It was only correct when element ids met very specific criteria.

It is obscure, most likely unused and not bit-exact compared to
libavcodec due to a different IDCT transform algorithm.

This fixes crashes in e.g. PNG decoding with SSE2 enabled. In fact, many
x86 optimizations for codecs assume that our buffer strides are 16-byte
aligned.

This fixes XMM register clobber problems on Win64.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>

Fixes bug 221.

CC: libav-stable@libav.org

Also slightly move around code not allocate a new frame if we won't
decode it. This prevents us from putting undecoded frames in frame
pointers, which (in mt decoding) other threads will use and wait on
as references, causing a deadlock (if we skipped decoding) or a crash
(if we didn't initialized next_framep[] at all).

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind

next_dts is used for estimating the dts of the next packet if it's
missing. Therefore, it makes no sense to set it from the pts of the last
decoded frame. Also it should be estimated from the current packet
duration/ticks_per_frame always, not only when a frame was successfully
decoded.

It more accurately describes what does this variable store.

It currently has different meanings at different times (dts of the last
read packet/pts of the last decoded frame). Reduce obfuscation by
storing pts of the decoded frame in the frame itself.

It's used to predict dts, not pts.

Current code compares the desired recording time with InputStream.pts,
which has a very unclear meaning. Change the code to use actual
timestamps of the frames passed to the encoder.

In several tests, one less frame is encoded, which is more correct.

In the idroq test one more frame is encoded, which is again more
correct.

Behavior with stream copy should be unchanged.

The actual number (1/1000) will probably require some
discussion/tweaking in the future, but should be good enough for now,
since the timestamps in AVSubtitle are in this timebase by definition.

The output is obviously not supposed to contain video (since only
-acodec copy is specified), but that only happens because of the way -t
handling is implemented currently.

It makes sense in some cases to split up the output packet to save on memory
usage (ape frames can be very large), but the current/default size is
arbitrary. Allowing the user to configure this gives more flexibility and
requires minimal additional code.

This will allow the decoder to return samples for the full packet, and it also
makes the decoded buffer pointers aligned.

Signed-off-by: Diego Biurrun <diego@biurrun.de>

Signed-off-by: Diego Biurrun <diego@biurrun.de>

Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>

FFALIGN doesn't work with non-powers-of-2.

This reverts commit 7ad1b612.
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>

Fixes seeking after decoder has already been flushed for codecs using
CODEC_CAP_DELAY.

Calculates based on total file size and wavetaillength from the header.
Falls back to multiplying finalframeblocks by 8 instead of 4 so that it will
at least be overestimating for 24-bit. Currently it can underestimate the
final packet size, leading to decoding errors.

This avoids buffer overread when the last packet size estimate is too small.

Signed-off-by: Diego Biurrun <diego@biurrun.de>

Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>

Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>

Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>

Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>

It is already set in avcodec_alloc_frame().
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>

Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>