Bug 94357

pulseaudio don't like the multi channel volume control

you have add PCM Capture Source in pulseaudio input conf files

https://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mixer/paths

what are the usage of two PCM Playback Volume Controls 


Card hw:1 'U5'/'VIA Technologies Inc. MAYA U5 at usb-0000:00:1a.0-1.3, full speed'
  Mixer name	: 'USB Mixer'
  Components	: 'USB040d:3401'
  Controls      : 23
  Simple ctrls  : 10
Simple mixer control 'Speaker',0
  Capabilities: pvolume pswitch pswitch-joined
  Playback channels: Front Left - Front Right - Rear Left - Rear Right - Front Center - Woofer
  Limits: Playback 0 - 30
  Mono:
  Front Left: Playback 16 [53%] [-21.00dB] [on]
  Front Right: Playback 16 [53%] [-21.00dB] [on]
  Rear Left: Playback 16 [53%] [-21.00dB] [on]
  Rear Right: Playback 16 [53%] [-21.00dB] [on]
  Front Center: Playback 16 [53%] [-21.00dB] [on]
  Woofer: Playback 16 [53%] [-21.00dB] [on]
Simple mixer control 'PCM',0
  Capabilities: volume
  Playback channels: Front Left - Front Right
  Capture channels: Front Left - Front Right
  Limits: 0 - 1
  Front Left: 1 [100%]
  Front Right: 1 [100%]
Simple mixer control 'PCM Capture Source',0
  Capabilities: enum
  Items: 'Microphone' 'Line In' 'Stereo Mix' 'Spdif Rx'
  Item0: 'Microphone'
Simple mixer control 'PCM',1
  Capabilities: volume
  Playback channels: Front Left - Front Right
  Capture channels: Front Left - Front Right
  Limits: 0 - 1
  Front Left: 1 [100%]
  Front Right: 1 [100%]

Can you help me fix this Raymond cuz i dont feel confident with linux yet?

do you mean this patch still not fix the problem ?

https://cgit.freedesktop.org/pulseaudio/pulseaudio/commit/src/modules/alsa?id=fcbbe34e001b9bee3e73a663e9bf9ddc11c0a1d5

You have to post output of alsa-info.sh and pactl list


https://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mixer/paths/analog-input.conf.common#n110

The easy way is to change PCM Capture Source to Capture Source in alsa driver, otherwise you have to add elememt PCM Capture Souce and those corresponding  option PCM Capture Source:inputs to pulseaudio conf files

here is alsa-info.sh: http://www.alsa-project.org/db/?f=4d08149b5fab1a43c9b0555ed457fd337eb258ea

and here is pactl list: http://paste.ubuntu.com/15272803/

http://git.alsa-project.org/?p=alsa-lib.git;a=blob;f=src/conf/cards/USB-Audio.conf;hb=HEAD


You need to define iec958 in above conf if device 0 is not digial optical output


Pulseaudio 2.1 profile use lfe of surround 5.1 speaker

speaker-test -c6 -t wav -D surround51:CARD=U5

Do 5.1 channels correct in speaker test? 


**** List of PLAYBACK Hardware Devices **** 

card 0: U5 [MAYA U5], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 

card 0: U5 [MAYA U5], device 1: USB Audio [USB Audio #1] Subdevices: 1/1 Subdevice #0: subdevice #0

 # If a device does not use the first PCM device for digital data, the device
 # number for the iec958 device can be changed here.
  USB-Audio.pcm.iec958_device {
          # "NoiseBlaster 3000" 42
          "USB Sound Blaster HD" 1

I have 5.1 card but only 2 speakrs and headphones

speaker-test -c6 -t wav -D surround51:CARD=U5 this command doesnt work becouse device is busy

i run it with pasuspender -- speaker-test -c6 -t wav -D surround51:CARD=U5 and i hear front left from FRONT CENTER and "front right" from SIDE RIGHT

But i dont get it why You want me to edit my alsa configs if my alsa works correct (when i uninstall pulseaudio) am i missing something here?

if i run 'pasuspender -- speaker-test -c2 -twav -Dhw:U5' i get voice everytime. But then i have go to pulseaudio gui settings and find working profile. Currently used profile likes to break everytime i stop music

surround21 is defined in http://git.alsa-project.org/?p=alsa-lib.git;a=blob;f=src/conf/pcm/surround21.conf;hb=HEAD



#  Hardware output from 2.1 speakers.
#  Samples must be positioned:
#    chn0 - front left
#    chn1 - front right
#    chn2 - LFE
#;hb=HEAD


	type route
	slave.pcm {
		@func refer
		name {
			@func concat
			strings [
				"cards."
				{
					@func card_driver
					card $CARD
				}
				".pcm.surround51." $DEV ":CARD=" $CARD
			]
		}
	}
	ttable.0.FL 1
	ttable.1.FR 1
	ttable.2.LFE 1
	hint {
		description "2.1 Surround output to Front and Subwoofer speakers"
		device $DEV
	}



both syrround21 and surround 51 also use route


USB-Audio.pcm.surround51.0 {
	@args [ CARD ]
	@args.CARD { type string }
	@func refer
	name {
		@func concat
		strings [
			"cards.USB-Audio."
			{ @func card_name card $CARD }
			".pcm.surround51:CARD=" $CARD
		]
	}
	default {
		type route
		ttable.0.0 1
		ttable.1.1 1
		ttable.2.4 1
		ttable.3.5 1
		ttable.4.2 1
		ttable.5.3 1
		slave {
			pcm {
				type hw
				card $CARD
				device 0
			}
			channels 6
		}
	}
}

can you post your usb audio device  040d:3401


lsusb -vvvv

http://pastebin.com/WFtsb113

http://audiotrack.net/en/index.php?document_srl=531&ckattempt=1&mid=usb



VIA’s latest EnvyUSB USB2.0 FS audio controller and high quality High Definition audio codec(DAC SNR 110dB) are mounted. This supports speakers of 2 channel(2.1 channel), 4 channel, 5.1 channel and has high quality playback sound of maximum 24bit/48kHz with supporting frequency of 16bit/24bit, 44.1kHz/48kHz. This provides optical output and input terminals which makes playback even at 24bit/48kHz supporting Blue Ray Control Protection function and also supports AC3 PassThrough function.


Seem only support two channels 24bit,  only 16bit for 6 channels

How about 4 channels? 


speaker-test -d -c4 -t wav -D surround40:CARD=U5

speaker-test -d -c3 -t wav -D surround21:CARD=U5

The default usb iec958 is device 0 for playback and capture 




**** List of CAPTURE Hardware Devices **** 
card 0: U5 [MAYA U5], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0

aplay --dump-hw-params -Dhw:CARD=U5,0 anystereo.wav

Simple mixer control 'PCM Capture Source',0 Capabilities: enum Items: 'Microphone' 'Line In' 'Stereo Mix' 'Spdif Rx' Item0: 'Microphone'

Do you mean digital input use same capture device 0 by slecting Spdif Rx?

AudioStreaming Interface Descriptor:
        bLength                14
        bDescriptorType        36
        bDescriptorSubtype      2 (FORMAT_TYPE)
        bFormatType             1 (FORMAT_TYPE_I)
        bNrChannels             6
        bSubframeSize           3
        bBitResolution         24
        bSamFreqType            2 Discrete
        tSamFreq[ 0]        48000
        tSamFreq[ 1]        44100


The descriptor seem support 6 channels 24bits

Profile: input:analog-stereo: Wejście Analogowe stereo (sinks: 0, sources: 1, priority: 60, available: yes) input:iec958-stereo: Wejście Cyfrowe stereo (IEC958) (sinks: 0, sources: 1, priority: 55, available: yes) output:analog-stereo: Wyjście Analogowe stereo (sinks: 1, sources: 0, priority: 6000, available: yes) output:analog-stereo+input:analog-stereo: Analogowy dupleks stereo (sinks: 1, sources: 1, priority: 6060, available: yes) output:analog-stereo+input:iec958-stereo: Wyjście Analogowe stereo + Wejście Cyfrowe stereo (IEC958) (sinks: 1, sources: 1, priority: 6055, available: yes) output:analog-surround-21: Wyjście Analogowe surround 2.1 (sinks: 1, sources: 0, priority: 800, available: yes) output:analog-surround-21+input:analog-stereo: Wyjście Analogowe surround 2.1 + Wejście Analogowe stereo (sinks: 1, sources: 1, priority: 860, available: yes) output:analog-surround-21+input:iec958-stereo: Wyjście Analogowe surround 2.1 + Wejście Cyfrowe stereo (IEC958) (sinks: 1, sources: 1, priority: 855, available: yes) output:analog-surround-41: Wyjście Analogowe surround 4.1 (sinks: 1, sources: 0, priority: 800, available: yes) output:analog-surround-41+input:analog-stereo: Wyjście Analogowe surround 4.1 + Wejście Analogowe stereo (sinks: 1, sources: 1, priority: 860, available: yes) output:analog-surround-41+input:iec958-stereo: Wyjście Analogowe surround 4.1 + Wejście Cyfrowe stereo (IEC958) (sinks: 1, sources: 1, priority: 855, available: yes) output:analog-surround-50: Wyjście Analogowe surround 5.0 (sinks: 1, sources: 0, priority: 700, available: yes) output:analog-surround-50+input:analog-stereo: Wyjście Analogowe surround 5.0 + Wejście Analogowe stereo (sinks: 1, sources: 1, priority: 760, available: yes) output:analog-surround-50+input:iec958-stereo: Wyjście Analogowe surround 5.0 + Wejście Cyfrowe stereo (IEC958) (sinks: 1, sources: 1, priority: 755, available: yes) output:analog-surround-51: Wyjście Analogowe surround 5.1 (sinks: 1, sources: 0, priority: 800, available: yes) output:analog-surround-51+input:analog-stereo: Wyjście Analogowe surround 5.1 + Wejście Analogowe stereo (sinks: 1, sources: 1, priority: 860, available: yes) output:analog-surround-51+input:iec958-stereo: Wyjście Analogowe surround 5.1 + Wejście Cyfrowe stereo (IEC958) (sinks: 1, sources: 1, priority: 855, available: yes) output:iec958-stereo: Wyjście Cyfrowe stereo (IEC958) (sinks: 1, sources: 0, priority: 5500, available: yes) output:iec958-stereo+input:analog-stereo: Wyjście Cyfrowe stereo (IEC958) + Wejście Analogowe stereo (sinks: 1, sources: 1, priority: 5560, available: yes) output:iec958-stereo+input:iec958-stereo: Cyfrowy dupleks stereo (IEC958) (sinks: 1, sources: 1, priority: 5555, available: yes) off: Wyłączone (sinks: 0, sources: 0, priority: 0, available: yes) 

Aktywny profil: output:analog-surround-41+input:analog-stereo

Seem all profile are available,

http://git.alsa-project.org/?p=alsa-lib.git;a=blob;f=src/conf/pcm/surround41.conf;hb=HEAD



#  Hardware output from 4.1 speakers.
#  Samples must be positioned:
#    chn0 - front left
#    chn1 - front right
#    chn2 - rear left
#    chn3 - rear right
 #    chn4 - LFE


       ttable.0.FL 1
          ttable.1.FR 1
           ttable.2.RL 1
           ttable.3.RR 1
           ttable.4.LFE 1
           hint {


do the following work as expected ?

speaker-test -d -c5 -t wav -D surround41:CARD=U5


speaker-test -d -c5 -t wav -D surround50:CARD=U5

speaker playback switch only have one value but speaker playback volume have 6 values


	control.14 {
		iface MIXER
		name 'Speaker Playback Switch'
		value true
		comment {
			access 'read write'
			type BOOLEAN
			count 1
		}
	}
	control.15 {
		iface MIXER
		name 'Speaker Playback Volume'
		value.0 29
		value.1 29
		value.2 28
		value.3 28
		value.4 28
		value.5 29
		comment {
			access 'read write'
			type INTEGER
			count 6
			range '0 - 30'
			dbmin -4500
			dbmax 0
			dbvalue.0 -150
			dbvalue.1 -150
			dbvalue.2 -300
			dbvalue.3 -300
			dbvalue.4 -300
			dbvalue.5 -150
		}
	}



Simple mixer control 'Speaker',0
  Capabilities: pvolume pswitch pswitch-joined
  Playback channels: Front Left - Front Right - Rear Left - Rear Right - Front Center - Woofer
  Limits: Playback 0 - 30
  Mono:
  Front Left: Playback 29 [97%] [-1.50dB] [on]
  Front Right: Playback 29 [97%] [-1.50dB] [on]
  Rear Left: Playback 28 [93%] [-3.00dB] [on]
  Rear Right: Playback 28 [93%] [-3.00dB] [on]
  Front Center: Playback 28 [93%] [-3.00dB] [on]
  Woofer: Playback 29 [97%] [-1.50dB] [on]
Simple mixer control 'PCM',0
  Capabilities: volume
  Playback channels: Front Left - Front Right
  Capture channels: Front Left - Front Right
  Limits: 0 - 1
  Front Left: 1 [100%]
  Front Right: 1 [100%]

pasuspender -- speaker-test -d -c4 -t wav -D surround40:CARD=U5

returns bad argument

pasuspender speaker-test -d -c3 -t wav -D surround21:CARD=U5
i hear output but only from FRONT LEFT


aplay --dump-hw-params -Dhw:CARD=U5,0 anystereo.wav
WORKS fine


I will give you good example what happend like 5 min ago. I was listening to spotify. Pause for 5 sec and PLAY again - Silence. I put my jack into another ports (rear, center) still silence then i back to my FRONT.

typed pact list sinks short - returned:

"32	alsa_output.usb-VIA_Technologies_Inc._MAYA_U5-00.analog-surround-41	module-alsa-card.c	s16le 5ch 44100Hz	RUNNING
"

But i have this LED diode on my USB CARD http://www.audiotrack.co.kr/images/maya_u5_en_11.jpg  It blinks when card is playing. But now it doesnt even when pactl responded with RUNNING 

then i typed again  pasuspender speaker-test -d -c3 -t wav -D surround21:CARD=U5
and after i CTRL+C my sound returned to spotify... i guess its some pulseaudio problem

http://www.audiotrack.co.kr/images/maya_u5_en_15.jpg

Im using 

Microphone 
Headphones
And Analog Front

Another thing i found lets say i choose profile 
http://imgur.com/4fG5L4b

I have silence on this profile even when music is playing from spotify.

i tryed to type pasuspender -- spearkertest ... few times. And after random quantity of tryes my sound is back on this profile so i dont have to "search" for working profile. Why is so? its surely pulseaudio thing...

> 
> 
> aplay --dump-hw-params -Dhw:CARD=U5,0 anystereo.wav
> WORKS fine
> 
> 
Can you post the output using hw:0,0 and hw:0,1 since --dump-hw-params dump the hardware capability of the devices

http://pastebin.com/KUb04gHj

On 0.0 i hear sound
on 0,1 there is silence

do red light of optical spdif turn on when you use hw:0,1

device 0 seem support 2 or 6 channels playback , 16 and 24 bits

device 1 only support 2 channles 16 bits

HW Params of device "hw:0,0":
-----------------
ACCESS:  MMAP_INTERLEAVED RW_INTERLEAVED
FORMAT:  S16_LE S24_3LE
SUBFORMAT:  STD
SAMPLE_BITS: [16 24]
FRAME_BITS: [32 144]
CHANNELS: [2 6]
RATE: [44100 48000]


HW Params of device "hw:0,1":
--------------------
ACCESS:  MMAP_INTERLEAVED RW_INTERLEAVED
FORMAT:  S16_LE
SUBFORMAT:  STD
SAMPLE_BITS: 16
FRAME_BITS: 32
CHANNELS: 2
RATE: [44100 48000

Please post the output of speaker test for surround51, surround21, surround41, option -d provide pcm dump of the route

do alsamixer or amixer change the volume of corresponding channel? 

Seem only one mute switch

Simple mixer control 'Speaker',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right - Rear Left - Rear Right - Front Center - Woofer Limits: Playback 0 - 30 Mono: Front Left: Playback 29 [97%] [-1.50dB] [on] Front Right: Playback 29 [97%] [-1.50dB] [on] Rear Left: Playback 28 [93%] [-3.00dB] [on] Rear Right: Playback 28 [93%] [-3.00dB] [on] Front Center: Playback 28 [93%] [-3.00dB] [on] Woofer: Playback 29 [97%] [-1.50dB] [on]

seen bug in alsa driver, no dB range for voeeslume control , this seem like switch, 

do it affect both playback and capture ?

control.4 {
		iface MIXER
		name 'PCM Volume'
		value.0 1
		value.1 1
		comment {
			access 'read write'
			type INTEGER
			count 2
			range '0 - 1'
		}
	}
	control.5 {
		iface MIXER
		name 'PCM Volume'
		index 1
		value.0 1
		value.1 1
		comment {
			access 'read write'
			type INTEGER
			count 2
			range '0 - 1'
		}
	}

http://pastebin.com/X1nqiUhL

http://pastebin.com/KfepYn9c

http://pastebin.com/YDK2MLhM

Yes my led blinks when aplay --dump-hw-params -Dhw:CARD=U5,1 anystereo.wav

This time i tryed to find something in pacmd, after http://pastebin.com/Ay0Bdqjw

sound appers.

I think pulseaudio suspend my card and cant wake it up even "pulseaudio thinks its not suspender" thats why my manualny commands works after few tries but thats annoying

Also another odd thing now i done something like this:
1. Start spotify (had sound)
2. Pause spotify
3. Start VLC Movie ( no sound)
4. Unpause spotify (had sound in spotify)
5. Start VLC (had sound in both spotify and VLC)

(In reply to Matthew from comment #31)
> Yes my led blinks when aplay --dump-hw-params -Dhw:CARD=U5,1 anystereo.wav

you need similar fix for iec958 device red laser turn on optical spdif when 

aplay  -Diec958:CARD=U5 anystereo.wav

http://git.alsa-project.org/?p=alsa-lib.git;a=commitdiff;h=a9b1299556598fed12d0d42c80aaa933aced8a42;hp=efda3e2bc766b4147f35f4b006d315ea7b3c55fe

can you post pulseaudio verbose log ?

any warning message when PCM Volume controls have no dB info


you need recompile alsa driver to produce debug output to find out those two PCM Volume control belong to which UnitID 

    bDescriptorSubtype      4 (MIXER_UNIT)
        bUnitID                10
        bNrInPins               3
        baSourceID( 0)          1  USB Streaming
        baSourceID( 1)          7  Feature Unit 8 Feature Unit 2 Microphone
        baSourceID( 2)         19  Feature Unit 15 Line Connector
        
		  
		bDescriptorSubtype      5 (SELECTOR_UNIT)
        bUnitID                14
        bNrInPins               4
        baSource( 0)            9 Feature  Unit 8 Feature Unit 2 Microphone
        baSource( 1)           18 Feature Unit 15 Line Connector
        baSource( 2)           26 Feature Unit 25 Analog Unit
        baSource( 3)           17 Feature Unit 16 SPDIF Interface

https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/sound/usb/mixer.c

https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/sound/usb/mixer.c

https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/sound/usb/mixer.c

when you compile alsa driver with debug=full

	usb_audio_dbg(state->chip, "[%d] FU [%s] ch = %d, val = %d/%d/%d\n",
		      cval->head.id, kctl->id.name, cval->channels,
		      cval->min, cval->max, cval->res);

usb print debug output in system log for those MU, PU, SU, ... and the cooresponding control name

Raymond i think i will pass, im linux user for 2 weeks and kinda dont have skills to do this like this already, maybe in some future. If alsa works fine for me i will just remove pulseaudio and problem is solved if there is no normal way to get this work properly

Thanks for Your time.

http://paste.ubuntu.com/15305893/

Maybe something like this is useful?

unit 10 is the mixer unit


Unit: 10
    Control: name="Line In Volume", index=0
    Info: id=10, control=9, cmask=0x20, channels=1, type="S16"
    Volume: min=0, max=0, dBmin=0, dBmax=0
  Unit: 10
    Control: name="Microphone Volume", index=1
    Info: id=10, control=8, cmask=0x1c, channels=3, type="S16"
    Volume: min=790, max=790, dBmin=308, dBmax=308
  Unit: 10
    Control: name="Microphone Volume", index=0
    Info: id=10, control=7, cmask=0xa, channels=2, type="S16"
    Volume: min=2313, max=2313, dBmin=903, dBmax=903
  Unit: 10
    Control: name="PCM Volume", index=1
    Info: id=10, control=5, cmask=0x24, channels=2, type="S16"
    Volume: min=-17536, max=-17536, dBmin=-6850, dBmax=-6850
  Unit: 10
    Control: name="PCM Volume", index=0
    Info: id=10, control=4, cmask=0x14, channels=2, type="S16"
    Volume: min=-17536, max=-17536, dBmin=-6850, dBmax=-6850

AudioControl Interface Descriptor:
        bLength                14
        bDescriptorType        36
        bDescriptorSubtype      4 (MIXER_UNIT)
      Warning: Descriptor too short
        bUnitID                10
        bNrInPins               3
        baSourceID( 0)          1
        baSourceID( 1)          7
        baSourceID( 2)         19
        bNrChannels             6
        wChannelConfig     0x003f
          Left Front (L)
          Right Front (R)
          Center Front (C)
          Low Freqency Enhancement (LFE)
          Left Surround (LS)
          Right Surround (RS)
        iChannelNames           0 
        bmControls         0x00
        bmControls         0x00
        bmControls         0x0a
        iMixer                 36

refer to USB Audio 2.0 specification
 
Table 4-11: Mixer Unit Descriptor 

bLength = 13 + bNrInPins + bmMixerControls

Sorry i have like 0 idea what to do with this

https://github.com/gregkh/usbutils/commit/31e20bde94abdbdf03c2b147fc83b0d9fd3d881d

seem only report 'descriptor too short' for UAC1 but not UAC2

-- GitLab Migration Automatic Message --

This bug has been migrated to freedesktop.org's GitLab instance and has been closed from further activity.

You can subscribe and participate further through the new bug through this link to our GitLab instance: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/issues/195.