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

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

libswresample takes care of that now.

Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

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

* qatar/master:
  mpegvideo_enc: only allocate output packet when we know there will be output
  Add names for more channel layouts to the channel layout map.
  sunrast: Add a sample request for RMP_RAW colormap.
  avcodec: do not override pts or duration from the audio encoder
  Add prores regression test.
  Enable already existing rso regression test.
  Add regression test for "sox" format muxer/demuxer.
  Add dpx encoding regression test.
  swscale: K&R formatting cosmetics for PowerPC code (part I/II)
  img2: Use ff_guess_image2_codec(filename) shorthand where appropriate.
  Clarify licensing information about files borrowed from libjpeg.
  Mark mutable static data const where appropriate.
  avplay: fix -threads option
  dvbsubdec: avoid undefined signed left shift in RGBA macro
  mlpdec: use av_log_ask_for_sample()
  gif: K&R formatting cosmetics
  png: make .long_name more descriptive
  movdec: Adjust keyframe flagging in fragmented files
  rv34: change most "int stride" into "ptrdiff_t stride".

Conflicts:
	avprobe.c
	ffplay.c
	libavcodec/mlpdec.c
	libavcodec/mpegvideo_enc.c
	libavcodec/pngenc.c
	libavcodec/x86/v210-init.c
	libavfilter/vf_boxblur.c
	libavfilter/vf_crop.c
	libavfilter/vf_drawtext.c
	libavfilter/vf_lut.c
	libavfilter/vf_overlay.c
	libavfilter/vf_pad.c
	libavfilter/vf_scale.c
	libavfilter/vf_select.c
	libavfilter/vf_setpts.c
	libavfilter/vf_settb.c
	libavformat/img2.c
	libavutil/audioconvert.c
	tests/codec-regression.sh
	tests/lavf-regression.sh
	tests/ref/lavf/dpx
	tests/ref/vsynth1/prores
	tests/ref/vsynth2/prores
Merged-by: Michael Niedermayer <michaelni@gmx.at>

This fixes seeking to before and after files with ffplay.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Its otherwise spaming every time one tries to seek to outside
the file.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Theres no usefull or even remotely complete information on it currently.
Which just leads to confusion.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Fixes a memleak.

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

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

This allows encoders to set pts and/or duration even if they do not use
CODEC_CAP_DELAY.

Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>

Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>

Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>

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

Mostly based on doc/examples/filtering.c. lavfi API is still limited to
"buffer feeding" instead of "frame feeding" at the moment, so this
example code sticks with it.

The AVOptions based default to threads auto in 2473a45c
works only if avplay does not use custom option handling
for -threads.

CC: <libav-stable@libav.org>

Samples buffer ref is allocated and loaded with the uninitialized data
pointers:
     av_asrc_buffer_add_buffer()
  -> av_asrc_buffer_add_samples()
  -> avfilter_get_audio_buffer_ref_from_arrays(data, ...)

...which leads to a crash with at least lavfi/ashowinfo in case of !NULL
(see the for loop while samplesref->data[plane]).

Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Janne Grunau <janne-libav@jannau.net>

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

Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Diego Biurrun <diego@biurrun.de>

For video, mark the first sample in a trun which doesn't have the
sample-is-non-sync-sample flag set as a keyframe.

In particular, the "sample does not depend on other samples" flag
isn't enough to make it a keyframe, since later frames still can
reference frames prior to that one (the flag only says that that
particular frame doesn't depend on other frames).

This fixes bug 215.
Signed-off-by: Martin Storsjö <martin@martin.st>

* qatar/master: (36 commits)
  adpcmenc: Use correct frame_size for Yamaha ADPCM.
  avcodec: add ff_samples_to_time_base() convenience function to internal.h
  adx parser: set duration
  mlp parser: set duration instead of frame_size
  gsm parser: set duration
  mpegaudio parser: set duration instead of frame_size
  (e)ac3 parser: set duration instead of frame_size
  flac parser: set duration instead of frame_size
  avcodec: add duration field to AVCodecParserContext
  avutil: add av_rescale_q_rnd() to allow different rounding
  pnmdec: remove useless .pix_fmts
  libmp3lame: support float and s32 sample formats
  libmp3lame: renaming, rearrangement, alignment, and comments
  libmp3lame: use the LAME default bit rate
  libmp3lame: use avpriv_mpegaudio_decode_header() for output frame parsing
  libmp3lame: cosmetics: remove some pointless comments
  libmp3lame: convert some debugging code to av_dlog()
  libmp3lame: remove outdated comment.
  libmp3lame: do not set coded_frame->key_frame.
  libmp3lame: improve error handling in MP3lame_encode_init()
  ...

Conflicts:
	doc/APIchanges
	libavcodec/libmp3lame.c
	libavcodec/pcxenc.c
	libavcodec/pnmdec.c
	libavcodec/pnmenc.c
	libavcodec/sgienc.c
	libavcodec/utils.c
	libavformat/hls.c
	libavutil/avutil.h
	libswscale/x86/swscale_mmx.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>

This prevents having to sign-extend on 64-bit systems with 32-bit ints,
such as x86-64. Also fixes crashes on systems where we don't do it and
arguments are not in registers, such as Win64 for all weight functions.

That is the logical order in which the functions are called,
and there is no longer any need of forward declarations.

libswresample is used in all cases.

Output packet size should match avctx->block_align. The target output packet
size is 1024 bytes.
Before:
mono   - 1024 samples -> 512 bytes
stereo - 2048 samples -> 2048 bytes
After:
mono   - 2048 samples -> 1024 bytes
stereo - 1024 samples -> 1024 bytes

Fixes memory leak when encoding at least with mpegvideo using the
new encode2 function.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>