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.

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]).

* 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  libmp3lame: remove outdated comment.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>

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>

This will allow parsers to export the duration of the current frame being
output, if known, instead of using AVCodecContext.frame_size.

The parser was fixed so this workaround should no longer
be necessary.
This allows using stream-copy to fix files with keyframes
incorrectly marked.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>

Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>

Also, only set bit rate for CBR.

also remove unneeded commented-out full frame data debugging

We now require at least libmp3lame 3.98.3.
lame_encode_buffer_interleaved() still doesn't work for mono, but it does not
"die"; it just expects a stereo interleaved buffer.

it is already set in avcodec_alloc_frame()

Signed-off-by: Stefano Sabatini <stefasab@gmail.com>

Also move the get_bit_rate() function, in order to avoid an undefined
symbol reference in avcodec_open2().

This avoids reading any old data in the AVIOContext buffer after
the seek, and indicates to the mpegts demuxer that we've seeked,
avoiding continuity check errors.
Signed-off-by: Martin Storsjö <martin@martin.st>

The channel layout must start with 0x, otherwise it is parsed as a
number of channels.

Set the pict_type/key_frame properties on the coded picture.