mirror of https://github.com/mpv-player/mpv
Further libaf documentation by Anders with some more updates by me.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@8892 b3059339-0415-0410-9bf9-f77b7e298cf2
This commit is contained in:
diego
parent
8229e28f90
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@ -197,6 +197,8 @@
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<LI><A HREF="sound.html#af_volume">2.3.2.3.5 Software volume control</A></LI>
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<LI><A HREF="sound.html#af_equalizer">2.3.2.3.6 Equalizer</A></LI>
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<LI><A HREF="sound.html#af_panning">2.3.2.3.7 Panning filter</A></LI>
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<LI><A HREF="sound.html#af_sub">2.3.2.3.8 Sub-woofer</A></LI>
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<LI><A HREF="sound.html#af_surround">2.3.2.3.9 Surround-sound decoder</A></LI>
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</UL>
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</LI>
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<LI><A HREF="sound.html#plugins">2.3.2.4 Audio plugins (deprecated)</A>
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@ -1010,7 +1010,7 @@ Activate a comma separated list of audio filters and their options.
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Available filters are:
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.
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.RSs
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.IPs resample[=srate[:sloppy][:fast]]
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.IPs resample[=srate[:sloppy][:type]]
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Changes the sample rate of the audio stream to an integer srate (Hz).
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It only supports the 16 bit little endian format.
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.IPs channels[=nch]
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@ -1034,7 +1034,7 @@ unsigned or signed
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.br
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le or be (little or big endian)
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.br
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.IPs "volume[=v:sc:pr:en]"
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.IPs "volume[=v:sc]"
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Select the output volume level.
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This filter is not reentrant and can therefore only be enabled once for every
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audio stream.
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@ -1045,15 +1045,25 @@ completely and +40dB equals a gain of 1000).
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The default gain is -20dB.
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.br
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sc: enable soft clipping.
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.REss
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.IPs "pan[=n:l01:l02:..l10:l11:l12:...ln0:ln1:ln2:...]"
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Mixes channels arbitrarily, see DOCS/sound.html for details.
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.RSss
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n: number of output channels (1 - 6).
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.br
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pr: enable probing of the volume level for each audio stream.
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Both the maximum and instantaneous volume is probed.
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The instantaneous volume can only be accessed through the runtime interface,
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but the maximum volume is printed at the end of the movie.
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This value can be used when transcoding movies to maximize the utilization
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of the dynamic range.
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lij: how much of input channel j is mixed into output channel i.
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.REss
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.IPs "sub[=fc:ch]"
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Add sub-woofer channel.
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.RSss
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fc: Cutoff frequency for low-pass filter (20Hz to 300Hz) default is 60Hz.
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.br
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en: enable and disable the volume control.
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ch: channel number for the sub-channel.
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.REss
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.IPs "surround[=d]"
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Decoder for matrix encoded surround sound, works on many 2 channel files.
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.RSss
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d: delay time in ms for the rear speakers (0ms to 1000ms) default is 15ms.
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.REss
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.IPs delay[=ch1:ch2:...]
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Delays the sound output.
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205
DOCS/sound.html
205
DOCS/sound.html
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@ -189,40 +189,71 @@
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<P>would set the output frequency of the resample filter to 11025Hz and downmix
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the audio to 1 channel using the pan filter.</P>
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<P>Most filters respond to the <CODE>-v</CODE> switch, which makes the filters
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print out status messages.</P>
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<P>The overall execution of the filter layer is controlled using the
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<CODE>-af-adv</CODE> switch. This switch has two suboptions:</P>
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<DL>
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<DT><CODE>force</CODE><DT>
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<DD>is an integer between 0 and 3 that controls how the filters are inserted
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and what speed/accuracy optimizations they use:
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<DD>is a Bit field that controls how the filters are inserted and what
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speed/accuracy optimizations they use:
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<DL>
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<DT>0</DT>
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<DT><CODE>0</CODE></DT>
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<DD>Use automatic insertion of filters and optimize according to CPU
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speed.</DD>
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<DT>1</DT>
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<DD>Use automatic insertion of filters and optimize for the highest speed.
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If this option is set the processing of the audio data will be done
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using fix point arithmetics. Warning: Some features in the audio filters
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will silently fail, and the sound quality may drop.</DD>
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<DT>2</DT>
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<DD>Use automatic insertion of filters and optimize for quality. If this
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option is set the processing of the audio data will be done using
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floating point instructions and is therefore quite CPU intensive, but
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gives a lot higher sound quality than fix point processing.</DD>
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<DT>3</DT>
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<DD>Use no automatic insertion of filters and no optimization. Warning: It
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may be possible to crash MPlayer using this setting.</DD>
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<DT><CODE>1</CODE></DT>
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<DD>Use automatic insertion of filters and optimize for the highest
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speed.<BR>
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<EM>Warning:</EM> Some features in the audio filters may silently fail,
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and the sound quality may drop.</DD>
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<DT><CODE>2</CODE></DT>
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<DD>Use automatic insertion of filters and optimize for quality.</DD>
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<DT><CODE>3</CODE></DT>
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<DD>Use no automatic insertion of filters and no optimization.<BR>
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<I>Warning:</I> It may be possible to crash MPlayer using this
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setting.</DD>
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<DT><CODE>4</CODE></DT>
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<DD>Use automatic insertion of filters according to 0 above, but use
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floating point processing when possible.</DD>
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<DT><CODE>5</CODE></DT>
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<DD>Use automatic insertion of filters according to 1 above, but use
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floating point processing when possible.</DD>
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<DT><CODE>6</CODE></DT>
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<DD>Use automatic insertion of filters according to 2 above, but use
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floating point processing when possible.</DD>
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<DT><CODE>7</CODE></DT>
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<DD>Use no automatic insertion of filters according to 3 above, and use
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floating point processing when possible.</DD>
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</DL>
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</DD>
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<DT><CODE>list</CODE></DT>
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<DD>is an alias for the -af switch.</DD>
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</DL>
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<P>The filter layer is also affected by the following generic switches:
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<DL>
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<DT><CODE>-v</CODE></DT>
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<DD>Increases the verbosity level and makes most filters print out extra
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status messages.</DD>
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<DT><CODE>-channels</CODE></DT>
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<DD>This option sets the number of output channels your sound card is using.
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It also affects the number of channels that are being decoded from the
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media. If the media contains less channels than requested the channels
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filter (see below) will automatically be inserted. The routing will be the
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default routing for the channels filter.</DD>
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<DT><CODE>-srate</CODE></DT>
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<DD>This option selects the sample rate of your sound card. If the sample
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frequency of your sound card is different from that of the current media,
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the resample filter (see below) will be inserted into the audio filter layer
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to compensate for the difference.</DD>
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<DT><CODE>-format</CODE><DT>
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<DD>This option sets the sample format of the audio filter layer and the sound
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card. If the requested sample format of your sound card is different from
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that of the current media, a format filter (see below) will be inserted to
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rectify the difference.</DD>
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</DL>
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<H5><A NAME="af_resample">2.3.2.3.1 Up/Down-sampling</A></H5>
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@ -233,7 +264,7 @@
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has three switches:</P>
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<DL>
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<DT><CODE>srate</CODE></DT>
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<DT><CODE>srate <8-192></CODE></DT>
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<DD>is an integer used for setting the output sample
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frequency in Hz. The valid range for this parameter is 8kHz to 192kHz. If
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the input and output sample frequency are the same or if this parameter is
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@ -244,12 +275,19 @@
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<DT><CODE>sloppy</CODE></DT>
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<DD>is an optional binary parameter that allows the output frequency to differ
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slightly from the frequency given by <CODE>srate</CODE>. This switch can be
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used if the startup of the playback is extremely slow.</DD>
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used if the startup of the playback is extremely slow. It is enabled by
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default.</DD>
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<DT><CODE>fast</CODE><DT>
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<DD>is an optional binary parameter that enables linear interpolation as
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resampling method. Linear interpolation is extremely fast, but suffers from
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poor sound quality especially when used for up-sampling.</DD>
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<DT><CODE>type <0-2></CODE><DT>
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<DD>is an optional integer between <CODE>0</CODE> and <CODE>2</CODE> that
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selects which resampling method to use. Here <CODE>0</CODE> represents
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linear interpolation as resampling method, <CODE>1</CODE> represents
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resampling using a poly-phase filter-bank and integer processing and
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<CODE>2</CODE> represents resampling using a poly-phase filter-bank and
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floating point processing. Linear interpolation is extremely fast, but
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suffers from poor sound quality especially when used for up-sampling. The
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best quality is given by <CODE>2</CODE> but this method also suffers from
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the highest CPU load.</DD>
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</DL>
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<P>Example:<BR>
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@ -268,19 +306,19 @@
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itself if not needed. The number of switches is dynamic:</P>
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<DL>
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<DT><CODE>nch</CODE></DT>
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<DD>is an integer between 1 and 6 that is used for setting the number of
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output channels. This switch is required, leaving it empty results in a
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runtime error.</DD>
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<DT><CODE>nch <1-6></CODE></DT>
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<DD>is an integer between <CODE>1</CODE> and <CODE>6</CODE> that is used for
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setting the number of output channels. This switch is required, leaving it
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empty results in a runtime error.</DD>
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<DT><CODE>nr</CODE></DT>
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<DD>is an integer between 1 and 6 that is used for specifying the number of
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routes. This parameter is optional. If it is omitted the default routing is
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used.</DD>
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<DT><CODE>nr <1-6></CODE></DT>
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<DD>is an integer between <CODE>1</CODE> and <CODE>6</CODE> that is used for
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specifying the number of routes. This parameter is optional. If it is
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omitted the default routing is used.</DD>
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<DT><CODE>from1:to1:from2:to2:from3:to3...</CODE></DT>
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<DD>are pairs of numbers between 0 and 5 that define where each channel should
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be routed.</DD>
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<DD>are pairs of numbers between <CODE>0</CODE> and <CODE>5</CODE> that define
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where each channel should be routed.</DD>
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</DL>
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<P>If only <CODE>nch</CODE> is given the default routing is used, it works as
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@ -311,11 +349,12 @@
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needed by the sound card or another filter.</P>
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<DL>
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<DT><CODE>bps</CODE></DT>
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<DD>can be 1, 2 or 4 and denotes the number of bytes per sample. This switch
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is required, leaving it empty results in a runtime error.</DD>
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<DT><CODE>bps <number></CODE></DT>
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<DD>can be <CODE>1</CODE>, <CODE>2</CODE> or <CODE>4</CODE> and denotes the
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number of bytes per sample. This switch is required, leaving it empty
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results in a runtime error.</DD>
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<DT><CODE>f</CODE></DT>
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<DT><CODE>f <format></CODE></DT>
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<DD>is a text string describing the sample format. The string is a
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concatenated mix of: <CODE>alaw</CODE>, <CODE>mulaw</CODE> or
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<CODE>imaadpcm</CODE>, <CODE>float</CODE> or <CODE>int</CODE>,
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@ -377,9 +416,9 @@
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background is gone. This filter has two switches:</P>
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<DL>
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<DT><CODE>v</CODE></DT>
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<DD>is a floating point number between -200 and +60 which represents the
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volume level in dB. The default level is -10dB.</DD>
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<DT><CODE>v <-200 - +60></CODE></DT>
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<DD>is a floating point number between <CODE>-200</CODE> and <CODE>+60</CODE>
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which represents the volume level in dB. The default level is -10dB.</DD>
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<DT><CODE>c</CODE></DT>
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<DD>is a binary control that turns soft clipping on and off. Soft-clipping can
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@ -402,7 +441,7 @@
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<H5><A NAME="af_equalizer">2.3.2.3.6 Equalizer</A></H5>
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<P> This filter is a 10 octave band graphic equalizer, implemented using 10 IIR
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<P>This filter is a 10 octave band graphic equalizer, implemented using 10 IIR
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band pass filters. This means that it works regardless of what type of audio
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is being played back. The center frequencies for the 10 bands are:</P>
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@ -427,12 +466,12 @@
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that band. This problem can be worked around by up-sampling the sound using
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the resample filter before it reaches this filter. </P>
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<P> This filter has 10 parameters:</P>
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<P>This filter has 10 parameters:</P>
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<DL>
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<DT><CODE>g1:g2:g3...g10</CODE></DT>
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<DD>are floating point numbers between -12 to +12dB representing the gain in
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dB for each frequency band.</DD>
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<DD>are floating point numbers between <CODE>-12<CODE> and <CODE>+12</CODE>
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representing the gain in dB for each frequency band.</DD>
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</DL>
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<P>Example:<BR>
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@ -457,14 +496,15 @@
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the number of output channels:</P>
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<DL>
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<DT><CODE>nch</CODE></DT>
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<DD>is an integer between 1 and 6 and is used for setting the number of output
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channels. This switch is required, leaving it empty results in a runtime
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error.</DD>
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<DT><CODE>nch <1-6></CODE></DT>
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<DD>is an integer between <CODE>1</CODE> and <CODE>6</CODE> and is used for
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setting the number of output channels. This switch is required, leaving it
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empty results in a runtime error.</DD>
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<DT><CODE>l00:l01:l02:..l10:l11:l12:...ln0:ln1:ln2:...</CODE></DT>
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<DD>are floating point values between 0 and 1. <CODE>l[i][j]</CODE> determines
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how much of input channel j is mixed into output channel i.</DD>
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<DD>are floating point values between <CODE>0</CODE> and <CODE>1</CODE>.
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<CODE>l[i][j]</CODE> determines how much of input channel j is mixed into
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output channel i.</DD>
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</DL>
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<P>Example 1:<BR>
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@ -480,6 +520,65 @@
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example).</P>
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<H5><A NAME="af_sub">2.3.2.3.8 Sub-woofer</A></H5>
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<P>This filter adds a sub woofer channel to the audio stream. The audio data
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used for creating the sub-woofer channel is an average of the sound in channel
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0 and channel 1. The resulting sound is then low-pass filtered by a a 4th
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order Butterworth filter with a default cutoff frequency of 60Hz and added to
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a separate channel in the audio stream. Warning: Disable this filter when you
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are playing DVDs with Dolby Digital 5.1 sound, otherwise this filter will
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disrupt the sound to the sub-woofer. This filter has two parameters:</P>
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<DL>
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<DT><CODE>fc <20-300></CODE></DT>
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<DD>is an optional floating point number used for setting the cutoff frequency
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for the filter in Hz. The valid range is 20Hz to 300Hz. For the best result
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try setting the cutoff frequency as low as possible. This will improve the
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stereo or surround sound experience. The default cutoff frequency is
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60Hz.</DD>
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<DT><CODE>ch <0-5></CODE></DT>
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<DD>is an optional integer between <CODE>0</CODE> and <CODE>5</CODE> which
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determines the channel number in which to insert the sub-channel audio.
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The default is channel number <CODE>5</CODE>. Observe that the number of
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channels will automatically be increased to <CODE>ch</CODE> if
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necessary.</DD>
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</DL>
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<P>Example:<BR>
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<CODE>mplayer -af sub=100:4 -channels 5 media.avi</CODE></P>
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<P>would add a sub-woofer channel with a cutoff frequency of 100Hz to output
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channel 4.</P>
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<H5><A NAME="af_surround">2.3.2.3.9 Surround-sound decoder</A></H5>
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<P>This filter is a decoder for matrix encoded surround sound. Dolby Surround is
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an example of a matrix encoded format. Many files with 2 channel audio
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actually contain matrixed surround sound. To use this feature you need a sound
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card supporting at least 4 channels. This filter has one parameter:</P>
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<DL>
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<DT><CODE>d <0-1000></CODE></DT>
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<DD>is an optional floating point number between <CODE>0</CODE> and
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<CODE>1000</CODE> used for setting the delay time in ms for the rear
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speakers. This delay should be set as follows: if d1 is the distance from
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the listening position to the front speakers and d2 is the distance from
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the listening position to the rear speakers, then the delay <CODE>d</CODE>
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should be set to 15ms if d1 <= d2 and to 15 + 5*(d1-d2) if d1 > d2.
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The default value for <CODE>d</CODE> is 20ms.</DD>
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</DL>
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<P>Example:<BR>
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<CODE>mplayer -af surround=15 -channels 4 media.avi</CODE></P>
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<P>would add a surround sound decoding with 15ms delay for the sound to the rear
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speakers.</P>
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<H2><STRONG>Note: Audio plugins have been deprecated by audio filters and will be
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removed soon.</STRONG></H2>
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Reference in New Issue