* qatar/master:
id3v2: remove pointless casts
id3v2: read TXXX frames with two calls to decode_str() instead of one.
id3v2: don't discard the whole tag when encountering empty frames.
libvpx: fix build with older libvpx versions.
ARM: check for inline asm 'y' operand modifier support
Conflicts:
libavcodec/libvpxenc.c
libavformat/id3v2.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This simplifies useage for users who are used to vpxenc.
We continue to support the previous options with the exception of the flags
for which it would be slightly messy.
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 ea6331f8bb
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).
VPX_ERROR_RESILIENT_DEFAULT and VPX_ERROR_RESILIENT_PARTITIONS weren't
defined before 4cb0ebe5b27d35ccc2a78c1d16f2622ddef21f74 (CommitDate: Tue
Jun 28 11:10:17 2011)
The inline asm added in bf5d46d uses the 'y' modifier which
is only supported from gcc 4.5. This check allows building
with older compilers.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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>
* 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>
* 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.
This patch set the limit to 16.
For information, thoses previous commits:
41f7e2d11d5cbb0e70a0
assumed it was either 30 or 32.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
