[MPlayer-dev-eng] DOCS bug
Diego Biurrun
diego at biurrun.de
Sat Jan 11 14:04:33 CET 2003
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>
> + <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>
> + <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>
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