Merge remote-tracking branch 'qatar/master'

* qatar/master: (26 commits)
  fate: use diff -b in oneline comparison
  Add missing version bumps and APIchanges/Changelog entries.
  lavfi: move buffer management function to a separate file.
  lavfi: move formats-related functions from default.c to formats.c
  lavfi: move video-related functions to a separate file.
  fate: make smjpeg a demux test
  fate: separate sierra-vmd audio and video tests
  fate: separate smacker audio and video tests
  libmp3lame: set supported channel layouts.
  avconv: automatically insert asyncts when -async is used.
  avconv: add support for audio filters.
  lavfi: add asyncts filter.
  lavfi: add aformat filter
  lavfi: add an audio buffer sink.
  lavfi: add an audio buffer source.
  buffersrc: add av_buffersrc_write_frame().
  buffersrc: fix invalid read in uninit if the fifo hasn't been allocated
  lavfi: rename vsrc_buffer.c to buffersrc.c
  avfiltergraph: reindent
  lavfi: add channel layout/sample rate negotiation.
  ...

Conflicts:
	Changelog
	doc/APIchanges
	doc/filters.texi
	ffmpeg.c
	ffprobe.c
	libavcodec/libmp3lame.c
	libavfilter/Makefile
	libavfilter/af_aformat.c
	libavfilter/allfilters.c
	libavfilter/avfilter.c
	libavfilter/avfilter.h
	libavfilter/avfiltergraph.c
	libavfilter/buffersrc.c
	libavfilter/defaults.c
	libavfilter/formats.c
	libavfilter/src_buffer.c
	libavfilter/version.h
	libavfilter/vf_yadif.c
	libavfilter/vsrc_buffer.c
	libavfilter/vsrc_buffer.h
	libavutil/avutil.h
	tests/fate/audio.mak
	tests/fate/demux.mak
	tests/fate/video.mak

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

3 files changed, 104 insertions, 2 deletions

diff --git a/doc/APIchanges b/doc/APIchanges
index 2d25d49114..d00416c9c6 100644
--- a/doc/APIchanges
+++ b/doc/APIchanges
@@ -27,13 +27,30 @@ API changes, most recent first:
 2012-03-26 - a67d9cf - lavfi 2.66.100
   Add avfilter_fill_frame_from_{audio_,}buffer_ref() functions.
 
-2012-xx-xx - xxxxxxx - lavc 54.13.1
+2012-05-15 - lavfi 2.17.0
+  Add support for audio filters
+  ac71230/a2cd9be - add video/audio buffer sink in a new installed
+                    header buffersink.h
+  720c6b7 - add av_buffersrc_write_frame(), deprecate
+            av_vsrc_buffer_add_frame()
+  ab16504 - add avfilter_copy_buf_props()
+  9453c9e - add extended_data to AVFilterBuffer
+  1b8c927 - add avfilter_get_audio_buffer_ref_from_arrays()
+
+2012-05-09 - lavu 51.30.0 - samplefmt.h
+  142e740 - add av_samples_copy()
+  6d7f617 - add av_samples_set_silence()
+
+2012-05-09 - a5117a2 - lavc 54.13.1
   For audio formats with fixed frame size, the last frame
   no longer needs to be padded with silence, libavcodec
   will handle this internally (effectively all encoders
   behave as if they had CODEC_CAP_SMALL_LAST_FRAME set).
 
-2012-xx-xx - xxxxxxx - lavr 0.0.1
+2012-05-07 - 828bd08 - lavc 54.13.0 - avcodec.h
+  Add sample_rate and channel_layout fields to AVFrame.
+
+2012-05-01 - 4010d72 - lavr 0.0.1
   Change AV_MIX_COEFF_TYPE_Q6 to AV_MIX_COEFF_TYPE_Q8.
 
 2012-04-25 - 3527a73 - lavu 51.29.0 - cpu.h
diff --git a/doc/ffmpeg.texi b/doc/ffmpeg.texi
index 6765121680..5e2f33a4bb 100644
--- a/doc/ffmpeg.texi
+++ b/doc/ffmpeg.texi
@@ -499,6 +499,11 @@ Set the audio codec. This is an alias for @code{-codec:a}.
 @item -sample_fmt[:@var{stream_specifier}] @var{sample_fmt} (@emph{output,per-stream})
 Set the audio sample format. Use @code{-sample_fmts} to get a list
 of supported sample formats.
+@item -af @var{filter_graph} (@emph{output})
+@var{filter_graph} is a description of the filter graph to apply to
+the input audio.
+Use the option "-filters" to show all the available filters (including
+also sources and sinks).  This is an alias for @code{-filter:a}.
 @end table
 
 @section Advanced Audio options:
@@ -781,6 +786,7 @@ Audio sync method. "Stretches/squeezes" the audio stream to match the timestamps
 the parameter is the maximum samples per second by which the audio is changed.
 -async 1 is a special case where only the start of the audio stream is corrected
 without any later correction.
+This option has been deprecated. Use the @code{asyncts} audio filter instead.
 @item -copyts
 Copy timestamps from input to output.
 @item -copytb @var{mode}
diff --git a/doc/filters.texi b/doc/filters.texi
index 2af7b37d07..ba8c9f46fb 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -211,6 +211,32 @@ amovie=input.mkv:si=5 [a5];
 [x3][a5] amerge" -c:a pcm_s16le output.mkv
 @end example
 
+@section aformat
+
+Convert the input audio to one of the specified formats. The framework will
+negotiate the most appropriate format to minimize conversions.
+
+The filter accepts the following named parameters:
+@table @option
+
+@item sample_fmts
+A comma-separated list of requested sample formats.
+
+@item sample_rates
+A comma-separated list of requested sample rates.
+
+@item channel_layouts
+A comma-separated list of requested channel layouts.
+
+@end table
+
+If a parameter is omitted, all values are allowed.
+
+For example to force the output to either unsigned 8-bit or signed 16-bit stereo:
+@example
+aformat=sample_fmts\=u8\,s16:channel_layouts\=stereo
+@end example
+
 @section anull
 
 Pass the audio source unchanged to the output.
@@ -502,6 +528,25 @@ volume=-12dB
 @end example
 @end itemize
 
+@section asyncts
+Synchronize audio data with timestamps by squeezing/stretching it and/or
+dropping samples/adding silence when needed.
+
+The filter accepts the following named parameters:
+@table @option
+
+@item compensate
+Enable stretching/squeezing the data to make it match the timestamps.
+
+@item min_delta
+Minimum difference between timestamps and audio data (in seconds) to trigger
+adding/dropping samples.
+
+@item max_comp
+Maximum compensation in samples per second.
+
+@end table
+
 @section resample
 Convert the audio sample format, sample rate and channel layout. This filter is
 not meant to be used directly.
@@ -721,6 +766,33 @@ anullsrc=r=48000:cl=4
 anullsrc=r=48000:cl=mono
 @end example
 
+@section abuffer
+Buffer audio frames, and make them available to the filter chain.
+
+This source is not intended to be part of user-supplied graph descriptions but
+for insertion by calling programs through the interface defined in
+@file{libavfilter/buffersrc.h}.
+
+It accepts the following named parameters:
+@table @option
+
+@item time_base
+Timebase which will be used for timestamps of submitted frames. It must be
+either a floating-point number or in @var{numerator}/@var{denominator} form.
+
+@item sample_rate
+Audio sample rate.
+
+@item sample_fmt
+Name of the sample format, as returned by @code{av_get_sample_fmt_name()}.
+
+@item channel_layout
+Channel layout of the audio data, in the form that can be accepted by
+@code{av_get_channel_layout()}.
+@end table
+
+All the parameters need to be explicitly defined.
+
 @c man end AUDIO SOURCES
 
 @chapter Audio Sinks
@@ -745,6 +817,13 @@ Null audio sink, do absolutely nothing with the input audio. It is
 mainly useful as a template and to be employed in analysis / debugging
 tools.
 
+@section abuffersink
+This sink is intended for programmatic use. Frames that arrive on this sink can
+be retrieved by the calling program using the interface defined in
+@file{libavfilter/buffersink.h}.
+
+This filter accepts no parameters.
+
 @c man end AUDIO SINKS
 
 @chapter Video Filters