wrong decibel data?

wrong decibel data?

[Nicolo' Chieffo]

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Hello, I have a laptop with an intel hda audio chipset (STAC92xx)

My problem is that when I put pulseaudio volume to 14% (or less) I
can't hear any audio coming from speakers or headphones: in fact the
alsamixer volume is set to 0%

I already filed a bug to pulseaudio [1] in which the developers say
that this is a common problem which is usually caused by the ALSA
audio driver which reports a wrong decibel value.

[1] http://pulseaudio.org/ticket/580

[-- Attachment #2: alsa-info.txt.hJD91ef9uH --]
[-- Type: application/octet-stream, Size: 24538 bytes --]

upload=true&script=true&cardinfo=
!!################################
!!ALSA Information Script v 0.4.59
!!################################

!!Script ran on: Fri Apr  2 18:21:33 UTC 2010


!!Linux Distribution
!!------------------

Ubuntu lucid (development branch) \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu lucid (development branch)"


!!DMI Information
!!---------------

Manufacturer:      Dell Inc.
Product Name:      Latitude E6400                  


!!Kernel Information
!!------------------

Kernel release:    2.6.32-19-generic
Operating System:  GNU/Linux
Architecture:      x86_64
Processor:         unknown
SMP Enabled:       Yes


!!ALSA Version
!!------------

Driver version:     1.0.21
Library version:    1.0.22
Utilities version:  1.0.22


!!Loaded ALSA modules
!!-------------------

snd_hda_intel


!!Sound Servers on this system
!!----------------------------

Pulseaudio:
      Installed - Yes (/usr/bin/pulseaudio)
      Running - Yes

ESound Daemon:
      Installed - Yes (/usr/bin/esd)
      Running - No


!!Soundcards recognised by ALSA
!!-----------------------------

 0 [Intel          ]: HDA-Intel - HDA Intel
                      HDA Intel at 0xf6adc000 irq 21


!!PCI Soundcards installed in the system
!!--------------------------------------

00:1b.0 Audio device: Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03)


!!Advanced information - PCI Vendor/Device/Susbsystem ID's
!!--------------------------------------------------------

00:1b.0 0403: 8086:293e (rev 03)
	Subsystem: 1028:0233


!!Modprobe options (Sound related)
!!--------------------------------

snd-atiixp-modem: index=-2
snd-intel8x0m: index=-2
snd-via82xx-modem: index=-2
snd-usb-audio: index=-2
snd-usb-us122l: index=-2
snd-usb-usx2y: index=-2
snd-usb-caiaq: index=-2
snd-cmipci: mpu_port=0x330 fm_port=0x388
snd-pcsp: index=-2


!!Loaded sound module options
!!--------------------------

!!Module: snd_hda_intel
	bdl_pos_adj : 1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1
	enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y
	enable_msi : 0
	id : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>
	index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1
	model : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>
	patch : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>
	position_fix : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0
	power_save : 0
	power_save_controller : Y
	probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1
	probe_only : N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N
	single_cmd : N


!!HDA-Intel Codec information
!!---------------------------
--startcollapse--

Codec: IDT 92HD71B7X
Address: 0
Function Id: 0x1
Vendor Id: 0x111d76b2
Subsystem Id: 0x10280233
Revision Id: 0x100302
No Modem Function Group found
Default PCM:
    rates [0x7e0]: 44100 48000 88200 96000 176400 192000
    bits [0xe]: 16 20 24
    formats [0x1]: PCM
Default Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1
Default Amp-Out caps: ofs=0x7f, nsteps=0x7f, stepsize=0x02, mute=1
GPIO: io=8, o=0, i=0, unsolicited=1, wake=1
  IO[0]: enable=1, dir=1, wake=0, sticky=0, data=1, unsol=0
  IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0
  IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0
  IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0
  IO[4]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0
  IO[5]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0
  IO[6]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0
  IO[7]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0
Power-Map: 0x01
Analog Loopback: 0x00
Node 0x0a [Pin Complex] wcaps 0x400181: Stereo
  Pincap 0x0000001c: OUT HP Detect
  Pin Default 0x0421101f: [Jack] HP Out at Ext Right
    Conn = 1/8, Color = Black
    DefAssociation = 0x1, Sequence = 0xf
  Pin-ctls: 0x00:
  Unsolicited: tag=01, enabled=1
  Connection: 3
     0x10 0x11* 0x17
Node 0x0b [Pin Complex] wcaps 0x400081: Stereo
  Pincap 0x00001724: IN Detect
    Vref caps: HIZ 50 GRD 80
  Pin Default 0x04a11221: [Jack] Mic at Ext Right
    Conn = 1/8, Color = Black
    DefAssociation = 0x2, Sequence = 0x1
  Pin-ctls: 0x24: IN VREF_80
  Unsolicited: tag=03, enabled=1
Node 0x0c [Pin Complex] wcaps 0x400081: Stereo
  Pincap 0x00001724: IN Detect
    Vref caps: HIZ 50 GRD 80
  Pin Default 0x40f000f0: [N/A] Other at Ext N/A
    Conn = Unknown, Color = Unknown
    DefAssociation = 0xf, Sequence = 0x0
  Pin-ctls: 0x00: VREF_HIZ
  Unsolicited: tag=00, enabled=0
Node 0x0d [Pin Complex] wcaps 0x400181: Stereo
  Pincap 0x00000014: OUT Detect
  Pin Default 0x90170110: [Fixed] Speaker at Int N/A
    Conn = Analog, Color = Unknown
    DefAssociation = 0x1, Sequence = 0x0
    Misc = NO_PRESENCE
  Pin-ctls: 0x40: OUT
  Unsolicited: tag=00, enabled=0
  Connection: 3
     0x10* 0x11 0x17
Node 0x0e [Pin Complex] wcaps 0x400081: Stereo
  Pincap 0x00001724: IN Detect
    Vref caps: HIZ 50 GRD 80
  Pin Default 0x23a1902e: [Jack] Mic at Sep Left
    Conn = 1/8, Color = Pink
    DefAssociation = 0x2, Sequence = 0xe
  Pin-ctls: 0x24: IN VREF_80
  Unsolicited: tag=04, enabled=1
Node 0x0f [Pin Complex] wcaps 0x400181: Stereo
  Pincap 0x00000014: OUT Detect
  Pin Default 0x23014250: [Jack] Line Out at Sep Left
    Conn = 1/8, Color = Green
    DefAssociation = 0x5, Sequence = 0x0
  Pin-ctls: 0x40: OUT
  Unsolicited: tag=02, enabled=1
  Connection: 3
     0x10* 0x11 0x17
Node 0x10 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L
  Amp-Out caps: N/A
  Amp-Out vals:  [0x5a 0x5a]
  Converter: stream=5, channel=0
  Power: setting=D0, actual=D0
  Delay: 13 samples
Node 0x11 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L
  Amp-Out caps: N/A
  Amp-Out vals:  [0x5a 0x5a]
  Converter: stream=5, channel=0
  Power: setting=D0, actual=D0
  Delay: 13 samples
Node 0x12 [Audio Input] wcaps 0x1d0541: Stereo
  Converter: stream=0, channel=0
  SDI-Select: 0
  Power: setting=D3, actual=D3
  Delay: 13 samples
  Connection: 1
     0x1c
  Processing caps: benign=0, ncoeff=0
Node 0x13 [Audio Input] wcaps 0x1d0541: Stereo
  Converter: stream=0, channel=0
  SDI-Select: 0
  Power: setting=D3, actual=D3
  Delay: 13 samples
  Connection: 1
     0x1d
  Processing caps: benign=0, ncoeff=0
Node 0x14 [Pin Complex] wcaps 0x400100: Mono
  Pincap 0x00000010: OUT
  Pin Default 0x40f000f0: [N/A] Other at Ext N/A
    Conn = Unknown, Color = Unknown
    DefAssociation = 0xf, Sequence = 0x0
  Pin-ctls: 0x00:
  Connection: 1
     0x16
Node 0x15 [Audio Selector] wcaps 0x300101: Stereo
  Connection: 3
     0x10* 0x11 0x17
Node 0x16 [Audio Mixer] wcaps 0x200100: Mono
  Connection: 1
     0x15
Node 0x17 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In
  Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1
  Amp-In vals:  [0x97 0x97] [0x97 0x97] [0x97 0x97] [0x97 0x97] [0x97 0x97]
  Connection: 5
     0x10 0x11 0x27 0x1a 0x1b
Node 0x18 [Pin Complex] wcaps 0x40000d: Stereo Amp-Out
  Amp-Out caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0
  Amp-Out vals:  [0x00 0x00]
  Pincap 0x00000020: IN
  Pin Default 0x90a000f0: [Fixed] Mic at Int N/A
    Conn = Unknown, Color = Unknown
    DefAssociation = 0xf, Sequence = 0x0
  Pin-ctls: 0x20: IN
Node 0x19 [Pin Complex] wcaps 0x40000d: Stereo Amp-Out
  Amp-Out caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0
  Amp-Out vals:  [0x00 0x00]
  Pincap 0x00000020: IN
  Pin Default 0x40f000f0: [N/A] Other at Ext N/A
    Conn = Unknown, Color = Unknown
    DefAssociation = 0xf, Sequence = 0x0
  Pin-ctls: 0x00:
Node 0x1a [Audio Selector] wcaps 0x30010d: Stereo Amp-Out
  Amp-Out caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0
  Amp-Out vals:  [0x00 0x00]
  Connection: 3
     0x0b* 0x0c 0x0e
Node 0x1b [Audio Selector] wcaps 0x30010d: Stereo Amp-Out
  Amp-Out caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0
  Amp-Out vals:  [0x00 0x00]
  Connection: 3
     0x0b* 0x0c 0x0e
Node 0x1c [Audio Selector] wcaps 0x30090d: Stereo Amp-Out R/L
  Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1
  Amp-Out vals:  [0x80 0x80]
  Connection: 4
     0x1a 0x17 0x18* 0x19
Node 0x1d [Audio Selector] wcaps 0x30090d: Stereo Amp-Out R/L
  Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1
  Amp-Out vals:  [0x80 0x80]
  Connection: 4
     0x1b* 0x17 0x18 0x19
Node 0x1e [Pin Complex] wcaps 0x400301: Stereo Digital
  Pincap 0x00000010: OUT
  Pin Default 0x4f0000f0: [N/A] Line Out at Ext UNKNOWN
    Conn = Unknown, Color = Unknown
    DefAssociation = 0xf, Sequence = 0x0
  Pin-ctls: 0x00:
  Connection: 1
     0x24
Node 0x1f [Pin Complex] wcaps 0x400701: Stereo Digital
  Pincap 0x00010010: OUT EAPD
  EAPD 0x0:
  Pin Default 0x4f0000f0: [N/A] Line Out at Ext UNKNOWN
    Conn = Unknown, Color = Unknown
    DefAssociation = 0xf, Sequence = 0x0
  Pin-ctls: 0x00:
  Power: setting=D0, actual=D0
  Connection: 2
     0x24* 0x25
Node 0x20 [Pin Complex] wcaps 0x400301: Stereo Digital
  Pincap 0x00000010: OUT
  Pin Default 0x40f000f7: [N/A] Other at Ext N/A
    Conn = Unknown, Color = Unknown
    DefAssociation = 0xf, Sequence = 0x7
  Pin-ctls: 0x00:
  Connection: 1
     0x25
Node 0x21 [Audio Output] wcaps 0x40211: Stereo Digital
  Converter: stream=0, channel=0
  Digital:
  Digital category: 0x0
  PCM:
    rates [0x7e0]: 44100 48000 88200 96000 176400 192000
    bits [0xe]: 16 20 24
    formats [0x5]: PCM AC3
  Delay: 4 samples
Node 0x22 [Audio Output] wcaps 0x40211: Stereo Digital
  Converter: stream=0, channel=0
  Digital:
  Digital category: 0x0
  PCM:
    rates [0x7e0]: 44100 48000 88200 96000 176400 192000
    bits [0xe]: 16 20 24
    formats [0x5]: PCM AC3
  Delay: 4 samples
Node 0x23 [Vendor Defined Widget] wcaps 0xf00000: Mono
Node 0x24 [Audio Selector] wcaps 0x300101: Stereo
  Connection: 3
     0x21* 0x1c 0x1d
Node 0x25 [Audio Selector] wcaps 0x300101: Stereo
  Connection: 3
     0x22* 0x1c 0x1d
Node 0x26 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out
  Amp-Out caps: ofs=0x03, nsteps=0x03, stepsize=0x17, mute=1
  Amp-Out vals:  [0x00]
Node 0x27 [Pin Complex] wcaps 0x400000: Mono
  Pincap 0x00000020: IN
  Pin Default 0x40f000f0: [N/A] Other at Ext N/A
    Conn = Unknown, Color = Unknown
    DefAssociation = 0xf, Sequence = 0x0
  Pin-ctls: 0x00:
Node 0x28 [Volume Knob Widget] wcaps 0x600000: Mono
  Volume-Knob: delta=1, steps=127, direct=1, val=127
  Connection: 2
     0x10 0x11
Codec: Intel G45 DEVCTG
Address: 2
Function Id: 0x1
Vendor Id: 0x80862802
Subsystem Id: 0x80860101
Revision Id: 0x100000
No Modem Function Group found
Default PCM:
    rates [0x0]:
    bits [0x0]:
    formats [0x0]:
Default Amp-In caps: N/A
Default Amp-Out caps: N/A
GPIO: io=0, o=0, i=0, unsolicited=0, wake=0
Node 0x02 [Audio Output] wcaps 0x6211: 8-Channels Digital
  Converter: stream=0, channel=0
  Digital:
  Digital category: 0x0
  PCM:
    rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000
    bits [0x1e]: 16 20 24 32
    formats [0x5]: PCM AC3
Node 0x03 [Pin Complex] wcaps 0x40739d: 8-Channels Digital Amp-Out CP
  Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1
  Amp-Out vals:  [0x00 0x00]
  Pincap 0x00000094: OUT Detect HDMI
  Pin Default 0x18560010: [Jack] Digital Out at Int HDMI
    Conn = Digital, Color = Unknown
    DefAssociation = 0x1, Sequence = 0x0
  Pin-ctls: 0x40: OUT
  Unsolicited: tag=08, enabled=1
  Connection: 1
     0x02
--endcollapse--


!!ALSA Device nodes
!!-----------------

crw-rw----+ 1 root audio 116, 9 Apr  2 19:27 /dev/snd/controlC0
crw-rw----+ 1 root audio 116, 8 Apr  2 19:27 /dev/snd/hwC0D0
crw-rw----+ 1 root audio 116, 7 Apr  2 19:27 /dev/snd/hwC0D2
crw-rw----+ 1 root audio 116, 6 Apr  2 19:27 /dev/snd/pcmC0D0c
crw-rw----+ 1 root audio 116, 5 Apr  2 19:52 /dev/snd/pcmC0D0p
crw-rw----+ 1 root audio 116, 4 Apr  2 19:27 /dev/snd/pcmC0D3p
crw-rw----+ 1 root audio 116, 3 Apr  2 19:27 /dev/snd/seq
crw-rw----+ 1 root audio 116, 2 Apr  2 19:27 /dev/snd/timer

/dev/snd/by-path:
total 0
drwxr-xr-x 2 root root  60 Apr  2 19:27 .
drwxr-xr-x 3 root root 220 Apr  2 19:27 ..
lrwxrwxrwx 1 root root  12 Apr  2 19:27 pci-0000:00:1b.0 -> ../controlC0


!!Aplay/Arecord output
!!------------

APLAY

**** List of PLAYBACK Hardware Devices ****
card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog]
  Subdevices: 0/1
  Subdevice #0: subdevice #0
card 0: Intel [HDA Intel], device 3: INTEL HDMI [INTEL HDMI]
  Subdevices: 1/1
  Subdevice #0: subdevice #0

ARECORD

**** List of CAPTURE Hardware Devices ****
card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog]
  Subdevices: 2/2
  Subdevice #0: subdevice #0
  Subdevice #1: subdevice #1

!!Amixer output
!!-------------

!!-------Mixer controls for card 0 [Intel]

Card hw:0 'Intel'/'HDA Intel at 0xf6adc000 irq 21'
  Mixer name	: 'Intel G45 DEVCTG'
  Components	: 'HDA:111d76b2,10280233,00100302 HDA:80862802,80860101,00100000'
  Controls      : 24
  Simple ctrls  : 15
Simple mixer control 'Master',0
  Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum
  Playback channels: Mono
  Limits: Playback 0 - 64
  Mono: Playback 27 [42%] [-27.75dB] [on]
Simple mixer control 'Headphone',0
  Capabilities: pvolume pswitch penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 64
  Mono:
  Front Left: Playback 64 [100%] [0.00dB] [on]
  Front Right: Playback 64 [100%] [0.00dB] [on]
Simple mixer control 'PCM',0
  Capabilities: pvolume penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 255
  Mono:
  Front Left: Playback 255 [100%] [0.00dB]
  Front Right: Playback 255 [100%] [0.00dB]
Simple mixer control 'Front',0
  Capabilities: pvolume pswitch penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 64
  Mono:
  Front Left: Playback 64 [100%] [0.00dB] [on]
  Front Right: Playback 64 [100%] [0.00dB] [on]
Simple mixer control 'Front Mic Jack Mode',0
  Capabilities: enum
  Items: 'Mic In' 'Line In'
  Item0: 'Mic In'
Simple mixer control 'Mic Jack Mode',0
  Capabilities: enum
  Items: 'Mic In' 'Line In'
  Item0: 'Mic In'
Simple mixer control 'IEC958',0
  Capabilities: pswitch pswitch-joined penum
  Playback channels: Mono
  Mono: Playback [off]
Simple mixer control 'Capture',0
  Capabilities: cvolume cswitch penum
  Capture channels: Front Left - Front Right
  Limits: Capture 0 - 15
  Front Left: Capture 0 [0%] [0.00dB] [off]
  Front Right: Capture 0 [0%] [0.00dB] [off]
Simple mixer control 'Capture',1
  Capabilities: cvolume cswitch penum
  Capture channels: Front Left - Front Right
  Limits: Capture 0 - 15
  Front Left: Capture 0 [0%] [0.00dB] [off]
  Front Right: Capture 0 [0%] [0.00dB] [off]
Simple mixer control 'Digital Mic',0
  Capabilities: cvolume penum
  Capture channels: Front Left - Front Right
  Limits: Capture 0 - 3
  Front Left: Capture 0 [0%] [0.00dB]
  Front Right: Capture 0 [0%] [0.00dB]
Simple mixer control 'Input Source',0
  Capabilities: cenum
  Items: 'Mic' 'Front Mic' 'Digital Mic'
  Item0: 'Digital Mic'
Simple mixer control 'Input Source',1
  Capabilities: cenum
  Items: 'Mic' 'Front Mic' 'Digital Mic'
  Item0: 'Mic'
Simple mixer control 'Mux',0
  Capabilities: cvolume penum
  Capture channels: Front Left - Front Right
  Limits: Capture 0 - 3
  Front Left: Capture 0 [0%] [0.00dB]
  Front Right: Capture 0 [0%] [0.00dB]
Simple mixer control 'Mux',1
  Capabilities: cvolume penum
  Capture channels: Front Left - Front Right
  Limits: Capture 0 - 3
  Front Left: Capture 0 [0%] [0.00dB]
  Front Right: Capture 0 [0%] [0.00dB]
Simple mixer control 'PC Beep',0
  Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum
  Playback channels: Mono
  Limits: Playback 0 - 3
  Mono: Playback 0 [0%] [-18.00dB] [off]


!!Alsactl output
!!-------------

--startcollapse--
state.Intel {
	control.1 {
		comment.access 'read write'
		comment.type INTEGER
		comment.count 2
		comment.range '0 - 64'
		comment.dbmin -4800
		comment.dbmax 0
		iface MIXER
		name 'Front Playback Volume'
		value.0 64
		value.1 64
	}
	control.2 {
		comment.access 'read write'
		comment.type BOOLEAN
		comment.count 2
		iface MIXER
		name 'Front Playback Switch'
		value.0 true
		value.1 true
	}
	control.3 {
		comment.access 'read write'
		comment.type ENUMERATED
		comment.count 1
		comment.item.0 'Mic In'
		comment.item.1 'Line In'
		iface MIXER
		name 'Mic Jack Mode'
		value 'Mic In'
	}
	control.4 {
		comment.access 'read write'
		comment.type ENUMERATED
		comment.count 1
		comment.item.0 'Mic In'
		comment.item.1 'Line In'
		iface MIXER
		name 'Front Mic Jack Mode'
		value 'Mic In'
	}
	control.5 {
		comment.access 'read write'
		comment.type BOOLEAN
		comment.count 1
		iface MIXER
		name 'PC Beep Playback Switch'
		value false
	}
	control.6 {
		comment.access 'read write'
		comment.type INTEGER
		comment.count 1
		comment.range '0 - 3'
		comment.dbmin -1800
		comment.dbmax 0
		iface MIXER
		name 'PC Beep Playback Volume'
		value 0
	}
	control.7 {
		comment.access 'read write'
		comment.type INTEGER
		comment.count 2
		comment.range '0 - 64'
		comment.dbmin -4800
		comment.dbmax 0
		iface MIXER
		name 'Headphone Playback Volume'
		value.0 64
		value.1 64
	}
	control.8 {
		comment.access 'read write'
		comment.type BOOLEAN
		comment.count 2
		iface MIXER
		name 'Headphone Playback Switch'
		value.0 true
		value.1 true
	}
	control.9 {
		comment.access 'read write'
		comment.type INTEGER
		comment.count 2
		comment.range '0 - 15'
		comment.dbmin 0
		comment.dbmax 2250
		iface MIXER
		name 'Capture Volume'
		value.0 0
		value.1 0
	}
	control.10 {
		comment.access 'read write'
		comment.type BOOLEAN
		comment.count 2
		iface MIXER
		name 'Capture Switch'
		value.0 false
		value.1 false
	}
	control.11 {
		comment.access 'read write'
		comment.type INTEGER
		comment.count 2
		comment.range '0 - 15'
		comment.dbmin 0
		comment.dbmax 2250
		iface MIXER
		name 'Capture Volume'
		index 1
		value.0 0
		value.1 0
	}
	control.12 {
		comment.access 'read write'
		comment.type BOOLEAN
		comment.count 2
		iface MIXER
		name 'Capture Switch'
		index 1
		value.0 false
		value.1 false
	}
	control.13 {
		comment.access 'read write'
		comment.type INTEGER
		comment.count 2
		comment.range '0 - 3'
		comment.dbmin 0
		comment.dbmax 3000
		iface MIXER
		name 'Digital Mic Capture Volume'
		value.0 0
		value.1 0
	}
	control.14 {
		comment.access 'read write'
		comment.type INTEGER
		comment.count 2
		comment.range '0 - 3'
		comment.dbmin 0
		comment.dbmax 3000
		iface MIXER
		name 'Mux Capture Volume'
		value.0 0
		value.1 0
	}
	control.15 {
		comment.access 'read write'
		comment.type INTEGER
		comment.count 2
		comment.range '0 - 3'
		comment.dbmin 0
		comment.dbmax 3000
		iface MIXER
		name 'Mux Capture Volume'
		index 1
		value.0 0
		value.1 0
	}
	control.16 {
		comment.access 'read write'
		comment.type ENUMERATED
		comment.count 1
		comment.item.0 Mic
		comment.item.1 'Front Mic'
		comment.item.2 'Digital Mic'
		iface MIXER
		name 'Input Source'
		value 'Digital Mic'
	}
	control.17 {
		comment.access 'read write'
		comment.type ENUMERATED
		comment.count 1
		comment.item.0 Mic
		comment.item.1 'Front Mic'
		comment.item.2 'Digital Mic'
		iface MIXER
		name 'Input Source'
		index 1
		value Mic
	}
	control.18 {
		comment.access 'read write'
		comment.type INTEGER
		comment.count 1
		comment.range '0 - 64'
		comment.dbmin -4800
		comment.dbmax 0
		iface MIXER
		name 'Master Playback Volume'
		value 27
	}
	control.19 {
		comment.access 'read write'
		comment.type BOOLEAN
		comment.count 1
		iface MIXER
		name 'Master Playback Switch'
		value true
	}
	control.20 {
		comment.access read
		comment.type IEC958
		comment.count 1
		iface MIXER
		name 'IEC958 Playback Con Mask'
		value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000'
	}
	control.21 {
		comment.access read
		comment.type IEC958
		comment.count 1
		iface MIXER
		name 'IEC958 Playback Pro Mask'
		value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000'
	}
	control.22 {
		comment.access 'read write'
		comment.type IEC958
		comment.count 1
		iface MIXER
		name 'IEC958 Playback Default'
		value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000'
	}
	control.23 {
		comment.access 'read write'
		comment.type BOOLEAN
		comment.count 1
		iface MIXER
		name 'IEC958 Playback Switch'
		value false
	}
	control.24 {
		comment.access 'read write user'
		comment.type INTEGER
		comment.count 2
		comment.range '0 - 255'
		comment.tlv '0000000100000008ffffec1400000014'
		comment.dbmin -5100
		comment.dbmax 0
		iface MIXER
		name 'PCM Playback Volume'
		value.0 255
		value.1 255
	}
}
--endcollapse--


!!All Loaded Modules
!!------------------

Module
cryptd
aes_x86_64
aes_generic
binfmt_misc
ppdev
snd_hda_codec_intelhdmi
snd_hda_codec_idt
fbcon
tileblit
font
bitblit
softcursor
vga16fb
vgastate
joydev
arc4
pcmcia
snd_hda_intel
snd_hda_codec
snd_hwdep
snd_pcm_oss
snd_mixer_oss
snd_pcm
snd_seq_dummy
snd_seq_oss
snd_seq_midi
snd_rawmidi
snd_seq_midi_event
snd_seq
iwlagn
iwlcore
snd_timer
snd_seq_device
i915
drm_kms_helper
dell_wmi
mac80211
uvcvideo
videodev
v4l1_compat
ricoh_mmc
v4l2_compat_ioctl32
snd
sdhci_pci
psmouse
sdhci
serio_raw
led_class
yenta_socket
rsrc_nonstatic
dell_laptop
dcdbas
cfg80211
pcmcia_core
lp
drm
i2c_algo_bit
soundcore
snd_page_alloc
intel_agp
parport
video
output
usbhid
hid
usb_storage
ohci1394
ahci
ieee1394
e1000e


!!Sysfs Files
!!-----------

/sys/class/sound/hwC0D0/init_pin_configs:
0x0a 0x0421101f
0x0b 0x04a11021
0x0c 0x40f000f0
0x0d 0x90170110
0x0e 0x23a1102e
0x0f 0x23011050
0x14 0x40f000f2
0x18 0x90a601a0
0x19 0x40f000f4
0x1e 0x40f000f5
0x1f 0x40f000f6
0x20 0x40f000f7
0x27 0x40f000f0

/sys/class/sound/hwC0D0/driver_pin_configs:
0x0a 0x0421101f
0x0b 0x04a11221
0x0c 0x40f000f0
0x0d 0x90170110
0x0e 0x23a1902e
0x0f 0x23014250
0x14 0x40f000f0
0x18 0x90a000f0
0x19 0x40f000f0
0x1e 0x4f0000f0
0x1f 0x4f0000f0

/sys/class/sound/hwC0D0/user_pin_configs:

/sys/class/sound/hwC0D0/init_verbs:

/sys/class/sound/hwC0D2/init_pin_configs:
0x03 0x18560010

/sys/class/sound/hwC0D2/driver_pin_configs:

/sys/class/sound/hwC0D2/user_pin_configs:

/sys/class/sound/hwC0D2/init_verbs:


!!ALSA/HDA dmesg
!!------------------

[   11.054133] [drm] Initialized i915 1.6.0 20080730 for 0000:00:02.0 on minor 0
[   11.054177] HDA Intel 0000:00:1b.0: PCI INT A -> GSI 21 (level, low) -> IRQ 21
[   11.054209] HDA Intel 0000:00:1b.0: setting latency timer to 64
[   11.056448] vga16fb: initializing
--
[   11.213314] Console: switching to colour frame buffer device 180x56
[   11.218019] input: HDA Digital PCBeep as /devices/pci0000:00/0000:00:1b.0/input/input12
[   11.241280] input: HDA Intel Mic at Sep Left Jack as /devices/pci0000:00/0000:00:1b.0/sound/card0/input13
[   11.241442] input: HDA Intel Mic at Ext Right Jack as /devices/pci0000:00/0000:00:1b.0/sound/card0/input14
[   11.241554] input: HDA Intel Line Out at Sep Left Jack as /devices/pci0000:00/0000:00:1b.0/sound/card0/input15
[   11.241662] input: HDA Intel HP Out at Ext Right Jack as /devices/pci0000:00/0000:00:1b.0/sound/card0/input16
[   11.501294] dell-wmi: Unknown key ffd1 pressed



[-- Attachment #3: Type: text/plain, Size: 160 bytes --]

_______________________________________________
Alsa-devel mailing list
Alsa-devel@alsa-project.org
http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

Re: wrong decibel data?

[Nicolo' Chieffo]

I just had a discussion with pulseaudio developers in which they told
me that if the speakers don't emit any audio when the ALSA volume is
0%, then the correct gain value should be -inf dB (a value lower than
-200 for pulseaudio means mute). Unfortunately my card has -48 dB.

They also said that if with -47.25 dB (which is what I get with ALSA
volume set to 2%) I can hear audio even if it's very very low, at -48
dB I should still be able to hear something (the scale is
logarithmic).

Their request is to change to -200 the gain reported from my card
driver when the ALSA volume is set to 0%.
Thanks

Re: wrong decibel data?

[Colin Guthrie]

'Twas brillig, and Nicolo' Chieffo at 02/04/10 21:25 did gyre and gimble:
> I just had a discussion with pulseaudio developers in which they told
> me that if the speakers don't emit any audio when the ALSA volume is
> 0%, then the correct gain value should be -inf dB (a value lower than
> -200 for pulseaudio means mute). Unfortunately my card has -48 dB.
> 
> They also said that if with -47.25 dB (which is what I get with ALSA
> volume set to 2%) I can hear audio even if it's very very low, at -48
> dB I should still be able to hear something (the scale is
> logarithmic).
> 
> Their request is to change to -200 the gain reported from my card
> driver when the ALSA volume is set to 0%.
> Thanks

FWIW, I've got the same/similar h/w with a cutoff at 14% in PA as you
have. I've been meaning to get this fixed for a while, but I'm
incredibly lazy with certain things that don't bother me practically, so
haven't followed it up yet.

Thanks for getting the ball rolling :)

If any specific debug is needed here, feel free to ask.

Col

-- 

Colin Guthrie
gmane(at)colin.guthr.ie
http://colin.guthr.ie/

Day Job:
  Tribalogic Limited [http://www.tribalogic.net/]
Open Source:
  Mandriva Linux Contributor [http://www.mandriva.com/]
  PulseAudio Hacker [http://www.pulseaudio.org/]
  Trac Hacker [http://trac.edgewall.org/]

Re: wrong decibel data?

[Nicolo' Chieffo]

I don't think that any specific debug is needed. pulseaudio developers
just told that you have to declare -200 dB when the sound is disabled,
that's it.
Is this a difficult thing to do? which files must be touched?

On Sat, Apr 3, 2010 at 12:09 PM, Colin Guthrie <gmane@colin.guthr.ie> wrote:
> 'Twas brillig, and Nicolo' Chieffo at 02/04/10 21:25 did gyre and gimble:
>> I just had a discussion with pulseaudio developers in which they told
>> me that if the speakers don't emit any audio when the ALSA volume is
>> 0%, then the correct gain value should be -inf dB (a value lower than
>> -200 for pulseaudio means mute). Unfortunately my card has -48 dB.
>>
>> They also said that if with -47.25 dB (which is what I get with ALSA
>> volume set to 2%) I can hear audio even if it's very very low, at -48
>> dB I should still be able to hear something (the scale is
>> logarithmic).
>>
>> Their request is to change to -200 the gain reported from my card
>> driver when the ALSA volume is set to 0%.
>> Thanks
>
> FWIW, I've got the same/similar h/w with a cutoff at 14% in PA as you
> have. I've been meaning to get this fixed for a while, but I'm
> incredibly lazy with certain things that don't bother me practically, so
> haven't followed it up yet.
>
> Thanks for getting the ball rolling :)
>
> If any specific debug is needed here, feel free to ask.
>
> Col
>
> --
>
> Colin Guthrie
> gmane(at)colin.guthr.ie
> http://colin.guthr.ie/
>
> Day Job:
>  Tribalogic Limited [http://www.tribalogic.net/]
> Open Source:
>  Mandriva Linux Contributor [http://www.mandriva.com/]
>  PulseAudio Hacker [http://www.pulseaudio.org/]
>  Trac Hacker [http://trac.edgewall.org/]
>
> _______________________________________________
> Alsa-devel mailing list
> Alsa-devel@alsa-project.org
> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
>
_______________________________________________
Alsa-devel mailing list
Alsa-devel@alsa-project.org
http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

Re: wrong decibel data?

[Raymond Yau]

2010/4/3 Nicolo' Chieffo <nicolo.chieffo@gmail.com>

> I don't think that any specific debug is needed. pulseaudio developers
> just told that you have to declare -200 dB when the sound is disabled,
> that's it.
> Is this a difficult thing to do? which files must be touched?
>
>


to proivide dB scale of pulse master volume control of the alsa mixer
applications

alsa-plugins/pulse/ctl_pulse.c

Stereo Support in APLAY

[Reddy, MR Swami]

Hello,
Could you plz confirm, if the ALSA drivers support the stereo format files
with 'aplay'?

PS: Iam using the 1.0.5 version

Thanks
Swami

Re: Stereo Support in APLAY

[Clemens Ladisch]

Reddy, MR Swami wrote:
> Could you plz confirm, if the ALSA drivers support the stereo format files
> with 'aplay'?

aplay supports stereo files; whether the driver supports this depends
on the driver.

I don't know what you problem is, but it's likely that you tried to
play a stereo file on a mono device or a mono file on a stereo-only
device, and that you used a device name like "hw:0" that disables all
automatic format conversions, instead of "default".


Regards,
Clemens

Re: wrong decibel data?

[Raymond Yau]

2010/4/3 Colin Guthrie <gmane@colin.guthr.ie>

> 'Twas brillig, and Nicolo' Chieffo at 02/04/10 21:25 did gyre and gimble:
> > I just had a discussion with pulseaudio developers in which they told
> > me that if the speakers don't emit any audio when the ALSA volume is
> > 0%, then the correct gain value should be -inf dB (a value lower than
> > -200 for pulseaudio means mute). Unfortunately my card has -48 dB.
> >
> > They also said that if with -47.25 dB (which is what I get with ALSA
> > volume set to 2%) I can hear audio even if it's very very low, at -48
> > dB I should still be able to hear something (the scale is
> > logarithmic).
> >
> > Their request is to change to -200 the gain reported from my card
> > driver when the ALSA volume is set to 0%.
> > Thanks
>
> FWIW, I've got the same/similar h/w with a cutoff at 14% in PA as you
> have. I've been meaning to get this fixed for a while, but I'm
> incredibly lazy with certain things that don't bother me practically, so
> haven't followed it up yet.
>
> Thanks for getting the ball rolling :)
>
> If any specific debug is needed here, feel free to ask.
>
> Col
>
>  <http://mailman.alsa-project.org/mailman/listinfo/alsa-devel>
>

if -48dB of alsa volume represent (100-14)% of the volume range of
pulseaudio control

i.e. the volume range of pulseaudio is only -63dB  , why do you use -200dB
?

Re: wrong decibel data?

[Raymond Yau]

2010/4/3 Colin Guthrie <gmane@colin.guthr.ie>

> 'Twas brillig, and Nicolo' Chieffo at 02/04/10 21:25 did gyre and gimble:
> > I just had a discussion with pulseaudio developers in which they told
> > me that if the speakers don't emit any audio when the ALSA volume is
> > 0%, then the correct gain value should be -inf dB (a value lower than
> > -200 for pulseaudio means mute). Unfortunately my card has -48 dB.
> >
> > They also said that if with -47.25 dB (which is what I get with ALSA
> > volume set to 2%) I can hear audio even if it's very very low, at -48
> > dB I should still be able to hear something (the scale is
> > logarithmic).
> >
> > Their request is to change to -200 the gain reported from my card
> > driver when the ALSA volume is set to 0%.
> > Thanks
>
> FWIW, I've got the same/similar h/w with a cutoff at 14% in PA as you
> have. I've been meaning to get this fixed for a while, but I'm
> incredibly lazy with certain things that don't bother me practically, so
> haven't followed it up yet.
>
> Thanks for getting the ball rolling :)
>
> If any specific debug is needed here, feel free to ask.
>
> Col
>
> --
>

since PA provide software atten/software gain ,the software atten of 16bits
audio is -48dB

I guess the problem is related to PA shift the 0dB point (i.e. the software
gain +15dfB to 0dB)

you have to ask PA developer the value of software dB gain provided by the
PA server

Re: wrong decibel data?

[Raymond Yau]

2010/4/3 Colin Guthrie <gmane@colin.guthr.ie>

>
>
> FWIW, I've got the same/similar h/w with a cutoff at 14% in PA as you
> have. I've been meaning to get this fixed for a while, but I'm
> incredibly lazy with certain things that don't bother me practically, so
> haven't followed it up yet.
>
> Thanks for getting the ball rolling :)
>
> If any specific debug is needed here, feel free to ask.
>
> Col
>
> for debugging  , please post the output of alsa-info.sh of your hardware
and output of "amixer -D pulse" when you change the volume control  to 0% by
"alsamixer -c 0"

Re: wrong decibel data?

[Raymond Yau]

2010/4/3 Colin Guthrie <gmane@colin.guthr.ie>

> 'Twas brillig, and Nicolo' Chieffo at 02/04/10 21:25 did gyre and gimble:
> > I just had a discussion with pulseaudio developers in which they told
> > me that if the speakers don't emit any audio when the ALSA volume is
> > 0%, then the correct gain value should be -inf dB (a value lower than
> > -200 for pulseaudio means mute). Unfortunately my card has -48 dB.
> >
> > They also said that if with -47.25 dB (which is what I get with ALSA
> > volume set to 2%) I can hear audio even if it's very very low, at -48
> > dB I should still be able to hear something (the scale is
> > logarithmic).
> >
> > Their request is to change to -200 the gain reported from my card
> > driver when the ALSA volume is set to 0%.
> > Thanks
>
> FWIW, I've got the same/similar h/w with a cutoff at 14% in PA as you
> have. I've been meaning to get this fixed for a while, but I'm
> incredibly lazy with certain things that don't bother me practically, so
> haven't followed it up yet.
>
> Thanks for getting the ball rolling :)
>
> If any specific debug is needed here, feel free to ask.
>
> Col
>
> --
>
>
some sound card may not have hardware volume control ,(e.g. those the mic of
those low cost web cam ) which use softvol plugin which has a default -51dB
min_dB

do you mean that you also need to change it to -200 dB ?

Re: wrong decibel data?

[Raymond Yau]

2010/4/3 Colin Guthrie <gmane@colin.guthr.ie>

> 'Twas brillig, and Nicolo' Chieffo at 02/04/10 21:25 did gyre and gimble:
> > I just had a discussion with pulseaudio developers in which they told
> > me that if the speakers don't emit any audio when the ALSA volume is
> > 0%, then the correct gain value should be -inf dB (a value lower than
> > -200 for pulseaudio means mute). Unfortunately my card has -48 dB.
> >
> > They also said that if with -47.25 dB (which is what I get with ALSA
> > volume set to 2%) I can hear audio even if it's very very low, at -48
> > dB I should still be able to hear something (the scale is
> > logarithmic).
> >
> > Their request is to change to -200 the gain reported from my card
> > driver when the ALSA volume is set to 0%.
> > Thanks
>
> FWIW, I've got the same/similar h/w with a cutoff at 14% in PA as you
> have. I've been meaning to get this fixed for a while, but I'm
> incredibly lazy with certain things that don't bother me practically, so
> haven't followed it up yet.
>
> Thanks for getting the ball rolling :)
>
> If any specific debug is needed here, feel free to ask.
>
> Col
>
>
I don't agree *Lennart Poettering say *"ALSA does not define any reference
point for the dB scale it exports in its mixer controls "

For most of the driver with dB scale,  the 0dB point is the reference point
,  the reason is PA developer choose a cubic mapping  "It's now cubic, which
is supposed to be feel more 'natural'."


https://tango.0pointer.de/pipermail/pulseaudio-discuss/2009-May/003898.html

>> Because PA exposes the same volume scale mapping on all hardware.

>> In 0.9.15 we mapped 0% on the volume scale to -90dB and 100% to 0dB,
in between the mapping between those percentages and the dB scale was
linear (i.e. to the effect that we had an overall logarithmic
mapping).

>> This mapping is not particularly well chosen since it gives too much
control over the uninteresting parts below -20dB and too little
control over the 'interesting' parts above -20dB. That's why I modified
the mapping in PA git. It's now cubic, which is supposed to be feel
more 'natural'.

>> So, coming back to how PA maps those volumes to ALSA: alsamixer will expose
the exact discrete volume steps of the sound card and show dB
information just as little help at the side. PA however controls the
volume in dB and if the hw doesn't provide the requested volume step
we go to the next higher and then attenuate by the remaining factor in
software. That way we can provide the same volume range and granularity on all
hardware with the same mapping from the UI to the attenuation
factor. So basically, while tha mapping from those percentages to the
loudness is different in PA and alsamixer, the dB scaling is mostly
the same -- except when it is not...

Re: wrong decibel data?

[Colin Guthrie]

'Twas brillig, and Raymond Yau at 16/04/10 14:48 did gyre and gimble:
> 2010/4/3 Colin Guthrie <gmane@colin.guthr.ie>

Sorry for the top post but....

Raymond. Your replies are often very disjoint. I see no relevance to my
message here and why you're replying here is beyond me.

It's totally unclear what point you are making and which parts of this
message are your comments and which are Lennart's original comments
that's you've just pasted in.

What are you trying to say?

Col

> I don't agree *Lennart Poettering say *"ALSA does not define any reference
> point for the dB scale it exports in its mixer controls "
> 
> For most of the driver with dB scale,  the 0dB point is the reference point
> ,  the reason is PA developer choose a cubic mapping  "It's now cubic, which
> is supposed to be feel more 'natural'."
> 
> 
> https://tango.0pointer.de/pipermail/pulseaudio-discuss/2009-May/003898.html
> 
>>> Because PA exposes the same volume scale mapping on all hardware.
> 
>>> In 0.9.15 we mapped 0% on the volume scale to -90dB and 100% to 0dB,
> in between the mapping between those percentages and the dB scale was
> linear (i.e. to the effect that we had an overall logarithmic
> mapping).
> 
>>> This mapping is not particularly well chosen since it gives too much
> control over the uninteresting parts below -20dB and too little
> control over the 'interesting' parts above -20dB. That's why I modified
> the mapping in PA git. It's now cubic, which is supposed to be feel
> more 'natural'.
> 
>>> So, coming back to how PA maps those volumes to ALSA: alsamixer will expose
> the exact discrete volume steps of the sound card and show dB
> information just as little help at the side. PA however controls the
> volume in dB and if the hw doesn't provide the requested volume step
> we go to the next higher and then attenuate by the remaining factor in
> software. That way we can provide the same volume range and granularity on all
> hardware with the same mapping from the UI to the attenuation
> factor. So basically, while tha mapping from those percentages to the
> loudness is different in PA and alsamixer, the dB scaling is mostly
> the same -- except when it is not...


-- 

Colin Guthrie
gmane(at)colin.guthr.ie
http://colin.guthr.ie/

Day Job:
  Tribalogic Limited [http://www.tribalogic.net/]
Open Source:
  Mandriva Linux Contributor [http://www.mandriva.com/]
  PulseAudio Hacker [http://www.pulseaudio.org/]
  Trac Hacker [http://trac.edgewall.org/]

Re: wrong decibel data?

[Raymond Yau]

2010/4/17 Colin Guthrie <gmane@colin.guthr.ie>

> 'Twas brillig, and Raymond Yau at 16/04/10 14:48 did gyre and gimble:
> > 2010/4/3 Colin Guthrie <gmane@colin.guthr.ie>
>
> Sorry for the top post but....
>
> Raymond. Your replies are often very disjoint. I see no relevance to my
> message here and why you're replying here is beyond me.
>
> It's totally unclear what point you are making and which parts of this
> message are your comments and which are Lennart's original comments
> that's you've just pasted in.
>
> What are you trying to say?
>
> Col
>

http://pulseaudio.org/ticket/580

>> Basically, if the signal is completely cut off, then the attenuation is
-inf dB.

you have to sum the dB gain of  the SPL of the speaker/headphone when you
want to calculate the dB ( sound pressure and the distance )


>> FWIW, I've got the same/similar h/w with a cutoff at 14% in PA as you
have. I've been meaning to get this fixed for a while, but I'm
incredibly lazy with certain things that don't bother me practically, so
haven't followed it up yet.

>> If any specific debug is needed here, feel free to ask.

Just reply since you still have similar problem , but it seem that you are
unwilling to provide info to debug your case , at least you have to provide
output of alsa-info.sh even when you have the same/similar h/w  ,

do your laptop has the volume knob ?

https://tango.0pointer.de/pipermail/pulseaudio-discuss/2009-May/003898.html

If pulseaudio provide per-application volume control, why the PA community
propose to throw away the volume slider of the application. (i.e. why only
allow pavucontrol to change the per-application volume ?

http://thread.gmane.org/gmane.comp.audio.pulseaudio.general/6521/focus=6524

>> Making the application volume sliders control the device volume isn't
good either, because if an application has a volume slider, the natural
assumption made by users is that the slider controls only that application's
volume.

>> Solution: throw away volume sliders in applications, and promote centralized volume management with volume applets and hardware controls.


I suggest PA should provide the pre-application volume control for the
application too since from the viewpoint of PA when PA have to control all
the volume , it is better not allow application to control alsa master
volume control. but there is no reason to remove the volume sliders in
application

http://git.alsa-project.org/?p=alsa-tools.git;a=blob_plain;f=*hwmixvolume*/README
<http://git.alsa-project.org/?p=alsa-tools.git;a=blob_plain;f=hwmixvolume/README>

The per-application volume control is quite similar to the per voice
volume control of those hardware mixing sound cards ,
although there are some differences

 the per voice volume control of hardware mixing sound card provide digital
gain/atten by the DSP ( not those gain/atten of  DAC )

http://thread.gmane.org/gmane.linux.alsa.devel/24638/focus=24707

Re: wrong decibel data?

[Colin Guthrie]

'Twas brillig, and Raymond Yau at 21/04/10 03:32 did gyre and gimble:
> 2010/4/17 Colin Guthrie <gmane@colin.guthr.ie>
> 
>> 'Twas brillig, and Raymond Yau at 16/04/10 14:48 did gyre and gimble:
>>> 2010/4/3 Colin Guthrie <gmane@colin.guthr.ie>
>>
>> Sorry for the top post but....
>>
>> Raymond. Your replies are often very disjoint. I see no relevance to my
>> message here and why you're replying here is beyond me.
>>
>> It's totally unclear what point you are making and which parts of this
>> message are your comments and which are Lennart's original comments
>> that's you've just pasted in.
>>
>> What are you trying to say?
>>
>> Col
>>
> 
> http://pulseaudio.org/ticket/580
> 
>>> Basically, if the signal is completely cut off, then the attenuation is
> -inf dB.
> 
> you have to sum the dB gain of  the SPL of the speaker/headphone when you
> want to calculate the dB ( sound pressure and the distance )
> 
> 
>>> FWIW, I've got the same/similar h/w with a cutoff at 14% in PA as you
> have. I've been meaning to get this fixed for a while, but I'm
> incredibly lazy with certain things that don't bother me practically, so
> haven't followed it up yet.
> 
>>> If any specific debug is needed here, feel free to ask.
> 
> Just reply since you still have similar problem , but it seem that you are
> unwilling to provide info to debug your case , at least you have to provide
> output of alsa-info.sh even when you have the same/similar h/w  ,


What debug have you asked me for? It's impossible to tell from the
vaguely relevant quotations (poorly formatted in email so that only half
of the quoted text actually looks like a quote) and random specification
sheet snippets you post that you're actually asking for anything at all!

If you want information, ask for it clearly and I'll do my best to
answer it.


> If pulseaudio provide per-application volume control, why the PA community
> propose to throw away the volume slider of the application. (i.e. why only
> allow pavucontrol to change the per-application volume ?
> 
> http://thread.gmane.org/gmane.comp.audio.pulseaudio.general/6521/focus=6524

That is a discussion, not a statement of intent. As I stated on that
thread, I personally disagree with Tanu's position, but if you are
interested in this discussion, feel free to join in on that list. It's
nothing to do with this thread at all however (it's purely a UI and user
expectation of cause + effect), so let's not drag this thread further
into the land of obscure tangents.

>>> Making the application volume sliders control the device volume isn't
> good either, because if an application has a volume slider, the natural
> assumption made by users is that the slider controls only that application's
> volume.
> 
>>> Solution: throw away volume sliders in applications, and promote centralized volume management with volume applets and hardware controls.
> 
> 
> I suggest PA should provide the pre-application volume control for the
> application too since from the viewpoint of PA when PA have to control all
> the volume , it is better not allow application to control alsa master
> volume control. but there is no reason to remove the volume sliders in
> application

I suggest you discuss this issue on the actual thread itself rather than
pull it in to a completely different one. I do however think you've
missed the point, but like I say that is something to discuss there, not
here.

Col

-- 

Colin Guthrie
gmane(at)colin.guthr.ie
http://colin.guthr.ie/

Day Job:
  Tribalogic Limited [http://www.tribalogic.net/]
Open Source:
  Mandriva Linux Contributor [http://www.mandriva.com/]
  PulseAudio Hacker [http://www.pulseaudio.org/]
  Trac Hacker [http://trac.edgewall.org/]

Re: wrong decibel data?

[Raymond Yau]

2010/4/21 Colin Guthrie <gmane@colin.guthr.ie>

> 'Twas brillig, and Raymond Yau at 21/04/10 03:32 did gyre and gimble:
> > 2010/4/17 Colin Guthrie <gmane@colin.guthr.ie>
> >
> >> 'Twas brillig, and Raymond Yau at 16/04/10 14:48 did gyre and gimble:
> >>> 2010/4/3 Colin Guthrie <gmane@colin.guthr.ie>
> >>
> >> Sorry for the top post but....
> >>
> >> Raymond. Your replies are often very disjoint. I see no relevance to my
> >> message here and why you're replying here is beyond me.
> >>
> >> It's totally unclear what point you are making and which parts of this
> >> message are your comments and which are Lennart's original comments
> >> that's you've just pasted in.
> >>
> >> What are you trying to say?
> >>
> >> Col
> >>
> >
> > http://pulseaudio.org/ticket/580
> >
> >>> Basically, if the signal is completely cut off, then the attenuation is
> > -inf dB.
> >
> > you have to sum the dB gain of  the SPL of the speaker/headphone when you
> > want to calculate the dB ( sound pressure and the distance )
> >
> >
> >>> FWIW, I've got the same/similar h/w with a cutoff at 14% in PA as you
> > have. I've been meaning to get this fixed for a while, but I'm
> > incredibly lazy with certain things that don't bother me practically, so
> > haven't followed it up yet.
> >
> >>> If any specific debug is needed here, feel free to ask.
> >
> > Just reply since you still have similar problem , but it seem that you
> are
> > unwilling to provide info to debug your case , at least you have to
> provide
> > output of alsa-info.sh even when you have the same/similar h/w  ,
>
>
> What debug have you asked me for?


Just output of alsa-info.sh as stated in my previous email

and please confirm that PA provide software gain


> It's impossible to tell from the
> vaguely relevant quotations (poorly formatted in email so that only half
> of the quoted text actually looks like a quote) and random specification
> sheet snippets you post that you're actually asking for anything at all!
>
> If you want information, ask for it clearly and I'll do my best to
> answer it.
>
>
> > If pulseaudio provide per-application volume control, why the PA
> community
> > propose to throw away the volume slider of the application. (i.e. why
> only
> > allow pavucontrol to change the per-application volume ?
> >
> >
> http://thread.gmane.org/gmane.comp.audio.pulseaudio.general/6521/focus=6524
>
> That is a discussion, not a statement of intent. As I stated on that
> thread, I personally disagree with Tanu's position, but if you are
> interested in this discussion, feel free to join in on that list. It's
> nothing to do with this thread at all however (it's purely a UI and user
> expectation of cause + effect), so let's not drag this thread further
> into the land of obscure tangents.
>
> >>> Making the application volume sliders control the device volume isn't
> > good either, because if an application has a volume slider, the natural
> > assumption made by users is that the slider controls only that
> application's
> > volume.
> >
> >>> Solution: throw away volume sliders in applications, and promote
> centralized volume management with volume applets and hardware controls.
> >
> >
> > I suggest PA should provide the pre-application volume control for the
> > application too since from the viewpoint of PA when PA have to control
> all
> > the volume , it is better not allow application to control alsa master
> > volume control. but there is no reason to remove the volume sliders in
> > application
>
> I suggest you discuss this issue on the actual thread itself rather than
> pull it in to a completely different one. I do however think you've
> missed the point, but like I say that is something to discuss there, not
> here.
>


I just make suggestion only since I don't know whether it is feasible  to
implement IFACE_PCM in ctl_pulse.c but I strongly recommended PA provide dB
scale for the Master Volume control for the pulse device ( alsa-pulse plugin
) in order for the mixer application to provide a more useful info to the
user


>
> Col
>
>

Re: wrong decibel data?

[Colin Guthrie]

[-- Attachment #1: Type: text/plain, Size: 713 bytes --]

'Twas brillig, and Raymond Yau at 22/04/10 02:16 did gyre and gimble:
> Just output of alsa-info.sh as stated in my previous email

Oh I've only just noticed this reply (prompted by a comment on another
related thread).

Apologies, but I'd not seen this request until now.

I attach my alsa-info.sh output.


> and please confirm that PA provide software gain

Not sure precisely what you mean by this in this context.


Col


-- 

Colin Guthrie
gmane(at)colin.guthr.ie
http://colin.guthr.ie/

Day Job:
  Tribalogic Limited [http://www.tribalogic.net/]
Open Source:
  Mandriva Linux Contributor [http://www.mandriva.com/]
  PulseAudio Hacker [http://www.pulseaudio.org/]
  Trac Hacker [http://trac.edgewall.org/]

[-- Attachment #2: alsa-info.txt --]
[-- Type: text/plain, Size: 20557 bytes --]

upload=true&script=true&cardinfo=
!!################################
!!ALSA Information Script v 0.4.59
!!################################

!!Script ran on: Wed May 26 22:15:09 UTC 2010


!!Linux Distribution
!!------------------

Mandriva Linux release 2010.1 (Cooker) for x86_64 Kernel 2.6.33.4-desktop-2mnb on a Dual-processor x86_64 / \l LSB_VERSION=lsb-4.0-amd64:lsb-4.0-noarch DISTRIB_ID=MandrivaLinux DISTRIB_DESCRIPTION="Mandriva Linux 2010.1" Mandriva Linux release 2010.1 (Official) for x86_64 Mandriva Linux release 2010.1 (Official) for x86_64 Mandriva Linux release 2010.1 (Official) for x86_64 Mandriva Linux release 2010.1 (Official) for x86_64 Mandriva Linux release 2010.1 (Official) for x86_64


!!DMI Information
!!---------------

Manufacturer:      Dell Inc.
Product Name:      MM061                           


!!Kernel Information
!!------------------

Kernel release:    2.6.33.4-desktop-2mnb
Operating System:  GNU/Linux
Architecture:      x86_64
Processor:         x86_64
SMP Enabled:       Yes


!!ALSA Version
!!------------

Driver version:     1.0.21
Library version:    1.0.23
Utilities version:  1.0.23


!!Loaded ALSA modules
!!-------------------

snd_hda_intel


!!Sound Servers on this system
!!----------------------------

Pulseaudio:
      Installed - Yes (/usr/bin/pulseaudio)
      Running - Yes

ESound Daemon:
      Installed - Yes (/usr/bin/esd)
      Running - No

Jack:
      Installed - Yes (/usr/bin/jackd)
      Running - No


!!Soundcards recognised by ALSA
!!-----------------------------

 0 [Intel          ]: HDA-Intel - HDA Intel
                      HDA Intel at 0xefebc000 irq 27


!!PCI Soundcards installed in the system
!!--------------------------------------

00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 01)


!!Advanced information - PCI Vendor/Device/Susbsystem ID's
!!--------------------------------------------------------

00:1b.0 0403: 8086:27d8 (rev 01)
	Subsystem: 1028:01bd


!!Modprobe options (Sound related)
!!--------------------------------

snd-ac97-codec: power_save=1


!!HDA-Intel Codec information
!!---------------------------
--startcollapse--

Codec: SigmaTel STAC9200
Address: 0
Function Id: 0x1
Vendor Id: 0x83847690
Subsystem Id: 0x102801bd
Revision Id: 0x102201
No Modem Function Group found
Default PCM:
    rates [0x7e0]: 44100 48000 88200 96000 176400 192000
    bits [0xe]: 16 20 24
    formats [0x1]: PCM
Default Amp-In caps: N/A
Default Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1
GPIO: io=4, o=0, i=0, unsolicited=1, wake=1
  IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0
  IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0
  IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0
  IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0
Node 0x02 [Audio Output] wcaps 0xd0401: Stereo
  Device: name="STAC92xx Analog", type="Audio", device=0
  Converter: stream=5, channel=0
  Power: setting=D0, actual=D0
  Delay: 13 samples
Node 0x03 [Audio Input] wcaps 0x1d0541: Stereo
  Device: name="STAC92xx Analog", type="Audio", device=0
  Converter: stream=0, channel=0
  SDI-Select: 0
  Power: setting=D0, actual=D0
  Delay: 13 samples
  Connection: 1
     0x0a
  Processing caps: benign=0, ncoeff=0
Node 0x04 [Audio Input] wcaps 0x140311: Stereo Digital
  Converter: stream=0, channel=0
  SDI-Select: 0
  Digital:
  Digital category: 0x0
  PCM:
    rates [0x160]: 44100 48000 96000
    bits [0xe]: 16 20 24
    formats [0x5]: PCM AC3
  Delay: 4 samples
  Connection: 1
     0x08
Node 0x05 [Audio Output] wcaps 0x40211: Stereo Digital
  Control: name="IEC958 Playback Con Mask", index=0, device=0
  Control: name="IEC958 Playback Pro Mask", index=0, device=0
  Control: name="IEC958 Playback Default", index=0, device=0
  Control: name="IEC958 Playback Switch", index=0, device=0
  Control: name="IEC958 Default PCM Playback Switch", index=0, device=0
  Device: name="STAC92xx Digital", type="SPDIF", device=1
  Converter: stream=5, channel=0
  Digital:
  Digital category: 0x0
  PCM:
    rates [0x1e0]: 44100 48000 88200 96000
    bits [0xe]: 16 20 24
    formats [0x5]: PCM AC3
  Delay: 4 samples
Node 0x06 [Vendor Defined Widget] wcaps 0xf30201: Stereo Digital
  Delay: 3 samples
Node 0x07 [Audio Selector] wcaps 0x300901: Stereo R/L
  Connection: 3
     0x02* 0x08 0x0a
Node 0x08 [Pin Complex] wcaps 0x430681: Stereo Digital
  Pincap 0x00010024: IN EAPD Detect
  EAPD 0x0:
  Pin Default 0x40c003fa: [N/A] SPDIF In at Ext N/A
    Conn = Unknown, Color = Unknown
    DefAssociation = 0xf, Sequence = 0xa
    Misc = NO_PRESENCE
  Pin-ctls: 0x00:
  Unsolicited: tag=00, enabled=0
  Power: setting=D0, actual=D0
  Delay: 3 samples
Node 0x09 [Pin Complex] wcaps 0x400301: Stereo Digital
  Pincap 0x00000010: OUT
  Pin Default 0x01441340: [Jack] SPDIF Out at Ext Rear
    Conn = RCA, Color = Black
    DefAssociation = 0x4, Sequence = 0x0
    Misc = NO_PRESENCE
  Pin-ctls: 0x40: OUT
  Connection: 2
     0x05* 0x0a
Node 0x0a [Audio Selector] wcaps 0x30090d: Stereo Amp-Out R/L
  Control: name="Capture Volume", index=0, device=0
  Control: name="Capture Switch", index=0, device=0
  Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1
  Amp-Out vals:  [0x00 0x00]
  Connection: 1
     0x0c
Node 0x0b [Audio Selector] wcaps 0x300105: Stereo Amp-Out
  Control: name="Master Playback Volume", index=0, device=0
  Control: name="Master Playback Switch", index=0, device=0
  Amp-Out caps: N/A
  Amp-Out vals:  [0x1e 0x1e]
  Connection: 1
     0x07
Node 0x0c [Audio Selector] wcaps 0x30010d: Stereo Amp-Out
  Control: name="Mux Capture Volume", index=0, device=0
  Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0
  Amp-Out vals:  [0x03 0x03]
  Connection: 5
     0x10* 0x0f 0x0e 0x0d 0x12
Node 0x0d [Pin Complex] wcaps 0x400181: Stereo
  Pincap 0x0000003f: IN OUT HP Detect Trigger ImpSense
  Pin Default 0x0421121f: [Jack] HP Out at Ext Right
    Conn = 1/8, Color = Black
    DefAssociation = 0x1, Sequence = 0xf
  Pin-ctls: 0x00:
  Unsolicited: tag=01, enabled=1
  Connection: 1
     0x0b
Node 0x0e [Pin Complex] wcaps 0x400181: Stereo
  Pincap 0x0000003f: IN OUT HP Detect Trigger ImpSense
  Pin Default 0x90170310: [Fixed] Speaker at Int N/A
    Conn = Analog, Color = Unknown
    DefAssociation = 0x1, Sequence = 0x0
    Misc = NO_PRESENCE
  Pin-ctls: 0x40: OUT
  Unsolicited: tag=00, enabled=0
  Connection: 1
     0x0b
Node 0x0f [Pin Complex] wcaps 0x400181: Stereo
  Pincap 0x00000037: IN OUT Detect Trigger ImpSense
  Pin Default 0x408003fb: [N/A] Line In at Ext N/A
    Conn = Unknown, Color = Unknown
    DefAssociation = 0xf, Sequence = 0xb
    Misc = NO_PRESENCE
  Pin-ctls: 0x20: IN
  Unsolicited: tag=00, enabled=0
  Connection: 1
     0x0b
Node 0x10 [Pin Complex] wcaps 0x400181: Stereo
  Pincap 0x00001737: IN OUT Detect Trigger ImpSense
    Vref caps: HIZ 50 GRD 80
  Pin Default 0x04a11020: [Jack] Mic at Ext Right
    Conn = 1/8, Color = Black
    DefAssociation = 0x2, Sequence = 0x0
  Pin-ctls: 0x24: IN VREF_80
  Unsolicited: tag=02, enabled=1
  Connection: 1
     0x0b
Node 0x11 [Pin Complex] wcaps 0x400104: Mono Amp-Out
  Amp-Out caps: N/A
  Amp-Out vals:  [0x00]
  Pincap 0x00000010: OUT
  Pin Default 0x401003fc: [N/A] Speaker at Ext N/A
    Conn = Unknown, Color = Unknown
    DefAssociation = 0xf, Sequence = 0xc
    Misc = NO_PRESENCE
  Pin-ctls: 0x00:
  Connection: 1
     0x13
Node 0x12 [Pin Complex] wcaps 0x400001: Stereo
  Pincap 0x00000020: IN
  Pin Default 0x403003fd: [N/A] CD at Ext N/A
    Conn = Unknown, Color = Unknown
    DefAssociation = 0xf, Sequence = 0xd
    Misc = NO_PRESENCE
  Pin-ctls: 0x20: IN
Node 0x13 [Audio Mixer] wcaps 0x200100: Mono
  Connection: 1
     0x07
Node 0x14 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out
  Amp-Out caps: ofs=0x03, nsteps=0x03, stepsize=0x17, mute=1
  Amp-Out vals:  [0x00]
Codec: Conexant ID 2bfa
Address: 1
Function Id: 0x2
Vendor Id: 0x14f12bfa
Subsystem Id: 0x14f100c3
Revision Id: 0x90000
Modem Function Group: 0x2
--endcollapse--


!!ALSA Device nodes
!!-----------------

crw-rw----+ 1 root audio 116,  0 May 26 15:08 /dev/snd/controlC0
crw-rw----+ 1 root audio 116,  4 May 26 15:08 /dev/snd/hwC0D0
crw-rw----+ 1 root audio 116,  5 May 26 15:08 /dev/snd/hwC0D1
crw-rw----+ 1 root audio 116, 24 May 26 15:08 /dev/snd/pcmC0D0c
crw-rw----+ 1 root audio 116, 16 May 26 22:03 /dev/snd/pcmC0D0p
crw-rw----+ 1 root audio 116, 17 May 26 15:08 /dev/snd/pcmC0D1p
crw-rw----+ 1 root audio 116,  1 May 26 15:08 /dev/snd/seq
crw-rw----+ 1 root audio 116, 33 May 26 15:08 /dev/snd/timer

/dev/snd/by-path:
total 0
drwxr-xr-x 2 root root  60 May 26 15:08 .
drwxr-xr-x 3 root root 220 May 26 15:08 ..
lrwxrwxrwx 1 root root  12 May 26 15:08 pci-0000:00:1b.0 -> ../controlC0


!!ALSA configuration files
!!------------------------

!!User specific config file (~/.asoundrc)

pcm.a52x {
    @args [ CARD ]
    @args.CARD {
        type integer
        default 0
    }
    type a52
    card $CARD
#    channels 2
    bitrate 640
    slavepcm "iec958"
}

pcm.!xsurround51 {
        type a52
        channels 6
        bitrate 640
        slavepcm "iec958"
}




!!Aplay/Arecord output
!!------------

APLAY

**** List of PLAYBACK Hardware Devices ****
card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog]
  Subdevices: 0/1
  Subdevice #0: subdevice #0
card 0: Intel [HDA Intel], device 1: STAC92xx Digital [STAC92xx Digital]
  Subdevices: 1/1
  Subdevice #0: subdevice #0

ARECORD

**** List of CAPTURE Hardware Devices ****
card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog]
  Subdevices: 1/1
  Subdevice #0: subdevice #0

!!Amixer output
!!-------------

!!-------Mixer controls for card 0 [Intel]

Card hw:0 'Intel'/'HDA Intel at 0xefebc000 irq 27'
  Mixer name	: 'SigmaTel STAC9200'
  Components	: 'HDA:83847690,102801bd,00102201 HDA:14f12bfa,14f100c3,00090000'
  Controls      : 11
  Simple ctrls  : 6
Simple mixer control 'Master',0
  Capabilities: pvolume pswitch penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 30 [97%] [-1.50dB] [on]
  Front Right: Playback 30 [97%] [-1.50dB] [on]
Simple mixer control 'PCM',0
  Capabilities: pvolume penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 255
  Mono:
  Front Left: Playback 253 [99%] [0.40dB]
  Front Right: Playback 253 [99%] [0.40dB]
Simple mixer control 'IEC958',0
  Capabilities: pswitch pswitch-joined penum
  Playback channels: Mono
  Mono: Playback [off]
Simple mixer control 'IEC958 Default PCM',0
  Capabilities: pswitch pswitch-joined penum
  Playback channels: Mono
  Mono: Playback [on]
Simple mixer control 'Capture',0
  Capabilities: cvolume cswitch penum
  Capture channels: Front Left - Front Right
  Limits: Capture 0 - 15
  Front Left: Capture 0 [0%] [0.00dB] [on]
  Front Right: Capture 0 [0%] [0.00dB] [on]
Simple mixer control 'Mux',0
  Capabilities: cvolume penum
  Capture channels: Front Left - Front Right
  Limits: Capture 0 - 4
  Front Left: Capture 3 [75%] [30.00dB]
  Front Right: Capture 3 [75%] [30.00dB]


!!Alsactl output
!!-------------

--startcollapse--
state.Intel {
	control.1 {
		comment.access 'read write'
		comment.type INTEGER
		comment.count 2
		comment.range '0 - 31'
		comment.dbmin -4650
		comment.dbmax 0
		iface MIXER
		name 'Master Playback Volume'
		value.0 30
		value.1 30
	}
	control.2 {
		comment.access 'read write'
		comment.type BOOLEAN
		comment.count 2
		iface MIXER
		name 'Master Playback Switch'
		value.0 true
		value.1 true
	}
	control.3 {
		comment.access 'read write'
		comment.type INTEGER
		comment.count 2
		comment.range '0 - 15'
		comment.dbmin 0
		comment.dbmax 2250
		iface MIXER
		name 'Capture Volume'
		value.0 0
		value.1 0
	}
	control.4 {
		comment.access 'read write'
		comment.type BOOLEAN
		comment.count 2
		iface MIXER
		name 'Capture Switch'
		value.0 true
		value.1 true
	}
	control.5 {
		comment.access 'read write'
		comment.type INTEGER
		comment.count 2
		comment.range '0 - 4'
		comment.dbmin 0
		comment.dbmax 4000
		iface MIXER
		name 'Mux Capture Volume'
		value.0 3
		value.1 3
	}
	control.6 {
		comment.access read
		comment.type IEC958
		comment.count 1
		iface MIXER
		name 'IEC958 Playback Con Mask'
		value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000'
	}
	control.7 {
		comment.access read
		comment.type IEC958
		comment.count 1
		iface MIXER
		name 'IEC958 Playback Pro Mask'
		value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000'
	}
	control.8 {
		comment.access 'read write'
		comment.type IEC958
		comment.count 1
		iface MIXER
		name 'IEC958 Playback Default'
		value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000'
	}
	control.9 {
		comment.access 'read write'
		comment.type BOOLEAN
		comment.count 1
		iface MIXER
		name 'IEC958 Playback Switch'
		value false
	}
	control.10 {
		comment.access 'read write'
		comment.type BOOLEAN
		comment.count 1
		iface MIXER
		name 'IEC958 Default PCM Playback Switch'
		value true
	}
	control.11 {
		comment.access 'read write user'
		comment.type INTEGER
		comment.count 2
		comment.range '0 - 255'
		comment.tlv '0000000100000008ffffec1400000014'
		comment.dbmin -5100
		comment.dbmax 0
		iface MIXER
		name 'PCM Playback Volume'
		value.0 253
		value.1 253
	}
}
--endcollapse--


!!All Loaded Modules
!!------------------

Module
sit
tunnel4
cryptd
aes_x86_64
aes_generic
af_packet
fuse
ipt_IFWLOG
ipt_psd
cls_flow
cls_fw
cls_u32
sch_htb
sch_hfsc
sch_ingress
sch_sfq
xt_time
xt_connlimit
xt_realm
iptable_raw
xt_comment
xt_recent
xt_policy
ipt_ULOG
ipt_REJECT
ipt_REDIRECT
ipt_NETMAP
ipt_MASQUERADE
ipt_ECN
ipt_ecn
ipt_CLUSTERIP
ipt_ah
ipt_addrtype
nf_nat_tftp
nf_nat_snmp_basic
nf_nat_sip
nf_nat_pptp
nf_nat_proto_gre
nf_nat_irc
nf_nat_h323
nf_nat_ftp
nf_nat_amanda
ts_kmp
nf_conntrack_amanda
nf_conntrack_sane
nf_conntrack_tftp
nf_conntrack_sip
nf_conntrack_proto_sctp
nf_conntrack_pptp
nf_conntrack_proto_gre
nf_conntrack_netlink
nf_conntrack_netbios_ns
nf_conntrack_irc
nf_conntrack_h323
nf_conntrack_ftp
ipt_set
ipt_SET
ip_set_nethash
ip_set_iptreemap
ip_set_iptree
ip_set_ipporthash
ip_set_portmap
ip_set_macipmap
ip_set_ipmap
ip_set_iphash
ip_set
xt_TPROXY
nf_tproxy_core
xt_tcpmss
xt_pkttype
xt_physdev
xt_owner
xt_NFQUEUE
xt_NFLOG
nfnetlink_log
xt_multiport
xt_MARK
xt_mark
xt_mac
xt_limit
xt_length
xt_iprange
xt_helper
xt_hashlimit
xt_DSCP
xt_dscp
xt_dccp
xt_conntrack
xt_CONNMARK
xt_connmark
xt_CLASSIFY
ipt_LOG
xt_tcpudp
xt_state
iptable_nat
nf_nat
nf_conntrack_ipv4
nf_defrag_ipv4
nf_conntrack
iptable_mangle
nfnetlink
iptable_filter
ip_tables
x_tables
nfsd
exportfs
nfs
lockd
fscache
nfs_acl
auth_rpcgss
sunrpc
ipv6
vboxnetadp
vboxnetflt
vboxdrv
snd_hda_codec_idt
rfcomm
snd_hda_intel
snd_hda_codec
snd_hwdep
snd_seq_dummy
arc4
snd_seq_oss
iTCO_wdt
iTCO_vendor_support
snd_seq_midi_event
snd_seq
b44
snd_seq_device
ssb
snd_pcm_oss
rng_core
ecb
sdhci_pci
pcmcia
snd_pcm
sco
sdhci
iwlagn
pcmcia_core
snd_timer
ohci1394
snd_mixer_oss
iwlcore
bridge
mmc_core
sr_mod
sg
mii
stp
led_class
snd
ieee1394
mac80211
soundcore
snd_page_alloc
bnep
dell_laptop
l2cap
i2c_i801
dcdbas
cfg80211
i915
drm_kms_helper
drm
i2c_algo_bit
i2c_core
btusb
bluetooth
rfkill
binfmt_misc
cpufreq_ondemand
cpufreq_conservative
loop
cpufreq_powersave
acpi_cpufreq
freq_table
nvram
dell_wmi
ac
video
battery
wmi
output
button
ehci_hcd
uhci_hcd
joydev
evdev
processor
thermal
usbcore
dm_snapshot
dm_zero
dm_mirror
dm_region_hash
dm_log
dm_mod
ata_generic
ide_pci_generic
ide_gd_mod
ide_core
pata_acpi
ahci
ata_piix
libata
sd_mod
scsi_mod
crc_t10dif
ext4
jbd2
crc16


!!Sysfs Files
!!-----------

/sys/class/sound/hwC0D0/init_pin_configs:
0x08 0x40f000f0
0x09 0x40f000f1
0x0d 0x0421101f
0x0e 0x90170110
0x0f 0x40f000f2
0x10 0x04a11020
0x11 0x40f000f3
0x12 0x40f000f4

/sys/class/sound/hwC0D0/driver_pin_configs:
0x08 0x40c003fa
0x09 0x01441340
0x0d 0x0421121f
0x0e 0x90170310
0x0f 0x408003fb
0x10 0x04a11020
0x11 0x401003fc
0x12 0x403003fd

/sys/class/sound/hwC0D0/user_pin_configs:

/sys/class/sound/hwC0D0/init_verbs:

/sys/class/sound/hwC0D1/init_pin_configs:
0x73 0x016a0000

/sys/class/sound/hwC0D1/driver_pin_configs:

/sys/class/sound/hwC0D1/user_pin_configs:

/sys/class/sound/hwC0D1/init_verbs:


!!ALSA/HDA dmesg
!!------------------

composite sync not supported
HDA Intel 0000:00:1b.0: PCI INT A -> GSI 21 (level, low) -> IRQ 21
HDA Intel 0000:00:1b.0: irq 27 for MSI/MSI-X
HDA Intel 0000:00:1b.0: setting latency timer to 64
Bluetooth: RFCOMM TTY layer initialized
--
   inputs: mic=0x10, fmic=0x0, line=0x0, fline=0x0, cd=0x0, aux=0x0
input: HDA Intel Mic at Ext Right Jack as /devices/pci0000:00/0000:00:1b.0/sound/card0/input8
input: HDA Intel HP Out at Ext Right Jack as /devices/pci0000:00/0000:00:1b.0/sound/card0/input9
ieee1394: Host added: ID:BUS[0-00:1023]  GUID[334fc0001e64c450]
--
uhci_hcd 0000:00:1d.0: PCI INT A disabled
HDA Intel 0000:00:1b.0: PCI INT A disabled
i915 0000:00:02.0: PCI INT A disabled
--
i915 0000:00:02.0: restoring config space at offset 0xf (was 0x100, writing 0x10b)
HDA Intel 0000:00:1b.0: restoring config space at offset 0xf (was 0x100, writing 0x10a)
HDA Intel 0000:00:1b.0: restoring config space at offset 0x4 (was 0xffa7c004, writing 0xefebc004)
HDA Intel 0000:00:1b.0: restoring config space at offset 0x3 (was 0x0, writing 0x10)
HDA Intel 0000:00:1b.0: restoring config space at offset 0x1 (was 0x100000, writing 0x100102)
pcieport 0000:00:1c.0: restoring config space at offset 0xf (was 0x100, writing 0x20100)
--
i915 0000:00:02.0: setting latency timer to 64
HDA Intel 0000:00:1b.0: PCI INT A -> GSI 21 (level, low) -> IRQ 21
HDA Intel 0000:00:1b.0: setting latency timer to 64
HDA Intel 0000:00:1b.0: irq 27 for MSI/MSI-X
uhci_hcd 0000:00:1d.0: PCI INT A -> GSI 20 (level, low) -> IRQ 20
--
uhci_hcd 0000:00:1d.0: PCI INT A disabled
HDA Intel 0000:00:1b.0: PCI INT A disabled
i915 0000:00:02.0: PCI INT A disabled
--
i915 0000:00:02.0: restoring config space at offset 0xf (was 0x100, writing 0x10b)
HDA Intel 0000:00:1b.0: restoring config space at offset 0xf (was 0x100, writing 0x10a)
HDA Intel 0000:00:1b.0: restoring config space at offset 0x4 (was 0xffa7c004, writing 0xefebc004)
HDA Intel 0000:00:1b.0: restoring config space at offset 0x3 (was 0x0, writing 0x10)
HDA Intel 0000:00:1b.0: restoring config space at offset 0x1 (was 0x100000, writing 0x100102)
pcieport 0000:00:1c.0: restoring config space at offset 0xf (was 0x100, writing 0x20100)
--
i915 0000:00:02.0: setting latency timer to 64
HDA Intel 0000:00:1b.0: PCI INT A -> GSI 21 (level, low) -> IRQ 21
HDA Intel 0000:00:1b.0: setting latency timer to 64
HDA Intel 0000:00:1b.0: irq 27 for MSI/MSI-X
uhci_hcd 0000:00:1d.0: PCI INT A -> GSI 20 (level, low) -> IRQ 20
--
uhci_hcd 0000:00:1d.0: PCI INT A disabled
HDA Intel 0000:00:1b.0: PCI INT A disabled
i915 0000:00:02.0: PCI INT A disabled
--
i915 0000:00:02.0: restoring config space at offset 0xf (was 0x100, writing 0x10b)
HDA Intel 0000:00:1b.0: restoring config space at offset 0xf (was 0x100, writing 0x10a)
HDA Intel 0000:00:1b.0: restoring config space at offset 0x4 (was 0xffa7c004, writing 0xefebc004)
HDA Intel 0000:00:1b.0: restoring config space at offset 0x3 (was 0x0, writing 0x10)
HDA Intel 0000:00:1b.0: restoring config space at offset 0x1 (was 0x100000, writing 0x100102)
pcieport 0000:00:1c.0: restoring config space at offset 0xf (was 0x100, writing 0x20100)
--
i915 0000:00:02.0: setting latency timer to 64
HDA Intel 0000:00:1b.0: PCI INT A -> GSI 21 (level, low) -> IRQ 21
HDA Intel 0000:00:1b.0: setting latency timer to 64
HDA Intel 0000:00:1b.0: irq 27 for MSI/MSI-X
uhci_hcd 0000:00:1d.0: PCI INT A -> GSI 20 (level, low) -> IRQ 20



[-- Attachment #3: Type: text/plain, Size: 160 bytes --]

_______________________________________________
Alsa-devel mailing list
Alsa-devel@alsa-project.org
http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

Re: wrong decibel data?

[Clemens Ladisch]

Colin Guthrie wrote:
> state.Intel {
> 	control.1 {
> 		comment.access 'read write'
> 		comment.type INTEGER
> 		comment.count 2
> 		comment.range '0 - 31'
> 		comment.dbmin -4650
> 		comment.dbmax 0
> 		iface MIXER
> 		name 'Master Playback Volume'
> 		value.0 30
> 		value.1 30
> 	}

This is the hardware volume control.

> 	control.11 {
> 		comment.access 'read write user'
> 		comment.type INTEGER
> 		comment.count 2
> 		comment.range '0 - 255'
> 		comment.tlv '0000000100000008ffffec1400000014'
> 		comment.dbmin -5100
> 		comment.dbmax 0
> 		iface MIXER
> 		name 'PCM Playback Volume'
> 		value.0 253
> 		value.1 253
> 	}

This is the emulated software volume control that is created by the
softvol plugin.  This control gets recreated by "alsactl restore" even
when the plugin is not running.

Might it be possible that PA is trying to use this, but that it doesn't
have any effect because PA is using PCM device hw:0?  (Try unloading
snd-hda-intel and then deleting that entry from /etc/asound.state.)


Regards,
Clemens

Re: wrong decibel data?

[Colin Guthrie]

'Twas brillig, and Clemens Ladisch at 27/05/10 14:48 did gyre and gimble:
> Colin Guthrie wrote:
>> state.Intel {
>> 	control.1 {
>> 		comment.access 'read write'
>> 		comment.type INTEGER
>> 		comment.count 2
>> 		comment.range '0 - 31'
>> 		comment.dbmin -4650
>> 		comment.dbmax 0
>> 		iface MIXER
>> 		name 'Master Playback Volume'
>> 		value.0 30
>> 		value.1 30
>> 	}
> 
> This is the hardware volume control.
> 
>> 	control.11 {
>> 		comment.access 'read write user'
>> 		comment.type INTEGER
>> 		comment.count 2
>> 		comment.range '0 - 255'
>> 		comment.tlv '0000000100000008ffffec1400000014'
>> 		comment.dbmin -5100
>> 		comment.dbmax 0
>> 		iface MIXER
>> 		name 'PCM Playback Volume'
>> 		value.0 253
>> 		value.1 253
>> 	}
> 
> This is the emulated software volume control that is created by the
> softvol plugin.  This control gets recreated by "alsactl restore" even
> when the plugin is not running.
> 
> Might it be possible that PA is trying to use this, but that it doesn't
> have any effect because PA is using PCM device hw:0?  (Try unloading
> snd-hda-intel and then deleting that entry from /etc/asound.state.)

PA should play nice with the softvol plugin so I don't think this is the
bit that is at fault.

I strongly suspect that the reason has already been correctly identified
a while ago, which is that this card considers -48dB silent where as PA
assumes this level is -200dB. I believe it was Raymond who pointed out
the -48dB level in the HDA spec before on this list.

I'm not sure of the internals, but things do indeed go silent when the
volume reaches the magic -48dB mark (which is around the 14% mark with
the current cubic mapping).

I suspect that if I were to define infinity to be 48.0 in PA, everything
would work nicely.

What I think is then ultimately needed is a way to ensure that everyone
sings from the same hymn sheet regarding the real world value of -inf dB.


I was hoping Lennart would have commented on this thread by now, so I'll
try and prod him to get some proper input as I'm very much flailing
around wildly in the dark!


Col

-- 

Colin Guthrie
gmane(at)colin.guthr.ie
http://colin.guthr.ie/

Day Job:
  Tribalogic Limited [http://www.tribalogic.net/]
Open Source:
  Mandriva Linux Contributor [http://www.mandriva.com/]
  PulseAudio Hacker [http://www.pulseaudio.org/]
  Trac Hacker [http://trac.edgewall.org/]

Re: wrong decibel data?

[Colin Guthrie]

'Twas brillig, and Colin Guthrie at 27/05/10 15:43 did gyre and gimble:
> PA should play nice with the softvol plugin so I don't think this is the
> bit that is at fault.
> 
> I strongly suspect that the reason has already been correctly identified
> a while ago, which is that this card considers -48dB silent where as PA
> assumes this level is -200dB. I believe it was Raymond who pointed out
> the -48dB level in the HDA spec before on this list.
> 
> I'm not sure of the internals, but things do indeed go silent when the
> volume reaches the magic -48dB mark (which is around the 14% mark with
> the current cubic mapping).
> 
> I suspect that if I were to define infinity to be 48.0 in PA, everything
> would work nicely.

I'm wrong (surprise, surprise!). Changing this value makes very little
difference.

FWIW, the -200dB thing is just a work around for systems that do not
define INFINITY. On my system this is defined. in bits/inf.h

So the value of 200dB mentioned by Raymond is not actually used
generally speaking.

Even if I undef and define INFINITY to be 48, the actual shut down at
-48 seems to happen regardless.

Looking at this in more depth, it seems that the problem is such that
the Master control of my card controls things down to -46.5dB. Once that
has hit 0, the PCM control takes over and goes down to -51dB

It seems that PA's full scale is determined as the multiplication of
thse two elements; therefore: -97.5dB.

However as things basically go silent at -48dB, the inclusion of Master
and PCM controls represent too large a range.

So either the 48dB of the HDA spec should be split between Master and
PCM more evenly, (e.g. 24dB each) or PCM should be removed and Master
should just have the full 48dB range. I don't really understand the
Master<->PCM relationship, but this is certainly and issue.


So my second guess (having now abandoned the idea of a problem in PA's
-inf definition) is that fixing the above, should fix things :)

Col

-- 

Colin Guthrie
gmane(at)colin.guthr.ie
http://colin.guthr.ie/

Day Job:
  Tribalogic Limited [http://www.tribalogic.net/]
Open Source:
  Mandriva Linux Contributor [http://www.mandriva.com/]
  PulseAudio Hacker [http://www.pulseaudio.org/]
  Trac Hacker [http://trac.edgewall.org/]

Re: wrong decibel data?

[Raymond Yau]

2010/5/28 Colin Guthrie <gmane@colin.guthr.ie>

> 'Twas brillig, and Colin Guthrie at 27/05/10 15:43 did gyre and gimble:
> > PA should play nice with the softvol plugin so I don't think this is the
> > bit that is at fault.
> >
> > I strongly suspect that the reason has already been correctly identified
> > a while ago, which is that this card considers -48dB silent where as PA
> > assumes this level is -200dB. I believe it was Raymond who pointed out
> > the -48dB level in the HDA spec before on this list.
> >
>

if  floating point 0.0 is -inf dB , and 1.0 is 0dB ,

0.5 is -6dB , 0.25 is -12 dB , 0.125 is -24dB and 0.0625 is -48dB

how can PA master volume control at 10~15% equivalent to HDA 's -48dB ?

Can you provide the pulseaudio log when you change the volume from 100% to
0% ?

Re: wrong decibel data?

[Colin Guthrie]

[-- Attachment #1: Type: text/plain, Size: 5135 bytes --]

'Twas brillig, and Raymond Yau at 06/06/10 01:12 did gyre and gimble:
> 2010/5/28 Colin Guthrie <gmane@colin.guthr.ie>
> 
>> 'Twas brillig, and Colin Guthrie at 27/05/10 15:43 did gyre and gimble:
>>> PA should play nice with the softvol plugin so I don't think this is the
>>> bit that is at fault.
>>>
>>> I strongly suspect that the reason has already been correctly identified
>>> a while ago, which is that this card considers -48dB silent where as PA
>>> assumes this level is -200dB. I believe it was Raymond who pointed out
>>> the -48dB level in the HDA spec before on this list.
>>>
>>
> 
> if  floating point 0.0 is -inf dB , and 1.0 is 0dB ,
> 
> 0.5 is -6dB , 0.25 is -12 dB , 0.125 is -24dB and 0.0625 is -48dB

This is just a pure mapping from dB->linear, but as this linear mapping
is generally not "natural" there are several different approaches to
presenting this to users. In PA, a cubic mapping is used on top of this
basic conversion, to map to the percentage scale (0.0 to 1.0 if you like).

So I'm not sure what point you're making by providing these numbers. Can
you explain?

> how can PA master volume control at 10~15% equivalent to HDA 's -48dB ?

Not sure what you mean here, but I suspect it's the cubic mapping that
is confusing you.

Here is the function in PA's pulse/volume.c:

 pa_volume_t pa_sw_volume_from_linear(double v) {

    if (v <= 0.0)
        return PA_VOLUME_MUTED;

    /*
     * We use a cubic mapping here, as suggested and discussed here:
     *
     * http://www.robotplanet.dk/audio/audio_gui_design/
     *
http://lists.linuxaudio.org/pipermail/linux-audio-dev/2009-May/thread.html#23151
     *
     * We make sure that the conversion to linear and back yields the
     * same volume value! That's why we need the lround() below!
     */

    return (pa_volume_t) lround(cbrt(v) * PA_VOLUME_NORM);
 }


> Can you provide the pulseaudio log when you change the volume from 100% to
> 0% ?

I can provide the actual log output if you like but here is the output
from the following command:

(for i in 100 95 90 85 80 75 70 65 60 55 50 45 40 35 30 25 20 19 18 17
16 15 14 13 12 11 10 9 8 7 6 5 4 3 2 1 0; do echo
"============================== $i% =============="; amixer set Master
$i%>/dev/null; amixer -c0 get Master; amixer -c0 get PCM; echo
"PulseAudio:"; pacmd list-sinks| grep "index: 11" -A 13; done) 2>&1
>volume-test.txt


This basically sets the volume (via alsa->pulse plugin, but that's just
for convenience!) all the way down to 0. I get more fine grained below
20 as that is the interesting zone.


The output shows the scaling from 100% where both Master and PCM are
0dB, up to the point at around 16% where the Master channel is maxed out
at -46.5dB (0%) and PA starts instead manipulating the PCM control to
get more of it's range (up until this point PCM was only used to gain
more accuracy - i.e. when Master alone did not provide as fine grained a
setting as was needed; in this scenario, PA will use the PCM control to
get more accuracy and, if needed, it will also use software scaling on
top of that: for more info about this works see:
http://pulseaudio.org/wiki/PulseAudioStoleMyVolumes).

As you can see from the output, by the time we reach 2%, both Master and
PCM are fully maxed out at -46.5dB and -51dB respectively. At this point
PA wants a volume of  -101.93dB, so that means that -101.93 -
(-51+-46.5=-96.5) = -5.43 dB is performed in software.

By 1%, the software component of the reduction increases to -23.47dB to
give a total of -119.97dB. By 0% we reach -inf dB


Obviously the fact that the chip basically cuts off any audio when the
Master slider hits 0 (or perhaps when the combined volume reaches -48dB
- it's hard to tell) doesn't really play nicely with the real value of
-inf which we attempt to reach.



I'm not sure where this problem needs to be fixed. Oviously having a
Master and PCM slider whose range is far greater than the value of -48dB
is pointless. This configuration means that there are numerous "zero
point" configurations of the two sliders beyond which any further change
in value is useless. So disregarding PA completely, this setup is not ideal.

When PA is used, the value for -inf is actually configured by the system
and we attempt to scale to -inf (albeit via a cubic mapping from
percentage). If the volume literally cuts out at -48dB when dealing with
the h/w mixers, then there is a problem, but by the same token if the
-48dB level really is more like the silence we want to represent, then
perhaps trying to scale to -inf is pointless in itself and really the
range of the scale used in PA should be adjusted.

I don't know enough about this side of things to comment more accurately
than this, so when LinuxTag is over, hopefully Lennart can comment a bit
on this thread to add his opinions to the mix.

Col


Col





-- 

Colin Guthrie
gmane(at)colin.guthr.ie
http://colin.guthr.ie/

Day Job:
  Tribalogic Limited [http://www.tribalogic.net/]
Open Source:
  Mandriva Linux Contributor [http://www.mandriva.com/]
  PulseAudio Hacker [http://www.pulseaudio.org/]
  Trac Hacker [http://trac.edgewall.org/]

[-- Attachment #2: volume-test.txt --]
[-- Type: text/plain, Size: 34248 bytes --]

============================== 100% ==============
Simple mixer control 'Master',0
  Capabilities: pvolume pswitch penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 31 [100%] [0.00dB] [on]
  Front Right: Playback 31 [100%] [0.00dB] [on]
Simple mixer control 'PCM',0
  Capabilities: pvolume penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 255
  Mono:
  Front Left: Playback 255 [100%] [0.00dB]
  Front Right: Playback 255 [100%] [0.00dB]
PulseAudio:
  * index: 11
	name: <alsa_output.pci-0000_00_1b.0.analog-stereo>
	driver: <module-alsa-card.c>
	flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY FLAT_VOLUME DYNAMIC_LATENCY
	state: IDLE
	suspend cause: 
	priority: 9959
	volume: 0: 100% 1: 100%
	        0: 0.00 dB 1: 0.00 dB
	        balance 0.00
	base volume: 100%
	             0.00 dB
	volume steps: 65537
	muted: no
============================== 95% ==============
Simple mixer control 'Master',0
  Capabilities: pvolume pswitch penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 31 [100%] [0.00dB] [on]
  Front Right: Playback 31 [100%] [0.00dB] [on]
Simple mixer control 'PCM',0
  Capabilities: pvolume penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 255
  Mono:
  Front Left: Playback 249 [98%] [-1.20dB]
  Front Right: Playback 249 [98%] [-1.20dB]
PulseAudio:
  * index: 11
	name: <alsa_output.pci-0000_00_1b.0.analog-stereo>
	driver: <module-alsa-card.c>
	flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY FLAT_VOLUME DYNAMIC_LATENCY
	state: IDLE
	suspend cause: 
	priority: 9959
	volume: 0:  95% 1:  95%
	        0: -1.34 dB 1: -1.34 dB
	        balance 0.00
	base volume: 100%
	             0.00 dB
	volume steps: 65537
	muted: no
============================== 90% ==============
Simple mixer control 'Master',0
  Capabilities: pvolume pswitch penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 30 [97%] [-1.50dB] [on]
  Front Right: Playback 30 [97%] [-1.50dB] [on]
Simple mixer control 'PCM',0
  Capabilities: pvolume penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 255
  Mono:
  Front Left: Playback 249 [98%] [-1.20dB]
  Front Right: Playback 249 [98%] [-1.20dB]
PulseAudio:
  * index: 11
	name: <alsa_output.pci-0000_00_1b.0.analog-stereo>
	driver: <module-alsa-card.c>
	flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY FLAT_VOLUME DYNAMIC_LATENCY
	state: IDLE
	suspend cause: 
	priority: 9959
	volume: 0:  90% 1:  90%
	        0: -2.75 dB 1: -2.75 dB
	        balance 0.00
	base volume: 100%
	             0.00 dB
	volume steps: 65537
	muted: no
============================== 85% ==============
Simple mixer control 'Master',0
  Capabilities: pvolume pswitch penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 29 [94%] [-3.00dB] [on]
  Front Right: Playback 29 [94%] [-3.00dB] [on]
Simple mixer control 'PCM',0
  Capabilities: pvolume penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 255
  Mono:
  Front Left: Playback 249 [98%] [-1.20dB]
  Front Right: Playback 249 [98%] [-1.20dB]
PulseAudio:
  * index: 11
	name: <alsa_output.pci-0000_00_1b.0.analog-stereo>
	driver: <module-alsa-card.c>
	flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY FLAT_VOLUME DYNAMIC_LATENCY
	state: IDLE
	suspend cause: 
	priority: 9959
	volume: 0:  85% 1:  85%
	        0: -4.23 dB 1: -4.23 dB
	        balance 0.00
	base volume: 100%
	             0.00 dB
	volume steps: 65537
	muted: no
============================== 80% ==============
Simple mixer control 'Master',0
  Capabilities: pvolume pswitch penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 28 [90%] [-4.50dB] [on]
  Front Right: Playback 28 [90%] [-4.50dB] [on]
Simple mixer control 'PCM',0
  Capabilities: pvolume penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 255
  Mono:
  Front Left: Playback 249 [98%] [-1.20dB]
  Front Right: Playback 249 [98%] [-1.20dB]
PulseAudio:
  * index: 11
	name: <alsa_output.pci-0000_00_1b.0.analog-stereo>
	driver: <module-alsa-card.c>
	flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY FLAT_VOLUME DYNAMIC_LATENCY
	state: IDLE
	suspend cause: 
	priority: 9959
	volume: 0:  80% 1:  80%
	        0: -5.81 dB 1: -5.81 dB
	        balance 0.00
	base volume: 100%
	             0.00 dB
	volume steps: 65537
	muted: no
============================== 75% ==============
Simple mixer control 'Master',0
  Capabilities: pvolume pswitch penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 27 [87%] [-6.00dB] [on]
  Front Right: Playback 27 [87%] [-6.00dB] [on]
Simple mixer control 'PCM',0
  Capabilities: pvolume penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 255
  Mono:
  Front Left: Playback 248 [97%] [-1.40dB]
  Front Right: Playback 248 [97%] [-1.40dB]
PulseAudio:
  * index: 11
	name: <alsa_output.pci-0000_00_1b.0.analog-stereo>
	driver: <module-alsa-card.c>
	flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY FLAT_VOLUME DYNAMIC_LATENCY
	state: IDLE
	suspend cause: 
	priority: 9959
	volume: 0:  75% 1:  75%
	        0: -7.50 dB 1: -7.50 dB
	        balance 0.00
	base volume: 100%
	             0.00 dB
	volume steps: 65537
	muted: no
============================== 70% ==============
Simple mixer control 'Master',0
  Capabilities: pvolume pswitch penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 25 [81%] [-9.00dB] [on]
  Front Right: Playback 25 [81%] [-9.00dB] [on]
Simple mixer control 'PCM',0
  Capabilities: pvolume penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 255
  Mono:
  Front Left: Playback 254 [100%] [0.20dB]
  Front Right: Playback 254 [100%] [0.20dB]
PulseAudio:
  * index: 11
	name: <alsa_output.pci-0000_00_1b.0.analog-stereo>
	driver: <module-alsa-card.c>
	flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY FLAT_VOLUME DYNAMIC_LATENCY
	state: IDLE
	suspend cause: 
	priority: 9959
	volume: 0:  70% 1:  70%
	        0: -9.29 dB 1: -9.29 dB
	        balance 0.00
	base volume: 100%
	             0.00 dB
	volume steps: 65537
	muted: no
============================== 65% ==============
Simple mixer control 'Master',0
  Capabilities: pvolume pswitch penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 24 [77%] [-10.50dB] [on]
  Front Right: Playback 24 [77%] [-10.50dB] [on]
Simple mixer control 'PCM',0
  Capabilities: pvolume penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 255
  Mono:
  Front Left: Playback 252 [99%] [0.60dB]
  Front Right: Playback 252 [99%] [0.60dB]
PulseAudio:
  * index: 11
	name: <alsa_output.pci-0000_00_1b.0.analog-stereo>
	driver: <module-alsa-card.c>
	flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY FLAT_VOLUME DYNAMIC_LATENCY
	state: IDLE
	suspend cause: 
	priority: 9959
	volume: 0:  65% 1:  65%
	        0: -11.22 dB 1: -11.22 dB
	        balance 0.00
	base volume: 100%
	             0.00 dB
	volume steps: 65537
	muted: no
============================== 60% ==============
Simple mixer control 'Master',0
  Capabilities: pvolume pswitch penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 23 [74%] [-12.00dB] [on]
  Front Right: Playback 23 [74%] [-12.00dB] [on]
Simple mixer control 'PCM',0
  Capabilities: pvolume penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 255
  Mono:
  Front Left: Playback 249 [98%] [-1.20dB]
  Front Right: Playback 249 [98%] [-1.20dB]
PulseAudio:
  * index: 11
	name: <alsa_output.pci-0000_00_1b.0.analog-stereo>
	driver: <module-alsa-card.c>
	flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY FLAT_VOLUME DYNAMIC_LATENCY
	state: IDLE
	suspend cause: 
	priority: 9959
	volume: 0:  60% 1:  60%
	        0: -13.31 dB 1: -13.31 dB
	        balance 0.00
	base volume: 100%
	             0.00 dB
	volume steps: 65537
	muted: no
============================== 55% ==============
Simple mixer control 'Master',0
  Capabilities: pvolume pswitch penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 21 [68%] [-15.00dB] [on]
  Front Right: Playback 21 [68%] [-15.00dB] [on]
Simple mixer control 'PCM',0
  Capabilities: pvolume penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 255
  Mono:
  Front Left: Playback 253 [99%] [0.40dB]
  Front Right: Playback 253 [99%] [0.40dB]
PulseAudio:
  * index: 11
	name: <alsa_output.pci-0000_00_1b.0.analog-stereo>
	driver: <module-alsa-card.c>
	flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY FLAT_VOLUME DYNAMIC_LATENCY
	state: IDLE
	suspend cause: 
	priority: 9959
	volume: 0:  55% 1:  55%
	        0: -15.58 dB 1: -15.58 dB
	        balance 0.00
	base volume: 100%
	             0.00 dB
	volume steps: 65537
	muted: no
============================== 50% ==============
Simple mixer control 'Master',0
  Capabilities: pvolume pswitch penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 19 [61%] [-18.00dB] [on]
  Front Right: Playback 19 [61%] [-18.00dB] [on]
Simple mixer control 'PCM',0
  Capabilities: pvolume penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 255
  Mono:
  Front Left: Playback 255 [100%] [0.00dB]
  Front Right: Playback 255 [100%] [0.00dB]
PulseAudio:
  * index: 11
	name: <alsa_output.pci-0000_00_1b.0.analog-stereo>
	driver: <module-alsa-card.c>
	flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY FLAT_VOLUME DYNAMIC_LATENCY
	state: IDLE
	suspend cause: 
	priority: 9959
	volume: 0:  50% 1:  50%
	        0: -18.06 dB 1: -18.06 dB
	        balance 0.00
	base volume: 100%
	             0.00 dB
	volume steps: 65537
	muted: no
============================== 45% ==============
Simple mixer control 'Master',0
  Capabilities: pvolume pswitch penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 18 [58%] [-19.50dB] [on]
  Front Right: Playback 18 [58%] [-19.50dB] [on]
Simple mixer control 'PCM',0
  Capabilities: pvolume penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 255
  Mono:
  Front Left: Playback 249 [98%] [-1.20dB]
  Front Right: Playback 249 [98%] [-1.20dB]
PulseAudio:
  * index: 11
	name: <alsa_output.pci-0000_00_1b.0.analog-stereo>
	driver: <module-alsa-card.c>
	flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY FLAT_VOLUME DYNAMIC_LATENCY
	state: IDLE
	suspend cause: 
	priority: 9959
	volume: 0:  45% 1:  45%
	        0: -20.81 dB 1: -20.81 dB
	        balance 0.00
	base volume: 100%
	             0.00 dB
	volume steps: 65537
	muted: no
============================== 40% ==============
Simple mixer control 'Master',0
  Capabilities: pvolume pswitch penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 16 [52%] [-22.50dB] [on]
  Front Right: Playback 16 [52%] [-22.50dB] [on]
Simple mixer control 'PCM',0
  Capabilities: pvolume penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 255
  Mono:
  Front Left: Playback 249 [98%] [-1.20dB]
  Front Right: Playback 249 [98%] [-1.20dB]
PulseAudio:
  * index: 11
	name: <alsa_output.pci-0000_00_1b.0.analog-stereo>
	driver: <module-alsa-card.c>
	flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY FLAT_VOLUME DYNAMIC_LATENCY
	state: IDLE
	suspend cause: 
	priority: 9959
	volume: 0:  40% 1:  40%
	        0: -23.88 dB 1: -23.88 dB
	        balance 0.00
	base volume: 100%
	             0.00 dB
	volume steps: 65537
	muted: no
============================== 35% ==============
Simple mixer control 'Master',0
  Capabilities: pvolume pswitch penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 13 [42%] [-27.00dB] [on]
  Front Right: Playback 13 [42%] [-27.00dB] [on]
Simple mixer control 'PCM',0
  Capabilities: pvolume penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 255
  Mono:
  Front Left: Playback 254 [100%] [0.20dB]
  Front Right: Playback 254 [100%] [0.20dB]
PulseAudio:
  * index: 11
	name: <alsa_output.pci-0000_00_1b.0.analog-stereo>
	driver: <module-alsa-card.c>
	flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY FLAT_VOLUME DYNAMIC_LATENCY
	state: IDLE
	suspend cause: 
	priority: 9959
	volume: 0:  35% 1:  35%
	        0: -27.36 dB 1: -27.36 dB
	        balance 0.00
	base volume: 100%
	             0.00 dB
	volume steps: 65537
	muted: no
============================== 30% ==============
Simple mixer control 'Master',0
  Capabilities: pvolume pswitch penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 11 [35%] [-30.00dB] [on]
  Front Right: Playback 11 [35%] [-30.00dB] [on]
Simple mixer control 'PCM',0
  Capabilities: pvolume penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 255
  Mono:
  Front Left: Playback 249 [98%] [-1.20dB]
  Front Right: Playback 249 [98%] [-1.20dB]
PulseAudio:
  * index: 11
	name: <alsa_output.pci-0000_00_1b.0.analog-stereo>
	driver: <module-alsa-card.c>
	flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY FLAT_VOLUME DYNAMIC_LATENCY
	state: IDLE
	suspend cause: 
	priority: 9959
	volume: 0:  30% 1:  30%
	        0: -31.37 dB 1: -31.37 dB
	        balance 0.00
	base volume: 100%
	             0.00 dB
	volume steps: 65537
	muted: no
============================== 25% ==============
Simple mixer control 'Master',0
  Capabilities: pvolume pswitch penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 7 [23%] [-36.00dB] [on]
  Front Right: Playback 7 [23%] [-36.00dB] [on]
Simple mixer control 'PCM',0
  Capabilities: pvolume penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 255
  Mono:
  Front Left: Playback 255 [100%] [0.00dB]
  Front Right: Playback 255 [100%] [0.00dB]
PulseAudio:
  * index: 11
	name: <alsa_output.pci-0000_00_1b.0.analog-stereo>
	driver: <module-alsa-card.c>
	flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY FLAT_VOLUME DYNAMIC_LATENCY
	state: IDLE
	suspend cause: 
	priority: 9959
	volume: 0:  25% 1:  25%
	        0: -36.12 dB 1: -36.12 dB
	        balance 0.00
	base volume: 100%
	             0.00 dB
	volume steps: 65537
	muted: no
============================== 20% ==============
Simple mixer control 'Master',0
  Capabilities: pvolume pswitch penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 4 [13%] [-40.50dB] [on]
  Front Right: Playback 4 [13%] [-40.50dB] [on]
Simple mixer control 'PCM',0
  Capabilities: pvolume penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 255
  Mono:
  Front Left: Playback 248 [97%] [-1.40dB]
  Front Right: Playback 248 [97%] [-1.40dB]
PulseAudio:
  * index: 11
	name: <alsa_output.pci-0000_00_1b.0.analog-stereo>
	driver: <module-alsa-card.c>
	flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY FLAT_VOLUME DYNAMIC_LATENCY
	state: IDLE
	suspend cause: 
	priority: 9959
	volume: 0:  20% 1:  20%
	        0: -41.94 dB 1: -41.94 dB
	        balance 0.00
	base volume: 100%
	             0.00 dB
	volume steps: 65537
	muted: no
============================== 19% ==============
Simple mixer control 'Master',0
  Capabilities: pvolume pswitch penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 3 [10%] [-42.00dB] [on]
  Front Right: Playback 3 [10%] [-42.00dB] [on]
Simple mixer control 'PCM',0
  Capabilities: pvolume penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 255
  Mono:
  Front Left: Playback 249 [98%] [-1.20dB]
  Front Right: Playback 249 [98%] [-1.20dB]
PulseAudio:
  * index: 11
	name: <alsa_output.pci-0000_00_1b.0.analog-stereo>
	driver: <module-alsa-card.c>
	flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY FLAT_VOLUME DYNAMIC_LATENCY
	state: IDLE
	suspend cause: 
	priority: 9959
	volume: 0:  19% 1:  19%
	        0: -43.27 dB 1: -43.27 dB
	        balance 0.00
	base volume: 100%
	             0.00 dB
	volume steps: 65537
	muted: no
============================== 18% ==============
Simple mixer control 'Master',0
  Capabilities: pvolume pswitch penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 2 [6%] [-43.50dB] [on]
  Front Right: Playback 2 [6%] [-43.50dB] [on]
Simple mixer control 'PCM',0
  Capabilities: pvolume penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 255
  Mono:
  Front Left: Playback 250 [98%] [-1.00dB]
  Front Right: Playback 250 [98%] [-1.00dB]
PulseAudio:
  * index: 11
	name: <alsa_output.pci-0000_00_1b.0.analog-stereo>
	driver: <module-alsa-card.c>
	flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY FLAT_VOLUME DYNAMIC_LATENCY
	state: IDLE
	suspend cause: 
	priority: 9959
	volume: 0:  18% 1:  18%
	        0: -44.68 dB 1: -44.68 dB
	        balance 0.00
	base volume: 100%
	             0.00 dB
	volume steps: 65537
	muted: no
============================== 17% ==============
Simple mixer control 'Master',0
  Capabilities: pvolume pswitch penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 1 [3%] [-45.00dB] [on]
  Front Right: Playback 1 [3%] [-45.00dB] [on]
Simple mixer control 'PCM',0
  Capabilities: pvolume penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 255
  Mono:
  Front Left: Playback 250 [98%] [-1.00dB]
  Front Right: Playback 250 [98%] [-1.00dB]
PulseAudio:
  * index: 11
	name: <alsa_output.pci-0000_00_1b.0.analog-stereo>
	driver: <module-alsa-card.c>
	flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY FLAT_VOLUME DYNAMIC_LATENCY
	state: IDLE
	suspend cause: 
	priority: 9959
	volume: 0:  17% 1:  17%
	        0: -46.17 dB 1: -46.17 dB
	        balance 0.00
	base volume: 100%
	             0.00 dB
	volume steps: 65537
	muted: no
============================== 16% ==============
Simple mixer control 'Master',0
  Capabilities: pvolume pswitch penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 0 [0%] [-46.50dB] [on]
  Front Right: Playback 0 [0%] [-46.50dB] [on]
Simple mixer control 'PCM',0
  Capabilities: pvolume penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 255
  Mono:
  Front Left: Playback 249 [98%] [-1.20dB]
  Front Right: Playback 249 [98%] [-1.20dB]
PulseAudio:
  * index: 11
	name: <alsa_output.pci-0000_00_1b.0.analog-stereo>
	driver: <module-alsa-card.c>
	flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY FLAT_VOLUME DYNAMIC_LATENCY
	state: IDLE
	suspend cause: 
	priority: 9959
	volume: 0:  16% 1:  16%
	        0: -47.75 dB 1: -47.75 dB
	        balance 0.00
	base volume: 100%
	             0.00 dB
	volume steps: 65537
	muted: no
============================== 15% ==============
Simple mixer control 'Master',0
  Capabilities: pvolume pswitch penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 0 [0%] [-46.50dB] [on]
  Front Right: Playback 0 [0%] [-46.50dB] [on]
Simple mixer control 'PCM',0
  Capabilities: pvolume penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 255
  Mono:
  Front Left: Playback 241 [95%] [-2.80dB]
  Front Right: Playback 241 [95%] [-2.80dB]
PulseAudio:
  * index: 11
	name: <alsa_output.pci-0000_00_1b.0.analog-stereo>
	driver: <module-alsa-card.c>
	flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY FLAT_VOLUME DYNAMIC_LATENCY
	state: IDLE
	suspend cause: 
	priority: 9959
	volume: 0:  15% 1:  15%
	        0: -49.43 dB 1: -49.43 dB
	        balance 0.00
	base volume: 100%
	             0.00 dB
	volume steps: 65537
	muted: no
============================== 14% ==============
Simple mixer control 'Master',0
  Capabilities: pvolume pswitch penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 0 [0%] [-46.50dB] [on]
  Front Right: Playback 0 [0%] [-46.50dB] [on]
Simple mixer control 'PCM',0
  Capabilities: pvolume penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 255
  Mono:
  Front Left: Playback 232 [91%] [-4.60dB]
  Front Right: Playback 232 [91%] [-4.60dB]
PulseAudio:
  * index: 11
	name: <alsa_output.pci-0000_00_1b.0.analog-stereo>
	driver: <module-alsa-card.c>
	flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY FLAT_VOLUME DYNAMIC_LATENCY
	state: IDLE
	suspend cause: 
	priority: 9959
	volume: 0:  14% 1:  14%
	        0: -51.23 dB 1: -51.23 dB
	        balance 0.00
	base volume: 100%
	             0.00 dB
	volume steps: 65537
	muted: no
============================== 13% ==============
Simple mixer control 'Master',0
  Capabilities: pvolume pswitch penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 0 [0%] [-46.50dB] [on]
  Front Right: Playback 0 [0%] [-46.50dB] [on]
Simple mixer control 'PCM',0
  Capabilities: pvolume penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 255
  Mono:
  Front Left: Playback 222 [87%] [-6.60dB]
  Front Right: Playback 222 [87%] [-6.60dB]
PulseAudio:
  * index: 11
	name: <alsa_output.pci-0000_00_1b.0.analog-stereo>
	driver: <module-alsa-card.c>
	flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY FLAT_VOLUME DYNAMIC_LATENCY
	state: IDLE
	suspend cause: 
	priority: 9959
	volume: 0:  13% 1:  13%
	        0: -53.16 dB 1: -53.16 dB
	        balance 0.00
	base volume: 100%
	             0.00 dB
	volume steps: 65537
	muted: no
============================== 12% ==============
Simple mixer control 'Master',0
  Capabilities: pvolume pswitch penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 0 [0%] [-46.50dB] [on]
  Front Right: Playback 0 [0%] [-46.50dB] [on]
Simple mixer control 'PCM',0
  Capabilities: pvolume penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 255
  Mono:
  Front Left: Playback 212 [83%] [-8.60dB]
  Front Right: Playback 212 [83%] [-8.60dB]
PulseAudio:
  * index: 11
	name: <alsa_output.pci-0000_00_1b.0.analog-stereo>
	driver: <module-alsa-card.c>
	flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY FLAT_VOLUME DYNAMIC_LATENCY
	state: IDLE
	suspend cause: 
	priority: 9959
	volume: 0:  12% 1:  12%
	        0: -55.25 dB 1: -55.25 dB
	        balance 0.00
	base volume: 100%
	             0.00 dB
	volume steps: 65537
	muted: no
============================== 11% ==============
Simple mixer control 'Master',0
  Capabilities: pvolume pswitch penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 0 [0%] [-46.50dB] [on]
  Front Right: Playback 0 [0%] [-46.50dB] [on]
Simple mixer control 'PCM',0
  Capabilities: pvolume penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 255
  Mono:
  Front Left: Playback 200 [78%] [-11.00dB]
  Front Right: Playback 200 [78%] [-11.00dB]
PulseAudio:
  * index: 11
	name: <alsa_output.pci-0000_00_1b.0.analog-stereo>
	driver: <module-alsa-card.c>
	flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY FLAT_VOLUME DYNAMIC_LATENCY
	state: IDLE
	suspend cause: 
	priority: 9959
	volume: 0:  11% 1:  11%
	        0: -57.52 dB 1: -57.52 dB
	        balance 0.00
	base volume: 100%
	             0.00 dB
	volume steps: 65537
	muted: no
============================== 10% ==============
Simple mixer control 'Master',0
  Capabilities: pvolume pswitch penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 0 [0%] [-46.50dB] [on]
  Front Right: Playback 0 [0%] [-46.50dB] [on]
Simple mixer control 'PCM',0
  Capabilities: pvolume penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 255
  Mono:
  Front Left: Playback 188 [74%] [-13.40dB]
  Front Right: Playback 188 [74%] [-13.40dB]
PulseAudio:
  * index: 11
	name: <alsa_output.pci-0000_00_1b.0.analog-stereo>
	driver: <module-alsa-card.c>
	flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY FLAT_VOLUME DYNAMIC_LATENCY
	state: IDLE
	suspend cause: 
	priority: 9959
	volume: 0:  10% 1:  10%
	        0: -60.00 dB 1: -60.00 dB
	        balance 0.00
	base volume: 100%
	             0.00 dB
	volume steps: 65537
	muted: no
============================== 9% ==============
Simple mixer control 'Master',0
  Capabilities: pvolume pswitch penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 0 [0%] [-46.50dB] [on]
  Front Right: Playback 0 [0%] [-46.50dB] [on]
Simple mixer control 'PCM',0
  Capabilities: pvolume penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 255
  Mono:
  Front Left: Playback 174 [68%] [-16.20dB]
  Front Right: Playback 174 [68%] [-16.20dB]
PulseAudio:
  * index: 11
	name: <alsa_output.pci-0000_00_1b.0.analog-stereo>
	driver: <module-alsa-card.c>
	flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY FLAT_VOLUME DYNAMIC_LATENCY
	state: IDLE
	suspend cause: 
	priority: 9959
	volume: 0:   9% 1:   9%
	        0: -62.74 dB 1: -62.74 dB
	        balance 0.00
	base volume: 100%
	             0.00 dB
	volume steps: 65537
	muted: no
============================== 8% ==============
Simple mixer control 'Master',0
  Capabilities: pvolume pswitch penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 0 [0%] [-46.50dB] [on]
  Front Right: Playback 0 [0%] [-46.50dB] [on]
Simple mixer control 'PCM',0
  Capabilities: pvolume penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 255
  Mono:
  Front Left: Playback 159 [62%] [-19.20dB]
  Front Right: Playback 159 [62%] [-19.20dB]
PulseAudio:
  * index: 11
	name: <alsa_output.pci-0000_00_1b.0.analog-stereo>
	driver: <module-alsa-card.c>
	flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY FLAT_VOLUME DYNAMIC_LATENCY
	state: IDLE
	suspend cause: 
	priority: 9959
	volume: 0:   8% 1:   8%
	        0: -65.81 dB 1: -65.81 dB
	        balance 0.00
	base volume: 100%
	             0.00 dB
	volume steps: 65537
	muted: no
============================== 7% ==============
Simple mixer control 'Master',0
  Capabilities: pvolume pswitch penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 0 [0%] [-46.50dB] [on]
  Front Right: Playback 0 [0%] [-46.50dB] [on]
Simple mixer control 'PCM',0
  Capabilities: pvolume penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 255
  Mono:
  Front Left: Playback 142 [56%] [-22.60dB]
  Front Right: Playback 142 [56%] [-22.60dB]
PulseAudio:
  * index: 11
	name: <alsa_output.pci-0000_00_1b.0.analog-stereo>
	driver: <module-alsa-card.c>
	flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY FLAT_VOLUME DYNAMIC_LATENCY
	state: IDLE
	suspend cause: 
	priority: 9959
	volume: 0:   7% 1:   7%
	        0: -69.29 dB 1: -69.29 dB
	        balance 0.00
	base volume: 100%
	             0.00 dB
	volume steps: 65537
	muted: no
============================== 6% ==============
Simple mixer control 'Master',0
  Capabilities: pvolume pswitch penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 0 [0%] [-46.50dB] [on]
  Front Right: Playback 0 [0%] [-46.50dB] [on]
Simple mixer control 'PCM',0
  Capabilities: pvolume penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 255
  Mono:
  Front Left: Playback 121 [47%] [-26.80dB]
  Front Right: Playback 121 [47%] [-26.80dB]
PulseAudio:
  * index: 11
	name: <alsa_output.pci-0000_00_1b.0.analog-stereo>
	driver: <module-alsa-card.c>
	flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY FLAT_VOLUME DYNAMIC_LATENCY
	state: IDLE
	suspend cause: 
	priority: 9959
	volume: 0:   6% 1:   6%
	        0: -73.31 dB 1: -73.31 dB
	        balance 0.00
	base volume: 100%
	             0.00 dB
	volume steps: 65537
	muted: no
============================== 5% ==============
Simple mixer control 'Master',0
  Capabilities: pvolume pswitch penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 0 [0%] [-46.50dB] [on]
  Front Right: Playback 0 [0%] [-46.50dB] [on]
Simple mixer control 'PCM',0
  Capabilities: pvolume penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 255
  Mono:
  Front Left: Playback 98 [38%] [-31.40dB]
  Front Right: Playback 98 [38%] [-31.40dB]
PulseAudio:
  * index: 11
	name: <alsa_output.pci-0000_00_1b.0.analog-stereo>
	driver: <module-alsa-card.c>
	flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY FLAT_VOLUME DYNAMIC_LATENCY
	state: IDLE
	suspend cause: 
	priority: 9959
	volume: 0:   5% 1:   5%
	        0: -78.06 dB 1: -78.06 dB
	        balance 0.00
	base volume: 100%
	             0.00 dB
	volume steps: 65537
	muted: no
============================== 4% ==============
Simple mixer control 'Master',0
  Capabilities: pvolume pswitch penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 0 [0%] [-46.50dB] [on]
  Front Right: Playback 0 [0%] [-46.50dB] [on]
Simple mixer control 'PCM',0
  Capabilities: pvolume penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 255
  Mono:
  Front Left: Playback 69 [27%] [-37.20dB]
  Front Right: Playback 69 [27%] [-37.20dB]
PulseAudio:
  * index: 11
	name: <alsa_output.pci-0000_00_1b.0.analog-stereo>
	driver: <module-alsa-card.c>
	flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY FLAT_VOLUME DYNAMIC_LATENCY
	state: IDLE
	suspend cause: 
	priority: 9959
	volume: 0:   4% 1:   4%
	        0: -83.87 dB 1: -83.87 dB
	        balance 0.00
	base volume: 100%
	             0.00 dB
	volume steps: 65537
	muted: no
============================== 3% ==============
Simple mixer control 'Master',0
  Capabilities: pvolume pswitch penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 0 [0%] [-46.50dB] [on]
  Front Right: Playback 0 [0%] [-46.50dB] [on]
Simple mixer control 'PCM',0
  Capabilities: pvolume penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 255
  Mono:
  Front Left: Playback 31 [12%] [-44.80dB]
  Front Right: Playback 31 [12%] [-44.80dB]
PulseAudio:
  * index: 11
	name: <alsa_output.pci-0000_00_1b.0.analog-stereo>
	driver: <module-alsa-card.c>
	flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY FLAT_VOLUME DYNAMIC_LATENCY
	state: IDLE
	suspend cause: 
	priority: 9959
	volume: 0:   3% 1:   3%
	        0: -91.36 dB 1: -91.36 dB
	        balance 0.00
	base volume: 100%
	             0.00 dB
	volume steps: 65537
	muted: no
============================== 2% ==============
Simple mixer control 'Master',0
  Capabilities: pvolume pswitch penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 0 [0%] [-46.50dB] [on]
  Front Right: Playback 0 [0%] [-46.50dB] [on]
Simple mixer control 'PCM',0
  Capabilities: pvolume penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 255
  Mono:
  Front Left: Playback 0 [0%] [-51.00dB]
  Front Right: Playback 0 [0%] [-51.00dB]
PulseAudio:
  * index: 11
	name: <alsa_output.pci-0000_00_1b.0.analog-stereo>
	driver: <module-alsa-card.c>
	flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY FLAT_VOLUME DYNAMIC_LATENCY
	state: IDLE
	suspend cause: 
	priority: 9959
	volume: 0:   2% 1:   2%
	        0: -101.93 dB 1: -101.93 dB
	        balance 0.00
	base volume: 100%
	             0.00 dB
	volume steps: 65537
	muted: no
============================== 1% ==============
Simple mixer control 'Master',0
  Capabilities: pvolume pswitch penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 0 [0%] [-46.50dB] [on]
  Front Right: Playback 0 [0%] [-46.50dB] [on]
Simple mixer control 'PCM',0
  Capabilities: pvolume penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 255
  Mono:
  Front Left: Playback 0 [0%] [-51.00dB]
  Front Right: Playback 0 [0%] [-51.00dB]
PulseAudio:
  * index: 11
	name: <alsa_output.pci-0000_00_1b.0.analog-stereo>
	driver: <module-alsa-card.c>
	flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY FLAT_VOLUME DYNAMIC_LATENCY
	state: IDLE
	suspend cause: 
	priority: 9959
	volume: 0:   1% 1:   1%
	        0: -119.97 dB 1: -119.97 dB
	        balance 0.00
	base volume: 100%
	             0.00 dB
	volume steps: 65537
	muted: no
============================== 0% ==============
Simple mixer control 'Master',0
  Capabilities: pvolume pswitch penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 0 [0%] [-46.50dB] [on]
  Front Right: Playback 0 [0%] [-46.50dB] [on]
Simple mixer control 'PCM',0
  Capabilities: pvolume penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 255
  Mono:
  Front Left: Playback 0 [0%] [-51.00dB]
  Front Right: Playback 0 [0%] [-51.00dB]
PulseAudio:
  * index: 11
	name: <alsa_output.pci-0000_00_1b.0.analog-stereo>
	driver: <module-alsa-card.c>
	flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY FLAT_VOLUME DYNAMIC_LATENCY
	state: IDLE
	suspend cause: 
	priority: 9959
	volume: 0:   0% 1:   0%
	        0: -inf dB 1: -inf dB
	        balance 0.00
	base volume: 100%
	             0.00 dB
	volume steps: 65537
	muted: no

[-- Attachment #3: Type: text/plain, Size: 160 bytes --]

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Re: wrong decibel data?

[Raymond Yau]

2010/6/7 Colin Guthrie <gmane@colin.guthr.ie>

> >
> > if  floating point 0.0 is -inf dB , and 1.0 is 0dB ,
> >
> > 0.5 is -6dB , 0.25 is -12 dB , 0.125 is -24dB and 0.0625 is -48dB
>
>
> This is just a pure mapping from dB->linear, but as this linear mapping
> is generally not "natural" there are several different approaches to
> presenting this to users. In PA, a cubic mapping is used on top of this
> basic conversion, to map to the percentage scale (0.0 to 1.0 if you like).
>

Seem that my value is still wrong , every -6dB half the volume

so 1.5 is +6dB , 1.0 is 0dB, -6dB is 0.5 , -12 dB is 0.25, -18dB is 0.125,
-24dB is 0.0625

if shift 16 bit digital audio data to right by 1 bit is -6dB ,   -96dB is
0.000015



>
> So I'm not sure what point you're making by providing these numbers. Can
> you explain?
>
> > how can PA master volume control at 10~15% equivalent to HDA 's -48dB ?
>
> Not sure what you mean here, but I suspect it's the cubic mapping that
> is confusing you.
>
> Here is the function in PA's pulse/volume.c:
>
>  pa_volume_t pa_sw_volume_from_linear(double v) {
>
>    if (v <= 0.0)
>        return PA_VOLUME_MUTED;
>
>    /*
>     * We use a cubic mapping here, as suggested and discussed here:
>     *
>     * http://www.robotplanet.dk/audio/audio_gui_design/
>     *
>
> http://lists.linuxaudio.org/pipermail/linux-audio-dev/2009-May/thread.html#23151
>     *
>     * We make sure that the conversion to linear and back yields the
>     * same volume value! That's why we need the lround() below!
>     */
>
>    return (pa_volume_t) lround(cbrt(v) * PA_VOLUME_NORM);
>  }
>
>
can you provide the forumla for the "Master" volume control of pulse device
? ctl.pulse

there are 65536 steps ,
where are  0dB  , -6dB  and -inf dB on this cubic mapping ?

Re: wrong decibel data?

[Colin Guthrie]

'Twas brillig, and Raymond Yau at 08/06/10 01:47 did gyre and gimble:
>> Here is the function in PA's pulse/volume.c:
>>
>>  pa_volume_t pa_sw_volume_from_linear(double v) {
>>
>>    if (v <= 0.0)
>>        return PA_VOLUME_MUTED;
>>
>>    /*
>>     * We use a cubic mapping here, as suggested and discussed here:
>>     *
>>     * http://www.robotplanet.dk/audio/audio_gui_design/
>>     *
>>
>> http://lists.linuxaudio.org/pipermail/linux-audio-dev/2009-May/thread.html#23151
>>     *
>>     * We make sure that the conversion to linear and back yields the
>>     * same volume value! That's why we need the lround() below!
>>     */
>>
>>    return (pa_volume_t) lround(cbrt(v) * PA_VOLUME_NORM);
>>  }
>>
>>
> can you provide the forumla for the "Master" volume control of pulse device
> ? ctl.pulse
> 
> there are 65536 steps ,
> where are  0dB  , -6dB  and -inf dB on this cubic mapping ?

Well 0 dB and -inf dB are obvious: 100% and 0% respectively.

-6dB is about 80%:

1. Convert to Linear:
	10 ^ (-6 / 20.0) = 10 ^ -0.3 = 0.501187
2. Cube Root:
	 0.7943

Here is the code from volume.c that does this (combined with the
function already pasted above). It's probably easier just to follow the
code rather than have me describe it (no doubt poorly!):

static double dB_to_linear(double v) {
    return pow(10.0, v / 20.0);
}

pa_volume_t pa_sw_volume_from_dB(double dB) {
    if (isinf(dB) < 0 || dB <= PA_DECIBEL_MININFTY)
        return PA_VOLUME_MUTED;

    return pa_sw_volume_from_linear(dB_to_linear(dB));
}



The final stage in pa_sw_volume_from_linear() (the * by PA_VOLUME_NORM
and the lround()) is just converting to PA's own internal representation.

HTHs

Just for clarity, the reverse of for my approximate cut off at 16% would be:
 16% = 0.16
 0.16 ^ 3 = 0.004096
 20 * log(0.004096) = -47.75dB



Col


-- 

Colin Guthrie
gmane(at)colin.guthr.ie
http://colin.guthr.ie/

Day Job:
  Tribalogic Limited [http://www.tribalogic.net/]
Open Source:
  Mandriva Linux Contributor [http://www.mandriva.com/]
  PulseAudio Hacker [http://www.pulseaudio.org/]
  Trac Hacker [http://trac.edgewall.org/]

Re: wrong decibel data?

[Raymond Yau]

2010/6/8 Colin Guthrie <gmane@colin.guthr.ie>

> 'Twas brillig, and Raymond Yau at 08/06/10 01:47 did gyre and gimble:
> >> Here is the function in PA's pulse/volume.c:
> >>
> >>  pa_volume_t pa_sw_volume_from_linear(double v) {
> >>
> >>    if (v <= 0.0)
> >>        return PA_VOLUME_MUTED;
> >>
> >>    /*
> >>     * We use a cubic mapping here, as suggested and discussed here:
> >>     *
> >>     * http://www.robotplanet.dk/audio/audio_gui_design/
> >>     *
> >>
> >>
> http://lists.linuxaudio.org/pipermail/linux-audio-dev/2009-May/thread.html#23151
> >>     *
> >>     * We make sure that the conversion to linear and back yields the
> >>     * same volume value! That's why we need the lround() below!
> >>     */
> >>
> >>    return (pa_volume_t) lround(cbrt(v) * PA_VOLUME_NORM);
> >>  }
> >>
> >>
> > can you provide the forumla for the "Master" volume control of pulse
> device
> > ? ctl.pulse
> >
> > there are 65536 steps ,
> > where are  0dB  , -6dB  and -inf dB on this cubic mapping ?
>
> Well 0 dB and -inf dB are obvious: 100% and 0% respectively.
>
> -6dB is about 80%:
>
> 1. Convert to Linear:
>        10 ^ (-6 / 20.0) = 10 ^ -0.3 = 0.501187
> 2. Cube Root:
>         0.7943
>
> Here is the code from volume.c that does this (combined with the
> function already pasted above). It's probably easier just to follow the
> code rather than have me describe it (no doubt poorly!):
>
> static double dB_to_linear(double v) {
>    return pow(10.0, v / 20.0);
> }
>
> pa_volume_t pa_sw_volume_from_dB(double dB) {
>    if (isinf(dB) < 0 || dB <= PA_DECIBEL_MININFTY)
>        return PA_VOLUME_MUTED;
>
>    return pa_sw_volume_from_linear(dB_to_linear(dB));
> }
>
>
>
> The final stage in pa_sw_volume_from_linear() (the * by PA_VOLUME_NORM
> and the lround()) is just converting to PA's own internal representation.
>
> HTHs
>
> Just for clarity, the reverse of for my approximate cut off at 16% would
> be:
>  16% = 0.16
>  0.16 ^ 3 = 0.004096
>  20 * log(0.004096) = -47.75dB
>
>
Can you explain how PA handle the volume controls of ac97 codec ?

PCM -34.5dB to +*12* dB
Master -46.5dB to 0dB

The total dB range (PCM + MASTER) is -81dB to +*12*dB


Most user concern about recording without distrotion. (i.e. best result when
Capture Volume at 0dB , PCM and Master Volume at 0dB ) and they need where
are 0dB points

Re: wrong decibel data?

[Colin Guthrie]

'Twas brillig, and Raymond Yau at 10/06/10 04:16 did gyre and gimble:
> Can you explain how PA handle the volume controls of ac97 codec ?
> 
> PCM -34.5dB to +*12* dB
> Master -46.5dB to 0dB
> 
> The total dB range (PCM + MASTER) is -81dB to +*12*dB
> 
> 
> Most user concern about recording without distrotion. (i.e. best result when
> Capture Volume at 0dB , PCM and Master Volume at 0dB ) and they need where
> are 0dB points

I'm not 100% sure how this is handled, but I know it's not ignored.
You'll have to ask Lennart directly or dig in the code to see for sure.

Col


-- 

Colin Guthrie
gmane(at)colin.guthr.ie
http://colin.guthr.ie/

Day Job:
  Tribalogic Limited [http://www.tribalogic.net/]
Open Source:
  Mandriva Linux Contributor [http://www.mandriva.com/]
  PulseAudio Hacker [http://www.pulseaudio.org/]
  Trac Hacker [http://trac.edgewall.org/]

Re: wrong decibel data?

[Lennart Poettering]

On Thu, 10.06.10 17:11, Colin Guthrie (gmane@colin.guthr.ie) wrote:

> 
> 'Twas brillig, and Raymond Yau at 10/06/10 04:16 did gyre and gimble:
> > Can you explain how PA handle the volume controls of ac97 codec ?
> > 
> > PCM -34.5dB to +*12* dB
> > Master -46.5dB to 0dB
> > 
> > The total dB range (PCM + MASTER) is -81dB to +*12*dB
> > 
> > 
> > Most user concern about recording without distrotion. (i.e. best result when
> > Capture Volume at 0dB , PCM and Master Volume at 0dB ) and they need where
> > are 0dB points
> 
> I'm not 100% sure how this is handled, but I know it's not ignored.
> You'll have to ask Lennart directly or dig in the code to see for sure.

If the ALSA volume range is -x dB to +y dB, then the PA volume range
will be -x-y dB to 0dB (i.e. shifted by -y dB). On top of that most
volume controls should then mark the ALSA 0dB point as "base" volume on
the slider, at what PA then calls -y dB.

That way we will expose 0dB as maximum hw amplitude uniformly on all
sound cards and have a special point on the slider that is hinted to be
the "comfort" point.

This is all explained on
http://pulseaudio.org/wiki/WritingVolumeControlUIs#BaseVolumes

Lennart

-- 
Lennart Poettering                        Red Hat, Inc.
lennart [at] poettering [dot] net
http://0pointer.net/lennart/           GnuPG 0x1A015CC4

Re: wrong decibel data?

[Raymond Yau]

2010/6/13 Lennart Poettering <mznyfn@0pointer.de>

> On Thu, 10.06.10 17:11, Colin Guthrie (gmane@colin.guthr.ie) wrote:
>
> >
> > 'Twas brillig, and Raymond Yau at 10/06/10 04:16 did gyre and gimble:
> > > Can you explain how PA handle the volume controls of ac97 codec ?
> > >
> > > PCM -34.5dB to +*12* dB
> > > Master -46.5dB to 0dB
> > >
> > > The total dB range (PCM + MASTER) is -81dB to +*12*dB
> > >
> > >
> > > Most user concern about recording without distrotion. (i.e. best result
> when
> > > Capture Volume at 0dB , PCM and Master Volume at 0dB ) and they need
> where
> > > are 0dB points
> >
> > I'm not 100% sure how this is handled, but I know it's not ignored.
> > You'll have to ask Lennart directly or dig in the code to see for sure.
>
> If the ALSA volume range is -x dB to +y dB, then the PA volume range
> will be -x-y dB to 0dB (i.e. shifted by -y dB). On top of that most
> volume controls should then mark the ALSA 0dB point as "base" volume on
> the slider, at what PA then calls -y dB.
>
> That way we will expose 0dB as maximum hw amplitude uniformly on all
> sound cards and have a special point on the slider that is hinted to be
> the "comfort" point.
>
> This is all explained on
> http://pulseaudio.org/wiki/WritingVolumeControlUIs#BaseVolumes
>


shifted +12dB to 0dB is completely wrong

+12dB is 4.00 if 0dB is 1.0 (floating point )

you can performed clubsoda 's  experiement as in
https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/581650

>> "This is important because if I use those maximum settings and play any
audio which peaks above -12dB, the output will clip and sound distorted."


if your sound card have ac97 codec ., you can use audacity to record the
output from hw:0,0 and you will see clipping occur when you set "PCM" volume
above 0dB

Re: wrong decibel data?

[Colin Guthrie]

'Twas brillig, and Raymond Yau at 14/06/10 01:25 did gyre and gimble:
> if your sound card have ac97 codec ., you can use audacity to record the
> output from hw:0,0 and you will see clipping occur when you set "PCM" volume
> above 0dB

So the standard response is "don't do that then" :)

That's why the base volume is shown to the user via GUIs so that they
can gauge the best point on the slider to use. Currently there is no
indication with alsa sliders at which point the 0dB "sweet spot" lies.

IMO this is a good improvement for usability.

Col

-- 

Colin Guthrie
gmane(at)colin.guthr.ie
http://colin.guthr.ie/

Day Job:
  Tribalogic Limited [http://www.tribalogic.net/]
Open Source:
  Mandriva Linux Contributor [http://www.mandriva.com/]
  PulseAudio Hacker [http://www.pulseaudio.org/]
  Trac Hacker [http://trac.edgewall.org/]

Re: wrong decibel data?

[Raymond Yau]

2010/6/14 Colin Guthrie <gmane@colin.guthr.ie>

> 'Twas brillig, and Raymond Yau at 14/06/10 01:25 did gyre and gimble:
> > if your sound card have ac97 codec ., you can use audacity to record the
> > output from hw:0,0 and you will see clipping occur when you set "PCM"
> volume
> > above 0dB
>
> So the standard response is "don't do that then" :)
>
> That's why the base volume is shown to the user via GUIs so that they
> can gauge the best point on the slider to use. Currently there is no
> indication with alsa sliders at which point the 0dB "sweet spot" lies.
>
> IMO this is a good improvement for usability.
>
> Col
>
>
This is bug in pulseaudio in dB calculation because it shift the ALSA 's dB
value to PA 's own dB value

I guess none of PA developers has AC97 sound card

Re: wrong decibel data?

[Raymond Yau]

2010/6/14 Colin Guthrie <gmane@colin.guthr.ie>

> 'Twas brillig, and Raymond Yau at 14/06/10 01:25 did gyre and gimble:
> > if your sound card have ac97 codec ., you can use audacity to record the
> > output from hw:0,0 and you will see clipping occur when you set "PCM"
> volume
> > above 0dB
>
> So the standard response is "don't do that then" :)
>
> That's why the base volume is shown to the user via GUIs so that they
> can gauge the best point on the slider to use. Currently there is no
> indication with alsa sliders at which point the 0dB "sweet spot" lies.
>
> IMO this is a good improvement for usability.
>
> Col
>


The correct way is to provide the real ALSA 's 0dB point (Playback volume)
for the user of AC97 sound card so that they can record without any
distortion using line in with the loopback cable connected to line out.

Re: wrong decibel data?

[Colin Guthrie]

'Twas brillig, and Raymond Yau at 14/06/10 09:45 did gyre and gimble:
> 2010/6/14 Colin Guthrie <gmane@colin.guthr.ie>
> 
>> 'Twas brillig, and Raymond Yau at 14/06/10 01:25 did gyre and gimble:
>>> if your sound card have ac97 codec ., you can use audacity to record the
>>> output from hw:0,0 and you will see clipping occur when you set "PCM"
>> volume
>>> above 0dB
>>
>> So the standard response is "don't do that then" :)
>>
>> That's why the base volume is shown to the user via GUIs so that they
>> can gauge the best point on the slider to use. Currently there is no
>> indication with alsa sliders at which point the 0dB "sweet spot" lies.
>>
>> IMO this is a good improvement for usability.
>>
>> Col
>>
> 
> 
> The correct way is to provide the real ALSA 's 0dB point (Playback volume)
> for the user of AC97 sound card so that they can record without any
> distortion using line in with the loopback cable connected to line out.

Raymond, can you please, please stop replying to messages multiple
times. It makes threads extremely hard to follow and makes replying to
your points consistently next to impossible.


>From Lennarts description, the 0dB point (according to alsa) IS the base
volume and thus is shown to the user as such. Am I missing something?

The fact that the 0dB point is shifted is fairly irrelevant to the user.

Col

-- 

Colin Guthrie
gmane(at)colin.guthr.ie
http://colin.guthr.ie/

Day Job:
  Tribalogic Limited [http://www.tribalogic.net/]
Open Source:
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  Trac Hacker [http://trac.edgewall.org/]

Re: wrong decibel data?

[Lennart Poettering]

On Mon, 14.06.10 16:45, Raymond Yau (superquad.vortex2@gmail.com) wrote:

> The correct way is to provide the real ALSA 's 0dB point (Playback volume)
> for the user of AC97 sound card so that they can record without any
> distortion using line in with the loopback cable connected to line out.

Jeez, man, I explained that in my original reply.

If I may quote myself:

   'On top of that most volume controls should then mark the ALSA 0dB
   point as "base" volume on the slider, at what PA then calls -y dB.

   That way we will expose 0dB as maximum hw amplitude uniformly on all
   sound cards and have a special point on the slider that is hinted to be
   the "comfort" point.

   This is all explained on
   http://pulseaudio.org/wiki/WritingVolumeControlUIs#BaseVolumes'

See? It's all explained there. We still show the ALSA 0dB point on our
sliders, we just don't call it "0dB" but "base volume".

Lennart

-- 
Lennart Poettering                        Red Hat, Inc.
lennart [at] poettering [dot] net
http://0pointer.net/lennart/           GnuPG 0x1A015CC4

Re: wrong decibel data?

[James Courtier-Dutton]

On 14 June 2010 09:33, Colin Guthrie <gmane@colin.guthr.ie> wrote:
> 'Twas brillig, and Raymond Yau at 14/06/10 01:25 did gyre and gimble:
>> if your sound card have ac97 codec ., you can use audacity to record the
>> output from hw:0,0 and you will see clipping occur when you set "PCM" volume
>> above 0dB
>
> So the standard response is "don't do that then" :)
>
> That's why the base volume is shown to the user via GUIs so that they
> can gauge the best point on the slider to use. Currently there is no
> indication with alsa sliders at which point the 0dB "sweet spot" lies.
>

What do you mean.
If you use "alsamixer", dB values are shown so it is easy to find the
0dB "sweet spot".
I think it is pulse audio that hides this information when it combines
two alsa mixer controls into one pulseaudio control.

Re: wrong decibel data?

[Raymond Yau]

2010/6/14 James Courtier-Dutton <james.dutton@gmail.com>

> On 14 June 2010 09:33, Colin Guthrie <gmane@colin.guthr.ie> wrote:
> > 'Twas brillig, and Raymond Yau at 14/06/10 01:25 did gyre and gimble:
> >> if your sound card have ac97 codec ., you can use audacity to record the
> >> output from hw:0,0 and you will see clipping occur when you set "PCM"
> volume
> >> above 0dB
> >
> > So the standard response is "don't do that then" :)
> >
> > That's why the base volume is shown to the user via GUIs so that they
> > can gauge the best point on the slider to use. Currently there is no
> > indication with alsa sliders at which point the 0dB "sweet spot" lies.
> >
>
> What do you mean.
> If you use "alsamixer", dB values are shown so it is easy to find the
> 0dB "sweet spot".
> I think it is pulse audio that hides this information when it combines
> two alsa mixer controls into one pulseaudio control.
>


The base volume seem to be the software 0dB point , (no software
gain/atten), but the user want the hardware 0dB point (no hardware
gain/atten if the hardware can provide hardware gain

This hardware 0dB point is extremely important when you want to record using
line in and line out

Re: wrong decibel data?

[James Courtier-Dutton]

On 14 June 2010 10:54, Raymond Yau <superquad.vortex2@gmail.com> wrote:
> 2010/6/14 James Courtier-Dutton <james.dutton@gmail.com>
>> If you use "alsamixer", dB values are shown so it is easy to find the
>> 0dB "sweet spot".
>> I think it is pulse audio that hides this information when it combines
>> two alsa mixer controls into one pulseaudio control.
>>
>
>
> The base volume seem to be the software 0dB point , (no software
> gain/atten), but the user want the hardware 0dB point (no hardware
> gain/atten if the hardware can provide hardware gain
>
> This hardware 0dB point is extremely important when you want to record using
> line in and line out

alsamixer gives the hardware 0dB point.

Re: wrong decibel data?

[Raymond Yau]

2010/6/14 James Courtier-Dutton <james.dutton@gmail.com>

> On 14 June 2010 10:54, Raymond Yau <superquad.vortex2@gmail.com> wrote:
> > 2010/6/14 James Courtier-Dutton <james.dutton@gmail.com>
> >> If you use "alsamixer", dB values are shown so it is easy to find the
> >> 0dB "sweet spot".
> >> I think it is pulse audio that hides this information when it combines
> >> two alsa mixer controls into one pulseaudio control.
> >>
> >
> >
> > The base volume seem to be the software 0dB point , (no software
> > gain/atten), but the user want the hardware 0dB point (no hardware
> > gain/atten if the hardware can provide hardware gain
> >
> > This hardware 0dB point is extremely important when you want to record
> using
> > line in and line out
>
> alsamixer gives the hardware 0dB point.
>

you are right ,

For ac97 Base volume is the real hardware  0dB point of PCM volume, and
Volume NORM is just the max_dB of +12dB which PA labeled it as 100% aka 0dB
aka Volume NORM

For HDA , there is no yellow region because max_dB is 0dB , Base Volume and
NORM is at the same point

http://en.wikipedia.org/wiki/Line_level

How can I get distortion free recording by line in of ac97 sound connected
to line out of HDA onborad and vice versa ?

http://pulseaudio.org/wiki/WritingVolumeControlUIs#Colouredvolumesliders

Re: wrong decibel data?

[Lennart Poettering]

On Mon, 14.06.10 17:54, Raymond Yau (superquad.vortex2@gmail.com) wrote:

> The base volume seem to be the software 0dB point , (no software
> gain/atten), but the user want the hardware 0dB point (no hardware
> gain/atten if the hardware can provide hardware gain

jeez man, this is nonsense.

In PA we call the max hw amplification 0dB. And then we extend things
both ways in software. And finally we mark the ALSA 0dB spot (i.e. the
*hardware* 0dB spot) as "base volume".

There is nothing lost here.

And you get a cookie if you read this:

http://pulseaudio.org/wiki/WritingVolumeControlUIs#BaseVolumes

and this:

http://pulseaudio.org/wiki/WritingVolumeControlUIs#Colouredvolumesliders

And then you'll notice that we actually thought about hw and sw volume
ranges quite a bit and nothing is lost. For example, our recommended
color-coded UI will mark the ALSA 0dB spot with a color transition from
green to yellow, to give the user an idea that things get worse beyond
that point, even if he doesn't understand dB or hw or sw volumes.

Please, just drop this thread now, and just read the wiki. Thanks.

Lennart

-- 
Lennart Poettering                        Red Hat, Inc.
lennart [at] poettering [dot] net
http://0pointer.net/lennart/           GnuPG 0x1A015CC4

Re: wrong decibel data?

[Colin Guthrie]

'Twas brillig, and James Courtier-Dutton at 14/06/10 09:56 did gyre and
gimble:
> On 14 June 2010 09:33, Colin Guthrie <gmane@colin.guthr.ie> wrote:
>> 'Twas brillig, and Raymond Yau at 14/06/10 01:25 did gyre and gimble:
>>> if your sound card have ac97 codec ., you can use audacity to record the
>>> output from hw:0,0 and you will see clipping occur when you set "PCM" volume
>>> above 0dB
>>
>> So the standard response is "don't do that then" :)
>>
>> That's why the base volume is shown to the user via GUIs so that they
>> can gauge the best point on the slider to use. Currently there is no
>> indication with alsa sliders at which point the 0dB "sweet spot" lies.
>>
> 
> What do you mean.
> If you use "alsamixer", dB values are shown so it is easy to find the
> 0dB "sweet spot".
> I think it is pulse audio that hides this information when it combines
> two alsa mixer controls into one pulseaudio control.

But it doesn't hide it. It's shown very clearly in the volume control
GUIs as the Base Volume.

Do you really think that most users look at the sliders to find the 0dB
point? Does gnome-alsa-mixer (the old one) expose this information? No.
Does kmix? No. So the vast, vast majority of users do not know where the
0dB point is unless they use alsamixer.... and even if the user is
advanced enough to use alsamixer, then I'd still say a proportion of
users are just looking at how far up the slider is rather than looking
specifically for 0dB.

So I'd argue the exact opposite of your claim. That with the base volume
clearly presented in the GUI, the h/w 0dB spot is much, much more
obvious to the vast majority of users.

I really think this is a vast improvement over a complex balancing act
of getting two different sliders setup to get distortion free audio!

Col

Caveat: I've not yet made kmix show the base volume, so it still suffers
from the problem of masking this important information from the user.



-- 

Colin Guthrie
gmane(at)colin.guthr.ie
http://colin.guthr.ie/

Day Job:
  Tribalogic Limited [http://www.tribalogic.net/]
Open Source:
  Mandriva Linux Contributor [http://www.mandriva.com/]
  PulseAudio Hacker [http://www.pulseaudio.org/]
  Trac Hacker [http://trac.edgewall.org/]

Re: wrong decibel data?

[James Courtier-Dutton]

On 14 June 2010 11:22, Colin Guthrie <gmane@colin.guthr.ie> wrote:
> 'Twas brillig, and James Courtier-Dutton at 14/06/10 09:56 did gyre and
> gimble:
>> If you use "alsamixer", dB values are shown so it is easy to find the
>> 0dB "sweet spot".
>> I think it is pulse audio that hides this information when it combines
>> two alsa mixer controls into one pulseaudio control.
>
> But it doesn't hide it. It's shown very clearly in the volume control
> GUIs as the Base Volume.
>
> Do you really think that most users look at the sliders to find the 0dB
> point? Does gnome-alsa-mixer (the old one) expose this information? No.
> Does kmix? No. So the vast, vast majority of users do not know where the
> 0dB point is unless they use alsamixer.... and even if the user is
> advanced enough to use alsamixer, then I'd still say a proportion of
> users are just looking at how far up the slider is rather than looking
> specifically for 0dB.
>
> So I'd argue the exact opposite of your claim. That with the base volume
> clearly presented in the GUI, the h/w 0dB spot is much, much more
> obvious to the vast majority of users.
>
> I really think this is a vast improvement over a complex balancing act
> of getting two different sliders setup to get distortion free audio!
>
> Col

One has very different problems with capture than one does with playback.
With capture it is important to identify which are analog controls
(applied to the analog part of the circuit) and which are digital
controls (applied to the digital part of the circuit)
So, for capture one might wish to adjust the analog control so that
the signal going into the ADC is a suitable level, but once the signal
is digital, one should really not adjust it further, and just record
what you have.
If one was to combine these two capture controls in one PA control, it
would just be wrong.

I think there is some indication with the name of the control. It
sometimes has "Analog" or "Digital" attached to it.
I think this would be better if alsa reported the "Analog" or
"Digital" as meta data, like the dB Scales.
PA could then make more informed decisions for capture. I.e. only
display the "Analog" controls, and hide the digital ones, setting them
to 0dB.
That would provide the most distortion free capture.
I think it would also be useful if the alsa driver also reported meta
data indicating how the controls are connected together, because then
PA would have even more information to make better decisions.
For example, USB audio devices have this information, but it is not
sent to user space.

Re: wrong decibel data?

[Colin Guthrie]

'Twas brillig, and James Courtier-Dutton at 14/06/10 11:46 did gyre and
gimble:
> On 14 June 2010 11:22, Colin Guthrie <gmane@colin.guthr.ie> wrote:
>> 'Twas brillig, and James Courtier-Dutton at 14/06/10 09:56 did gyre and
>> gimble:
>>> If you use "alsamixer", dB values are shown so it is easy to find the
>>> 0dB "sweet spot".
>>> I think it is pulse audio that hides this information when it combines
>>> two alsa mixer controls into one pulseaudio control.
>>
>> But it doesn't hide it. It's shown very clearly in the volume control
>> GUIs as the Base Volume.
>>
>> Do you really think that most users look at the sliders to find the 0dB
>> point? Does gnome-alsa-mixer (the old one) expose this information? No.
>> Does kmix? No. So the vast, vast majority of users do not know where the
>> 0dB point is unless they use alsamixer.... and even if the user is
>> advanced enough to use alsamixer, then I'd still say a proportion of
>> users are just looking at how far up the slider is rather than looking
>> specifically for 0dB.
>>
>> So I'd argue the exact opposite of your claim. That with the base volume
>> clearly presented in the GUI, the h/w 0dB spot is much, much more
>> obvious to the vast majority of users.
>>
>> I really think this is a vast improvement over a complex balancing act
>> of getting two different sliders setup to get distortion free audio!
>>
>> Col
> 
> One has very different problems with capture than one does with playback.
> With capture it is important to identify which are analog controls
> (applied to the analog part of the circuit) and which are digital
> controls (applied to the digital part of the circuit)
> So, for capture one might wish to adjust the analog control so that
> the signal going into the ADC is a suitable level, but once the signal
> is digital, one should really not adjust it further, and just record
> what you have.
> If one was to combine these two capture controls in one PA control, it
> would just be wrong.
> 
> I think there is some indication with the name of the control. It
> sometimes has "Analog" or "Digital" attached to it.
> I think this would be better if alsa reported the "Analog" or
> "Digital" as meta data, like the dB Scales.
> PA could then make more informed decisions for capture. I.e. only
> display the "Analog" controls, and hide the digital ones, setting them
> to 0dB.
> That would provide the most distortion free capture.
> I think it would also be useful if the alsa driver also reported meta
> data indicating how the controls are connected together, because then
> PA would have even more information to make better decisions.
> For example, USB audio devices have this information, but it is not
> sent to user space.


Sounds like that would actually fit in with the current logic (correct
me if I'm wrong). PA adjusts multiple sliders in a left-to-right
fashion, trying to achieve the ultimate volume the user has requested by
setting the left most first and checking if it's "accurate enough". If
it is, then it stops there, but if not then it tries to adjust the
control to the right. This is repeated until we are "accurate enough"
with any further adjustements made in software if needed.

If all the analog controls are lined up to the left followed by all the
digital controls to the right, then the goal of "leaving the digital
controls alone if at all possible" would be a by product (I think).

At present Master will always be left most, and PCM will always sit to
the right of it.

In addition to Lennart's previously posted link on Volume Control GUIs
(for the lazyweb: http://pulseaudio.org/wiki/WritingVolumeControlUIs),
this approach is explained here:
http://pulseaudio.org/wiki/PulseAudioStoleMyVolumes


Col

-- 

Colin Guthrie
gmane(at)colin.guthr.ie
http://colin.guthr.ie/

Day Job:
  Tribalogic Limited [http://www.tribalogic.net/]
Open Source:
  Mandriva Linux Contributor [http://www.mandriva.com/]
  PulseAudio Hacker [http://www.pulseaudio.org/]
  Trac Hacker [http://trac.edgewall.org/]

Re: wrong decibel data?

[Alan Horstmann]

On Monday 14 June 2010 11:46, James Courtier-Dutton wrote:
> On 14 June 2010 11:22, Colin Guthrie <gmane@colin.guthr.ie> wrote:
> > 'Twas brillig, and James Courtier-Dutton at 14/06/10 09:56 did gyre and
> >
> > gimble:
> >> If you use "alsamixer", dB values are shown so it is easy to find the
> >> 0dB "sweet spot".
> >> I think it is pulse audio that hides this information when it combines
> >> two alsa mixer controls into one pulseaudio control.
> >
> > But it doesn't hide it. It's shown very clearly in the volume control
> > GUIs as the Base Volume.
> >
> > Do you really think that most users look at the sliders to find the 0dB
> > point? Does gnome-alsa-mixer (the old one) expose this information? No.
> > Does kmix? No. So the vast, vast majority of users do not know where the
> > 0dB point is unless they use alsamixer.... and even if the user is
> > advanced enough to use alsamixer, then I'd still say a proportion of
> > users are just looking at how far up the slider is rather than looking
> > specifically for 0dB.
> >
> > So I'd argue the exact opposite of your claim. That with the base volume
> > clearly presented in the GUI, the h/w 0dB spot is much, much more
> > obvious to the vast majority of users.
> >
> > I really think this is a vast improvement over a complex balancing act
> > of getting two different sliders setup to get distortion free audio!
> >
> > Col
>
> One has very different problems with capture than one does with playback.
> With capture it is important to identify which are analog controls
> (applied to the analog part of the circuit) and which are digital
> controls (applied to the digital part of the circuit)
> So, for capture one might wish to adjust the analog control so that
> the signal going into the ADC is a suitable level, but once the signal
> is digital, one should really not adjust it further, and just record
> what you have.
> If one was to combine these two capture controls in one PA control, it
> would just be wrong.
>
> I think there is some indication with the name of the control. It
> sometimes has "Analog" or "Digital" attached to it.
> I think this would be better if alsa reported the "Analog" or
> "Digital" as meta data, like the dB Scales.
> PA could then make more informed decisions for capture. I.e. only
> display the "Analog" controls, and hide the digital ones, setting them
> to 0dB.
> That would provide the most distortion free capture.
> I think it would also be useful if the alsa driver also reported meta
> data indicating how the controls are connected together, because then
> PA would have even more information to make better decisions.
> For example, USB audio devices have this information, but it is not
> sent to user space.

I tried unsuccessfully to argue the importance of the distinction between 
analogue and digital gain controls for capture some time ago in respect of 
ice1712 based cards fitted with 24-bit AK4524 ADAC chips (or something 
similar).  The chip provides analogue gain pre-ADC, and software attenuation 
post-ADC.  They are however set via a single register, which prevents 
needless gain+attenuation.  Originally they were exposed as 2 separate (but 
interacting) controls, giving the user the ability to set true '0dB' 
no-gain-no-software-attenuation point (for pro use).  The proposal was made 
to merge into a single control, and my view against did not prevail.  Thus 
from 1.0.14 the analogue and digital controls portions are indistinguishable 
on this card, unless you happen to know the 'magic number' that is the value 
corresponding to the transition from analogue to digital on the scale.

If a mixer app were unknowingly to consider the full-scale value to be the 
normal operating point, and scale from there, the results would be poor.

Alan

Re: wrong decibel data?

[Raymond Yau]

2010/6/14 James Courtier-Dutton <james.dutton@gmail.com>

> On 14 June 2010 11:22, Colin Guthrie <gmane@colin.guthr.ie> wrote:
> > 'Twas brillig, and James Courtier-Dutton at 14/06/10 09:56 did gyre and
> > gimble:
> >> If you use "alsamixer", dB values are shown so it is easy to find the
> >> 0dB "sweet spot".
> >> I think it is pulse audio that hides this information when it combines
> >> two alsa mixer controls into one pulseaudio control.
> >
> > But it doesn't hide it. It's shown very clearly in the volume control
> > GUIs as the Base Volume.
> >
> > Do you really think that most users look at the sliders to find the 0dB
> > point? Does gnome-alsa-mixer (the old one) expose this information? No.
> > Does kmix? No. So the vast, vast majority of users do not know where the
> > 0dB point is unless they use alsamixer.... and even if the user is
> > advanced enough to use alsamixer, then I'd still say a proportion of
> > users are just looking at how far up the slider is rather than looking
> > specifically for 0dB.
> >
> > So I'd argue the exact opposite of your claim. That with the base volume
> > clearly presented in the GUI, the h/w 0dB spot is much, much more
> > obvious to the vast majority of users.
> >
> > I really think this is a vast improvement over a complex balancing act
> > of getting two different sliders setup to get distortion free audio!
> >
> > Col
>
> One has very different problems with capture than one does with playback.
> With capture it is important to identify which are analog controls
> (applied to the analog part of the circuit) and which are digital
> controls (applied to the digital part of the circuit)
> So, for capture one might wish to adjust the analog control so that
> the signal going into the ADC is a suitable level, but once the signal
> is digital, one should really not adjust it further, and just record
> what you have.
> If one was to combine these two capture controls in one PA control, it
> would just be wrong.
>
>
The AC97 recording from line-in problem seem not related to capture gain
since you can set capture volume to 0dB

The HDA 's "PCM" softvol plugin is different from AC97 "PCM" Playback volume

But you can change the softvol plugin to add gain to emulate the clipping in
software side if PA developers did not have ac97 sound card ( clipping occur
in hardware side )

  /usr/share/alsa/cards/HDA-Intel.conf

HDA-Intel.pcm.front.0 {
    @args [ CARD ]
    @args.CARD {
        type string
    }
    type softvol
    slave.pcm {
        type hw
        card $CARD
    }
    control {
        name "PCM Playback Volume"
        card $CARD
    }
+      min_dB -46.5
+     max_dB 12.0
+      resolution 32
}

Re: wrong decibel data?

[Colin Guthrie]

'Twas brillig, and Raymond Yau at 14/06/10 13:36 did gyre and gimble:
> 2010/6/14 James Courtier-Dutton <james.dutton@gmail.com>
> 
>> On 14 June 2010 11:22, Colin Guthrie <gmane@colin.guthr.ie> wrote:
>>> 'Twas brillig, and James Courtier-Dutton at 14/06/10 09:56 did gyre and
>>> gimble:
>>>> If you use "alsamixer", dB values are shown so it is easy to find the
>>>> 0dB "sweet spot".
>>>> I think it is pulse audio that hides this information when it combines
>>>> two alsa mixer controls into one pulseaudio control.
>>>
>>> But it doesn't hide it. It's shown very clearly in the volume control
>>> GUIs as the Base Volume.
>>>
>>> Do you really think that most users look at the sliders to find the 0dB
>>> point? Does gnome-alsa-mixer (the old one) expose this information? No.
>>> Does kmix? No. So the vast, vast majority of users do not know where the
>>> 0dB point is unless they use alsamixer.... and even if the user is
>>> advanced enough to use alsamixer, then I'd still say a proportion of
>>> users are just looking at how far up the slider is rather than looking
>>> specifically for 0dB.
>>>
>>> So I'd argue the exact opposite of your claim. That with the base volume
>>> clearly presented in the GUI, the h/w 0dB spot is much, much more
>>> obvious to the vast majority of users.
>>>
>>> I really think this is a vast improvement over a complex balancing act
>>> of getting two different sliders setup to get distortion free audio!
>>>
>>> Col
>>
>> One has very different problems with capture than one does with playback.
>> With capture it is important to identify which are analog controls
>> (applied to the analog part of the circuit) and which are digital
>> controls (applied to the digital part of the circuit)
>> So, for capture one might wish to adjust the analog control so that
>> the signal going into the ADC is a suitable level, but once the signal
>> is digital, one should really not adjust it further, and just record
>> what you have.
>> If one was to combine these two capture controls in one PA control, it
>> would just be wrong.
>>
>>
> The AC97 recording from line-in problem seem not related to capture gain
> since you can set capture volume to 0dB
> 
> The HDA 's "PCM" softvol plugin is different from AC97 "PCM" Playback volume
> 
> But you can change the softvol plugin to add gain to emulate the clipping in
> software side if PA developers did not have ac97 sound card ( clipping occur
> in hardware side )
> 
>   /usr/share/alsa/cards/HDA-Intel.conf
> 
> HDA-Intel.pcm.front.0 {
>     @args [ CARD ]
>     @args.CARD {
>         type string
>     }
>     type softvol
>     slave.pcm {
>         type hw
>         card $CARD
>     }
>     control {
>         name "PCM Playback Volume"
>         card $CARD
>     }
> +      min_dB -46.5
> +     max_dB 12.0
> +      resolution 32
> }

I've made this change on my system and while previously my UI had no
"Base Volume" displayed (because all my "h/w" (I include softvol in
that) controls had their dB value >0.

Now that this change is live, I have a base volume present in my GUI (at
around the 64% mark with the cubic scale we've already discussed). When
I set my volume ot the base volume, the h/w controls are all set to 0dB
which is exactly as expected.

I fail to see the point here? The base volume is clearly exposed to the
as the recommended point on the scale at which no clipping occurs.

I really don't get where your complaint is.

Col





-- 

Colin Guthrie
gmane(at)colin.guthr.ie
http://colin.guthr.ie/

Day Job:
  Tribalogic Limited [http://www.tribalogic.net/]
Open Source:
  Mandriva Linux Contributor [http://www.mandriva.com/]
  PulseAudio Hacker [http://www.pulseaudio.org/]
  Trac Hacker [http://trac.edgewall.org/]

Re: wrong decibel data?

[James Courtier-Dutton]

On 14 June 2010 15:17, Colin Guthrie <gmane@colin.guthr.ie> wrote:
>
> I've made this change on my system and while previously my UI had no
> "Base Volume" displayed (because all my "h/w" (I include softvol in
> that) controls had their dB value >0.
>
> Now that this change is live, I have a base volume present in my GUI (at
> around the 64% mark with the cubic scale we've already discussed). When
> I set my volume ot the base volume, the h/w controls are all set to 0dB
> which is exactly as expected.
>
> I fail to see the point here? The base volume is clearly exposed to the
> as the recommended point on the scale at which no clipping occurs.
>
> I really don't get where your complaint is.
>

Well, if you can define "Volume" in a way that lets you understand
this then fine.
For me, all these controls are not adjusting "Volume", they are
adjusting "Gain", so why are they even called "Base Volume".

Re: wrong decibel data?

[Colin Guthrie]

'Twas brillig, and James Courtier-Dutton at 14/06/10 16:27 did gyre and
gimble:
> On 14 June 2010 15:17, Colin Guthrie <gmane@colin.guthr.ie> wrote:
>>
>> I've made this change on my system and while previously my UI had no
>> "Base Volume" displayed (because all my "h/w" (I include softvol in
>> that) controls had their dB value >0.
>>
>> Now that this change is live, I have a base volume present in my GUI (at
>> around the 64% mark with the cubic scale we've already discussed). When
>> I set my volume ot the base volume, the h/w controls are all set to 0dB
>> which is exactly as expected.
>>
>> I fail to see the point here? The base volume is clearly exposed to the
>> as the recommended point on the scale at which no clipping occurs.
>>
>> I really don't get where your complaint is.
>>
> 
> Well, if you can define "Volume" in a way that lets you understand
> this then fine.
> For me, all these controls are not adjusting "Volume", they are
> adjusting "Gain", so why are they even called "Base Volume".

It's all about context. While "Gain" may be a more accurate terminology,
the vast majority of users wont really understand this. The term
"Volume" is much more readily understandable by the unwashed masses.

At the end of the day, "relative volume adjustments" and "gain" are the
same thing but I guess people who understand and appreicate what 0dB
means would think of gain as the net change of a system of controls from
input to output, but think of volume as something absolute - e.g. a
volume of 11 would sound the same regardless of source, where as a gain
of +4dB would be meaningless in itself and be entirely dependant on the
level of the source..... but trying to explain all this in a simple GUI
you grannie can use is not really worth the effort IMHO.

Col



-- 

Colin Guthrie
gmane(at)colin.guthr.ie
http://colin.guthr.ie/

Day Job:
  Tribalogic Limited [http://www.tribalogic.net/]
Open Source:
  Mandriva Linux Contributor [http://www.mandriva.com/]
  PulseAudio Hacker [http://www.pulseaudio.org/]
  Trac Hacker [http://trac.edgewall.org/]

Re: wrong decibel data?

[Mark Brown]

On Mon, Jun 14, 2010 at 04:27:32PM +0100, James Courtier-Dutton wrote:

> Well, if you can define "Volume" in a way that lets you understand
> this then fine.
> For me, all these controls are not adjusting "Volume", they are
> adjusting "Gain", so why are they even called "Base Volume".

"Volume" has a specific meaning in the ALSA ABI.

Re: wrong decibel data?

[Raymond Yau]

2010/6/14 James Courtier-Dutton <james.dutton@gmail.com>

> On 14 June 2010 15:17, Colin Guthrie <gmane@colin.guthr.ie> wrote:
> >
> > I've made this change on my system and while previously my UI had no
> > "Base Volume" displayed (because all my "h/w" (I include softvol in
> > that) controls had their dB value >0.
> >
> > Now that this change is live, I have a base volume present in my GUI (at
> > around the 64% mark with the cubic scale we've already discussed). When
> > I set my volume ot the base volume, the h/w controls are all set to 0dB
> > which is exactly as expected.
> >
> > I fail to see the point here? The base volume is clearly exposed to the
> > as the recommended point on the scale at which no clipping occurs.
> >
> > I really don't get where your complaint is.
> >
>

alsamixer -D pulse does not show that sweet spot to the user



>
> Well, if you can define "Volume" in a way that lets you understand
> this then fine.
> For me, all these controls are not adjusting "Volume", they are
> adjusting "Gain", so why are they even called "Base Volume".
>

If you play music using aplay or mplayer with modified front device , you
should notice that the playback signal after passing the softvol plugin is
already clipped when you set the softvol to 12dB (software gain is the cause
of this clipping )

Re: wrong decibel data?

[Raymond Yau]

2010/6/14 Colin Guthrie <gmane@colin.guthr.ie>

> 'Twas brillig, and Raymond Yau at 14/06/10 13:36 did gyre and gimble:
> > 2010/6/14 James Courtier-Dutton <james.dutton@gmail.com>
> >
> >> On 14 June 2010 11:22, Colin Guthrie <gmane@colin.guthr.ie> wrote:
> >>> 'Twas brillig, and James Courtier-Dutton at 14/06/10 09:56 did gyre and
> >>> gimble:
> >>>> If you use "alsamixer", dB values are shown so it is easy to find the
> >>>> 0dB "sweet spot".
> >>>> I think it is pulse audio that hides this information when it combines
> >>>> two alsa mixer controls into one pulseaudio control.
> >>>
> >>> But it doesn't hide it. It's shown very clearly in the volume control
> >>> GUIs as the Base Volume.
> >>>
> >>> Do you really think that most users look at the sliders to find the 0dB
> >>> point? Does gnome-alsa-mixer (the old one) expose this information? No.
> >>> Does kmix? No. So the vast, vast majority of users do not know where
> the
> >>> 0dB point is unless they use alsamixer.... and even if the user is
> >>> advanced enough to use alsamixer, then I'd still say a proportion of
> >>> users are just looking at how far up the slider is rather than looking
> >>> specifically for 0dB.
> >>>
> >>> So I'd argue the exact opposite of your claim. That with the base
> volume
> >>> clearly presented in the GUI, the h/w 0dB spot is much, much more
> >>> obvious to the vast majority of users.
> >>>
> >>> I really think this is a vast improvement over a complex balancing act
> >>> of getting two different sliders setup to get distortion free audio!
> >>>
> >>> Col
> >>
> >> One has very different problems with capture than one does with
> playback.
> >> With capture it is important to identify which are analog controls
> >> (applied to the analog part of the circuit) and which are digital
> >> controls (applied to the digital part of the circuit)
> >> So, for capture one might wish to adjust the analog control so that
> >> the signal going into the ADC is a suitable level, but once the signal
> >> is digital, one should really not adjust it further, and just record
> >> what you have.
> >> If one was to combine these two capture controls in one PA control, it
> >> would just be wrong.
> >>
> >>
> > The AC97 recording from line-in problem seem not related to capture gain
> > since you can set capture volume to 0dB
> >
> > The HDA 's "PCM" softvol plugin is different from AC97 "PCM" Playback
> volume
> >
> > But you can change the softvol plugin to add gain to emulate the clipping
> in
> > software side if PA developers did not have ac97 sound card ( clipping
> occur
> > in hardware side )
> >
> >   /usr/share/alsa/cards/HDA-Intel.conf
> >
> > HDA-Intel.pcm.front.0 {
> >     @args [ CARD ]
> >     @args.CARD {
> >         type string
> >     }
> >     type softvol
> >     slave.pcm {
> >         type hw
> >         card $CARD
> >     }
> >     control {
> >         name "PCM Playback Volume"
> >         card $CARD
> >     }
> > +      min_dB -46.5
> > +     max_dB 12.0
> > +      resolution 32
> > }
>
> I've made this change on my system and while previously my UI had no
> "Base Volume" displayed (because all my "h/w" (I include softvol in
> that) controls had their dB value >0.
>

as your card has no h/w gain,  > 0dB , but the gain in softvol plugin is a
software gain (i.e. in the red region in PA "s volume scale

how can base_volume display in gnome volume control (unamplified) ?

BTW ,  -46.5dB to 0dB of softvol plugin is software atten ( not h/w atten )




>
> Now that this change is live, I have a base volume present in my GUI (at
> around the 64% mark with the cubic scale we've already discussed). When
> I set my volume ot the base volume, the h/w controls are all set to 0dB
> which is exactly as expected.
>


>
> I fail to see the point here? The base volume is clearly exposed to the
> as the recommended point on the scale at which no clipping occurs.
>
> I really don't get where your complaint is.
>
> Col
>
>
>
>

Re: wrong decibel data?

[Colin Guthrie]

'Twas brillig, and Raymond Yau at 22/06/10 03:31 did gyre and gimble:
> 2010/6/14 Colin Guthrie <gmane@colin.guthr.ie>
>> I've made this change on my system and while previously my UI had no
>> "Base Volume" displayed (because all my "h/w" (I include softvol in
>> that) controls had their dB value >0.
>>
> 
> as your card has no h/w gain,  > 0dB , but the gain in softvol plugin is a
> software gain (i.e. in the red region in PA "s volume scale
> 
> how can base_volume display in gnome volume control (unamplified) ?
> 
> BTW ,  -46.5dB to 0dB of softvol plugin is software atten ( not h/w atten )

As I said above, anything that comes from alsa is considered h/w
amplification for the purposes of PA's volume scale. It's not practical
to differentiate them.

Does any card actually configure softvol, by default, to provide any
gain, > 0dB for outputs? If so, then this is IMO a bad idea.

Again, it's ultimately related to your range-checking bugbear. Users
would want to know when both hardware and software amplification kicks
in. In PA this is represented clearly by the 'Base Volume' marker for
the case of "h/w amplification" and volumes >0dB/100% in the "software
amplification".

Not all PA UIs allow this >0dB/100% slider: gnome-volume-control being
one that does and something that as you know, I made an effort to
standardise recently. I've not (yet) actually made much in the way of
progress on this due to various time constraints, but the principle of
what values to use is laid down.

FWIW, even within itself g-v-c is inconsistent. The full GUI allows up
to this 150%/~+11dB mark (it works in percentages which is ugly - I'll
be changing that), but the applet only goes up to 0dB which is rather
annoying. I'll try and fix that if noone beats me to it.

Col



-- 

Colin Guthrie
gmane(at)colin.guthr.ie
http://colin.guthr.ie/

Day Job:
  Tribalogic Limited [http://www.tribalogic.net/]
Open Source:
  Mandriva Linux Contributor [http://www.mandriva.com/]
  PulseAudio Hacker [http://www.pulseaudio.org/]
  Trac Hacker [http://trac.edgewall.org/]

Re: wrong decibel data?

[Raymond Yau]

2010/6/22 Colin Guthrie <gmane@colin.guthr.ie>

> 'Twas brillig, and Raymond Yau at 22/06/10 03:31 did gyre and gimble:
> > 2010/6/14 Colin Guthrie <gmane@colin.guthr.ie>
> >> I've made this change on my system and while previously my UI had no
> >> "Base Volume" displayed (because all my "h/w" (I include softvol in
> >> that) controls had their dB value >0.
> >>
> >
> > as your card has no h/w gain,  > 0dB , but the gain in softvol plugin is
> a
> > software gain (i.e. in the red region in PA "s volume scale
> >
> > how can base_volume display in gnome volume control (unamplified) ?
> >
> > BTW ,  -46.5dB to 0dB of softvol plugin is software atten ( not h/w atten
> )
>
> As I said above, anything that comes from alsa is considered h/w
> amplification for the purposes of PA's volume scale. It's not practical
> to differentiate them


Software gain is different from hardware gain.

Clipping due to software gain +12dB cannot compensated by -12dB by hardware
atten


.
>
> Does any card actually configure softvol, by default, to provide any
> gain, > 0dB for outputs? If so, then this is IMO a bad idea.
>


-51dB to 0dB is also software atten of softvol plugin

if PA did not want to differeniate hardware/software gain/atten
Just  set this softvol PCM to 0dB and use PA 's own software gain/atten or
don't use front device for HDA

Re: wrong decibel data?

[Colin Guthrie]

'Twas brillig, and Raymond Yau at 22/06/10 16:29 did gyre and gimble:
> 2010/6/22 Colin Guthrie <gmane@colin.guthr.ie>
>> As I said above, anything that comes from alsa is considered h/w
>> amplification for the purposes of PA's volume scale. It's not practical
>> to differentiate them
> 
> 
> Software gain is different from hardware gain.

Really? Wow, thanks for that :p

> Clipping due to software gain +12dB cannot compensated by -12dB by hardware
> atten

Another obvious statement, but not really anything to do with the
discussion.

This only becomes a problem if the softvol plugin is configured to go
>0dB. And if it is, then I've got to ask why....

You can configure all sorts of crazy and weird shit in alsa if you care
to, but it's totally impractical for something higher in the stack to
deal with all the nuances of the ultimate results of that configuration.
As far as anything further up the stack is concerned, if it's
represented in alsa, it's "hardware".

Just like softmodems, I don't care further up the stack whether or not
the functionality is implemented in firmware or software, I just care
that I have an interface to use a modem.

So, yes, of course you could configure softvol plugin to do nuts things
in ALSA if you're that way inclined. I don't think anyone who is not
trying to do weird things will do that, however, and I don't know of any
particular h/w that is setup in a weird way by default either.

>> Does any card actually configure softvol, by default, to provide any
>> gain, > 0dB for outputs? If so, then this is IMO a bad idea.
>>
> 
> -51dB to 0dB is also software atten of softvol plugin

So that's not > 0dB then is it?

Col



-- 

Colin Guthrie
gmane(at)colin.guthr.ie
http://colin.guthr.ie/

Day Job:
  Tribalogic Limited [http://www.tribalogic.net/]
Open Source:
  Mandriva Linux Contributor [http://www.mandriva.com/]
  PulseAudio Hacker [http://www.pulseaudio.org/]
  Trac Hacker [http://trac.edgewall.org/]

Re: wrong decibel data?

[Raymond Yau]

2010/6/23 Colin Guthrie <gmane@colin.guthr.ie>

> 'Twas brillig, and Raymond Yau at 22/06/10 16:29 did gyre and gimble:
> > 2010/6/22 Colin Guthrie <gmane@colin.guthr.ie>
> >> As I said above, anything that comes from alsa is considered h/w
> >> amplification for the purposes of PA's volume scale. It's not practical
> >> to differentiate them
> >
> >
> > Software gain is different from hardware gain.
>
> Really? Wow, thanks for that :p
>
> > Clipping due to software gain +12dB cannot compensated by -12dB by
> hardware
> > atten
>
> Another obvious statement, but not really anything to do with the
> discussion.
>
> This only becomes a problem if the softvol plugin is configured to go
> >0dB. And if it is, then I've got to ask why....
>
> You can configure all sorts of crazy and weird shit in alsa if you care
> to, but it's totally impractical for something higher in the stack to
> deal with all the nuances of the ultimate results of that configuration.
> As far as anything further up the stack is concerned, if it's
> represented in alsa, it's "hardware".
>
> Just like softmodems, I don't care further up the stack whether or not
> the functionality is implemented in firmware or software, I just care
> that I have an interface to use a modem.
>
> So, yes, of course you could configure softvol plugin to do nuts things
> in ALSA if you're that way inclined. I don't think anyone who is not
> trying to do weird things will do that, however, and I don't know of any
> particular h/w that is setup in a weird way by default either.
>
> >> Does any card actually configure softvol, by default, to provide any
> >> gain, > 0dB for outputs? If so, then this is IMO a bad idea.
> >>
> >
> > -51dB to 0dB is also software atten of softvol plugin
>
> So that's not > 0dB then is it?
>
> Col
>
>
>
>
As you have modified /usr/share/alsa/cards/HDA-Intel.conf

HDA-Intel.pcm.front.0 {
    @args [ CARD ]
    @args.CARD {
        type string
    }
    type softvol
    slave.pcm {
        type hw
        card $CARD
    }
    control {
        name "PCM Playback Volume"
        card $CARD
    }

you should notice that the front device has a softvol plugin with name "PCM
Playback Volume"

why PA server still insist to open front device for capturing ?

http://thread.gmane.org/gmane.linux.alsa.devel/67912/focus=68248

As Takashi had already mention that

"I agree that the capture from "front" PCM isn't considered as valid.
The "front", "rear", "center_lfe" definitions are rather for multi-channel
playbacks. The capture on these channels aren't useful in most cases."

you can perform an experiement

pcm.test {
    type softvol
    slave.pcm "hw:0,0"
    control {
        name "PA Playback Volume"
        card 0
    }

arecord -D test -f CD -d 10 -v test.wav

you will find "PA Playback Volume" in playback screen of alsamixer

I am not sure the softvol control created when PA open front device
for playback and capture is used for playback or capture

BTW , PA still using "front" to open CTL device too

Re: wrong decibel data?

[Colin Guthrie]

'Twas brillig, and Raymond Yau at 23/06/10 02:15 did gyre and gimble:
> why PA server still insist to open front device for capturing ?

Couldn't tell you. Perhaps Lennart is unaware that it is considered
invalid? I'll ask him.

> http://thread.gmane.org/gmane.linux.alsa.devel/67912/focus=68248
> 
> As Takashi had already mention that
> 
> "I agree that the capture from "front" PCM isn't considered as valid.
> The "front", "rear", "center_lfe" definitions are rather for multi-channel
> playbacks. The capture on these channels aren't useful in most cases."
> 
> you can perform an experiement
> 
> pcm.test {
>     type softvol
>     slave.pcm "hw:0,0"
>     control {
>         name "PA Playback Volume"
>         card 0
>     }
> 
> arecord -D test -f CD -d 10 -v test.wav
> 
> you will find "PA Playback Volume" in playback screen of alsamixer
> 
> I am not sure the softvol control created when PA open front device
> for playback and capture is used for playback or capture
> 
> BTW , PA still using "front" to open CTL device too

You'll have to ask Lennart directly about that bit.

I can't speak for him with regards to the full setup of the mixer path
probing stuff, so perhaps "front" should be edited out of this for
paths. It would likely be a fairly trivial change to configuration files
rather than any code changes.

Col

-- 

Colin Guthrie
gmane(at)colin.guthr.ie
http://colin.guthr.ie/

Day Job:
  Tribalogic Limited [http://www.tribalogic.net/]
Open Source:
  Mandriva Linux Contributor [http://www.mandriva.com/]
  PulseAudio Hacker [http://www.pulseaudio.org/]
  Trac Hacker [http://trac.edgewall.org/]

Re: wrong decibel data?

[Raymond Yau]

2010/6/23 Colin Guthrie <gmane@colin.guthr.ie>

> 'Twas brillig, and Raymond Yau at 23/06/10 02:15 did gyre and gimble:
> > why PA server still insist to open front device for capturing ?
>
> Couldn't tell you. Perhaps Lennart is unaware that it is considered
> invalid? I'll ask him.
>
> > http://thread.gmane.org/gmane.linux.alsa.devel/67912/focus=68248
> >
> > As Takashi had already mention that
> >
> > "I agree that the capture from "front" PCM isn't considered as valid.
> > The "front", "rear", "center_lfe" definitions are rather for
> multi-channel
> > playbacks. The capture on these channels aren't useful in most cases."
> >
> > you can perform an experiement
> >
> > pcm.test {
> >     type softvol
> >     slave.pcm "hw:0,0"
> >     control {
> >         name "PA Playback Volume"
> >         card 0
> >     }
> >
> > arecord -D test -f CD -d 10 -v test.wav
> >
> > you will find "PA Playback Volume" in playback screen of alsamixer
> >
> > I am not sure the softvol control created when PA open front device
> > for playback and capture is used for playback or capture
> >
> > BTW , PA still using "front" to open CTL device too
>
> You'll have to ask Lennart directly about that bit.
>

arecord -Dfront:0 -v -f cd | aplay -Dfront:0 -v -f cd

the softvol control was seem to used by arecord and aplay concurrently

Using front device for capture
1)  HDA, the problem is softvol plugin,
2)  emu10k1 , the problem is the hook controls in front playback device
which change the route of volume of the DSP hardware mixer
3)



>
> I can't speak for him with regards to the full setup of the mixer path
> probing stuff, so perhaps "front" should be edited out of this for
> paths. It would likely be a fairly trivial change to configuration files
> rather than any code changes.
>
> Col
>
>
Well , He had already hardcoded to use "hw:card_number" in dbverify.c , it
is only PA server still try to open mixer device "front"

The tooltip of gnome volume control in Fedora 13 clearly indiacte that 100%
is 0dB for intel8x0 which use ac97 codec  when you put the cursor on top of
the speaker icon at the system bar

https://bugzilla.gnome.org/show_bug.cgi?id=618551

according to gnome-media-developer  Bastien Nocera

In any case, we only display what PulseAudio tells us, so you should poke the
PulseAudio devs about this.

This is the code in question:
gdouble
gvc_mixer_stream_get_decibel (GvcMixerStream *stream)
{
        g_return_val_if_fail (GVC_IS_MIXER_STREAM (stream), 0);

        return pa_sw_volume_to_dB(
                        (pa_volume_t)
gvc_channel_map_get_volume(stream->priv->channel_map)[VOLUME]);
}



if you don't have any ac97 sound card , you can user virtualbox which has an
emulated intel8x0 sound card

Re: wrong decibel data?

[Colin Guthrie]

'Twas brillig, and Raymond Yau at 28/06/10 02:47 did gyre and gimble:
> 2010/6/23 Colin Guthrie <gmane@colin.guthr.ie>
>> I can't speak for him with regards to the full setup of the mixer path
>> probing stuff, so perhaps "front" should be edited out of this for
>> paths. It would likely be a fairly trivial change to configuration files
>> rather than any code changes.
>>
>>
> Well , He had already hardcoded to use "hw:card_number" in dbverify.c , it
> is only PA server still try to open mixer device "front"

Yeah dbverify is a very simple tool that doesn't really warrant an
advanced interface. There may be other reasons to use hw: directly too.


> The tooltip of gnome volume control in Fedora 13 clearly indiacte that 100%
> is 0dB for intel8x0 which use ac97 codec  when you put the cursor on top of
> the speaker icon at the system bar
> 
> https://bugzilla.gnome.org/show_bug.cgi?id=618551
> 
> according to gnome-media-developer  Bastien Nocera
> 
> In any case, we only display what PulseAudio tells us, so you should poke the
> PulseAudio devs about this.
> 
> This is the code in question:
> gdouble
> gvc_mixer_stream_get_decibel (GvcMixerStream *stream)
> {
>         g_return_val_if_fail (GVC_IS_MIXER_STREAM (stream), 0);
> 
>         return pa_sw_volume_to_dB(
>                         (pa_volume_t)
> gvc_channel_map_get_volume(stream->priv->channel_map)[VOLUME]);
> }
> 
> 
> 
> if you don't have any ac97 sound card , you can user virtualbox which has an
> emulated intel8x0 sound card

Dude, we've talked about this already. The PA server applies no software
amplification and has set the controls provided by ALSA to their
maximum, which we call 0dB.

This approach allows us to fit in with a slider system that has 0dB at
it's top end and we can also show the ALSA 0dB point as the "Base
Volume" in our UIs.

Now you could argue (which you obviously are) that the dB of the
underlying card should still be exposed in the purest sense, but that
does make for more complicated calculations and confusing repercussions
elsewhere, including the mapping from percentage to dB not being
consistent (e.g. sometimes 100% is 0dB, sometimes it's 12dB etc.).

You could also argue that any controls that go >0dB should just be
ignored by PA - i.e. we never control the mixers in a way that would go
over 0dB unless we're going above our 100%. That may be a more valid
approach, but it needs a standard way to represent volumes >100% which
would then fall into the zone of influence of your other current
bugbear: the range checking thing in alsa-lib/alsamixer. Ultimately this
boils down to the fact that the 0dB point in some alsa mixers is located
at a random percentage < 100%. This is fundamentally the problem we are
trying to combat and make it clear to the user where that point is.


So as I've said before, the current approach is IMO the best way to fit
in with the semantics currently available. If you change those semantics
and provide mechanisms to represent a tri-range (min, 100%/0dB, max)
then we could probably adjust to fit in with those semantics. But none
currently exist. So we'll stick with how it's done now for the time being.


Col
-- 

Colin Guthrie
gmane(at)colin.guthr.ie
http://colin.guthr.ie/

Day Job:
  Tribalogic Limited [http://www.tribalogic.net/]
Open Source:
  Mandriva Linux Contributor [http://www.mandriva.com/]
  PulseAudio Hacker [http://www.pulseaudio.org/]
  Trac Hacker [http://trac.edgewall.org/]

Re: wrong decibel data?

[Raymond Yau]

2010/6/28 Colin Guthrie <gmane@colin.guthr.ie>

> 'Twas brillig, and Raymond Yau at 28/06/10 02:47 did gyre and gimble:
>
> > The tooltip of gnome volume control in Fedora 13 clearly indiacte that
> 100%
> > is 0dB for intel8x0 which use ac97 codec  when you put the cursor on top
> of
> > the speaker icon at the system bar
> >
> > https://bugzilla.gnome.org/show_bug.cgi?id=618551
> >
> > according to gnome-media-developer  Bastien Nocera
> >
> > In any case, we only display what PulseAudio tells us, so you should poke
> the
> > PulseAudio devs about this.
> >
> > This is the code in question:
> > gdouble
> > gvc_mixer_stream_get_decibel (GvcMixerStream *stream)
> > {
> >         g_return_val_if_fail (GVC_IS_MIXER_STREAM (stream), 0);
> >
> >         return pa_sw_volume_to_dB(
> >                         (pa_volume_t)
> > gvc_channel_map_get_volume(stream->priv->channel_map)[VOLUME]);
> > }
> >
> >
> >
> > if you don't have any ac97 sound card , you can user virtualbox which has
> an
> > emulated intel8x0 sound card
>
> Dude, we've talked about this already. The PA server applies no software
> amplification and has set the controls provided by ALSA to their
> maximum, which we call 0dB.
>

This is completely wrong when you rename ALSA driver's max_dB to PA 's 0dB

Most applications does not provide any software gain/atten too, when the
ALSA drivers 's volume control is at Max dB (e.g. +12dB for ac97 PCM ) , it
is still 12dB as long as you changed the hardware volume control to 12dB,
it can be 0dB only if PA can provide -12dB (software atten)


the dB value of playback/capture in  gnome-volume-control and "alsamixer
-c0" should be the same for any sound card.

a simple test case is using aplay which did not provide any software
gain/atten

For the ac97 sound card with PCM -34.5dB to +12 dB ( assume that you have
already set the ac97 "Master" to 0dB )

1) aplay -Dhw:0,0 any.wav with "alsamixer -c0 " "Master" at 0.dB and "PCM"
at 12.dB

2) aplay -Dpulse any.wav with gnome volume control at max(0dB) 100%

In both case, "alsamixer -c0" tell you that it is +12dB in playback gain
but gnome-volume-control tell you that it is 0dB gain

if you use  decibel meter, voltmeter or oscilloscope to measure the output
signal
those meters should tell you those two signals are at same level

PA should not rename the dB value on volume scale just because the
computation of the dB value is complicated since user cannot rely on this
shifted dB scale to perform distortion free line level playback/recording
(this require to set both playback and capture volume control at 0dB)

In simple word,  PA is actually provide a wrong dB value to gnome volume
control for those driver which the max_dB is > 0

PA 's renamin max_dB to 0dB  is not limited to those drivers with max_dB > 0


it also affect those driver with max_dB < 0

http://thread.gmane.org/gmane.linux.alsa.devel/73114/focus=73125

Original range (127): -63.5 .. 0 dB
Limited range  (100): -63.5 .. -13.5 dB


http://git.alsa-project.org/?p=alsa-lib.git;a=commitdiff;h=30ad5ed04017f7e77b25cf40f18c26396903cd23;hp=19892334499ed21ed4dc30084ad8700253f9cb2f

http://git.0pointer.de/?p=pulseaudio.git;a=commit;h=7bc5cd78454bee5990d29a0ba04482a505a08346

PA cannot rename -13.5dB as 0dB after the driver limited the max_dB to
-13.5dB by reducing the number of steps of the control




>> No. -inf dB is -inf dB. If the card does not go down to -inf dB, PA will
extend the range in >> software. If a card only goes down to -12dB minimum
the range -12dB -> -infdB will be >> done in software. Lennart explained
this very clearly to you already.

It is unlikely for PA just provide software atten for this range (-inf dB to
min dB) since PA volume control has 65536 steps

As most sound cards only provide 32 to 256 steps by hardware (e.g.
DAC/ADC/codec)

The value returned by get_playback_dB() may different from set_playback_dB()
when the driver cannot set the dB value between two hardware steps  (e.g.
ac97 codec  1.5dB per step , so you cannot set playback volume to -0.1dB by
using hw volume control alone,   this is still true even when some HDA codec
support 0.25 dB per step )

This mean that if PA want to set playback volume to -0.1dB,
it have to set the hardware control to 0dB and peform software atten -0.1dB
or set the hardware control to -1.5dB and perform software gain +1.4dB

Can you explain whether PA perform software atten in the above case ?


PA seem did not care about the difference of variable "f" before and after
calling snd_mixer_selem_set_playback_dB() and
snd_mixer_selem_get_playback_dB() to perform necessary software gain/atten
(difference of f )


static int element_set_volume(pa_alsa_element *e, snd_mixer_t *m, const
pa_channel_map *cm, pa_cvolume *v) {


        if (e->has_dB) {
            long value = to_alsa_dB(f);

            if (e->direction == PA_ALSA_DIRECTION_OUTPUT) {
                /* If we call set_play_volume() without checking first
                 * if the channel is available, ALSA behaves ver
                 * strangely and doesn't fail the call */
                if (snd_mixer_selem_has_playback_channel(me, c)) {
                    if ((r = snd_mixer_selem_set_playback_dB(me, c, value,
+1)) >= 0)
                        r = snd_mixer_selem_get_playback_dB(me, c, &value);
                } else
                    r = -1;
            } else {
                if (snd_mixer_selem_has_capture_channel(me, c)) {
                    if ((r = snd_mixer_selem_set_capture_dB(me, c, value,
+1)) >= 0)
                        r = snd_mixer_selem_get_capture_dB(me, c, &value);
                } else
                    r = -1;
            }

            if (r < 0)
                continue;

#ifdef HAVE_VALGRIND_MEMCHECK_H
            VALGRIND_MAKE_MEM_DEFINED(&value, sizeof(value));
#endif

            f = from_alsa_dB(value);

        } else {

Re: wrong decibel data?

[Lennart Poettering]

On Mon, 14.06.10 11:46, James Courtier-Dutton (james.dutton@gmail.com) wrote:

> One has very different problems with capture than one does with playback.
> With capture it is important to identify which are analog controls
> (applied to the analog part of the circuit) and which are digital
> controls (applied to the digital part of the circuit)
> So, for capture one might wish to adjust the analog control so that
> the signal going into the ADC is a suitable level, but once the signal
> is digital, one should really not adjust it further, and just record
> what you have.
> If one was to combine these two capture controls in one PA control, it
> would just be wrong.

Well, we apply volumes going from the "outer end" to the "inner end" of
the pipeline. That should have the affect that the biggest part is done
on the analog elements (i.e. "outer") and only minimal attenuation on the
digital elements (i.e. "inner").

In short: this should be covered nicely by PA already, at least if the
ALSA mixer used somewhat standard names for things.

> I think there is some indication with the name of the control. It
> sometimes has "Analog" or "Digital" attached to it.
> I think this would be better if alsa reported the "Analog" or
> "Digital" as meta data, like the dB Scales.
> PA could then make more informed decisions for capture. I.e. only
> display the "Analog" controls, and hide the digital ones, setting them
> to 0dB.

I think it is advisable to to attenuation in hw if possible, simply to
minimize CPU load. Hence we try to do the biggest volume adjustment in
analog hw, and everything we cannot apply there we do in digital hw, and
finally everything else in sw.

Lennart

-- 
Lennart Poettering                        Red Hat, Inc.
lennart [at] poettering [dot] net
http://0pointer.net/lennart/           GnuPG 0x1A015CC4

Re: wrong decibel data?

[Lennart Poettering]

On Mon, 14.06.10 11:22, Colin Guthrie (gmane@colin.guthr.ie) wrote:

> 
> 'Twas brillig, and James Courtier-Dutton at 14/06/10 09:56 did gyre and
> gimble:
> > On 14 June 2010 09:33, Colin Guthrie <gmane@colin.guthr.ie> wrote:
> >> 'Twas brillig, and Raymond Yau at 14/06/10 01:25 did gyre and gimble:
> >>> if your sound card have ac97 codec ., you can use audacity to record the
> >>> output from hw:0,0 and you will see clipping occur when you set "PCM" volume
> >>> above 0dB
> >>
> >> So the standard response is "don't do that then" :)
> >>
> >> That's why the base volume is shown to the user via GUIs so that they
> >> can gauge the best point on the slider to use. Currently there is no
> >> indication with alsa sliders at which point the 0dB "sweet spot" lies.
> >>
> > 
> > What do you mean.
> > If you use "alsamixer", dB values are shown so it is easy to find the
> > 0dB "sweet spot".
> > I think it is pulse audio that hides this information when it combines
> > two alsa mixer controls into one pulseaudio control.
> 
> But it doesn't hide it. It's shown very clearly in the volume control
> GUIs as the Base Volume.

Let me also stress that "dB" is not at all understandable to most
people. It is a very technical unit, and showing 0dB in the UI just like
that won't be very helpful for most people.

That's why we thought about this, and are recommending a color coded
slider to be exposed in the UI which encodes the range information in a
sane way that is intuitively understandable by users. In green, in
yellow and in red. (meaning hw attenuated, hw amplified, sw amplified
ranges)

http://pulseaudio.org/wiki/WritingVolumeControlUIs#Colouredvolumesliders

Lennart

-- 
Lennart Poettering                        Red Hat, Inc.
lennart [at] poettering [dot] net
http://0pointer.net/lennart/           GnuPG 0x1A015CC4

Re: wrong decibel data?

[Raymond Yau]

2010/6/15 Lennart Poettering <mznyfn@0pointer.de>

> On Mon, 14.06.10 11:22, Colin Guthrie (gmane@colin.guthr.ie) wrote:
>
> >
> > 'Twas brillig, and James Courtier-Dutton at 14/06/10 09:56 did gyre and
> > gimble:
> > > On 14 June 2010 09:33, Colin Guthrie <gmane@colin.guthr.ie> wrote:
> > >> 'Twas brillig, and Raymond Yau at 14/06/10 01:25 did gyre and gimble:
> > >>> if your sound card have ac97 codec ., you can use audacity to record
> the
> > >>> output from hw:0,0 and you will see clipping occur when you set "PCM"
> volume
> > >>> above 0dB
> > >>
> > >> So the standard response is "don't do that then" :)
> > >>
> > >> That's why the base volume is shown to the user via GUIs so that they
> > >> can gauge the best point on the slider to use. Currently there is no
> > >> indication with alsa sliders at which point the 0dB "sweet spot" lies.
> > >>
> > >
> > > What do you mean.
> > > If you use "alsamixer", dB values are shown so it is easy to find the
> > > 0dB "sweet spot".
> > > I think it is pulse audio that hides this information when it combines
> > > two alsa mixer controls into one pulseaudio control.
> >
> > But it doesn't hide it. It's shown very clearly in the volume control
> > GUIs as the Base Volume.
>
> Let me also stress that "dB" is not at all understandable to most
> people. It is a very technical unit, and showing 0dB in the UI just like
> that won't be very helpful for most people.
>
> That's why we thought about this, and are recommending a color coded
> slider to be exposed in the UI which encodes the range information in a
> sane way that is intuitively understandable by users. In green, in
> yellow and in red. (meaning hw attenuated, hw amplified, sw amplified
> ranges)
>
> http://pulseaudio.org/wiki/WritingVolumeControlUIs#Colouredvolumesliders
>
> Lennart
>
>
Clipping may occur at recording if there is gain above hardware 0dB point  ,
the region should be coloured as red ,

What is the meaning of  the yellow region ?

For HDA base volume seem to be at the same point as the norm volume since
max_db of playback is 0dB

+12dB(400%) is even larger than the software gain 150% of PA

Re: wrong decibel data?

[Colin Guthrie]

'Twas brillig, and Raymond Yau at 15/06/10 00:43 did gyre and gimble:
> +12dB(400%) is even larger than the software gain 150% of PA

Software gain of PA 150% = ~+11dB, so not as different as you imply.

I've explained the cubic mapping already, so please don't use arbitrary,
differently calculated percentages when comparing things. It's like
comparing apples to oranges.

Col

-- 

Colin Guthrie
gmane(at)colin.guthr.ie
http://colin.guthr.ie/

Day Job:
  Tribalogic Limited [http://www.tribalogic.net/]
Open Source:
  Mandriva Linux Contributor [http://www.mandriva.com/]
  PulseAudio Hacker [http://www.pulseaudio.org/]
  Trac Hacker [http://trac.edgewall.org/]

Re: wrong decibel data?

[Lennart Poettering]

On Mon, 14.06.10 09:56, James Courtier-Dutton (james.dutton@gmail.com) wrote:

> 
> On 14 June 2010 09:33, Colin Guthrie <gmane@colin.guthr.ie> wrote:
> > 'Twas brillig, and Raymond Yau at 14/06/10 01:25 did gyre and gimble:
> >> if your sound card have ac97 codec ., you can use audacity to record the
> >> output from hw:0,0 and you will see clipping occur when you set "PCM" volume
> >> above 0dB
> >
> > So the standard response is "don't do that then" :)
> >
> > That's why the base volume is shown to the user via GUIs so that they
> > can gauge the best point on the slider to use. Currently there is no
> > indication with alsa sliders at which point the 0dB "sweet spot" lies.
> >
> 
> What do you mean.
> If you use "alsamixer", dB values are shown so it is easy to find the
> 0dB "sweet spot".

Whether dB is shown or not has nothing to do with PA. Some UIs show it,
others don't. And most UIs do show the "base volume" too, which refers
to the ALSA 0dB "comfort" point. There is nothing lost here. We just
simplified things a littl and made the volume range infinite, as well as
we unified what things look like on various hardware.

> I think it is pulse audio that hides this information when it combines
> two alsa mixer controls into one pulseaudio control.

No, we don't. The final "base volume" will be put where all sliders that
are merged are at 0dB.

Lennart

-- 
Lennart Poettering                        Red Hat, Inc.
lennart [at] poettering [dot] net
http://0pointer.net/lennart/           GnuPG 0x1A015CC4

Re: wrong decibel data?

[Raymond Yau]

2010/6/7 Colin Guthrie <gmane@colin.guthr.ie>

> 'Twas brillig, and Raymond Yau at 06/06/10 01:12 did gyre and gimble:
> > 2010/5/28 Colin Guthrie <gmane@colin.guthr.ie>
> >
> >> 'Twas brillig, and Colin Guthrie at 27/05/10 15:43 did gyre and gimble:
> >>> PA should play nice with the softvol plugin so I don't think this is
> the
> >>> bit that is at fault.
> >>>
> >>> I strongly suspect that the reason has already been correctly
> identified
> >>> a while ago, which is that this card considers -48dB silent where as PA
> >>> assumes this level is -200dB. I believe it was Raymond who pointed out
> >>> the -48dB level in the HDA spec before on this list.
> >>>
> >>
> >
> > if  floating point 0.0 is -inf dB , and 1.0 is 0dB ,
> >
> > 0.5 is -6dB , 0.25 is -12 dB , 0.125 is -24dB and 0.0625 is -48dB
>
> This is just a pure mapping from dB->linear, but as this linear mapping
> is generally not "natural" there are several different approaches to
> presenting this to users. In PA, a cubic mapping is used on top of this
> basic conversion, to map to the percentage scale (0.0 to 1.0 if you like).
>
> So I'm not sure what point you're making by providing these numbers. Can
> you explain?
>
> > how can PA master volume control at 10~15% equivalent to HDA 's -48dB ?
>
> Not sure what you mean here, but I suspect it's the cubic mapping that
> is confusing you.
>

>> FWIW, I've got the same/similar h/w with a cutoff at 14% in PA as you
have.

According to your email , you mention than your hardware has a cutoff at 14%
in PA

the mixer application programmer  can select any kind of scaling in the
slider

do you mean 14% in pavucontrol or Master volume control of "pulse" device ?

but pavucontrol did not provide any dB value at any point

Re: wrong decibel data?

[Colin Guthrie]

'Twas brillig, and Raymond Yau at 08/06/10 05:01 did gyre and gimble:
> According to your email , you mention than your hardware has a cutoff at 14%
> in PA

It's actually closer to about 16%... but in that area yes.

> the mixer application programmer  can select any kind of scaling in the
> slider

Indeed, but pulseaudio exposes a pa_volume_t which is basically a linear
scale. This upper bound is exposed via a constant: PA_VOLUME_MAX

pulse/volume.h:113:#define PA_VOLUME_MAX ((pa_volume_t) UINT32_MAX-1)

And the "100%" aka 0dB point is represented by another constant,
PA_VOLUME_NORM:

pulse/volume.h:107:#define PA_VOLUME_NORM ((pa_volume_t) 0x10000U)

The scale between 0 and PA_VOLUME_NORM is linear via pa_volume_t but
this is mapped via the cubic mapping internally when controling the
volume of the alsa device.

So volume control GUIs and the ALSA plugin use this scale in a linear
fashon to represent the rand 0% (pa_volume_t = 0) to 100% (pa_volume_t =
PA_VOLUME_NORM).

> do you mean 14% in pavucontrol or Master volume control of "pulse" device ?

Both. They both use the range 0 ... PA_VOLUME_NORM to represent 0% -
100% and thus they correspond with each other fine[1]

> but pavucontrol did not provide any dB value at any point

It doesn't display the dB value, but perhaps it should? It's more of a
developer tool anyway (tho' used by a lot of users until desktop
provided tools catch up) so exposing dB here would make sense IMO. I'll
cook up a patch.



Col


1. Note that this may change at some point in the future as we define
how much to show the user as an "overdrive" of volume. As you may know,
gnome-volume-control allows a setting up to 150%, but this isn't
refelected in the alsa plugin. Ultimately we decided to define a new
constant "PA_VOLUME_OVERDRIVE" (or similar, I can't remember the exact
name) and set this to be +11dB (via the current cubic mapping function).
This approximates to the 150% offered in g-v-c.

-- 

Colin Guthrie
gmane(at)colin.guthr.ie
http://colin.guthr.ie/

Day Job:
  Tribalogic Limited [http://www.tribalogic.net/]
Open Source:
  Mandriva Linux Contributor [http://www.mandriva.com/]
  PulseAudio Hacker [http://www.pulseaudio.org/]
  Trac Hacker [http://trac.edgewall.org/]

Re: wrong decibel data?

[Raymond Yau]

2010/5/27 Colin Guthrie <gmane@colin.guthr.ie>

> 'Twas brillig, and Clemens Ladisch at 27/05/10 14:48 did gyre and gimble:
> > Colin Guthrie wrote:
> >> state.Intel {
> >>      control.1 {
> >>              comment.access 'read write'
> >>              comment.type INTEGER
> >>              comment.count 2
> >>              comment.range '0 - 31'
> >>              comment.dbmin -4650
> >>              comment.dbmax 0
> >>              iface MIXER
> >>              name 'Master Playback Volume'
> >>              value.0 30
> >>              value.1 30
> >>      }
> >
> > This is the hardware volume control.
> >
> >>      control.11 {
> >>              comment.access 'read write user'
> >>              comment.type INTEGER
> >>              comment.count 2
> >>              comment.range '0 - 255'
> >>              comment.tlv '0000000100000008ffffec1400000014'
> >>              comment.dbmin -5100
> >>              comment.dbmax 0
> >>              iface MIXER
> >>              name 'PCM Playback Volume'
> >>              value.0 253
> >>              value.1 253
> >>      }
> >
> > This is the emulated software volume control that is created by the
> > softvol plugin.  This control gets recreated by "alsactl restore" even
> > when the plugin is not running.
> >
> > Might it be possible that PA is trying to use this, but that it doesn't
> > have any effect because PA is using PCM device hw:0?  (Try unloading
> > snd-hda-intel and then deleting that entry from /etc/asound.state.)
>
> PA should play nice with the softvol plugin so I don't think this is the
> bit that is at fault.
>

For AC97 codec

PCM -34.5dB to +12 dB
Master -46.5dB to 0dB

The total dB range is -81dB to +12dB

For HDA

No idea how to know the master volume control is a virtual master volume
control

The softvol plugin -51dB to -0dB is software atten of the input digital
signal before pass to the HDA sound card

PA seem has its own software atten/gain and mixing of the digital signal
from the PA clients



>
> I strongly suspect that the reason has already been correctly identified
> a while ago, which is that this card considers -48dB silent where as PA
> assumes this level is -200dB. I believe it was Raymond who pointed out
> the -48dB level in the HDA spec before on this list.
>

Different HDA codecs have different dB range according to the HDA spec


>
> I'm not sure of the internals, but things do indeed go silent when the
> volume reaches the magic -48dB mark (which is around the 14% mark with
> the current cubic mapping).
>
> I suspect that if I were to define infinity to be 48.0 in PA, everything
> would work nicely.
>

I have doubt since I can still hear sound when using ac97 codec of my au8830
below -48dB and using  baudline (require OSS emulation) also indicate that
there is signal recorded by using loopback of ac97 codec


>
> What I think is then ultimately needed is a way to ensure that everyone
> sings from the same hymn sheet regarding the real world value of -inf dB.
>
>
> I was hoping Lennart would have commented on this thread by now, so I'll
> try and prod him to get some proper input as I'm very much flailing
> around wildly in the dark!
>
>
> Col
>
>

Re: wrong decibel data?

[Raymond Yau]

2010/5/27 Clemens Ladisch <clemens@ladisch.de>

> Colin Guthrie wrote:
> > state.Intel {
> >       control.1 {
> >               comment.access 'read write'
> >               comment.type INTEGER
> >               comment.count 2
> >               comment.range '0 - 31'
> >               comment.dbmin -4650
> >               comment.dbmax 0
> >               iface MIXER
> >               name 'Master Playback Volume'
> >               value.0 30
> >               value.1 30
> >       }
>
> This is the hardware volume control.
>
> >       control.11 {
> >               comment.access 'read write user'
> >               comment.type INTEGER
> >               comment.count 2
> >               comment.range '0 - 255'
> >               comment.tlv '0000000100000008ffffec1400000014'
> >               comment.dbmin -5100
> >               comment.dbmax 0
> >               iface MIXER
> >               name 'PCM Playback Volume'
> >               value.0 253
> >               value.1 253
> >       }
>
> This is the emulated software volume control that is created by the
> softvol plugin.  This control gets recreated by "alsactl restore" even
> when the plugin is not running.
>
> Might it be possible that PA is trying to use this, but that it doesn't
> have any effect because PA is using PCM device hw:0?  (Try unloading
> snd-hda-intel and then deleting that entry from /etc/asound.state.)
>
>
> Regards,
> Clemens
>

if I make a customised device "test" with softvol plugin with name "P
Playback Volume" in .asounrd

aplay -D test.wav

alsamixer -c0 show the "P" control and volume change as expected

Then change the name from "P Playback Volume" to "Q Playack Volume" and play
audio through device "test" again

alsamixer -c0 show both "P" and "Q" controls

"P" control can go up/down with no effect of course

but "Q" stayed at 100%

Is this a bug ?

Re: wrong decibel data?

[Raymond Yau]

2010/4/3 Nicolo' Chieffo <nicolo.chieffo@gmail.com>

> Hello, I have a laptop with an intel hda audio chipset (STAC92xx)
>
> My problem is that when I put pulseaudio volume to 14% (or less) I
> can't hear any audio coming from speakers or headphones: in fact the
> alsamixer volume is set to 0%
>
> I already filed a bug to pulseaudio [1] in which the developers say
> that this is a common problem which is usually caused by the ALSA
> audio driver which reports a wrong decibel value.
>
> [1] http://pulseaudio.org/ticket/580
>
>
please provide the output of "amixer -D pulse" when you set the volume
control of alsamixer to (-3dB , -6dB , -12 dB , -24 dB and -48dB ) using
"alsamixer -c 0"

Re: wrong decibel data?

[Nicolo' Chieffo]

I've already attached my alsa-info.log in my first post.

* alsamixer -48 dB (0%)
Simple mixer control 'Master',0
  Capabilities: pvolume pswitch pswitch-joined penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 65536
  Mono:
  Front Left: Playback 10151 [15%] [on]
  Front Right: Playback 10151 [15%] [on]

* alsamixer -24 dB (50%)
Simple mixer control 'Master',0
  Capabilities: pvolume pswitch pswitch-joined penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 65536
  Mono:
  Front Left: Playback 25693 [39%] [on]
  Front Right: Playback 25693 [39%] [on]

* alsamixer -12 dB (75%):
Simple mixer control 'Master',0
  Capabilities: pvolume pswitch pswitch-joined penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 65536
  Mono:
  Front Left: Playback 40409 [62%] [on]
  Front Right: Playback 40409 [62%] [on]

* alsamixer -6 dB (88%):
Simple mixer control 'Master',0
  Capabilities: pvolume pswitch pswitch-joined penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 65536
  Mono:
  Front Left: Playback 50872 [78%] [on]
  Front Right: Playback 50872 [78%] [on]

* alsamixer -3 dB (94%):
Simple mixer control 'Master',0
  Capabilities: pvolume pswitch pswitch-joined penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 65536
  Mono:
  Front Left: Playback 57079 [87%] [on]
  Front Right: Playback 57079 [87%] [on]

* alsamixer 0 dB (100%):
Simple mixer control 'Master',0
  Capabilities: pvolume pswitch pswitch-joined penum
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 65536
  Mono:
  Front Left: Playback 64044 [98%] [on]
  Front Right: Playback 64044 [98%] [on]

Re: wrong decibel data?

[Raymond Yau]

2010/4/6 Nicolo' Chieffo <nicolo.chieffo@gmail.com>

> I've already attached my alsa-info.log in my first post.
>
>
> * alsamixer -3 dB (94%):
> Simple mixer control 'Master',0
>  Capabilities: pvolume pswitch pswitch-joined penum
>  Playback channels: Front Left - Front Right
>  Limits: Playback 0 - 65536
>  Mono:
>  Front Left: Playback 57079 [87%] [on]
>  Front Right: Playback 57079 [87%] [on]
>
> * alsamixer 0 dB (100%):
> Simple mixer control 'Master',0
>  Capabilities: pvolume pswitch pswitch-joined penum
>  Playback channels: Front Left - Front Right
>  Limits: Playback 0 - 65536
>  Mono:
>  Front Left: Playback 64044 [98%] [on]
>  Front Right: Playback 64044 [98%] [on]
>

you should notice that 0dB of Pulseaudio volume control is not at the
maximum dB
i.e. (  floating point 1.0 is not at 100% any more , seem that floating
point 0 (-inf dB) is also not at 0% point too )

http://thread.gmane.org/gmane.linux.alsa.devel/70545/focus=70585

>>   As mentioned for the PA case I decided to shift 0dB to max
amplification in any case, which I think is a workable way to avoid this
problem.


how about "dbmeasure pulse" and "dbmeasure hw:0,0" ?

Re: wrong decibel data?

[Nicolo' Chieffo]

On Wed, Apr 7, 2010 at 2:34 AM, Raymond Yau <superquad.vortex2@gmail.com> wrote:
> how about "dbmeasure pulse" and "dbmeasure hw:0,0" ?

I don't have a line out and in plug, only headphones and mic. Does it
work? At which volume should I put the mic?

Re: wrong decibel data?

[Nicolo' Chieffo]

This is what I'm doing:
- got a jack-jack audio cable
- connected the headphone to the mic
- open alsamixer and put everything to 0 dB
- verified that the pass-throw works by using the sound recorder

./dbmeasure plughw:0 dbmeasure_hw.log

exits with this error:

Iteration 1, level is 0 (-inf dB).
Volume level measured (0) was too high or too low.
Please adjust mixer so that initial level is between 0.8 and 0.97.
Test run canceled.


I've also tried to put the mic level at 100% (22.50 dB) but the output
is the same.

Re: wrong decibel data?

[Raymond Yau]

2010/4/7 Nicolo' Chieffo <nicolo.chieffo@gmail.com>

> This is what I'm doing:
> - got a jack-jack audio cable
> - connected the headphone to the mic
> - open alsamixer and put everything to 0 dB
> - verified that the pass-throw works by using the sound recorder
>
> ./dbmeasure plughw:0 dbmeasure_hw.log
>
> exits with this error:
>

Have you ask  the author of "dbmeasure" ?


>
> Iteration 1, level is 0 (-inf dB).
> Volume level measured (0) was too high or too low.
> Please adjust mixer so that initial level is between 0.8 and 0.97.
> Test run canceled.
>
>
> I've also tried to put the mic level at 100% (22.50 dB) but the output
> is the same.
>

your card seem to support 16/20/24bits

if you right shift your 16/24 bits digital audio by 16/24 bits , the digital
data are always zero
so any software atten on a 16/24bits sound card is still a finite negative
number dB instead of -inf  dB

that 's why the professional user prefer 32bits sound card

Re: wrong decibel data?

[Raymond Yau]

2010/4/6 Nicolo' Chieffo <nicolo.chieffo@gmail.com>

> I've already attached my alsa-info.log in my first post.
>
>
>
> * alsamixer 0 dB (100%):
> Simple mixer control 'Master',0
>  Capabilities: pvolume pswitch pswitch-joined penum
>  Playback channels: Front Left - Front Right
>  Limits: Playback 0 - 65536
>  Mono:
>  Front Left: Playback 64044 [98%] [on]
>  Front Right: Playback 64044 [98%] [on]
>


Most users only care about 0dB point since any software gain is the cause of
clipping (distortion) when the volume control provide software gain


http://en.wikipedia.org/wiki/Clipping_%28audio%29

In digital signal
processing<http://en.wikipedia.org/wiki/Digital_signal_processing>,
clipping occurs when the signal is restricted by the range of a chosen
representation. For example in a system using 16-bit
signed<http://en.wikipedia.org/wiki/Signed-digit_representation>integers,
32767 is the largest positive value that can be represented, and
if during processing the amplitude of the signal is doubled,
sample<http://en.wikipedia.org/wiki/Sample_%28signal%29>values of
32000 should become 64000, but instead they are truncated to the
maximum, 32767.

Re: wrong decibel data?

[Nicolo' Chieffo]

I'm sorry but I found out that I gave you wrong values of "amixer -D
pulse": I just discovered that when I lower the alsa master volume, in
some occasions the PCM volume goes to 99%, so now I will put it to
100% and check the pulse volume again.

0 dB    =  65536 [100%]
-3 dB   =  57520 [88%]
-6 dB   =  52057 [79%]
-12 dB =  41034 [63%]
-24 dB =  26090 [40%]
-48 dB = 10151 [15%]

So again, the only problem is that -48 dB IS mute, but it's not
declared as mute.
This is the real issue.

Re: wrong decibel data?

[Raymond Yau]

2010/4/8 Nicolo' Chieffo <nicolo.chieffo@gmail.com>

> I'm sorry but I found out that I gave you wrong values of "amixer -D
> pulse": I just discovered that when I lower the alsa master volume, in
> some occasions the PCM volume goes to 99%, so now I will put it to
> 100% and check the pulse volume again.
>
> 0 dB    =  65536 [100%]
> -3 dB   =  57520 [88%]
> -6 dB   =  52057 [79%]
> -12 dB =  41034 [63%]
> -24 dB =  26090 [40%]
> -48 dB = 10151 [15%]
>
> So again, the only problem is that -48 dB IS mute, but it's not
> declared as mute.
> This is the real issue.
>

Refer to HD audio specification

7.3.4.10 Amplifier Capabilities

Mute Capable (1 bit) reports if the respective amplifier is capable of
muting. Muting implies a –infinity gain (no sound passes), but the actual
performance is determined by the hardware.

http://thread.gmane.org/gmane.linux.alsa.devel/39262/focus=16100

>> Clemens Ladisch (developer) 2004-11-09 09:26

>>    The AD1981B's datasheet says that the maximum attenuation is 46.5 dB
(which
>> conforms to the AC'97 specification).
>> To mute, use Mute.


refer to your codec datasheet , the volume knob widget also affect the
volume if it is configured in direct mode


• DIRECT MODE
• In Direct mode, the Volume Knob widget directly controls the volume of all
of the DACs in the part. The volume in the Volume Knob widget acts as the
master volume and limits the maximum volume for each of the DAC amplifiers.
The amp gain for each of the DACs can also be adjusted using the DAC
amplifiers. However, the actual gain for an individual DAC will be the sum
of the Volume Knob volume and the DAC amplifier volume. For example, if the
DAC amplifier gain is set to 0x7F (0dB) and the Volume Knob volume is set to
0x3F (-48dB) the resulting gain would be -48dB. If the combination of gains
is less than -95.25dB
(the equivalent to a value of 0x0 for the DAC or Volume Knob volume
settings) then the actual gain will be -95.25dB. For example, if the Volume
Knob is set to 0x3F (-48dB) and the DAC amplifier volume is set to 0x1F
(-72dB) then the DAC volume will be set to -95.25dB.

Re: wrong decibel data?

[Nicolo' Chieffo]

On Fri, Apr 9, 2010 at 1:11 AM, Raymond Yau <superquad.vortex2@gmail.com> wrote:
> Mute Capable (1 bit) reports if the respective amplifier is capable of
> muting. Muting implies a –infinity gain (no sound passes), but the actual
> performance is determined by the hardware.
>
> http://thread.gmane.org/gmane.linux.alsa.devel/39262/focus=16100

I think you missed the point, I'll explain again.
I'm not complaining that at 0% the audio is not mute, as the bug you linked.

>>> Clemens Ladisch (developer) 2004-11-09 09:26
>>>    The AD1981B's datasheet says that the maximum attenuation is 46.5 dB
>>> (which conforms to the AC'97 specification).
>>> To mute, use Mute.

In fact my problem is that at 0% I get mute, but I shouldn't (as
reported by that developer), since the max attenuation is not -inf,
but -48 dB.
So if you prefer, you could see this issue from a different point of
view: at -48 dB the audio should be still audible, but it's not.

so you have to decide where the issue resides (but definitely not in pulseaudio)
a) it is ok that the audio is mute at 0% (in this case you have to
declare -inf dB)
b) the audio level is really -48 dB (in this case the volume shouldn't
be cut off completely, but simply quite low)
Which one do you prefer?
_______________________________________________
Alsa-devel mailing list
Alsa-devel@alsa-project.org
http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

Re: wrong decibel data?

[Raymond Yau]

2010/4/9 Nicolo' Chieffo <nicolo.chieffo@gmail.com>

> On Fri, Apr 9, 2010 at 1:11 AM, Raymond Yau <superquad.vortex2@gmail.com>
> wrote:
> > Mute Capable (1 bit) reports if the respective amplifier is capable of
> > muting. Muting implies a –infinity gain (no sound passes), but the actual
> > performance is determined by the hardware.
> >
> > http://thread.gmane.org/gmane.linux.alsa.devel/39262/focus=16100
>
> I think you missed the point, I'll explain again.
> I'm not complaining that at 0% the audio is not mute, as the bug you
> linked.
>
> >>> Clemens Ladisch (developer) 2004-11-09 09:26
> >>>    The AD1981B's datasheet says that the maximum attenuation is 46.5 dB
> >>> (which conforms to the AC'97 specification).
> >>> To mute, use Mute.
>
> In fact my problem is that at 0% I get mute, but I shouldn't (as
> reported by that developer), since the max attenuation is not -inf,
> but -48 dB.
> So if you prefer, you could see this issue from a different point of
> view: at -48 dB the audio should be still audible, but it's not.
>
> so you have to decide where the issue resides (but definitely not in
> pulseaudio)
> a) it is ok that the audio is mute at 0% (in this case you have to
> declare -inf dB)
> b) the audio level is really -48 dB (in this case the volume shouldn't
> be cut off completely, but simply quite low)
> Which one do you prefer?
>


All the alsa drvier are using information in the codec datasheet to
implement the dB scale

All the AC97 codec datasheet, DAC datahseet and HDA codec specifcation and
datasheet are regarded the mute switch as -inf dB , the minimum dB are a
finite negative number because the codec , DAC and HDA codec have finite
bits of resolution when convert digital signal to analog signal

For your case if you change DAC amplifier volume control to -inf dB , how
can you calculate the sum of Volume knob widget and  DAC ampilifer volume
control since the datasheet explicitly mention that the max atten is 95.25dB


may be pulseaudio did not add the volume knob widget in the calculation

I don't know how PA calculate the softvol plugin "PCM volume control" since
you mention that PA seem not maintain "PCM volume control" at 0dB

Re: wrong decibel data?

[Raymond Yau]

2010/4/9 Nicolo' Chieffo <nicolo.chieffo@gmail.com>

> On Fri, Apr 9, 2010 at 1:11 AM, Raymond Yau <superquad.vortex2@gmail.com>
> wrote:
> > Mute Capable (1 bit) reports if the respective amplifier is capable of
> > muting. Muting implies a –infinity gain (no sound passes), but the actual
> > performance is determined by the hardware.
> >
> > http://thread.gmane.org/gmane.linux.alsa.devel/39262/focus=16100
>
> I think you missed the point, I'll explain again.
> I'm not complaining that at 0% the audio is not mute, as the bug you
> linked.
>
> >>> Clemens Ladisch (developer) 2004-11-09 09:26
> >>>    The AD1981B's datasheet says that the maximum attenuation is 46.5 dB
> >>> (which conforms to the AC'97 specification).
> >>> To mute, use Mute.
>
> In fact my problem is that at 0% I get mute, but I shouldn't (as
> reported by that developer), since the max attenuation is not -inf,
> but -48 dB.
> So if you prefer, you could see this issue from a different point of
> view: at -48 dB the audio should be still audible, but it's not.
>
> so you have to decide where the issue resides (but definitely not in
> pulseaudio)
> a) it is ok that the audio is mute at 0% (in this case you have to
> declare -inf dB)
> b) the audio level is really -48 dB (in this case the volume shouldn't
> be cut off completely, but simply quite low)
> Which one do you prefer?
>

But you still not provide the result of dbmeasure of "pulse" , "hw:0" and
"plug:front"

Re: wrong decibel data?

[Nicolo' Chieffo]

As I've already told you, I'm not able to run dbmeasure because it
says it can't detect any audio.
I will investigate...

Re: wrong decibel data?

[Raymond Yau]

2010/4/9 Nicolo' Chieffo <nicolo.chieffo@gmail.com>

> As I've already told you, I'm not able to run dbmeasure because it
> says it can't detect any audio.
> I will investigate...
>

I cannot run dbmeasure on my au8830 or HDA sound card too.

you have to ask the author of dbmeasure.c

First glance at the source code , it seem the program expect the driver
support exactly one second buffer but neither au8830 or HDA driver can offer
a buffer size of exactly one second

May be just only work on his USB audio only

Re: wrong decibel data?

[Raymond Yau]

2010/4/9 Nicolo' Chieffo <nicolo.chieffo@gmail.com>

> On Fri, Apr 9, 2010 at 1:11 AM, Raymond Yau <superquad.vortex2@gmail.com>
> wrote:
> > Mute Capable (1 bit) reports if the respective amplifier is capable of
> > muting. Muting implies a –infinity gain (no sound passes), but the actual
> > performance is determined by the hardware.
> >
> > http://thread.gmane.org/gmane.linux.alsa.devel/39262/focus=16100
>
> I think you missed the point, I'll explain again.
> I'm not complaining that at 0% the audio is not mute, as the bug you
> linked.
>
> >>> Clemens Ladisch (developer) 2004-11-09 09:26
> >>>    The AD1981B's datasheet says that the maximum attenuation is 46.5 dB
> >>> (which conforms to the AC'97 specification).
> >>> To mute, use Mute.
>
> In fact my problem is that at 0% I get mute, but I shouldn't (as
> reported by that developer), since the max attenuation is not -inf,
> but -48 dB.
> So if you prefer, you could see this issue from a different point of
> view: at -48 dB the audio should be still audible, but it's not.
>
> so you have to decide where the issue resides (but definitely not in
> pulseaudio)
> a) it is ok that the audio is mute at 0% (in this case you have to
> declare -inf dB)
> b) the audio level is really -48 dB (in this case the volume shouldn't
> be cut off completely, but simply quite low)
> Which one do you prefer?
>

The DR of 16bit audio is 96dB

http://en.wikipedia.org/wiki/Dynamic_range

The 16-bit compact disc <http://en.wikipedia.org/wiki/Compact_disc> has a
theoretical dynamic range of about 96
dB[7]<http://en.wikipedia.org/wiki/Dynamic_range#cite_note-Fries2005-6>(or
about 98 dB for sinusoidal signals, per the formula
[6] <http://en.wikipedia.org/wiki/Dynamic_range#cite_note-seeber-5>).
Digital audio with 20-bit digitization is theoretically capable of 120 dB
dynamic range; similarly, 24-bit digital audio calculates to 144 dB dynamic
range.[4]<http://en.wikipedia.org/wiki/Dynamic_range#cite_note-HuberRunstein2005-3>All
digital audio recording and playback chains include input and output
converters and associated analog circuitry, significantly limiting practical
dynamic range. Observed 16-bit digital audio dynamic range is about 90 dB

The SNR of your codec is 95dB SNR

This mean that PA not using all the sum of the volume of the amplifer and
volume knob widget to calculate the dB value of your codec

Re: wrong decibel data?

[Nicolo' Chieffo]

You are saying that the range for my card is 96 dB, do you mean that
the audio specs are wrong?
I mean, at 0 the volume is not audible, so I'm wondering why the -inf
"switch register" is not activated (inside the chip)...

Re: wrong decibel data?

[Nicolo' Chieffo]

DMBEASURE

There's still no way to run the measurement with "pulse" because of
the previous error:Iteration 1, level is 0 (-inf dB).
   Volume level measured (0) was too high or too low.
   Please adjust mixer so that initial level is between 0.8 and 0.97.
   Test run canceled.
Are you sure that the test has sense using the pulse device?

"./dbmeasure plug:front plug_front.csv" works, but seems to never end:
   Iteration 3297, level is 0.0414072 (-27.6585 dB).
   Iteration 3298, level is 0.0414074 (-27.6584 dB).
   Iteration 3299, level is 0.0414064 (-27.6587 dB).
   Iteration 3300, level is 0.041406 (-27.6587 dB).
   .....


"./dbmeasure hw:0 hw.csv" fails with this message: "Cannot set sample
format: Invalid argument"

Re: wrong decibel data?

[Raymond Yau]

2010/4/10 Nicolo' Chieffo <nicolo.chieffo@gmail.com>

> DMBEASURE
>
> There's still no way to run the measurement with "pulse" because of
> the previous error:Iteration 1, level is 0 (-inf dB).
>    Volume level measured (0) was too high or too low.
>   Please adjust mixer so that initial level is between 0.8 and 0.97.
>   Test run canceled.
> Are you sure that the test has sense using the pulse device?
>
> "./dbmeasure plug:front plug_front.csv" works, but seems to never end:
>   Iteration 3297, level is 0.0414072 (-27.6585 dB).
>   Iteration 3298, level is 0.0414074 (-27.6584 dB).
>   Iteration 3299, level is 0.0414064 (-27.6587 dB).
>   Iteration 3300, level is 0.041406 (-27.6587 dB).
>   .....
>
>
> "./dbmeasure hw:0 hw.csv" fails with this message: "Cannot set sample
> format: Invalid argument"
>

The DR range of software ( 16 bits audio data ) need to be same as the range
of hardware volume control if you want  ths slider move at same scale

i.e. if PA only use one hardware volume control of  -48dB to 0 but the DR
range of 16 bits audio is 96dB , PA have to scale up the dB value by two

The best way is of course to sum the DB range of  all  hardware volume
controls in the audio path of the codec  to get a DB range of value close to
-96dB to 0 (i.e. need to use the value of the Volume Knob widget )

However the softvol plugin is in the software side , you have to verify the
step of the solfware volume control "PCM Playback volume" behave as it
expected

The easy way is to play a full amplitude of sine wave to "front" device and
record it using an analog loop cable with 0dB capture gain , try to lower
the PCM volume and observe the amplitude of the recorded sine wave is
decreased at the correct ratio

Re: wrong decibel data?

[Raymond Yau]

2010/4/10 Nicolo' Chieffo <nicolo.chieffo@gmail.com>

> DMBEASURE
>
> There's still no way to run the measurement with "pulse" because of
> the previous error:Iteration 1, level is 0 (-inf dB).
>    Volume level measured (0) was too high or too low.
>   Please adjust mixer so that initial level is between 0.8 and 0.97.
>   Test run canceled.
> Are you sure that the test has sense using the pulse device?
>
> "./dbmeasure plug:front plug_front.csv" works, but seems to never end:
>   Iteration 3297, level is 0.0414072 (-27.6585 dB).
>   Iteration 3298, level is 0.0414074 (-27.6584 dB).
>   Iteration 3299, level is 0.0414064 (-27.6587 dB).
>   Iteration 3300, level is 0.041406 (-27.6587 dB).
>   .....
>
>
> "./dbmeasure hw:0 hw.csv" fails with this message: "Cannot set sample
> format: Invalid argument"
>

The sum the dB range of these two hardware volume controls are already -96dB
to 0 as same as the DR of 16 bits digital audio data is 96dB

the point is that PA developer think the dB range of the pulseaudio master
volume control is -inf dB to 0dB

    control.18 {
        comment.access 'read write'
        comment.type INTEGER
        comment.count 1
        comment.range '0 - 64'
        comment.dbmin -4800
        comment.dbmax 0
        iface MIXER
        name 'Master Playback Volume'
        value 27
    }

    control.7 {
        comment.access 'read write'
        comment.type INTEGER
        comment.count 2
        comment.range '0 - 64'
        comment.dbmin -4800
        comment.dbmax 0
        iface MIXER
        name 'Headphone Playback Volume'
        value.0 64
        value.1 64
    }

Re: wrong decibel data?

[Nicolo' Chieffo]

On Sat, Apr 10, 2010 at 11:27 AM, Raymond Yau
<superquad.vortex2@gmail.com> wrote:
> the point is that PA developer think the dB range of the pulseaudio master
> volume control is -inf dB to 0dB

The range is 200 dB, since they use that value as -inf.
Can you suggest something to tell to the pulseaudio developers?

Re: wrong decibel data?

[Raymond Yau]

2010/4/10 Nicolo' Chieffo <nicolo.chieffo@gmail.com>

> On Sat, Apr 10, 2010 at 11:27 AM, Raymond Yau
> <superquad.vortex2@gmail.com> wrote:
> > the point is that PA developer think the dB range of the pulseaudio
> master
> > volume control is -inf dB to 0dB
>
> The range is 200 dB, since they use that value as -inf.
> Can you suggest something to tell to the pulseaudio developers?
>

As your reported bug is "pulseaudio master volume is scaled differently as
alsa master"

the answer is because "pulseaudio master volume" has a different dB ranage
from the alsa master volume control

Re: wrong decibel data?

[Raymond Yau]

2010/4/10 Nicolo' Chieffo <nicolo.chieffo@gmail.com>

> On Sat, Apr 10, 2010 at 11:27 AM, Raymond Yau
> <superquad.vortex2@gmail.com> wrote:
> > the point is that PA developer think the dB range of the pulseaudio
> master
> > volume control is -inf dB to 0dB
>
> The range is 200 dB, since they use that value as -inf.
> Can you suggest something to tell to the pulseaudio developers?
>

http://pulseaudio.org/ticket/580

"pulseaudio master volume is scaled differently as alsa master"

If I were pulseaudio developer , I will close your bugs as marked it as
invalid since the "Pulseaudio master volume control" is not as same as "ALSA
master volume control"

Re: wrong decibel data?

[Nicolo' Chieffo]

I'm a bit confused... it's impossible that both bugs are invalid

Re: wrong decibel data?

[Raymond Yau]

2010/4/11 Nicolo' Chieffo <nicolo.chieffo@gmail.com>

> I'm a bit confused... it's impossible that both bugs are invalid
>

if you still have any question in software atten , you should ask the expert
-. the openal developer since openal rely on dB to calculate the correct
sound level for distance *attenuation*

Re: wrong decibel data?

[Raymond Yau]

2010/4/9 Nicolo' Chieffo <nicolo.chieffo@gmail.com>

> You are saying that the range for my card is 96 dB, do you mean that
> the audio specs are wrong?
> I mean, at 0 the volume is not audible, so I'm wondering why the -inf
> "switch register" is not activated (inside the chip)...
>

Have you ever studied the 92HD71B datasheet ?

In the Figure 11. 92HD71B Widget Diagram , NID 10 ( DAC) -95.25 dB to 0dB
with 0.75 dB step

You have to ask alsa developer why the control did not return this range

Re: wrong decibel data?

[Raymond Yau]

2010/4/10 Werner Van Belle <werner@yellowcouch.org>

> Raymond Yau wrote:
> > The dynamic range of 16bit audio is 96dB
> > http://en.wikipedia.org/wiki/Dynamic_range
> >
> Yes about that. I always wondered how they come up with 96dB ?
>
> A perceived doubling of volume is normally assumed to be +3dB,
> (log_10(2)=0.3) which means that if you have 16 bit audio you have 16
> 'doublings', or in essence only 48 dB. Even worse, since the last bit is
> a sign bit, you essentially can only achieve a dynamic range of 45dB !
>
> Now, I know this is off topic, but I never heard any good explanation
> why CD audio is suddenly 45 dB ?  If anybody knows, please share your
> thoughts !
>
> Wkr,
>
>
>
http://en.wikipedia.org/wiki/Decibel

When referring to measurements of amplitude it is usual to consider the
ratio of the squares of *A*1 (measured amplitude) and *A*0 (reference
amplitude). This is because in most applications power is proportional to
the square of amplitude,

in electrical circuit , dissipated power is typically proportional to the
square of voltage or current

Re: wrong decibel data?

[Werner Van Belle]

[-- Attachment #1.1: Type: text/plain, Size: 1167 bytes --]

Raymond Yau wrote:
>> Yes about that. I always wondered how they come up with 96dB ?
>>
>> A perceived doubling of volume is normally assumed to be +3dB,
>> (log_10(2)=0.3) which means that if you have 16 bit audio you have 16
>> 'doublings', or in essence only 48 dB. Even worse, since the last bit is
>> a sign bit, you essentially can only achieve a dynamic range of 45dB !
>>
>> Now, I know this is off topic, but I never heard any good explanation
>> why CD audio is suddenly 45 dB ?  If anybody knows, please share your
>> thoughts !
>>     
> http://en.wikipedia.org/wiki/Decibel
>
> When referring to measurements of amplitude it is usual to consider the
> ratio of the squares of *A*1 (measured amplitude) and *A*0 (reference
> amplitude). This is because in most applications power is proportional to
> the square of amplitude,
>
> in electrical circuit , dissipated power is typically proportional to the
> square of voltage or current
>   
Okay, that would then only make for a factor two. Instead of 45dB one
gets 90dB, this is still not 96dB as ordinarily claimed ?

Wkr,

Werner,-

-- 
http://werner.yellowcouch.org/



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Re: wrong decibel data?

[Raymond Yau]

2010/4/10 Werner Van Belle <werner@yellowcouch.org>

> Raymond Yau wrote:
> >> Yes about that. I always wondered how they come up with 96dB ?
> >>
> >> A perceived doubling of volume is normally assumed to be +3dB,
> >> (log_10(2)=0.3) which means that if you have 16 bit audio you have 16
> >> 'doublings', or in essence only 48 dB. Even worse, since the last bit is
> >> a sign bit, you essentially can only achieve a dynamic range of 45dB !
> >>
> >> Now, I know this is off topic, but I never heard any good explanation
> >> why CD audio is suddenly 45 dB ?  If anybody knows, please share your
> >> thoughts !
> >>
> > http://en.wikipedia.org/wiki/Decibel
> >
> > When referring to measurements of amplitude it is usual to consider the
> > ratio of the squares of *A*1 (measured amplitude) and *A*0 (reference
> > amplitude). This is because in most applications power is proportional to
> > the square of amplitude,
> >
> > in electrical circuit , dissipated power is typically proportional to the
> > square of voltage or current
> >
> Okay, that would then only make for a factor two. Instead of 45dB one
> gets 90dB, this is still not 96dB as ordinarily claimed ?
>
> Wkr,
>
> Werner,-
>
>
You are assume all DAC can provide perfect conversion of the digital signal
to analog signal .

that why the professional use try to compare the SNR of the sound card

I just want to point out that the dB range of the hardware volume controls
of codec/DAC cannot be -inf dB to 0dB since they just convert fixed entropy
of digital information to analog signal

The pulseaudio server did not sum up the dB range of all hardware volume
controls of those amplifier /attenuator in the audio path and this is why
the sliders of the volume controls cannot move at same scale since the DR of
digital data and dB range of the hardware volume control are not the same ,
so this is not really a driver bug , it is due to PA only use onet of the
hardware volume controls in the audio path to calculate the dB

if PA really provide software gain according to, that mean when the hardware
slider stayed at 0dB and all the gain are provide the PA server by software

Re: wrong decibel data?

[Raymond Yau]

2010/4/10 Werner Van Belle <werner@yellowcouch.org>

> Raymond Yau wrote:
> > The dynamic range of 16bit audio is 96dB
> > http://en.wikipedia.org/wiki/Dynamic_range
> >
> Yes about that. I always wondered how they come up with 96dB ?
>
> A perceived doubling of volume is normally assumed to be +3dB,
> (log_10(2)=0.3) which means that if you have 16 bit audio you have 16
> 'doublings', or in essence only 48 dB. Even worse, since the last bit is
> a sign bit, you essentially can only achieve a dynamic range of 45dB !
>
> Now, I know this is off topic, but I never heard any good explanation
> why CD audio is suddenly 45 dB ?  If anybody knows, please share your
> thoughts !
>
> Wkr,
>
>
Even when you are using floating point number

Floating point numbers provide a way to trade off signal-to-noise ratio for
an increase in dynamic range. For n bit floating-point numbers, with n-m
bits in the mantissa and m bits in the exponent

DR is still a finite number
http://en.wikipedia.org/wiki/Signal-to-noise_ratio

Re: wrong decibel data?

[Werner Van Belle]

[-- Attachment #1.1: Type: text/plain, Size: 397 bytes --]

Raymond Yau wrote:
> Floating point numbers provide a way to trade off signal-to-noise ratio for
> an increase in dynamic range. For n bit floating-point numbers, with n-m
> bits in the mantissa and m bits in the exponent
>   
That is an excellent point !
It is however not how data is stored on CD's, nor sent to many
soundcards :-(

Werner,-

-- 
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Re: wrong decibel data?

[Raymond Yau]

2010/4/9 Nicolo' Chieffo <nicolo.chieffo@gmail.com>

> On Fri, Apr 9, 2010 at 1:11 AM, Raymond Yau <superquad.vortex2@gmail.com>
> wrote:
> > Mute Capable (1 bit) reports if the respective amplifier is capable of
> > muting. Muting implies a –infinity gain (no sound passes), but the actual
> > performance is determined by the hardware.
> >
> > http://thread.gmane.org/gmane.linux.alsa.devel/39262/focus=16100
>
> I think you missed the point, I'll explain again.
> I'm not complaining that at 0% the audio is not mute, as the bug you
> linked.
>
> >>> Clemens Ladisch (developer) 2004-11-09 09:26
> >>>    The AD1981B's datasheet says that the maximum attenuation is 46.5 dB
> >>> (which conforms to the AC'97 specification).
> >>> To mute, use Mute.
>
> In fact my problem is that at 0% I get mute, but I shouldn't (as
> reported by that developer), since the max attenuation is not -inf,
> but -48 dB.
> So if you prefer, you could see this issue from a different point of
> view: at -48 dB the audio should be still audible, but it's not.
>
> so you have to decide where the issue resides (but definitely not in
> pulseaudio)
> a) it is ok that the audio is mute at 0% (in this case you have to
> declare -inf dB)
> b) the audio level is really -48 dB (in this case the volume shouldn't
> be cut off completely, but simply quite low)
> Which one do you prefer?
>

http://en.wikipedia.org/wiki/Decibel

In electronics, the decibel is often used to express power or amplitude
ratios (gains), in preference to arithmetic ratios or percentages. One
advantage is that the total decibel gain of a series of components (such as
amplifers and attenuators) can be calculated simply by summing the decibel
gains of the individual components.

Re: wrong decibel data?

[Raymond Yau]

2010/4/9 Nicolo' Chieffo <nicolo.chieffo@gmail.com>

> On Fri, Apr 9, 2010 at 1:11 AM, Raymond Yau <superquad.vortex2@gmail.com>
> wrote:
> > Mute Capable (1 bit) reports if the respective amplifier is capable of
> > muting. Muting implies a –infinity gain (no sound passes), but the actual
> > performance is determined by the hardware.
> >
> > http://thread.gmane.org/gmane.linux.alsa.devel/39262/focus=16100
>
> I think you missed the point, I'll explain again.
> I'm not complaining that at 0% the audio is not mute, as the bug you
> linked.
>
> >>> Clemens Ladisch (developer) 2004-11-09 09:26
> >>>    The AD1981B's datasheet says that the maximum attenuation is 46.5 dB
> >>> (which conforms to the AC'97 specification).
> >>> To mute, use Mute.
>
> In fact my problem is that at 0% I get mute, but I shouldn't (as
> reported by that developer), since the max attenuation is not -inf,
> but -48 dB.
> So if you prefer, you could see this issue from a different point of
> view: at -48 dB the audio should be still audible, but it's not.
>
> so you have to decide where the issue resides (but definitely not in
> pulseaudio)
> a) it is ok that the audio is mute at 0% (in this case you have to
> declare -inf dB)
> b) the audio level is really -48 dB (in this case the volume shouldn't
> be cut off completely, but simply quite low)
> Which one do you prefer?
>

if you had read studied the bug report

"On my Compaq R3070 laptop setting Master and Master Mono controls to 0 does
not result in no sound out. I still get audio at a low level."

The user only set Master and Master mono controls to 0.

http://www.analog.com/en/audiovideo-products/audio-codecs/ad1981b/products/product.html

Refer to Figure 1 THe functional BLock Diagram , you should notice that
there is another volume control in the audio path from DAC to LINE_OUT

The Master Volume and Master mono control have dB range from -46.5dB to 0dB

But the PCM Out volume control has dB range of -34.5dB to +12 dB

That is you have to sum the atten of the hardware volume controls in the
audio path

Re: wrong decibel data?

[Raymond Yau]

2010/4/8 Nicolo' Chieffo <nicolo.chieffo@gmail.com>

> I'm sorry but I found out that I gave you wrong values of "amixer -D
> pulse": I just discovered that when I lower the alsa master volume, in
> some occasions the PCM volume goes to 99%, so now I will put it to
> 100% and check the pulse volume again.
>
> 0 dB    =  65536 [100%]
> -3 dB   =  57520 [88%]
> -6 dB   =  52057 [79%]
> -12 dB =  41034 [63%]
> -24 dB =  26090 [40%]
> -48 dB = 10151 [15%]


You have to ask PA developer since  "PCM" is a softvol control with access
'user' and dbmin -51dB

    control.24 {
        comment.access 'read write user'
        comment.type INTEGER
        comment.count 2
        comment.range '0 - 255'
        comment.tlv '0000000100000008ffffec1400000014'
        comment.dbmin -5100
        comment.dbmax 0
        iface MIXER
        name 'PCM Playback Volume'
        value.0 255
        value.1 255
    }


HDA-Intel.pcm.front.0 {
	@args [ CARD ]
	@args.CARD {
		type string
	}
	type softvol
	slave.pcm {
		type hw
		card $CARD
	}
	control {
		name "PCM Playback Volume"
		card $CARD
	}
}

Re: wrong decibel data?

[Raymond Yau]

2010/4/8 Nicolo' Chieffo <nicolo.chieffo@gmail.com>

> I'm sorry but I found out that I gave you wrong values of "amixer -D
> pulse": I just discovered that when I lower the alsa master volume, in
> some occasions the PCM volume goes to 99%, so now I will put it to
> 100% and check the pulse volume again.
>
> 0 dB    =  65536 [100%]
> -3 dB   =  57520 [88%]
> -6 dB   =  52057 [79%]
> -12 dB =  41034 [63%]
> -24 dB =  26090 [40%]
> -48 dB = 10151 [15%]
>
> So again, the only problem is that -48 dB IS mute, but it's not
> declared as mute.
> This is the real issue.
>

For AC97 codec , both Master and PCM are hardware volume controls

Register 02 - control the stereo master volume
Register 18 - PCM out volume

But on HDA , PCM is most likely a software volume control just like
pulseaudio volume control