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

This fix is not ideal as it still limits the multithreading on field pictures
to the 2nd field only.
Ill try to fix it properly to allow both fields to decode concurrently but this
needs more work.

This bug exists since and was caused by:
commit ea6331f8
Author: Ronald S. Bultje <rsbultje@gmail.com>
Date:   Mon Jun 20 10:24:33 2011 -0400

    h264-mt: fix deadlock in packets with multiple slices (e.g. MP4).

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: Reimar Döffinger <Reimar.Doeffinger@gmx.de>

Fixes Ticket498
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Fixes Ticket518

Fixes Ticket526

From 52.503s (~40fps) to 27.973sec (~80fps) decoding of 480p sintel
trailer, i.e. a ~2x speedup overall, on a Nexus S.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

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

* qatar/master:
  mpc8: check output buffer size before decoding
  mpc7: return error if packet is too small.
  mpc7: check output buffer size before decoding
  nellymoserdec: allocate float_buf only when decoding to int16
  nellymoserdec: use NELLY_BUF_LEN instead of 128
  nellymoserdec: use NELLY_BLOCK_LEN instead of 64 when appropriate.
  nellymoserdec: allow user to request SAMPLE_FMT_FLT for output samples.
  nellymoser: check output buffer size before decoding
  win32: improve threading algorithm warning

Conflicts:
	libavcodec/nellymoserdec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>

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

* cus/stable:
  ffplay: use libswresample instead of av_audio_convert
  audioconvert: add av_get_default_channel_layout public function
  ffplay: use avctx->channels and avctx->freq before avcodec_open2 consistently
  ffplay: remove now unnecessary request_channels, we set it now with options
  ffplay: set request_channels to 2
Merged-by: Michael Niedermayer <michaelni@gmx.at>

Previously ffplay expected SDL_AudioOpen to provide the requested sample rate
and channel number. This is no longer a requirement because this patch replaces
the audio convert function with libswresample's swr_convert which is capable of
handling different sample formats, sample rates and different number of
channels and different channel layouts.

The patch also removes the hardcoded 16bit samples assumption and uses
av_get_bytes_per_sample almost everywhere. The only exceptions are
the update_sample_display and video_audio_display functions, it
seemed too much of a headache to make them generic.

We also fix a tiny bug in sdl_audio_callback, we ensure that the number of
bytes when we put silence in the buffer is a multiple of the frame size.

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>

This patch set the limit to 16.

For information, thoses previous commits:
41f7e2d1
5cbb0e70
assumed it was either 30 or 32.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

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

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

Prefer av_opt_set_defaults() instead.

If no threading is requested do not issue warning about unsupported
threading algorithm.