Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Since SDL has no audio buffer fullness info, one can get a much precise audio
clock based on the last time of the audio callback and the elapsed time since.

To achieve this I introduced the audio_current_pts and audio_current_pts_drift
variables (similar to video_current_pts and video_current_pts_drift) and
calculate them in the end of the audio callback, when VideoState->audio_clock
is already updated. The reference time I use is from the start of the audio
callback, because this way the amount of time used for audio decoding is not
interfereing with calculation.

I also replaced the audio_write_get_buf_size function with a calculated
variable because when the audio frame decoding is in progress audio_buf_size
and audio_buf_index are not stable, so using them from other threads are not a
good idea.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

* qatar/master:
  aacenc: Fix number of coefficients used in a LFE channel.
  aacenc: Fix a segfault with grouped psymodel.
Merged-by: Michael Niedermayer <michaelni@gmx.at>

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

The previous default value 0x0 was not good, since it is not even
valid.

In apply_unsharp(), when y is >= height, prevent out-of-buffer reading
from src, read from the last buffer line in src2 instead.

The check was implemented in the original unsharp libmpcodecs code and
lost in the port.

This also fixes output discrepancy between the two filters.

The delogo filter has been natively integrated into libavfilter,
simplify.

The ported filter supports named option parsing and more YUV formats.

Signed-off-by: Stefano Sabatini <stefano.sabatini-lala@poste.it>

Signed-off-by: Stefano Sabatini <stefano.sabatini-lala@poste.it>

The spec states:

 *  Only the lowest 12 spectral coefficients of any LFE may be non-zero

We were using the 12 lowest *bands*.

10l: Forgot about TYPE_LFE being after TYPE_CPE. Which causes a
segfault when encoding 5.1

Allow to cache more than one frame (e.g. for filters which return
more than one frame when avfilter_request_frame() is called on them),
and do not discard previously cached frames when a new one is added.

The new function provides a more flexible interface for peeking at a
FIFO buffer data.

In lavfi_read_header(), use the pad index designated in the inout for
linking an output to a sink, rather than always 0. Fix link creation
for filters with more than one output (e.g. the split filter).

Custom PAR num/denum are coded -1.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>

FFmpeg writes data_size as AU_UNKNOWN_SIZE, make demuxer not
fail when data_size is set to this value.
Should fix trac issue #394.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>

* qatar/master:
  swscale: add dithering to yuv2yuvX_altivec_real
  rv34: free+allocate buffer instead of reallocating it to preserve alignment
  h264: add missing brackets.
  swscale: use 15-bit intermediates for 9/10-bit scaling.
Merged-by: Michael Niedermayer <michaelni@gmx.at>

This is based on the original work by Baptiste Coudurier.

This is based on the original work by Baptiste Coudurier.

This is based on the original work by Baptiste Coudurier.

Signed-off-by: Stefano Sabatini <stefano.sabatini-lala@poste.it>

Also drop obfuscating CHROMA_WIDTH/HEIGHT macros.

More consistent with the original libmpcodecs code, and the name
"unsharpen" was confusing.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>