Alsa SoC Audio(part 2)

 

6 Audio Clocking
This text describes the audio clocking terms in ASoC and digital audio in
general. Note: Audio clocking can be complex!

Master Clock
------------
Every audio subsystem is driven by a master clock (sometimes referred to as MCLK
or SYSCLK). This audio master clock can be derived from a number of sources
(e.g. crystal, PLL, CPU clock) and is responsible for producing the correct
audio playback and capture sample rates.

Some master clocks (e.g. PLLs and CPU based clocks) are configurable in that
their speed can be altered by software (depending on the system use and to save
power). Other master clocks are fixed at a set frequency (i.e. crystals).

DAI Clocks
----------
The Digital Audio Interface is usually driven by a Bit Clock (often referred to
as BCLK). This clock is used to drive the digital audio data across the link
between the codec and CPU.

The DAI also has a frame clock to signal the start of each audio frame. This
clock is sometimes referred to as LRC (left right clock) or FRAME. This clock
runs at exactly the sample rate (LRC = Rate).

Bit Clock can be generated as follows:-

BCLK = MCLK / x

or

BCLK = LRC * x

or

BCLK = LRC * Channels * Word Size

This relationship depends on the codec or SoC CPU in particular. In general
it is best to configure BCLK to the lowest possible speed (depending on your
rate, number of channels and word size) to save on power.

It is also desirable to use the codec (if possible) to drive (or master) the
audio clocks as it usually gives more accurate sample rates than the CPU.

                                        
7 Dynamic Audio Power Management for Portable Devices
1. Description
==============

Dynamic Audio Power Management (DAPM) is designed to allow portable
Linux devices to use the minimum amount of power within the audio
subsystem at all times. It is independent of other kernel PM and as
such, can easily co-exist with the other PM systems.

DAPM is also completely transparent to all user space applications as
all power switching is done within the ASoC core. No code changes or
recompiling are required for user space applications. DAPM makes power
switching decisions based upon any audio stream (capture/playback)
activity and audio mixer settings within the device.

DAPM spans the whole machine. It covers power control within the entire
audio subsystem, this includes internal codec power blocks and machine
level power systems.

There are 4 power domains within DAPM

    1. Codec domain - VREF, VMID (core codec and audio power)
       Usually controlled at codec probe/remove and suspend/resume, although
       can be set at stream time if power is not needed for sidetone, etc.

    2. Platform/Machine domain - physically connected inputs and outputs
       Is platform/machine and user action specific, is configured by the
       machine driver and responds to asynchronous events e.g when HP
       are inserted

    3. Path domain - audio susbsystem signal paths
       Automatically set when mixer and mux settings are changed by the user.
       e.g. alsamixer, amixer.

    4. Stream domain - DACs and ADCs.
       Enabled and disabled when stream playback/capture is started and
       stopped respectively. e.g. aplay, arecord.

All DAPM power switching decisions are made automatically by consulting an audio
routing map of the whole machine. This map is specific to each machine and
consists of the interconnections between every audio component (including
internal codec components). All audio components that effect power are called
widgets hereafter.


2. DAPM Widgets
===============

Audio DAPM widgets fall into a number of types:-

o Mixer      - Mixes several analog signals into a single analog signal.
o Mux        - An analog switch that outputs only one of many inputs.
o PGA        - A programmable gain amplifier or attenuation widget.
o ADC        - Analog to Digital Converter
o DAC        - Digital to Analog Converter
o Switch     - An analog switch
o Input      - A codec input pin
o Output     - A codec output pin
o Headphone - Headphone (and optional Jack)
o Mic        - Mic (and optional Jack)
o Line       - Line Input/Output (and optional Jack)
o Speaker    - Speaker
o Pre        - Special PRE widget (exec before all others)
o Post       - Special POST widget (exec after all others)

(Widgets are defined in include/sound/soc-dapm.h)

Widgets are usually added in the codec driver and the machine driver. There are
convience macros defined in soc-dapm.h that can be used to quickly build a
list of widgets of the codecs and machines DAPM widgets.

Most widgets have a name, register, shift and invert. Some widgets have extra
parameters for stream name and kcontrols.


2.1 Stream Domain Widgets
-------------------------

Stream Widgets relate to the stream power domain and only consist of ADCs
(analog to digital converters) and DACs (digital to analog converters).

Stream widgets have the following format:-

SND_SOC_DAPM_DAC(name, stream name, reg, shift, invert),

NOTE: the stream name must match the corresponding stream name in your codec
snd_soc_codec_dai.

e.g. stream widgets for HiFi playback and capture

SND_SOC_DAPM_DAC("HiFi DAC", "HiFi Playback", REG, 3, 1),
SND_SOC_DAPM_ADC("HiFi ADC", "HiFi Capture", REG, 2, 1),


2.2 Path Domain Widgets
-----------------------

Path domain widgets have a ability to control or affect the audio signal or
audio paths within the audio subsystem. They have the following form:-

SND_SOC_DAPM_PGA(name, reg, shift, invert, controls, num_controls)

Any widget kcontrols can be set using the controls and num_controls members.

e.g. Mixer widget (the kcontrols are declared first)

/* Output Mixer */
static const snd_kcontrol_new_t wm8731_output_mixer_controls[] = {
SOC_DAPM_SINGLE("Line Bypass Switch", WM8731_APANA, 3, 1, 0),
SOC_DAPM_SINGLE("Mic Sidetone Switch", WM8731_APANA, 5, 1, 0),
SOC_DAPM_SINGLE("HiFi Playback Switch", WM8731_APANA, 4, 1, 0),
};

SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1, wm8731_output_mixer_controls,
ARRAY_SIZE(wm8731_output_mixer_controls)),


2.3 Platform/Machine domain Widgets
-----------------------------------

Machine widgets are different from codec widgets in that they don't have a
codec register bit associated with them. A machine widget is assigned to each
machine audio component (non codec) that can be independently powered. e.g.

o Speaker Amp
o Microphone Bias
o Jack connectors

A machine widget can have an optional call back.

e.g. Jack connector widget for an external Mic that enables Mic Bias
when the Mic is inserted:-

static int spitz_mic_bias(struct snd_soc_dapm_widget* w, int event)
{
if(SND_SOC_DAPM_EVENT_ON(event))
   set_scoop_gpio(&spitzscoop2_device.dev, SPITZ_SCP2_MIC_BIAS);
else
   reset_scoop_gpio(&spitzscoop2_device.dev, SPITZ_SCP2_MIC_BIAS);

return 0;
}

SND_SOC_DAPM_MIC("Mic Jack", spitz_mic_bias),


2.4 Codec Domain
----------------

The codec power domain has no widgets and is handled by the codecs DAPM event
handler. This handler is called when the codec powerstate is changed wrt to any
stream event or by kernel PM events.


2.5 Virtual Widgets
-------------------

Sometimes widgets exist in the codec or machine audio map that don't have any
corresponding soft power control. In this case it is necessary to create
a virtual widget - a widget with no control bits e.g.

SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_DAPM_NOPM, 0, 0, NULL, 0),

This can be used to merge to signal paths together in software.

After all the widgets have been defined, they can then be added to the DAPM
subsystem individually with a call to snd_soc_dapm_new_control().


3. Codec Widget Interconnections
================================

Widgets are connected to each other within the codec and machine by audio paths
(called interconnections). Each interconnection must be defined in order to
create a map of all audio paths between widgets.

This is easiest with a diagram of the codec (and schematic of the machine audio
system), as it requires joining widgets together via their audio signal paths.

e.g., from the WM8731 output mixer (wm8731.c)

The WM8731 output mixer has 3 inputs (sources)

1. Line Bypass Input
2. DAC (HiFi playback)
3. Mic Sidetone Input

Each input in this example has a kcontrol associated with it (defined in example
above) and is connected to the output mixer via it's kcontrol name. We can now
connect the destination widget (wrt audio signal) with it's source widgets.

/* output mixer */
{"Output Mixer", "Line Bypass Switch", "Line Input"},
{"Output Mixer", "HiFi Playback Switch", "DAC"},
{"Output Mixer", "Mic Sidetone Switch", "Mic Bias"},

So we have :-

Destination Widget <=== Path Name <=== Source Widget

Or:-

Sink, Path, Source

Or :-

"Output Mixer" is connected to the "DAC" via the "HiFi Playback Switch".

When there is no path name connecting widgets (e.g. a direct connection) we
pass NULL for the path name.

Interconnections are created with a call to:-

snd_soc_dapm_connect_input(codec, sink, path, source);

Finally, snd_soc_dapm_new_widgets(codec) must be called after all widgets and
interconnections have been registered with the core. This causes the core to
scan the codec and machine so that the internal DAPM state matches the
physical state of the machine.


3.1 Machine Widget Interconnections
-----------------------------------
Machine widget interconnections are created in the same way as codec ones and
directly connect the codec pins to machine level widgets.

e.g. connects the speaker out codec pins to the internal speaker.

/* ext speaker connected to codec pins LOUT2, ROUT2 */
{"Ext Spk", NULL , "ROUT2"},
{"Ext Spk", NULL , "LOUT2"},

This allows the DAPM to power on and off pins that are connected (and in use)
and pins that are NC respectively.


4 Endpoint Widgets
===================
An endpoint is a start or end point (widget) of an audio signal within the
machine and includes the codec. e.g.

o Headphone Jack
o Internal Speaker
o Internal Mic
o Mic Jack
o Codec Pins

When a codec pin is NC it can be marked as not used with a call to

snd_soc_dapm_set_endpoint(codec, "Widget Name", 0);

The last argument is 0 for inactive and 1 for active. This way the pin and its
input widget will never be powered up and consume power.

This also applies to machine widgets. e.g. if a headphone is connected to a
jack then the jack can be marked active. If the headphone is removed, then
the headphone jack can be marked inactive.


5 DAPM Widget Events
====================

Some widgets can register their interest with the DAPM core in PM events.
e.g. A Speaker with an amplifier registers a widget so the amplifier can be
powered only when the spk is in use.

/* turn speaker amplifier on/off depending on use */
static int corgi_amp_event(struct snd_soc_dapm_widget *w, int event)
{
if (SND_SOC_DAPM_EVENT_ON(event))
   set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_APM_ON);
else
   reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_APM_ON);

return 0;
}

/* corgi machine dapm widgets */
static const struct snd_soc_dapm_widget wm8731_dapm_widgets =
SND_SOC_DAPM_SPK("Ext Spk", corgi_amp_event);

Please see soc-dapm.h for all other widgets that support events.


5.1 Event types
---------------

The following event types are supported by event widgets.

/* dapm event types */
#define SND_SOC_DAPM_PRE_PMU 0x1 /* before widget power up */
#define SND_SOC_DAPM_POST_PMU 0x2 /* after widget power up */
#define SND_SOC_DAPM_PRE_PMD 0x4 /* before widget power down */
#define SND_SOC_DAPM_POST_PMD 0x8 /* after widget power down */
#define SND_SOC_DAPM_PRE_REG 0x10 /* before audio path setup */
#define SND_SOC_DAPM_POST_REG 0x20 /* after audio path setup */
           
           
8 Audio Pops and Clicks
Pops and clicks are unwanted audio artifacts caused by the powering up and down
of components within the audio subsystem. This is noticeable on PCs when an
audio module is either loaded or unloaded (at module load time the sound card is
powered up and causes a popping noise on the speakers).

Pops and clicks can be more frequent on portable systems with DAPM. This is
because the components within the subsystem are being dynamically powered
depending on the audio usage and this can subsequently cause a small pop or
click every time a component power state is changed.


Minimising Playback Pops and Clicks
===================================
Playback pops in portable audio subsystems cannot be completely eliminated
currently, however future audio codec hardware will have better pop and click
suppression. Pops can be reduced within playback by powering the audio
components in a specific order. This order is different for startup and
shutdown and follows some basic rules:-

Startup Order :- DAC --> Mixers --> Output PGA --> Digital Unmute

Shutdown Order :- Digital Mute --> Output PGA --> Mixers --> DAC

This assumes that the codec PCM output path from the DAC is via a mixer and then
a PGA (programmable gain amplifier) before being output to the speakers.


Minimising Capture Pops and Clicks
==================================
Capture artifacts are somewhat easier to get rid as we can delay activating the
ADC until all the pops have occurred. This follows similar power rules to
playback in that components are powered in a sequence depending upon stream
startup or shutdown.

Startup Order - Input PGA --> Mixers --> ADC

Shutdown Order - ADC --> Mixers --> Input PGA


Zipper Noise
============
An unwanted zipper noise can occur within the audio playback or capture stream
when a volume control is changed near its maximum gain value. The zipper noise
is heard when the gain increase or decrease changes the mean audio signal
amplitude too quickly. It can be minimised by enabling the zero cross setting
for each volume control. The ZC forces the gain change to occur when the signal
crosses the zero amplitude line.