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SurroundSound

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NOTE: This article applies to surround sound sent to the analog outputs. For digital output, see DigitalOut.

surround51 and surround40 are the generic PCM definitions for 6 (aka 5.1) and 4 (aka 4.0) channel analog outputs. When the ALSA driver supports 5.1 or 4.0 analog output, the corresponding configuration file for ALSA-lib is transparently provided and includes the definition of surround51 and/or surround40. The usage is very simple. Just pass surround51 or surround40 as the PCM name...

aplay -Dsurround51 foo.wav

where foo.wav is a WAV file containing 6 channel stereo samples. Note that surround51 and surround40 are supposed to be analog, not for the digital AC3/DTS. They don't decode anything. They just support the multi-channel PCM.

Contents

Additional notes on surround PCM

from Greg Lee

Custom Routing of Signals to Surround Outputs

Note:There were problems with multichannel LADSPA plugins in (some?) versions earlier than ALSA 1.0.14rc2. If you experience any strange behavior, you may need to upgrade to that version or above. For lowpass filtering with LADSPA also see this howto.

With a little .asoundrc hackery, you can easily route signals around any way you like. See Playing stereo on surround sound setup (Howto) for an example of how to play a stereo signal over booth front and rear speakers simultaneously. The example can easily be adapted to suit your needs. The Howto tells us how to route our "normal" stereo signals to 4 or 5 speakers. It would be nice to process the signal for the subwoofer with a lowpass filter. Thats's where the LADSPA plugins come in handy:

pcm.lowpass {
     type ladspa
     slave.pcm    ...
     path "/usr/lib/ladspa"
     plugins [ {
          label lpf 
          input {
               controls [ 50 ]
          }
     } ]
}

This filter will cut off everything above 50 Hz. Only problem: How do I send only one channel (a mixture of right and left channel) through this filter? And where do I send the filtered signal? I have tried this setup:

# move channel 0 to channel 2
pcm.move0to2 {
     type route
     slave.pcm 3to6
     slave.channels 3
     ttable.0.2 1
}
# 3to6 has 3 input and 6 output channels
# the stereo signal is on ch. 0 and 1
# ch.0 is copied to 0 and 2  (front and rear)
# ch.1 is copied to 1 and 3  (front and rear)
# ch.0 and ch.1 will is mixed to ch.4 (center)
# ch.2 is routed to ch.5 (subwoofer)
pcm.3to6 {
     type route
     slave.pcm "surround51"
     slave.channels 6
     ttable.0.0 1
     ttable.1.1 1
     ttable.0.2 1
     ttable.1.3 1
     ttable.0.4 0.5
     ttable.1.4 0.5
     ttable.2.5 1
}

This seems to work from the lowpass filter to the output (I have tried aplay -D ```plug:lowpass``` sound.wav). But still the question - How to I get the data of one channel to the lowpass filter?

Update 18th April 2006 - since alsa 1.0.11rc2 the ladspa plugin can handle multichannel ladspa plugins. Using a 3 channel ladspa plugin where 2 channels are passed through and one has a low pass filter applied the effect desired above can be achieved.

Question: Is somebody able to describe how this lowpass filter only for the LFE channel could be implemented with these multichannel ladspa plugins?

Answer:

  1. After installation of libasound2 version 1.0.11-7\~bpo.1 in Debian Sarge.
  2. Sample config - input is 1 mono channel and sub channel (0-sub,1-normal mono):

pcm.ice2_11cutoffsub {
     type ladspa
     slave.pcm       ice_plug
     path    "/usr/lib/ladspa"
     plugins {
          0 {
               label lp4pole_fcrcia_oa
               policy none
               input.bindings.0 "Input";
               output.bindings.0 "Output";
               input {
                    controls       [ 300 0 ]
               }
          }
          1 {
               label delay_5s
               input.bindings.0 "Input";
               output.bindings.0 "Output";
               input {
                    controls [ 0 0 ]
               }
          }
     }
}

Other sample config - input is stereo channel and sub channel (0,1-stereo,2-sub):

pcm.ice2_21cutoffsub {
     type ladspa
     slave.pcm       ice_plug
     path    "/usr/lib/ladspa"
     plugins {
          0 {
               label lp4pole_fcrcia_oa
               policy none
               input.bindings.2 "Input";
               output.bindings.2 "Output";
               input {
                    controls       [ 300 0 ]
               }
          }
          1 {
               label delay_0.01s
               input.bindings.0 "Input";
               output.bindings.0 "Output";
               input {
                    controls [ 0 1 ]
               }
          }
     }
}

I do not see the code, but seem that you need to cover all ALSA channels with at least one plugin - at least in my alsa version. Seem - If you do not pass other channels thought any alsa plugin you will get silence on that channel. Check me!

Question: I select filter by hearing them all, but who know that is the best and why?

Using surroundXX PCMs with JACK

On my SBLive, the naked surround40 device cannot be used with JACK. There are two problems:

So we need to create a new virtual device that will be mmap()able and has 2 ins and 4 outs. TakashiIwai helped out with some magic incantations to add to your .asoundrc:

ctl.jack40 {
     type hw
     card 0
}
pcm.jack40 {
     # "asym" allows for different
     # handling of in/out devices
     type asym
     playback.pcm {
          # route for mmap workaround
          type route
          slave.pcm surround40
          ttable.0.0 1
          ttable.1.1 1
          ttable.2.2 1
          ttable.3.3 1
     }
     capture.pcm {
          # 2 channels only
          type hw
          card 0
     }
}

Now you can start jackd with two inputs and four outputs:

jackd -d alsa --device jack40 --inchannels 2 --outchannels 4

For 5.1 cards, there is probably a similar spell. I only have 2 stereo outs on my card, so I can't test other setups. If you get 5.1 or other configurations to work, please add them here. On the older (dual-DAC) CMedia CMI8738, the above asoundrc setup works with:

jackd -d alsa --device jack40 --playback --outchannels 4

and the --outchannels switch can be omitted. You can't get capture inputs at the same time (I'm assuming because the ADC is busy driving the rear channels) but you do get four-way audio outputs.

For 5.1 sound, this .asoundrc works just fine

ctl.jack51 {
    type hw
    card 0
}

pcm.jack51 {
    # "asym" allows for different
    # handling of in/out devices
    type asym
    playback.pcm {
         # route for mmap workaround
         type plug
         slave.pcm "surround51"
         slave.channels 6
         route_policy duplicate
    }
    capture.pcm {
        # 2 channels only
        type hw
        card 0
    }
}

Just start JACK with

jackd -d alsa --device jack51 --inchannels 2 --outchannels 6

Using Surround and Mplayer or Xine

I had a problem where speaker-test worked for my alsa configuration but Xine didn't. I also never got my 7.1 surround system to work past 4.0...until I used the following for Xine (.xine/config):

audio.device.alsa_surround40_device:plughw:0,1

and

audio.device.alsa_surround51_device:plughw:0,1

Mplayer is similar:

mplayer -channels 6 -ao alsa:mmap:noblock:device=hw=0.1  dvd://4

Now I have Center, LFE, rear, and front. I should have side as well, but I haven't tested that yet. You can check your /proc/asound/card0/pcmXp/info files (replace X with the device #) to play with the 0,1 (or 0.1) to try it out.

The first number is the card number and the second is the device # on the card. Usually the device with multiple devices is your surround, but I am still confused by this. When using plughw, it is backwards of what I expected in /proc, but when I set up sections in my .asoundrc, it seems to follow what is in /proc.

Hope that helps someone! It took me a year and a half and countless hours to get this to finally work on my 64-Bit Asus A8V Deluxe (via8237 -> snd-via82xx).

Troubleshooting

Are you sure your speakers are plugged in properly? Sound cards that are integrated into the motherboard often have only three jacks: line out, line in, and microphone. Most of the time, the front speakers miniplug should go in the line out jack and the rear speakers miniplug should go in the line in jack. On at least one system with the VT8237 integrated audio chip, the rear speakers miniplug should go in the microphone jack for proper surround output.

External links

Retrieved from "http://alsa.opensrc.org/SurroundSound"

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