.. _audio_filters: AUDIO FILTERS ============= Audio filters allow you to modify the audio stream and its properties. The syntax is: --af= Setup a chain of audio filters. *NOTE*: To get a full list of available audio filters, see ``--af=help``. Audio filters are managed in lists. There are a few commands to manage the filter list. --af-add= Appends the filters given as arguments to the filter list. --af-pre= Prepends the filters given as arguments to the filter list. --af-del= Deletes the filters at the given indexes. Index numbers start at 0, negative numbers address the end of the list (-1 is the last). --af-clr Completely empties the filter list. Available filters are: 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 16-bit integer and float in native-endian format as input. 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. Allow (1) or disallow (0) the output frequency to differ slightly from the frequency given by (default: 1). Can be used if the startup of the playback is extremely slow. Select which resampling method to use. :0: linear interpolation (fast, poor quality especially when upsampling) :1: polyphase filterbank and integer processing :2: polyphase filterbank and floating point processing (slow, best quality) *EXAMPLE*: ``mpv --af=resample=44100:0:0`` would set the output frequency of the resample filter to 44100Hz using exact output frequency scaling and linear interpolation. lavcresample[=srate[:length[:linear[:count[:cutoff]]]]] Changes the sample rate of the audio stream to an integer in Hz. It only supports the 16-bit native-endian format. the output sample rate length of the filter with respect to the lower sampling rate (default: 16) if 1 then filters will be linearly interpolated between polyphase entries log2 of the number of polyphase entries (..., 10->1024, 11->2048, 12->4096, ...) (default: 10->1024) cutoff frequency (0.0-1.0), default set depending upon filter length lavcac3enc[=tospdif[:bitrate[:minchn]]] Encode multi-channel audio to AC-3 at runtime using libavcodec. Supports 16-bit native-endian input format, maximum 6 channels. The output is big-endian when outputting a raw AC-3 stream, native-endian when outputting to S/PDIF. The output sample rate of this filter is same with the input sample rate. When input sample rate is 48kHz, 44.1kHz, or 32kHz, this filter directly use it. Otherwise a resampling filter is auto-inserted before this filter to make the input and output sample rate be 48kHz. You need to specify ``--channels=N`` to make the decoder decode audio into N-channel, then the filter can encode the N-channel input to AC-3. Output raw AC-3 stream if zero or not set, output to S/PDIF for passthrough when is set non-zero. The bitrate to encode the AC-3 stream. Set it to either 384 or 384000 to get 384kbits. Valid values: 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384, 448, 512, 576, 640. Default bitrate is based on the input channel number: :1ch: 96 :2ch: 192 :3ch: 224 :4ch: 384 :5ch: 448 :6ch: 448 If the input channel number is less than , the filter will detach itself (default: 5). sweep[=speed] Produces a sine sweep. <0.0-1.0> Sine function delta, use very low values to hear the sweep. sinesuppress[=freq:decay] Remove a sine at the specified frequency. Useful to get rid of the 50/60Hz noise on low quality audio equipment. It probably only works on mono input. The frequency of the sine which should be removed (in Hz) (default: 50) Controls the adaptivity (a larger value will make the filter adapt to amplitude and phase changes quicker, a smaller value will make the adaptation slower) (default: 0.0001). Reasonable values are around 0.001. bs2b[=option1:option2:...] Bauer stereophonic to binaural transformation using ``libbs2b``. Improves the headphone listening experience by making the sound similar to that from loudspeakers, allowing each ear to hear both channels and taking into account the distance difference and the head shadowing effect. It is applicable only to 2 channel audio. fcut=<300-1000> Set cut frequency in Hz. feed=<10-150> Set feed level for low frequencies in 0.1*dB. profile= Several profiles are available for convenience: :default: will be used if nothing else was specified (fcut=700, feed=45) :cmoy: Chu Moy circuit implementation (fcut=700, feed=60) :jmeier: Jan Meier circuit implementation (fcut=650, feed=95) If fcut or feed options are specified together with a profile, they will be applied on top of the selected profile. hrtf[=flag] Head-related transfer function: Converts multichannel audio to 2 channel output for headphones, preserving the spatiality of the sound. ==== =================================== Flag Meaning ==== =================================== m matrix decoding of the rear channel s 2-channel matrix decoding 0 no matrix decoding (default) ==== =================================== 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: === ========== No. frequency === ========== 0 31.25 Hz 1 62.50 Hz 2 125.00 Hz 3 250.00 Hz 4 500.00 Hz 5 1.00 kHz 6 2.00 kHz 7 4.00 kHz 8 8.00 kHz 9 16.00 kHz === ========== 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. :::...: floating point numbers representing the gain in dB for each frequency band (-12-12) *EXAMPLE*: ``mpv --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. channels=nch[:nr:from1:to1:from2:to2:from3:to3:...] Can be used for adding, removing, routing and copying audio channels. If only 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. number of output channels (1-8) number of routes (1-8) Pairs of numbers between 0 and 7 that define where to route each channel. *EXAMPLE*: ``mpv --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. ``mpv --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. format[=format] Convert between different sample formats. Automatically enabled when needed by the sound card or another filter. See also ``--format``. Sets the desired format. The general form is 'sbe', where 's' denotes the sign (either 's' for signed 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 mpv is running on). Valid values (amongst others) are: 's16le', 'u32be' and 'u24ne'. Exceptions to this rule that are also valid format specifiers: u8, s8, floatle, floatbe, floatne, mulaw, alaw, mpeg2, ac3 and imaadpcm. 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. This filter has a second feature: It measures the overall maximum sound level and prints out that level when mpv exits. This feature currently only works with floating-point data, use e.g. ``--af-adv=force=5``, or use ``--af=stats``. *NOTE*: This filter is not reentrant and can therefore only be enabled once for every audio stream. 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 (default: 0). 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. *WARNING*: This feature creates distortion and should be considered a last resort. *EXAMPLE*: ``mpv --af=volume=10.1:0 media.avi`` Would amplify the sound by 10.1dB and hard-clip if the sound level is too high. pan=n[:L00: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. number of output channels (1-8) How much of input channel i is mixed into output channel j (0-1). So in principle you first have n numbers saying what to do with the first input channel, then n numbers that act on the second input channel etc. If you do not specify any numbers for some input channels, 0 is assumed. *EXAMPLE*: ``mpv --af=pan=1:0.5:0.5 media.avi`` Would down-mix from stereo to mono. ``mpv --af=pan=3:1:0:0.5:0:1:0.5 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). sub[=fc:ch] 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. *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. 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. Determines the channel number in which to insert the sub-channel audio. Channel number can be between 0 and 7 (default: 5). Observe that the number of channels will automatically be increased to if necessary. *EXAMPLE*: ``mpv --af=sub=100:4 --channels=5 media.avi`` Would add a sub-woofer channel with a cutoff frequency of 100Hz to output channel 4. center Creates a center channel from the front channels. May currently be low quality as it does not implement a high-pass filter for proper extraction yet, but averages and halves the channels instead. Determines the channel number in which to insert the center channel. Channel number can be between 0 and 7 (default: 5). Observe that the number of channels will automatically be increased to if necessary. surround[=delay] 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. 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 should be set to 15ms if d1 <= d2 and to 15 + 5*(d1-d2) if d1 > d2. *EXAMPLE*: ``mpv --af=surround=15 --channels=4 media.avi`` Would add surround sound decoding with 15ms delay for the sound to the rear speakers. delay[=ch1:ch2:...] 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. ch1,ch2,... The delay in ms that should be imposed on each channel (floating point number between 0 and 1000). To calculate the required delay for the different channels do as follows: 1. 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). 2. Subtract the distances s1 to s5 from the maximum distance, i.e. ``s[i] = max(s) - s[i]; i = 1...5``. 3. Calculate the required delays in ms as ``d[i] = 1000*s[i]/342; i = 1...5``. *EXAMPLE*: ``mpv --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. export[=mmapped_file[:nsamples]] Exports the incoming signal to other processes using memory mapping (``mmap()``). Memory mapped areas contain a header: | int nch /\* number of channels \*/ | int size /\* buffer size \*/ | unsigned long long counter /\* Used to keep sync, updated every time new data is exported. \*/ The rest is payload (non-interleaved) 16 bit data. file to map data to (default: ``~/.mpv/mpv-af_export``) number of samples per channel (default: 512) *EXAMPLE*: ``mpv --af=export=/tmp/mpv-af_export:1024 media.avi`` Would export 1024 samples per channel to ``/tmp/mpv-af_export``. extrastereo[=mul] (Linearly) increases the difference between left and right channels which adds some sort of "live" effect to playback. 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. volnorm[=method:target] Maximizes the volume without distorting the sound. Sets the used method. 1 Use a single sample to smooth the variations via the standard weighted mean over past samples (default). 2 Use several samples to smooth the variations via the standard weighted mean over past samples. Sets the target amplitude as a fraction of the maximum for the sample type (default: 0.25). ladspa=file:label[:controls...] Load a LADSPA (Linux Audio Developer's Simple Plugin API) plugin. This filter is reentrant, so multiple LADSPA plugins can be used at once. Specifies the LADSPA plugin library file. If ``LADSPA_PATH`` is set, it searches for the specified file. If it is not set, you must supply a fully specified pathname.