Quite often, the original weights are multiple of 512. By prescaling them
by 1/512 when they are computed (once per frame), no intermediate shifting
is needed, and no prescaling on each call either.
The x86 code already used that trick.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
If either input or output layout is known and the channel counts match,
use the known layout for both. Otherwise choose the default layout based on
av_get_default_channel_layout().
Changed some FATE references due to some WAVE files now having a non-zero
channel mask.
This commit is dedicated to the audiophiles who can hear it when a
needle is dropped on the moon.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Plain POSIX malloc(0) is allowed to return either NULL or a
non-NULL pointer. The calling code should be ready to handle
a NULL return as a correct return (instead of a failure) if the size
to allocate was 0 - this makes sure the condition is handled
in a consistent way across platforms.
This also avoids calling posix_memalign(&ptr, 32, 0) on OS X,
which returns an invalid pointer (a non-NULL pointer that causes
crashes when passed to av_free).
Abort in debug mode, to help track down issues related to
incorrect handling of this case.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
avconv: use default alignment for audio buffer
avcodec: use align == 0 for default alignment in avcodec_fill_audio_frame()
avutil: use align == 0 for default alignment in audio sample buffer functions
avutil: allow NULL linesize in av_samples_fill_arrays() and av_samples_alloc()
avconv: remove OutputStream.picref.
avconv: only set SAR once on the decoded frame.
avcodec: validate the channel layout vs. channel count for decoders
audioconvert: make av_get_channel_layout accept composite names.
avutil: add av_get_packed_sample_fmt() and av_get_planar_sample_fmt()
Conflicts:
doc/APIchanges
ffmpeg.c
libavcodec/utils.c
libavcodec/version.h
libavutil/audioconvert.c
libavutil/audioconvert.h
libavutil/avutil.h
libavutil/samplefmt.c
libavutil/samplefmt.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Decode output must be converted to rgb24 to avoid CRC difference
due to palette being stored in machine endianness.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Signal that it can output a frame when there are frames on the main
input and EOF on the overlay input, but a frame is buffered -- e.g.
single picture overlay.
This will allow a workaround for cases where input timestamps are invalid or
when decoder delay of 1 packet or more confuses avconv into using the wrong
timestamps as a sync reference.
Since those are pseudo-palette formats, swscale does not write
into data[1], swscale must initialize the palette properly itself.
This lead to frames that actually decoded as all-gray before.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
This fixes that the GIF encoder crashes with it because
it has no palette.
And the arguments for the pseudopalette apply to gray8 as
much as to RGB8 etc.
In addition the changes required in lavfi should be needed anyway
when adding support for RGB8 etc.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
We can't use whether the input format is paletted to decide that
the output format has a palette in data[1], too, that makes no sense.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Resolution changes are usually only used to scale with
network bandwidth, the (full) resolution specified in the
RM header really is authoritative.
While I am not sure this is the best solution, it is a
conservative approach that still should fix the most
common cases.
In particular, it fixes aspect with the sample from trac
issue #785 (in MPlayer, ffplay seems to just ignore
sample aspect changes in mid-playback).
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
When decoding LATM, this function will not process extradata
but a different buffer.
It seems this was forgotten to update when LATM support
was added.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
