Pass the audio source unchanged to the output.
-not meant to be used directly, it is inserted automatically by libavfilter
-whenever conversion is needed. Use the @var{aformat} filter to force a specific
-conversion.
+@section aresample
+
+Resample the input audio to the specified sample rate.
+
+The filter accepts exactly one parameter, the output sample rate. If not
+specified then the filter will automatically convert between its input
+and output sample rates.
+
+For example, to resample the input audio to 44100Hz:
+@example
+aresample=44100
+@end example
+
+@section ashowinfo
+
+Show a line containing various information for each input audio frame.
+The input audio is not modified.
+
+The shown line contains a sequence of key/value pairs of the form
+@var{key}:@var{value}.
+
+A description of each shown parameter follows:
+
+@table @option
+@item n
+sequential number of the input frame, starting from 0
+
+@item pts
+presentation TimeStamp of the input frame, expressed as a number of
+time base units. The time base unit depends on the filter input pad, and
+is usually 1/@var{sample_rate}.
+
+@item pts_time
+presentation TimeStamp of the input frame, expressed as a number of
+seconds
+
+@item pos
+position of the frame in the input stream, -1 if this information in
+unavailable and/or meaningless (for example in case of synthetic audio)
+
+@item fmt
+sample format name
+
+@item chlayout
+channel layout description
+
+@item nb_samples
+number of samples (per each channel) contained in the filtered frame
+
+@item rate
+sample rate for the audio frame
+
+@item planar
+if the packing format is planar, 0 if packed
+
+@item checksum
+Adler-32 checksum (printed in hexadecimal) of all the planes of the input frame
+
+@item plane_checksum
+Adler-32 checksum (printed in hexadecimal) for each input frame plane,
+expressed in the form "[@var{c0} @var{c1} @var{c2} @var{c3} @var{c4} @var{c5}
+@var{c6} @var{c7}]"
+@end table
+
+@section asplit
+
+Pass on the input audio to two outputs. Both outputs are identical to
+the input audio.
+
+For example:
+@example
+[in] asplit[out0], showaudio[out1]
+@end example
+
+will create two separate outputs from the same input, one cropped and
+one padded.
+
+@section astreamsync
+
+Forward two audio streams and control the order the buffers are forwarded.
+
+The argument to the filter is an expression deciding which stream should be
+forwarded next: if the result is negative, the first stream is forwarded; if
+the result is positive or zero, the second stream is forwarded. It can use
+the following variables:
+
+@table @var
+@item b1 b2
+number of buffers forwarded so far on each stream
+@item s1 s2
+number of samples forwarded so far on each stream
+@item t1 t2
+current timestamp of each stream
+@end table
+
+The default value is @code{t1-t2}, which means to always forward the stream
+that has a smaller timestamp.
+
+Example: stress-test @code{amerge} by randomly sending buffers on the wrong
+input, while avoiding too much of a desynchronization:
+@example
+amovie=file.ogg [a] ; amovie=file.mp3 [b] ;
+[a] [b] astreamsync=(2*random(1))-1+tanh(5*(t1-t2)) [a2] [b2] ;
+[a2] [b2] amerge
+@end example
+
+@section earwax
+
+Make audio easier to listen to on headphones.
+
+This filter adds `cues' to 44.1kHz stereo (i.e. audio CD format) audio
+so that when listened to on headphones the stereo image is moved from
+inside your head (standard for headphones) to outside and in front of
+the listener (standard for speakers).
+
+Ported from SoX.
+
+@section pan
+
+Mix channels with specific gain levels. The filter accepts the output
+channel layout followed by a set of channels definitions.
+
+This filter is also designed to remap efficiently the channels of an audio
+stream.
+
+The filter accepts parameters of the form:
+"@var{l}:@var{outdef}:@var{outdef}:..."
+
+@table @option
+@item l
+output channel layout or number of channels
+
+@item outdef
+output channel specification, of the form:
+"@var{out_name}=[@var{gain}*]@var{in_name}[+[@var{gain}*]@var{in_name}...]"
+
+@item out_name
+output channel to define, either a channel name (FL, FR, etc.) or a channel
+number (c0, c1, etc.)
+
+@item gain
+multiplicative coefficient for the channel, 1 leaving the volume unchanged
+
+@item in_name
+input channel to use, see out_name for details; it is not possible to mix
+named and numbered input channels
+@end table
+
+If the `=' in a channel specification is replaced by `<', then the gains for
+that specification will be renormalized so that the total is 1, thus
+avoiding clipping noise.
+
+@subsection Mixing examples
+
+For example, if you want to down-mix from stereo to mono, but with a bigger
+factor for the left channel:
+@example
+pan=1:c0=0.9*c0+0.1*c1
+@end example
+
+A customized down-mix to stereo that works automatically for 3-, 4-, 5- and
+7-channels surround:
+@example
+pan=stereo: FL < FL + 0.5*FC + 0.6*BL + 0.6*SL : FR < FR + 0.5*FC + 0.6*BR + 0.6*SR
+@end example
+
+Note that @command{ffmpeg} integrates a default down-mix (and up-mix) system
+that should be preferred (see "-ac" option) unless you have very specific
+needs.
+
+@subsection Remapping examples
+
+The channel remapping will be effective if, and only if:
+
+@itemize
+@item gain coefficients are zeroes or ones,
+@item only one input per channel output,
+@end itemize
+
+If all these conditions are satisfied, the filter will notify the user ("Pure
+channel mapping detected"), and use an optimized and lossless method to do the
+remapping.
+
+For example, if you have a 5.1 source and want a stereo audio stream by
+dropping the extra channels:
+@example
+pan="stereo: c0=FL : c1=FR"
+@end example
+
+Given the same source, you can also switch front left and front right channels
+and keep the input channel layout:
+@example
+pan="5.1: c0=c1 : c1=c0 : c2=c2 : c3=c3 : c4=c4 : c5=c5"
+@end example
+
+If the input is a stereo audio stream, you can mute the front left channel (and
+still keep the stereo channel layout) with:
+@example
+pan="stereo:c1=c1"
+@end example
+
+Still with a stereo audio stream input, you can copy the right channel in both
+front left and right:
+@example
+pan="stereo: c0=FR : c1=FR"
+@end example
+
+@section silencedetect
+
+Detect silence in an audio stream.
+
+This filter logs a message when it detects that the input audio volume is less
+or equal to a noise tolerance value for a duration greater or equal to the
+minimum detected noise duration.
+
+The printed times and duration are expressed in seconds.
+
+@table @option
+@item duration, d
+Set silence duration until notification (default is 2 seconds).
+
+@item noise, n
+Set noise tolerance. Can be specified in dB (in case "dB" is appended to the
+specified value) or amplitude ratio. Default is -60dB, or 0.001.
+@end table
+
+Detect 5 seconds of silence with -50dB noise tolerance:
+@example
+silencedetect=n=-50dB:d=5
+@end example
+
+Complete example with @command{ffmpeg} to detect silence with 0.0001 noise
+tolerance in @file{silence.mp3}:
+@example
+ffmpeg -f lavfi -i amovie=silence.mp3,silencedetect=noise=0.0001 -f null -
+@end example
+
+@section volume
+
+Adjust the input audio volume.
+
+The filter accepts exactly one parameter @var{vol}, which expresses
+how the audio volume will be increased or decreased.
+
+Output values are clipped to the maximum value.
+
+If @var{vol} is expressed as a decimal number, the output audio
+volume is given by the relation:
+@example
+@var{output_volume} = @var{vol} * @var{input_volume}
+@end example
+
+If @var{vol} is expressed as a decimal number followed by the string
+"dB", the value represents the requested change in decibels of the
+input audio power, and the output audio volume is given by the
+relation:
+@example
+@var{output_volume} = 10^(@var{vol}/20) * @var{input_volume}
+@end example
+
+Otherwise @var{vol} is considered an expression and its evaluated
+value is used for computing the output audio volume according to the
+first relation.
+
+Default value for @var{vol} is 1.0.
+
+@subsection Examples
+
+@itemize
+@item
+Half the input audio volume:
+@example
+volume=0.5
+@end example
+
+The above example is equivalent to:
+@example
+volume=1/2
+@end example
+
+@item
+Decrease input audio power by 12 decibels:
+@example
+volume=-12dB
+@end example
+@end itemize
+
+ @section resample
+ Convert the audio sample format, sample rate and channel layout. This filter is
++not meant to be used directly.
+
@c man end AUDIO FILTERS
@chapter Audio Sources
+include $(SUBDIR)../config.mak
+
NAME = avfilter
FFLIBS = avutil swscale
-FFLIBS-$(CONFIG_MOVIE_FILTER) += avformat avcodec
+ FFLIBS-$(CONFIG_RESAMPLE_FILTER) += avresample
-HEADERS = avfilter.h \
+FFLIBS-$(CONFIG_ACONVERT_FILTER) += swresample
+FFLIBS-$(CONFIG_AMOVIE_FILTER) += avformat avcodec
+FFLIBS-$(CONFIG_ARESAMPLE_FILTER) += swresample
+FFLIBS-$(CONFIG_MOVIE_FILTER) += avformat avcodec
+FFLIBS-$(CONFIG_PAN_FILTER) += swresample
+FFLIBS-$(CONFIG_REMOVELOGO_FILTER) += avformat avcodec
+FFLIBS-$(CONFIG_MP_FILTER) += avcodec postproc
+
+HEADERS = asrc_abuffer.h \
+ avcodec.h \
+ avfilter.h \
avfiltergraph.h \
buffersink.h \
buffersrc.h \
defaults.o \
drawutils.o \
formats.o \
+ graphdump.o \
graphparser.o \
+ src_buffer.o \
+ transform.o \
vf_scale.o \
- vsrc_buffer.o \
+OBJS-$(CONFIG_AVCODEC) += avcodec.o
+OBJS-$(CONFIG_AVFORMAT) += lavfutils.o
+OBJS-$(CONFIG_SWSCALE) += lswsutils.o
+
+OBJS-$(CONFIG_ACONVERT_FILTER) += af_aconvert.o
+OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o
+OBJS-$(CONFIG_AMERGE_FILTER) += af_amerge.o
OBJS-$(CONFIG_ANULL_FILTER) += af_anull.o
+OBJS-$(CONFIG_ARESAMPLE_FILTER) += af_aresample.o
+OBJS-$(CONFIG_ASHOWINFO_FILTER) += af_ashowinfo.o
+OBJS-$(CONFIG_ASPLIT_FILTER) += af_asplit.o
+OBJS-$(CONFIG_ASTREAMSYNC_FILTER) += af_astreamsync.o
+OBJS-$(CONFIG_EARWAX_FILTER) += af_earwax.o
+OBJS-$(CONFIG_PAN_FILTER) += af_pan.o
+ OBJS-$(CONFIG_RESAMPLE_FILTER) += af_resample.o
+OBJS-$(CONFIG_SILENCEDETECT_FILTER) += af_silencedetect.o
+OBJS-$(CONFIG_VOLUME_FILTER) += af_volume.o
+OBJS-$(CONFIG_AEVALSRC_FILTER) += asrc_aevalsrc.o
+OBJS-$(CONFIG_AMOVIE_FILTER) += src_movie.o
OBJS-$(CONFIG_ANULLSRC_FILTER) += asrc_anullsrc.o
+OBJS-$(CONFIG_ABUFFERSINK_FILTER) += sink_buffer.o
OBJS-$(CONFIG_ANULLSINK_FILTER) += asink_anullsink.o
+OBJS-$(CONFIG_ASS_FILTER) += vf_ass.o
+OBJS-$(CONFIG_BBOX_FILTER) += bbox.o vf_bbox.o
+OBJS-$(CONFIG_BLACKDETECT_FILTER) += vf_blackdetect.o
OBJS-$(CONFIG_BLACKFRAME_FILTER) += vf_blackframe.o
OBJS-$(CONFIG_BOXBLUR_FILTER) += vf_boxblur.o
+OBJS-$(CONFIG_COLORMATRIX_FILTER) += vf_colormatrix.o
OBJS-$(CONFIG_COPY_FILTER) += vf_copy.o
OBJS-$(CONFIG_CROP_FILTER) += vf_crop.o
OBJS-$(CONFIG_CROPDETECT_FILTER) += vf_cropdetect.o
return;
initialized = 1;
+ REGISTER_FILTER (ACONVERT, aconvert, af);
+ REGISTER_FILTER (AFORMAT, aformat, af);
+ REGISTER_FILTER (AMERGE, amerge, af);
REGISTER_FILTER (ANULL, anull, af);
+ REGISTER_FILTER (ARESAMPLE, aresample, af);
+ REGISTER_FILTER (ASHOWINFO, ashowinfo, af);
+ REGISTER_FILTER (ASPLIT, asplit, af);
+ REGISTER_FILTER (ASTREAMSYNC, astreamsync, af);
+ REGISTER_FILTER (EARWAX, earwax, af);
+ REGISTER_FILTER (PAN, pan, af);
+ REGISTER_FILTER (SILENCEDETECT, silencedetect, af);
+ REGISTER_FILTER (VOLUME, volume, af);
+ REGISTER_FILTER (RESAMPLE, resample, af);
+ REGISTER_FILTER (ABUFFER, abuffer, asrc);
+ REGISTER_FILTER (AEVALSRC, aevalsrc, asrc);
+ REGISTER_FILTER (AMOVIE, amovie, asrc);
REGISTER_FILTER (ANULLSRC, anullsrc, asrc);
+ REGISTER_FILTER (ABUFFERSINK, abuffersink, asink);
REGISTER_FILTER (ANULLSINK, anullsink, asink);
+ REGISTER_FILTER (ASS, ass, vf);
+ REGISTER_FILTER (BBOX, bbox, vf);
+ REGISTER_FILTER (BLACKDETECT, blackdetect, vf);
REGISTER_FILTER (BLACKFRAME, blackframe, vf);
REGISTER_FILTER (BOXBLUR, boxblur, vf);
+ REGISTER_FILTER (COLORMATRIX, colormatrix, vf);
REGISTER_FILTER (COPY, copy, vf);
REGISTER_FILTER (CROP, crop, vf);
REGISTER_FILTER (CROPDETECT, cropdetect, vf);
projects / ffmpeg / commitdiff