[MPlayer-dev-eng] DOCS bug

[Diego Biurrun]

Anders Johansson wrote:
>>IIRC you added some features during the last days (subwoofer etc) that
>>might be worth documenting ;-) (hint, hint ;-)))
> 
> Patch attached.

Thanks, applied.

Diego

Some comments:

>        <DD>Use automatic insertion of filters and optimize for the highest speed.
> -        If this option is set the processing of the audio data will be done
> -        using fix point arithmetics. Warning: Some features in the audio filters
> -        will silently fail, and the sound quality may drop.</DD>
> +        Warning: Some features in the audio filters may
> +        silently fail, and the sound quality may drop.</DD>
>        <DT>2</DT>
> -      <DD>Use automatic insertion of filters and optimize for quality. If this
> -        option is set the processing of the audio data will be done using
> -        floating point instructions and is therefore quite CPU intensive, but
> -        gives a lot higher sound quality than fix point processing.</DD>
> +      <DD>Use automatic insertion of filters and optimize for quality.</DD>
>        <DT>3</DT>
>        <DD>Use no automatic insertion of filters and no optimization. Warning: It
>          may be possible to crash MPlayer using this setting.</DD>
> +      <DT>4</DT>
> +      <DD>Use automatic insertion of filters according to 0 above, but use
> +      floating point processing when possible.</DD>
> +      <DT>5</DT>
> +      <DD>Use automatic insertion of filters according to 1 above, but use
> +      floating point processing when possible.</DD>
> +      <DT>6</DT>
> +      <DD>Use automatic insertion of filters according to 2 above, but use
> +      floating point processing when possible.</DD>
> +      <DT>7</DT>
> +      <DD>Use no automatic insertion of filters according to 3 above, and use
> +      floating point processing when possible.</DD>
>      </DL>
>    </DD>
>    
> @@ -223,6 +228,30 @@
>    <DD>is an alias for the -af switch.</DD>
>  </DL>
>  
> +<P>The filter layer is also affected by the following generic switches:
> +<DL>
> +  <DT><CODE>-v</CODE></DT>
> +  <DD>Increases the verbosity level and makes most filters print out
> +    extra status in messages.</DD>
> +  <DT><CODE>-channels</CODE></DT>
> +  <DD>This option sets the number of output channels your sound-card
> +    is using. It also affects the number of channels that are being
> +    decoded from the media. If the media contains less channels than
> +    requested the channels filter (see below) will automatically
> +    inserted. The routing will be the default routing for the channels
> +    filter.</DD>
> +  <DT><CODE>-srate</CODE></DT>
> +  <DD>This option selects the sample rate of your sound-card. If the
> +    sample frequency of your sound-card is different from that of the
> +    current media, the resample filter (see below) will be inserted
> +    into the audio filter layer to compensate for the difference.</DD>
> +  <DT><CODE>-format</CODE><DT>
> +  <DD>This option sets the sample format of the audio filter layer and
> +    of the sound-card. If the requested sample format of your
> +    sound-card is different from that of the current media, a format
> +    filter (see below) will be inserted to rectify the
> +    difference.</DD>
> +</DL>
>  
>  <H5><A NAME="af_resample">2.3.2.3.1 Up/Down-sampling</A></H5>
>  
> @@ -244,12 +273,18 @@
>    <DT><CODE>sloppy</CODE></DT>
>    <DD>is an optional binary parameter that allows the output frequency to differ
>      slightly from the frequency given by <CODE>srate</CODE>. This switch can be
> -    used if the startup of the playback is extremely slow.</DD>
> +    used if the startup of the playback is extremely slow. This option
> +    is enabled by default.</DD>
>  
> -  <DT><CODE>fast</CODE><DT>
> -  <DD>is an optional binary parameter that enables linear interpolation as
> -    resampling method. Linear interpolation is extremely fast, but suffers from
> -    poor sound quality especially when used for up-sampling.</DD>
> +  <DT><CODE>type</CODE><DT>
> +  <DD>is an optional integer between 0 and 2 that selects which
> +    resampling method to use. Here 0 represents linear interpolation
> +    as resampling method, 1 represents resampling using a poly-phase
> +    filter-bank and integer processing and 2 represents resampling
> +    using a poly-phase filter-bank and floating point processing. Linear
> +    interpolation is extremely fast, but suffers from poor sound
> +    quality especially when used for up-sampling. The best quality is
> +    given by 2 but this method also suffers from the highest CPU load.</DD> 
>  </DL>
>  
>  <P>Example:<BR>
> @@ -364,7 +399,6 @@
>  <P>would delay front left and right by 10.5ms, the two rear channels and the sub
>    by 0ms and the center channel by 7ms.</P>
>  
> -
>  <H5><A NAME="af_volume">2.3.2.3.5 Software volume control</A></H5>
>  
>  <P>This filter is a software volume control. Use this filter with caution since
> @@ -478,6 +512,68 @@
>  <P>would give 3 channel output leaving channels 0 and 1 intact, and mix channels
>    0 and 1 into output channel 2 (which could be sent to a sub-woofer for
>    example).</P>
> +
> +
> +<H5><A NAME="af_sub">2.3.2.3.5 Sub-woofer</A></H5>
> +
> +<P>This filter adds a sub woofer channel to the audio stream. The
> +  audio data used for creating the sub-woofer channel is an average of
> +  the sound in channel 0 and channel 1. The resulting sound is then
> +  low-pass filtered by a a 4th order Butterworth filter with a default
> +  cutoff frequency of 60Hz, and added to a separate channel in the
> +  audio stream. Warning: Disable this filter when you are playing DVDs
> +  with Dolby Digital 5.1 sound, otherwise this filter will disrupt the
> +  sound to the sub-woofer. This filter has two parameters:</P>
> +
> +<DL>
> +  <DT><CODE>fc</CODE></DT>
> +  <DD>is an optional floating point number used for setting the cutoff
> +    frequency for the filter in Hz. The valid range is 20Hz to
> +    300Hz. For the best result try setting the cutoff frequency as low
> +    as possible. This will improve the stereo or surround sound
> +    experience. The default cutoff frequency is 60Hz.</DD>
> +
> +  <DT><CODE>ch</CODE></DT>
> +  <DD>is an optional integer between 0 and 5 which determines the
> +    channel number in which to insert the sub-channel audio. The
> +    default is channel number 5. Observe that the number of channels
> +    will automatically be increased to <CODE>ch</CODE> if
> +    necessary.</DD>
> +</DL>
> +
> +<P>Example:<BR>
> +  &nbsp;&nbsp;<CODE>mplayer -af sub=100:4 -channels 5 media.avi</CODE></P>
> +
> +<P>would add a sub-woofer channel with a cutoff frequency of 100Hz to
> +  output channel 4.</P>
> +
> +<H5><A NAME="af_surround">2.3.2.3.6 Surround-sound decoder</A></H5>
> +
> +<P> This filter is a decoder for matrix encoded surround sound. Dolby
> +  Surround is an example of a matrix encoded format. Many files with
> +  2 channel audio actually contain matrixed surround sound. To use
> +  this feature you need a sound card supporting at least 4 channels.
> +  This filter has one parameter:</P>
> +
> +<DL>
> +  <DT><CODE>d</CODE></DT>
> +  <DD>is an optional floating point number between 0 and 1000 used for
> +    setting the delay time in ms for the rear speakers. This delay
> +    should be set as follows: if d1 is the distance from the listening
> +    position to the front speakers and d2 is the distance from the
> +    listening position to the rear speakers, then the delay
> +    <CODE>d</CODE> should be set to 15ms if d1 <= d2 and to 15 +
> +    5*(d1-d2) if d1 > d2. The default value for <CODE>d</CODE> is
> +    20ms. </DD>
> +</DL>
> +
> +<P>Example:<BR>
> +  &nbsp;&nbsp;<CODE>mplayer -af surround=15 -channels 4 media.avi</CODE></P>
> +
> +<P>would add a surround sound decoding with 15ms delay for the sound
> +  to the rear speakers. </P>