MPlayer's audio interface is called libao2. It currently contains these drivers: DriverComment oss OSS (ioctl) driver (supports hardware AC3 passthrough) sdl SDL driver (supports sound daemons like ESD and ARTS) nas NAS (Network Audio System) driver alsa5 native ALSA 0.5 driver alsa9 native ALSA 0.9 driver (supports hardware AC3 passthrough) sun SUN audio driver (/dev/audio) for BSD and Solaris8 users macosx native MacOSX driver win32 native Win32 driver arts native ARTS driver (mostly for KDE users) esd native ESD driver (mostly for GNOME users) Linux sound card drivers have compatibility problems. This is because MPlayer relies on an in-built feature of properly coded sound drivers that enable them to maintain correct audio/video sync. Regrettably, some driver authors don't take the care to code this feature since it is not needed for playing MP3s or sound effects. Other media players like aviplay or xine possibly work out-of-the-box with these drivers because they use "simple" methods with internal timing. Measuring showed that their methods are not as efficient as MPlayer's. Using MPlayer with a properly written audio driver will never result in A/V desyncs related to the audio, except only with very badly created files (check the man page for workarounds). If you happen to have a bad audio driver, try the -autosync option, it should sort out your problems. See the man page for detailed information. If you have an OSS driver, first try -ao oss (this is the default). If you experience glitches, halts or anything out of the ordinary, try -ao sdl (NOTE: you need to have SDL libraries and header files installed). The SDL audio driver helps in a lot of cases and also supports ESD (GNOME) and ARTS (KDE). If you have ALSA version 0.5, then you almost always have to use -ao alsa5, since ALSA 0.5 has buggy OSS emulation code, and will crash MPlayer with a message like this: DEMUXER: Too many (945 in 8390980 bytes) video packets in the buffer! On Solaris, use the SUN audio driver with the -ao sun option, otherwise neither video nor audio will work. If the sound clicks when playing from CD-ROM, turn on IRQ unmasking, e.g. hdparm -u1 /dev/cdrom (man hdparm). This is generally beneficial and described in more detail in the CD-ROM section. On Linux, a 2.4.x kernel is highly recommended. Kernel 2.2 is not tested. Linux sound drivers are primarily provided by the free version of OSS. These drivers have been superseded by ALSA (Advanced Linux Sound Architecture) in the 2.5 development series. If your distribution does not already use ALSA you may wish to try their drivers if you experience sound problems. ALSA drivers are generally superior to OSS in compatibility, performance and features. But some sound cards are only supported by the commercial OSS drivers from 4Front Technologies. They also support several non-Linux systems. SOUND CARD DRIVER Max kHz Max Channels Max Opens the number of applications that are able to use the device at the same time. OSS/Free ALSA OSS/Pro other VIA onboard (686/A/B, 8233, 8235) via82cxxx_audio snd-via82xx 4-48 kHz or 48 kHz only, depending on the chipset Aureal Vortex 2 none none OK Linux Aureal Drivers buffer size increased to 32k 48 4.1 5+ SB Live! Analog OK, S/PDIF not working Both OK Both OK Creative's OSS driver (S/PDIF support) 192 4.0/5.1 32 SB 128 PCI (es1371) OK ? 48 stereo 2 SB AWE 64 max 44kHz 48kHz sounds bad 48 GUS PnP none OK OK 48 Gravis UltraSound ACE Gravis UltraSound MAX OK OK (?) 48 ESS 688 OK OK (?) 48 C-Media cards (CMI8338/8738) OK OK S/PDIF is supported with ALSA 0.9.x ? 44 stereo 1 Yamaha cards (*ymf*) not OK (?) (maybe -ao sdl) OK only with ALSA 0.5 with OSS emulation AND -ao sdl (!) (?) Cards with envy24 chips (like Terratec EWS88MT) ? ? OK ? PC Speaker or DAC OK none Linux PC speaker OSS driver The driver emulates 44.1, maybe more. mono 1 Feedback to this document is welcome. Please tell us how MPlayer and your sound card(s) worked together. The old audio plugins have been superseded by a new audio filter layer. Audio filters are used for changing the properties of the audio data before the sound reaches the sound card. The activation and deactivation of the filters is normally automated but can be overridden. The filters are activated when the properties of the audio data differ from those required by the sound card and deactivated if unnecessary. The -af filter1,filter2,... option is used to override the automatic activation of filters or to insert filters that are not automatically inserted. The filters will be executed as they appear in the comma separated list. Example: mplayer -af resample,pan movie.avi would run the sound through the resampling filter followed by the pan filter. Observe that the list must not contain any spaces, else it will fail. The filters often have options that change their behavior. These options are explained in detail in the sections below. A filter will execute using default settings if its options are omitted. Here is an example of how to use filters in combination with filter specific options: mplayer -af resample=11025,pan=1:0.5:0.5 -channels 1 -srate 11025 media.avi would set the output frequency of the resample filter to 11025Hz and downmix the audio to 1 channel using the pan filter. The overall execution of the filter layer is controlled using the -af-adv option. This option has two suboptions: force is a bit field that controls how the filters are inserted and what speed/accuracy optimizations they use: 0 Use automatic insertion of filters and optimize according to CPU speed. 1 Use automatic insertion of filters and optimize for the highest speed. Warning: Some features in the audio filters may silently fail, and the sound quality may drop. 2 Use automatic insertion of filters and optimize for quality. 3 Use no automatic insertion of filters and no optimization. Warning: It may be possible to crash MPlayer using this setting. 4 Use automatic insertion of filters according to 0 above, but use floating point processing when possible. 5 Use automatic insertion of filters according to 1 above, but use floating point processing when possible. 6 Use automatic insertion of filters according to 2 above, but use floating point processing when possible. 7 Use no automatic insertion of filters according to 3 above, and use floating point processing when possible. list is an alias for the -af option. The filter layer is also affected by the following generic options: -v Increases the verbosity level and makes most filters print out extra status messages. -channels This option sets the number of output channels you would like your sound card to use. 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 be inserted. The routing will be the default routing for the channels filter. -srate This option selects the sample rate you would like your sound card to use (of course the cards have limits on this). 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. -format This option sets the sample format between the audio filter layer and 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. MPlayer fully supports sound up/down-sampling through the resample filter. It 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 it is necessary, but it can also be explicitly enabled on the command line. It has three options: srate <8000-192000> is an integer used for setting the output sample frequency in Hz. The valid range for this parameter is 8kHz to 192kHz. 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. sloppy is an optional binary parameter that allows the output frequency to differ slightly from the frequency given by srate. This option can be used if the startup of the playback is extremely slow. It is enabled by default. type <0-2> 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. Example: 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. The channels filter can be used for adding and removing channels, it can also be used for routing or copying channels. It is automatically enabled when the output from the audio filter layer differs from the input layer or when it is requested by another filter. This filter unloads itself if not needed. The number of options is dynamic: nch <1-6> is an integer between 1 and 6 that is used for setting the number of output channels. This option is required, leaving it empty results in a runtime error. nr <1-6> is an integer between 1 and 6 that is used for specifying the number of routes. This parameter is optional. If it is omitted the default routing is used. from1:to1:from2:to2:from3:to3... are pairs of numbers between 0 and 5 that define where each channel should be routed. 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. Example 1: 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. Example 2: 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. The format filter converts between different sample formats. It is automatically enabled when needed by the sound card or another filter. bps <number> can be 1, 2 or 4 and denotes the number of bytes per sample. This option is required, leaving it empty results in a runtime error. f <format> is a text string describing the sample format. The string is a concatenated mix of: alaw, mulaw or imaadpcm, float or int, unsigned or signed, le or be (little or big endian). This option is required, leaving it empty results in a runtime error. Example: mplayer -af format=4:float media.avi would set the output format to 4 bytes per sample floating point data. The delay filter 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. This filter has a variable number of parameters: d1:d2:d3... are floating point numbers representing the delays in ms that should be imposed on the different channels. The minimum delay is 0ms and the maximum is 1000ms. To calculate the required delay for the different channels do as follows: 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 sub-woofer (you will not hear the difference anyway). Subtract the distances s1 to s5 from the maximum distance i.e. s[i] = max(s) - s[i]; i = 1...5 Calculate the required delays in ms as d[i] = 1000*s[i]/342; i = 1...5 Example: 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. Software volume control is implemented by the volume audio filter. 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 two options: v <-200 - +60> is a floating point number between -200 and +60 which represents the volume level in dB. The default level is 0dB. c is a binary control that turns soft clipping on and off. 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. Be aware that this feature creates distortion and should be considered a last resort. Example: 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. 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. The equalizer filter represents a 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: Band No.Center frequency 031.25 Hz 162.50 Hz 2125.0 Hz 3250.0 Hz 4500.0 Hz 51.000 kHz 62.000 kHz 74.000 kHz 88.000 kHz 916.00 kHz If the sample rate of the sound being played back 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 up-sampling the sound using the resample filter before it reaches this filter. This filter has 10 parameters: g1:g2:g3...g10 are floating point numbers between -12 and +12 representing the gain in dB for each frequency band. Example: 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. Use the pan filter to mix channels arbitrarily. It is basically a combination of the volume control and the channels filter. There are two major uses for this filter: Down-mixing many channels to only a few, stereo to mono for example. Varying 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: nch <1-6> is an integer between 1 and 6 and is used for setting the number of output channels. This option is required, leaving it empty results in a runtime error. l00:l01:l02:..l10:l11:l12:...ln0:ln1:ln2:... are floating point values between 0 and 1. l[i][j] determines how much of input channel j is mixed into output channel i. Example 1: mplayer -af pan=1:0.5:0.5 -channels 1 media.avi would down-mix from stereo to mono. Example 2: 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 sub-woofer for example). The sub 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 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: fc <20-300> 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. ch <0-5> 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 ch if necessary. Example: 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. Matrix encoded surround sound can be decoded by the surround filter. 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: d <0-1000> 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 d should be set to 15ms if d1 <= d2 and to 15 + 5*(d1-d2) if d1 > d2. The default value for d is 20ms. Example: mplayer -af surround=15 -channels 4 media.avi would add surround sound decoding with 15ms delay for the sound to the rear speakers. This audio filter 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, it's updated every time new data is exported.*/ The rest is payload (non-interleaved) 16bit data. mmapped_file The file you want this filter to export to. The default is to map to ~/.mplayer/mplayer-af_export. nsamples Number of samples per channel. The default is 512 samples. Example: mplayer -af export=/tmp/mplayer-af_export:1024 media.avi would export 1024 samples per channel to /tmp/mplayer-af_export. Audio plugins have been deprecated by audio filters and will be removed soon. MPlayer has support for audio plugins. Audio plugins can be used for changing the properties of the audio data before the sound reaches the sound card. They are enabled using the -aop switch which takes a list=plugin1,plugin2,... argument. The list argument is required and determines which plugins should be used and in which order they should be executed. Example: mplayer media.avi -aop list=resample,format would run the sound through the resampling plugin followed by the format plugin. The plugins can also have switches that change their behavior. These switches are explained in detail in the sections below. A plugin will execute using default settings if its switches are omitted. Here is an example of how to use plugins in combination with plugin specific switches: mplayer media.avi -aop list=resample,format:fout=44100:format=0x8 would set the output frequency of the resample plugin to 44100 Hz and the output format of the format plugin to AFMT_U8. Currently audio plugins can not be used in MEncoder. MPlayer fully supports up/downsampling of the sound. This plugin 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 44.1 kHz. Limitations in your hardware are not auto detected, so you have to specify the sample frequency explicitly. This plugin has one switch: fout which is used for setting the desired output sample frequency. It defaults to 48 kHz, and is given in Hz. Usage: mplayer media.avi -aop list=resample:fout=freq where freq is the frequency in Hz, like 44100. The output frequency should not be scaled up from the default value. Scaling up will cause the audio and video streams to be played in slow motion in addition to audio distortion. MPlayer has an audio plugin that can decode 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. Usage: mplayer media.avi -aop list=surround If your sound card driver does not support signed 16-bit int data type, this plugin can be used to change the format to one which your sound card can understand. It has one switch, format, which can be set to one of the numbers found in libao2/afmt.h. This plugin is hardly ever needed and is intended for advanced users. Keep in mind that this plugin only changes the sample format and not the sample frequency or the number of channels. Usage: mplayer media.avi -aop list=format:format=outfmt where outfmt is the required output format. This plugin delays the sound and is intended as an example of how to develop new plugins. It can not be used for anything useful from a users perspective and is mentioned here for the sake of completeness only. Do not use this plugin unless you are a developer. This plugin is a software replacement for the volume control, and can be used on machines with a broken mixer device. It can also be used if one wants to change the output volume of MPlayer without changing the PCM volume setting in the mixer. It has one switch volume that is used for setting the initial sound level. The initial sound level can be set to values between 0 and 255 and defaults to 101 which equals 0dB amplification. Use this plugin 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 plugin out and control the output level to your speakers with the master volume control of the mixer. 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. Usage: mplayer media.avi -aop list=volume:volume=0-255 This plugin also has compressor or "soft-clipping" capabilities. Compression can be used if the dynamic range of the sound is very high or if the dynamic range of the loudspeakers is very low. Be aware that this feature creates distortion and should be considered a last resort. Usage: mplayer media.avi -aop list=volume:softclip This plugin (linearly) increases the difference between left and right channels (like the XMMS extrastereo plugin) which gives some sort of "live" effect to playback. Usage: mplayer media.avi -aop list=extrastereo mplayer media.avi -aop list=extrastereo:mul=3.45 The default coefficient (mul) is a float number that defaults to 2.5. If you set it to 0.0, you will have mono sound (average of both channels). If you set it to 1.0, sound will be unchanged, if you set it to -1.0, left and right channels will be swapped. This plugin maximizes the volume without distorting the sound. Usage: mplayer media.avi -aop list=volnorm