| |
|
|
|
|
|
|
| |
- size variables were used in a confusing way
- incorrect size var use led to channel layouts not being set properly
- channel layouts were incorrectly mapped for >2-channel AAC
- bitrates not accepted by the encoder were discarded instead of being clamped
- some minor style/indentation fixes
|
| |
|
|
|
| |
This isn't necessary here, and for some reason broke only multichannel
AAC encoding when a channel layout tag was set.
|
| |
|
|
|
|
|
|
|
|
| |
Fixes error "libavcodec/audiotoolboxenc.c:294:50: error: use of undeclared
identifier 'kAudioCodecPropertyPacketSizeLimitForVBR'."
That was added to 10.9:
https://developer.apple.com/library/mac/releasenotes/General/APIDiffsMacOSX10_9/AudioUnit.html
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
|
|
|
Fixes trac #4828
|
