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author | diego <diego@b3059339-0415-0410-9bf9-f77b7e298cf2> | 2005-02-24 11:00:45 +0000 |
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committer | diego <diego@b3059339-0415-0410-9bf9-f77b7e298cf2> | 2005-02-24 11:00:45 +0000 |
commit | 201691be524b294e30ff5a935e625655bdf38555 (patch) | |
tree | e99530c09156cc865c89a0d258c9b47aa98e411f /DOCS/man | |
parent | 55aa474d7e70016da5f36edd2251de1b476a50ba (diff) | |
download | mpv-201691be524b294e30ff5a935e625655bdf38555.tar.bz2 mpv-201691be524b294e30ff5a935e625655bdf38555.tar.xz |
Move audio filter descriptions to the man page.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@14788 b3059339-0415-0410-9bf9-f77b7e298cf2
Diffstat (limited to 'DOCS/man')
-rw-r--r-- | DOCS/man/en/mplayer.1 | 335 |
1 files changed, 299 insertions, 36 deletions
diff --git a/DOCS/man/en/mplayer.1 b/DOCS/man/en/mplayer.1 index 778332bedf..de422906b7 100644 --- a/DOCS/man/en/mplayer.1 +++ b/DOCS/man/en/mplayer.1 @@ -3507,15 +3507,57 @@ To get a full list of available audio filters, see \-af help. Available filters are: . .TP -.B resample[=srate[:sloppy][:type]] -Changes the sample rate of the audio stream to an integer srate in Hz. +.B resample[=srate[:sloppy[:type]]] +Changes the sample rate of the audio stream. +Can be used if you have a fixed frequency sound card or if you are +stuck with an old sound card that is only capable of max 44.1kHz. +This filter is automatically enabled if necessary. It only supports the 16-bit little-endian format. +.br +.I NOTE: With MEncoder, you need to also use \-srate <srate>. +.PD 0 +.RSs +.IPs <srate> +output sample frequency in Hz. +The valid range for this parameter is 8000 to 192000. +If the input and output sample frequency are the same or if this +parameter is omitted the filter is automatically unloaded. +A high sample frequency normally improves the audio quality, +especially when used in combination with other filters. +.IPs <sloppy> +Allow (1) or disallow (0) the output frequency to differ slightly +from the frequency given by <srate> (default: 1). +Can be used if the startup of the playback is extremely slow. +.IPs <type> +Selects which resampling method to use. +.RSss +0: linear interpolation (fast, poor quality especially when upsampling) +.br +1: polyphase filterbank and integer processing +.br +2: polyphase filterbank and floating point processing (slow, best quality) +.REss +.PD 1 +.RE +.sp 1 +.RS +.I EXAMPLE: +.RE +.PD 0 +.RSs +.IPs "mplayer -af resample=44100:0:0" +would set the output frequency of the resample filter to 44100Hz using +exact output frequency scaling and linear interpolation. +.RE +.PD 1 . .TP .B lavcresample[=srate[:length[:linear[:count[:cutoff]]]]] -Changes the sample rate of the audio stream to an integer srate in Hz. +Changes the sample rate of the audio stream to an integer <srate> in Hz. It only supports the 16-bit little-endian format. +.br +.I NOTE: With MEncoder, you need to also use \-srate <srate>. .PD 0 .RSs @@ -3544,17 +3586,93 @@ Head-related transfer function: Converts multichannel audio to 2 channel output for headphones, preserving the spatiality of the sound. . .TP -.B channels[=nch] -Change the number of channels to <nch> output channels. -If the number of output channels is bigger than the number of input channels -empty channels are inserted (except when mixing from mono to stereo, then -the mono channel is repeated in both of the output channels). -If the number of output channels is smaller than the number of input channels -the exceeding channels are truncated. +.B equalizer=[g1:g2:g3:...:g10] +10 octave band graphic equalizer, implemented using 10 IIR band pass filters. +This means that it works regardless of what type of audio is being played back. +The center frequencies for the 10 bands are: +.sp 1 +.PD 0 +.RS +.IPs "No. frequency" +.IPs "0 31.25 Hz" +.IPs "1 62.50 Hz" +.IPs "2 125.00 Hz" +.IPs "3 250.00 Hz" +.IPs "4 500.00 Hz" +.IPs "5 1.00 kHz" +.IPs "6 2.00 kHz" +.IPs "7 4.00 kHz" +.IPs "8 8.00 kHz" +.IPs "9 16.00 kHz" +.RE +.PD 1 +.sp 1 +.RS +If the sample rate of the sound being played is lower than the center +frequency for a frequency band, then that band will be disabled. +A known bug with this filter is that the characteristics for the +uppermost band are not completely symmetric if the sample +rate is close to the center frequency of that band. +This problem can be worked around by upsampling the sound +using the resample filter before it reaches this filter. +.RE +.PD 0 +.RSs +.IPs <g1>:<g2>:<g3>:...:<g10> +floating point numbers representing the gain in dB +for each frequency band (-12\-12) +.RE +.sp 1 +.RS +.I EXAMPLE: +.RE +.RSs +.IPs "mplayer \-af equalizer=11:11:10:5:0:-12:0:5:12:12 media.avi" +Would amplify the sound in the upper and lower frequency region +while canceling it almost completely around 1kHz. +.RE +.PD 1 +. +.TP +.B channels=nch[:nr:from1:to1:from2:to2:from3:to3:...] +Can be used for adding, removing, routing and copying audio channels. +If only <nch> is given the default routing is used, it works as +follows: If the number of output channels is bigger than the number of +input channels empty channels are inserted (except mixing from mono to +stereo, then the mono channel is repeated in both of the output +channels). +If the number of output channels is smaller than the number +of input channels the exceeding channels are truncated. +.PD 0 +.RSs +.IPs <nch> +number of output channels (1\-6) +.IPs <nr>\ +number of routes (1\-6) +.IPs <from1:to1:from2:to2:from3:to3:...> +Pairs of numbers between 0 and 5 that define where to route each channel. +.RE +.sp 1 +.RS +.I EXAMPLE: +.RE +.RSs +.IPs "mplayer -af channels=4:4:0:1:1:0:2:2:3:3 media.avi" +Would change the number of channels to 4 and set up 4 routes that +swap channel 0 and channel 1 and leave channel 2 and 3 intact. +Observe that if media containing two channels was played back, channels +2 and 3 would contain silence but 0 and 1 would still be swapped. +.IPs "mplayer -af channels=6:4:0:0:0:1:0:2:0:3 media.avi" +Would change the number of channels to 6 and set up 4 routes +that copy channel 0 to channels 0 to 3. +Channel 4 and 5 will contain silence. +.RE +.PD 1 . .TP .B format[=format] -Change the current sample format. +Convert between different sample formats. +Automatically enabled when needed by the sound card or another filter. .PD 0 .RSs .IPs <format> @@ -3564,72 +3682,207 @@ or 'u' for unsigned), 'b' denotes the number of bits per sample (16, 24 or 32) and 'e' denotes the endianness ('le' means little-endian, 'be' big-endian and 'ne' the endianness of the computer MPlayer is running on). Valid values (amongst others) are: 's16le', 'u32be' and 'u24ne'. -Exceptions to this rule are: u8, s8, floatle, floatbe, floatne, mulaw, alaw, -mpeg2, ac3 and imaadpcm. +Exceptions to this rule that are also valid format specifiers: u8, s8, +floatle, floatbe, floatne, mulaw, alaw, mpeg2, ac3 and imaadpcm. .RE .PD 1 . .TP -.B volume[=v:sc] -Select the output volume level. -This filter is not reentrant and can therefore only be enabled once for every -audio stream. +.B volume[=v[:sc]] +Implements software volume control. +Use this filter with caution since it can reduce the signal +to noise ratio of the sound. +In most cases it is best to set the level for the PCM sound to max, +leave this filter out and control the output level to your +speakers with the master volume control of the mixer. +In case your sound card has a digital PCM mixer instead of an analog +one, and you hear distortion, use the MASTER mixer instead. +If there is an external amplifier connected to the computer (this +is almost always the case), the noise level can be minimized by +adjusting the master level and the volume knob on the amplifier +until the hissing noise in the background is gone. +.br +This filter has a second feature: It measures the overall maximum +sound level and prints out that level when MPlayer exits. +This volume estimate can be used for setting the sound level in +MEncoder such that the maximum dynamic range is utilized. +.br +.I NOTE: +This filter is not reentrant and can therefore only be enabled +once for every audio stream. .PD 0 .RSs .IPs <v>\ \ Sets the desired gain in dB for all channels in the stream -from -200dB to +60dB (where -200dB mutes the sound -completely and +60dB equals a gain of 1000). +from -200dB to +60dB, where -200dB mutes the sound +completely and +60dB equals a gain of 1000 (default: 0). .IPs <sc>\ -Enable soft clipping. +Turns soft clipping on (1) or off (0). +Soft-clipping can make the sound more smooth if very +high volume levels are used. +Enable this option if the dynamic range of the +loudspeakers is very low. +.br +.I WARNING: +This feature creates distortion and should be considered a last resort. +.RE +.sp 1 +.RS +.I EXAMPLE: +.RE +.RSs +.IPs "mplayer -af volume=10.1:0 media.avi" +would amplify the sound by 10.1dB and hard-clip if the +sound level is too high. .RE .PD 1 . .TP -.B pan[=n:l01:l02:...l10:l11:l12:...ln0:ln1:ln2:...] -Mixes channels arbitrarily, see DOCS/\:HTML/\:en/\:audio.html for details. -An example how to downmix a six-channel file to two channels with this -filter can be found in the examples section near the end of the man page. +.B pan=n[:l01:l02:...l10:l11:l12:...ln0:ln1:ln2:...] +Mixes channels arbitrarily. +Basically a combination of the volume and the channels filter +that can be used to down-mix many channels to only a few, +e.g.\& stereo to mono or vary the "width" of the center +speaker in a surround sound system. +This filter is hard to use, and will require some tinkering +before the desired result is obtained. +The number of options for this filter depends on +the number of output channels. +An example how to downmix a six-channel file to two channels with +this filter can be found in the examples section near the end. .PD 0 .RSs .IPs <n>\ \ number of input channels (1\-6) .IPs <lij> -How much of input channel j is mixed into output channel i. +How much of input channel j is mixed into output channel i (0\-1). +.RE +.sp 1 +.RS +.I EXAMPLE: +.RE +.RSs +.IPs "mplayer -af pan=1:0.5:0.5 -channels 1 media.avi" +Would down-mix from stereo to mono. +.IPs "mplayer -af pan=3:1:0:1:0.5:0.5 -channels 3 media.avi" +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 subwoofer for example). .RE .PD 1 . .TP .B sub[=fc:ch] -Add subwoofer channel. +Adds a subwoofer channel to the audio stream. +The audio data used for creating the subwoofer channel is +an average of the sound in channel 0 and channel 1. +The resulting sound is then low-pass filtered by a 4th order +Butterworth filter with a default cutoff frequency of 60Hz +and added to a separate channel in the audio stream. +.br +.I Warning: +Disable this filter when you are playing DVDs with Dolby +Digital 5.1 sound, otherwise this filter will disrupt +the sound to the subwoofer. .PD 0 .RSs .IPs <fc>\ -cutoff frequency for low-pass filter (20Hz to 300Hz) (default: 60Hz) +cutoff frequency in Hz for the low-pass filter (20Hz to 300Hz) (default: 60Hz) +For the best result try setting the cutoff frequency as low as possible. +This will improve the stereo or surround sound experience. .IPs <ch>\ -channel number for the sub-channel +Determines the channel number in which to insert the sub-channel audio. +Channel number can be between 0 and 5 (default: 5). +Observe that the number of channels will automatically +be increased to <ch> if necessary. +.RE +.sp 1 +.RS +.I EXAMPLE: +.RE +.RSs +.IPs "mplayer -af sub=100:4 -channels 5 media.avi" +would add a sub-woofer channel with a cutoff frequency of +100Hz to output channel 4. .RE .PD 1 . .TP .B surround[=delay] -Decoder for matrix encoded surround sound, works on many 2 channel files. +Decoder for matrix encoded surround sound like Dolby Surround. +Many files with 2 channel audio actually contain matrixed surround sound. +Requires a sound card supporting at least 4 channels. .PD 0 .RSs .IPs <delay> delay time in ms for the rear speakers (0 to 1000) (default: 20) +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 d should +be set to 15ms if d1 <= d2 and to 15 + 5*(d1-d2) if d1 > d2. +.RE +.sp 1 +.RS +.I EXAMPLE: +.RE +.RSs +.IPs "mplayer \-af surround=15 \-channels 4 media.avi" +Would add surround sound decoding with 15ms delay for the sound to the +rear speakers. .RE .PD 1 . .TP .B delay[=ch1:ch2:...] -Delays the sound output. -Specify the delay separately for each channel in milliseconds (floating point -number between 0 and 1000). +Delays the sound to the loudspeakers such that the sound from the +different channels arrives at the listening position simultaneously. +It is only useful if you have more than 2 loudspeakers. +.PD 0 +.RSs +.IPs ch1,ch2,... +The delay in ms that should be imposed on each channel +(floating point number between 0 and 1000). +.RE +.PD 1 +.sp 1 +.RS +To calculate the required delay for the different channels do as follows: +.IP 1. 3 +Measure the distance to the loudspeakers in meters in relation +to your listening position, giving you the distances s1 to s5 +(for a 5.1 system). There is no point in compensating for the +subwoofer (you will not hear the difference anyway). +.IP 2. 3 +Subtract the distances s1 to s5 from the maximum distance, +i.e.\& s[i] = max(s) - s[i]; i = 1...5. +.IP 3. +Calculate the required delays in ms as d[i] = 1000*s[i]/342; i = 1...5. +.RE +.PD 0 +.sp 1 +.RS +.I EXAMPLE: +.RE +.RSs +.IPs "mplayer \-af delay=10.5:10.5:0:0:7:0 media.avi" +Would delay front left and right by 10.5ms, the two rear channels +and the sub by 0ms and the center channel by 7ms. +.RE +.PD 1 . .TP .B export[=mmapped_file[:nsamples]] Exports the incoming signal to other processes using memory mapping (mmap()). +Memory mapped areas contain a header: +.sp 1 +.nf +int nch /*number of channels*/ +int size /*buffer size*/ +unsigned long long counter /*Used to keep sync, updated every + time new data is exported.*/ +.fi +.sp 1 +The rest is payload (non-interleaved) 16 bit data. .PD 0 .RSs .IPs <mmapped_file> @@ -3637,16 +3890,26 @@ file to map data to (default: ~/.mplayer/\:mplayer-af_export) .IPs <nsamples> number of samples per channel (default: 512) .RE +.sp 1 +.RS +.I EXAMPLE: +.RE +.RSs +.IPs "mplayer \-af export=/tmp/mplayer-af_export:1024 media.avi" +Would export 1024 samples per channel to '/tmp/mplayer-af_export'. +.RE .PD 1 . .TP .B extrastereo[=mul] -Increases the difference between left and right channels to add some -sort of "live" effect to playback. +(Linearly) increases the difference between left and right channels +which adds some sort of "live" effect to playback. .PD 0 .RSs .IPs <mul> -difference coefficient (default: 2.5) +Sets the difference coefficient (default: 2.5). +0.0 means mono sound (average of both channels), with 1.0 sound will be +unchanged, with -1.0 left and right channels will be swapped. .RE .PD 1 . @@ -3703,7 +3966,7 @@ from the upper left corner of the bigger image. .I NOTE: To get a full list of available video filters, see \-vf help. .sp 1 -Filters are managed in lists. +Video filters are managed in lists. There are a few commands to manage the filter list. . .TP |