ao.rst

AUDIO OUTPUT DRIVERS

Audio output drivers are interfaces to different audio output facilities. The syntax is:

--ao=<driver1,driver2,...[,]>

Specify a priority list of audio output drivers to be used.

If the list has a trailing ',', mpv will fall back on drivers not contained in the list.

This is an object settings list option. See `List Options`_ for details.

Note

See --ao=help for a list of compiled-in audio output drivers sorted by autoprobe order.

Note that the default audio output driver is subject to change, and must not be relied upon. If a certain AO needs to be used, it must be explicitly specified.

Available audio output drivers are:

alsa

ALSA audio output driver.

The following global options are supported by this audio output:

--alsa-resample=yes

Enable ALSA resampling plugin. (This is disabled by default, because some drivers report incorrect audio delay in some cases.)

--alsa-mixer-device=<device>

Set the mixer device used with ao-volume (default: default).

--alsa-mixer-name=<name>

Set the name of the mixer element (default: Master). This is for example PCM or Master.

--alsa-mixer-index=<number>

Set the index of the mixer channel (default: 0). Consider the output of "amixer scontrols", then the index is the number that follows the name of the element.

--alsa-non-interleaved

Allow output of non-interleaved formats (if the audio decoder uses this format). Currently disabled by default, because some popular ALSA plugins are utterly broken with non-interleaved formats.

--alsa-ignore-chmap

Don't read or set the channel map of the ALSA device - only request the required number of channels, and then pass the audio as-is to it. This option most likely should not be used. It can be useful for debugging, or for static setups with a specially engineered ALSA configuration (in this case you should always force the same layout with --audio-channels, or it will work only for files which use the layout implicit to your ALSA device).

--alsa-buffer-time=<microseconds>

Set the requested buffer time in microseconds. A value of 0 skips requesting anything from the ALSA API. This and the --alsa-periods option uses the ALSA near functions to set the requested parameters. If doing so results in an empty configuration set, setting these parameters is skipped.

Both options control the buffer size. A low buffer size can lead to higher CPU usage and audio dropouts, while a high buffer size can lead to higher latency in volume changes and other filtering.

--alsa-periods=<number>

Number of periods requested from the ALSA API. See --alsa-buffer-time for further remarks.

Warning

To get multichannel/surround audio, use --audio-channels=auto. The default for this option is auto-safe, which makes this audio output explicitly reject multichannel output, as there is no way to detect whether a certain channel layout is actually supported.

You can also try using the upmix plugin. This setup enables multichannel audio on the default device with automatic upmixing with shared access, so playing stereo and multichannel audio at the same time will work as expected.

oss

OSS audio output driver

jack

JACK (Jack Audio Connection Kit) audio output driver.

The following global options are supported by this audio output:

--jack-port=<name>

Connects to the ports with the given name (default: physical ports).

--jack-name=<client>

Client name that is passed to JACK (default: mpv). Useful if you want to have certain connections established automatically.

--jack-autostart=<yes|no>

Automatically start jackd if necessary (default: disabled). Note that this tends to be unreliable and will flood stdout with server messages.

--jack-connect=<yes|no>

Automatically create connections to output ports (default: enabled). When enabled, the maximum number of output channels will be limited to the number of available output ports.

--jack-std-channel-layout=<waveext|any>

Select the standard channel layout (default: waveext). JACK itself has no notion of channel layouts (i.e. assigning which speaker a given channel is supposed to map to) - it just takes whatever the application outputs, and reroutes it to whatever the user defines. This means the user and the application are in charge of dealing with the channel layout. waveext uses WAVE_FORMAT_EXTENSIBLE order, which, even though it was defined by Microsoft, is the standard on many systems. The value any makes JACK accept whatever comes from the audio filter chain, regardless of channel layout and without reordering. This mode is probably not very useful, other than for debugging or when used with fixed setups.

coreaudio (macOS only)

Native macOS audio output driver using AudioUnits and the CoreAudio sound server.

Automatically redirects to coreaudio_exclusive when playing compressed formats.

The following global options are supported by this audio output:

--coreaudio-change-physical-format=<yes|no>

Change the physical format to one similar to the requested audio format (default: no). This has the advantage that multichannel audio output will actually work. The disadvantage is that it will change the system-wide audio settings. This is equivalent to changing the Format setting in the Audio Devices dialog in the Audio MIDI Setup utility. Note that this does not affect the selected speaker setup.

--coreaudio-spdif-hack=<yes|no>

Try to pass through AC3/DTS data as PCM. This is useful for drivers which do not report AC3 support. It converts the AC3 data to float, and assumes the driver will do the inverse conversion, which means a typical A/V receiver will pick it up as compressed IEC framed AC3 stream, ignoring that it's marked as PCM. This disables normal AC3 passthrough (even if the device reports it as supported). Use with extreme care.

coreaudio_exclusive (macOS only)

Native macOS audio output driver using direct device access and exclusive mode (bypasses the sound server).

avfoundation (macOS only)

Native macOS audio output driver using AVSampleBufferAudioRenderer in AVFoundation, which supports spatial audio.

Warning

Turning on spatial audio may hang the playback if mpv is not started out of the bundle, though playback with spatial audio off always works.

audiounit (iOS only)

Native iOS audio output driver using AudioUnits and AudioToolbox.

openal

OpenAL audio output driver.

--openal-num-buffers=<2-128>

Specify the number of audio buffers to use. Lower values are better for lower CPU usage. Default: 4.

--openal-num-samples=<256-32768>

Specify the number of complete samples to use for each buffer. Higher values are better for lower CPU usage. Default: 8192.

--openal-direct-channels=<yes|no>

Enable OpenAL Soft's direct channel extension when available to avoid tinting the sound with ambisonics or HRTF. Default: yes.

pulse

PulseAudio audio output driver

The following global options are supported by this audio output:

--pulse-host=<host>

Specify the host to use. An empty <host> string uses a local connection, "localhost" uses network transfer (most likely not what you want).

--pulse-buffer=<1-2000|native>

Set the audio buffer size in milliseconds. A higher value buffers more data, and has a lower probability of buffer underruns. A smaller value makes the audio stream react faster, e.g. to playback speed changes. "native" lets the sound server determine buffers.

--pulse-latency-hacks=<yes|no>

Enable hacks to workaround PulseAudio timing bugs (default: yes). If enabled, mpv will do elaborate latency calculations on its own. If disabled, it will use PulseAudio automatically updated timing information. Disabling this might help with e.g. networked audio or some plugins, while enabling it might help in some unknown situations (it is currently enabled due to known bugs with PulseAudio 16.0).

--pulse-allow-suspended=<yes|no>

Allow mpv to use PulseAudio even if the sink is suspended (default: no). Can be useful if PulseAudio is running as a bridge to jack and mpv has its sink-input set to the one jack is using.

pipewire

PipeWire audio output driver

The following global options are supported by this audio output:

--pipewire-buffer=<1-2000|native>

Set the audio buffer size in milliseconds. A higher value buffers more data, and has a lower probability of buffer underruns. A smaller value makes the audio stream react faster, e.g. to playback speed changes. "native" lets the sound server determine buffers.

--pipewire-remote=<remote>

Specify the PipeWire remote daemon name to connect to via local UNIX sockets. An empty <remote> string uses the default remote named pipewire-0.

--pipewire-volume-mode=<channel|global>

Specify if the ao-volume property should apply to the channel volumes or the global volume. By default the channel volumes are used.

sdl

SDL 2.0+ audio output driver. Should work on any platform supported by SDL 2.0, but may require the SDL_AUDIODRIVER environment variable to be set appropriately for your system.

Note

This driver is for compatibility with extremely foreign environments, such as systems where none of the other drivers are available.

The following global options are supported by this audio output:

--sdl-buflen=<length>

Sets the audio buffer length in seconds. Is used only as a hint by the sound system. Playing a file with -v will show the requested and obtained exact buffer size. A value of 0 selects the sound system default.

null

Produces no audio output but maintains video playback speed. You can use --ao=null --ao-null-untimed for benchmarking.

The following global options are supported by this audio output:

--ao-null-untimed

Do not simulate timing of a perfect audio device. This means audio decoding will go as fast as possible, instead of timing it to the system clock.

--ao-null-buffer

Simulated buffer length in seconds.

--ao-null-outburst

Simulated chunk size in samples.

--ao-null-speed

Simulated audio playback speed as a multiplier. Usually, a real audio device will not go exactly as fast as the system clock. It will deviate just a little, and this option helps to simulate this.

--ao-null-latency

Simulated device latency. This is additional to EOF.

--ao-null-broken-eof

Simulate broken audio drivers, which always add the fixed device latency to the reported audio playback position.

--ao-null-broken-delay

Simulate broken audio drivers, which don't report latency correctly.

--ao-null-channel-layouts

If not empty, this is a , separated list of channel layouts the AO allows. This can be used to test channel layout selection.

--ao-null-format

Force the audio output format the AO will accept. If unset accepts any.

pcm

Raw PCM/WAVE file writer audio output

The following global options are supported by this audio output:

--ao-pcm-waveheader=<yes|no>

Include or do not include the WAVE header (default: included). When not included, raw PCM will be generated.

--ao-pcm-file=<filename>

Write the sound to <filename> instead of the default audiodump.wav. If no-waveheader is specified, the default is audiodump.pcm.

--ao-pcm-append=<yes|no>

Append to the file, instead of overwriting it. Always use this with the no-waveheader option - with waveheader it's broken, because it will write a WAVE header every time the file is opened.

sndio

Audio output to the OpenBSD sndio sound system

(Note: only supports mono, stereo, 4.0, 5.1 and 7.1 channel layouts.)

wasapi

Audio output to the Windows Audio Session API.

The following global options are supported by this audio output:

--wasapi-exclusive-buffer=<default|min|1-2000000>

Set buffer duration in exclusive mode (i.e., with --audio-exclusive=yes). default and min use the default and minimum device period reported by WASAPI, respectively. You can also directly specify the buffer duration in microseconds, in which case a duration shorter than the minimum device period will be rounded up to the minimum period.

The default buffer duration should provide robust playback in most cases, but reportedly on some devices there are glitches following stream resets under the default setting. In such cases, specifying a shorter duration might help.