* 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>
* qatar/master:
lavc: add video/audio/encoding flags to global_quality option
libvpxenc: use libvpx's own defaults for some parameters
vpxenc: add private options
Conflicts:
libavcodec/libvpxenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (34 commits)
dpcm: return error if packet is too small
dpcm: use smaller data types for static tables
dpcm: use sol_table_16 directly instead of through the DPCMContext.
dpcm: replace short with int16_t
dpcm: check to make sure channels is 1 or 2.
dpcm: misc pretty-printing
dpcm: remove unnecessary variable by using bytestream functions.
dpcm: move codec-specific variable declarations to their corresponding decoding blocks.
dpcm: consistently use the variable name 'n' for the next input byte.
dpcm: output AV_SAMPLE_FMT_U8 for Sol DPCM subcodecs 1 and 2.
dpcm: calculate and check actual output data size prior to decoding.
dpcm: factor out the stereo flag calculation
dpcm: cosmetics: rename channel_number to ch
avserver: Fix a bug where the socket is IPv4, but IPv6 is autoselected for the loopback address.
lavf: Avoid using av_malloc(0) in av_dump_format
dxva2_h264: pass the correct 8x8 scaling lists
dca: NEON optimised high freq VQ decoding
avcodec: reject audio packets with NULL data and non-zero size
dxva: Add ability to enable workaround for older ATI cards
latmenc: Set latmBufferFullness to largest value to indicate it is not used
...
Conflicts:
libavcodec/dxva2_h264.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
