Detailed changes v1.1.6 v1.1.7

Changelog between 1.1.6 and 1.1.7 releases

alsa-lib

Core

- Release v1.1.7

- conf/ucm: bytcr-rt5651: Add bytcr-rt5651-mono-spk-in2-mic-hp-swapped config

Add a longname profile for devices with a mono speaker, the Internal Mic

hooked up to IN2 and the left and right channels of their headphones

output swapped.

Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>

- conf/ucm: bytcr-rt5651: Add long-name UCM profiles

After recent kernel work, the kernel now sets a long-name for bytcr-rt5651

boards which indicates if a single (mono) speaker or stereo speakers are

used and if in1, in2, or in1 and 2 are used for the internal mic(s) (the

headset mic sofar is always on in3).

This commit adds UCM profiles for bytcr-rt5651 boards using these new

long-names, based on the generic bytcr-rt5651 profile.

Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>

- conf/ucm: bytcr-rt5651: Use generic SSP enable + disable sequence, support SSP0

Use the generic SSP enable sequence from bytcr/PlatformEnableSeq.conf,

for boards using SSP2 this is identical the code it replaces and this

adds support for boards using SSP0.

This fixes sound not working on Bay Trail CR tablets with a rt5651 codec.

This commit also calls the generic disable sequence on shutdown

(this is new).

Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>

- conf/ucm: chtnau8824: Add Cube iWork8 Air and Pipo W2S specific profiles

The Cube iWork8 Air and Pipo W2S tablets both only have a single speaker.

Add long-name profiles for them which are identical to the default

chtnau8824 profile, except that they include the nau8824/MonoSpeaker.conf

snippet instead of the nau8824/Speaker.conf one.

- conf/ucm: chtnau8824: Add UCM profile for chtnau8824 boards

Add UCM profile for chtnau8824 boards based on:

https://github.com/plbossart/UCM/blob/master/chtnau8824

Split into multiple files in the same way as this was done for the

bytcr-rt5640 support, re-using the existing ucm/PlatformEnableSeq.conf

and ucm/PlatformDisableSeq.conf files for the SST mixer settings.

- conf/ucm: bytcr-rt5640: Add long-name UCM profiles

With a recently merged kernel commit, the kernel now sets a long-name for

bytcr-rt5640 boards which indicates if a single (mono) speaker or stereo

speakers are used and wether dmic1, in1 or in3 is used for the internal

mic (the headset mic sofar is always in2).

This commit adds UCM profiles for bytcr-rt5640 boards using these new

long-names, based on the generic bytcr-rt5640 profile.

The added profiles have the unnecessary input / output options from the

generic profile removed leaving only 2 input and 2 output options, which

are automatically switched between by e.g. pulse based on jack-detect.

- conf/ucm: bytcr-rt5640: Add generic bytcr-rt5640 UCM profile

This commit adds the generic UCM profile for bytcr-rt5640 boards from:

https://github.com/plbossart/UCM, plus the fixes from this pull-req:

https://github.com/plbossart/UCM/pull/31

The profile has been split up into separate per input / output files to

allow for creation of long-name profiles with the specific input / output

combinations found on a board without needing to copy and paste things.

Note this profile exports all inputs and both stereo/mono speaker setups

even though a typical device will not use all. Ideally a long-name based

device specific profile made up of the various parts should be used

instead.

- configure: Fix forgotten ucm entry

The previous commit forgot to add to configure.ac. Fix it.

- conf/ucm: chtrt5645: At config for the Lenovo Ideapad Miix 320

The Lenovo Ideapad Miix 320 uses a digital mic connected to the DMIC2 input

(unlike the Asus T100HA which has it connected to the DMIC1 input), add a

long-name config specific for the Miix 320, which is a copy of the standard

chtrt5645 config with the internal analog mic section replaced with one

for a digital mic connected to the DMIC2 input.

- conf/ucm: chtrt5645: At config for the Asus T100HA

The Asus T100HA uses a digital mic rather then an analog one, add

long-name config specific for the T100HA, which is a copy of the standard

chtrt5645 config with the internal analog mic section replaced with one

for the digital mic found on the Asus T100HA.

- configure: remove src/conf/alsa.conf.d/Makefile

Control API

- control_hw: Fix issue when applying seccomp policy

When seccomp policy is applied to filter ioctl syscall with

SNDRV_CTL_IOCTL_TLV_COMMAND, SNDRV_CTL_IOCTL_TLV_READ and

SNDRV_CTL_IOCTL_TLV_WRITE in whiltelist, alsa-lib still breaks

in at snd_ctl_hw_elem_tlv().

The problem behind is because ioctl() takes unsigned long cmd

argument, and the signed bit of local int variable could cause

0xff bytes appended after casted to unsigned long.

In kernel, seccomp data struct takes 64 bits argument to check

against seccomp rules, these unexpected 0xff bytes could make

the rule check fail.

Fix the problem by passing unsigned int to ioctl.

- hcontrol: use position offset macro of TLV data

A series of SNDRV_CTL_TLVO_XXX macro was introduced for position offset

of TLV data. This commit applies a code optimization.

- control: use position offset macro of TLV data

A series of SNDRV_CTL_TLVO_XXX macro was introduced for position offset

of TLV data. This commit applies a code optimization.

PCM API

- pcm: ioplug: Transfer all available data

The snd_pcm_mmap_begin() call returns the amount of contiguous data,

which is less than the total available if it wraps around the buffer

boundary.

If we don't handle this split we leave stale data in the buffer that

should have been overwritten, as well as unread data in the io_plugin

that gets transferred on a subsequent call at the wrong offset.

- pcm: rate: Add error check for snd_pcm_avail_update()

Without these changes a negative error code returned by

snd_pcm_avail_update() will be not handled correctly.

With this patch the returned error code of snd_pcm_avail_update() will be

returned by snd_pcm_rate_avail_update().

- pcm: ioplug: Provide avail helper function for plugins

This function can be called without calling snd_pcm_avail_update().

The call to snd_pcm_avail_update() can take some time.

Therefore some developers would not like to call it from a real-time

context (e.g. from JACK client context).

- pcm: snd_interval_refine_first/last: exclude value only if also excluded before

Without this commit the following intervals [x y), (x y) were be

replaced to (y-1 y) by snd_interval_refine_last(). This was also done if

y-1 is part of the previous interval.

With this changes it will be replaced with [y-1 y) in case of y-1 is

part of the previous interval. A similar behavior will be used for

snd_interval_refine_first().

This solves the issue reported here:

https://bugzilla.opensuse.org/show_bug.cgi?id=1033179

and work arounded with commit

e736715 ("pcm: dmix: Disable var_periodsize as default").

I am able to reproduce the issue with a simplified aplay use case using

the following configuration:

pcm_slave.adr3_tdm_8ch {

pcm {

type hw

card "Loopback"

device 0

}

rate 48000

period_size 128

buffer_size 1024

channels 2

}

pcm.dshare_Playback_3 {

type dmix

ipc_key 600

ipc_perm 0660

ipc_gid audio

var_periodsize true

slave adr3_tdm_8ch

}

pcm.AdevAcousticoutSpeech {

type rate

slave.pcm dshare_Playback_3

slave.rate 48000

}

$ modprobe snd_aloop

$ aplay -v --period-size=352 -c2 -fS16_LE -r22500 -DAdevAcousticoutSpeech /dev/urandom

...

Rule 9 (0xffff91d1f230): PERIODS=(0 2) -> NONE BUFFER_SIZE=480 PERIOD_SIZE=[240 240]

refine_soft 'AdevAcousticoutSpeech' (end--22)

...

aplay: ../../alsa-utils-1.1.5/aplay/aplay.c:1390: set_params: Assertion `err >= 0' failed.

Aborted by signal Aborted...

The following stack trace shows where the -EINVAL will be thrown:

__snd_pcm_hw_params_set_period_size_near()

snd1_pcm_hw_param_set_near()

snd1_pcm_hw_param_set_last()

snd1_pcm_hw_refine_slave()

snd1_pcm_hw_refine_soft()

snd_pcm_hw_rule_div()

snd1_interval_refine()

This issue exists due to PERIODS does not include 2

Rule 9 (0xffff91d1f230): PERIODS=(0 9) -> (0 2) BUFFER_SIZE=[120 480]

PERIOD_SIZE=(240 241)

because of an invalid integer inverval of PERIOD_SIZE of (240 241).

This interval is set by snd_interval_refine_last().

- pcm: Define refine mask bits explicitly (and fix missing DSD support)

Instead of the expanded bit numbers like 0x81ffffff, list up the all

supported PCM bits explicitly for refine_masks[] in pcm_params.c.

This makes easier to update any additional formats or other

parameters, and easier to spot out missing ones.

Actually the GSM and DSD formats were missing; with this commit, they

are supported properly now.

- pcm: dmix: Fix hwptr updates at status call

In the commit 38a2d2eda880 ("pcm: dmix: Do not discard slave reported

delay in status result"), the PCM dmix hwptr update code was rewritten

to follow the slave PCM hwptr update. This is based on the similar

change in PCM dshare, the commit faf53c197cab.

There was a bug in the commit 38a2d2eda880 regarding the PCM state

change, and it was addressed in commit 3752e6b8733d ("pcm: dmix: Fix

the inconsistent PCM state"). However, we've hit yet another bug in

this commit. Namely, the hwptr update was forgotten in the

snd_pcm_dmix_sync_ptr0() function. So the hwptr value passed from

snd_pcm_dmix_status() isn't properly stored, and it screws up at some

long run occasionally.

This patch covers the bug by replacing with the right value.

Fixes: 38a2d2eda880 ("pcm: dmix: Do not discard slave reported delay in status result")

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=200013

- pcm ioplug: fix some coverity issues (switch, missing unlock in snd_pcm_ioplug_drain())

- pcm: add missing flags initialization for the fallback control data

- pcm: softvol: use position offset macro of TLV data

A series of SNDRV_CTL_TLVO_XXX macro was introduced for position offset

of TLV data. This commit applies a code optimization.

- pcm: hw: use position offset macro of TLV data

A series of SNDRV_CTL_TLVO_XXX macro was introduced for position offset

of TLV data. This commit applies a code optimization.

- pcm: softvol: Allow up to 90 dB of gain

The gain algorithm used in softvol can handle gain factors of up to

32767 which is slightly more than 90 dB, so allow a max_dB of 90 dB.

This doesn't affect existing asound.conf files, but does allow a

max_dB of up to 90 dB when needed.

Tested using Audacity that there is no undue distorsion or other

artefacts when 90 dB of gain is applied to a suitable signal (i.e.

a signal quiet enough not be clipped whan applying 90 dB of gain).

PCM Plugin API

- pcm: Fix header guard in pcm_plugin.h

Sequencer API

- seq: Fix signedness in MIDI encoder/decoder

The qlen field of struct snd_midi_event was declared as size_t while

status_events[] assigns the qlen to -1 indicating to skip. This leads

to the misinterpretation since size_t is unsigned, hence it passes the

check "dev.qlen > 0" incorrectly in snd_midi_event_encode_byte(),

which eventually results in a memory corruption.

Also, snd_midi_event_decode() doesn't consider about a negative qlen

value and tries to copy the size as is.

This patch fixes these issues: the first one is addressed by simply

replacing size_t with ssize_t in snd_midi_event struct. For the

latter, a check "qlen <= 0" is added to bail out; this is also good as

a slight optimization.

Reported-by: Prashant Malani <pmalani@chromium.org>

Topology API

- ASoC: topology: Add alias conf parameter names for hw_configs

Currently, some parameter names in conf differ from field names in struct.

These look like typos.

This commit suggests to add aliases for such parameters, so that the names

in conf are similar to names in struct. This solution is backwards

compatible.

If the difference between conf names and struct names is done on purpose -

this commit can be dropped.

- ASoC: topology: Add definitions for mclk_direction values

Current comment makes not clear the direction of mclk. Previously, similar

description caused a misunderstanding for bclk_master and fsync_master.

This commit solves the potential confusion the same way it is solved for

bclk_master and fsync_master.

- ASoC: topology: Add missing clock gating parameter when parsing hw_configs

Clock gating parameter is a part of `dai_fmt`. It is supported by

`alsa-lib` when creating a topology binary file, but ignored by kernel

when loading this topology file.

After applying this commit, the clock gating parameter is not ignored any

more. This solution is backwards compatible. The existing behaviour is

not broken, because by default the parameter value is 0 and is ignored.

snd_soc_tplg_hw_config.clock_gated = 0 => no effect

snd_soc_tplg_hw_config.clock_gated = 1 => SND_SOC_DAIFMT_GATED

snd_soc_tplg_hw_config.clock_gated = 2 => SND_SOC_DAIFMT_CONT

For example, the following config, based on

alsa-lib/src/conf/topology/broadwell/broadwell.conf, is now supported:

~~~~

SectionHWConfig."CodecHWConfig" {

id "1"

format "I2S" # physical audio format.

pm_gate_clocks "true" # clock can be gated

}

SectionLink."Codec" {

# used for binding to the physical link

id "0"

hw_configs [

"CodecHWConfig"

]

default_hw_conf_id "1"

}

~~~~

Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>

- ASoC: topology: Fix bclk and fsync inversion in set_link_hw_format()

The values of bclk and fsync are inverted WRT the codec. But the existing

solution already works for Broadwell, see the alsa-lib config:

`alsa-lib/src/conf/topology/broadwell/broadwell.conf`

This commit provides the backwards-compatible solution to fix this misuse.

This commit goes in pair with the corresponding patch for linux.

Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>

Tested-by: Pan Xiuli <xiuli.pan@linux.intel.com>

Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>

Use Case Manager API

- ucm: adding the folder of card_long_name when finding verb conf file

The board configuration file and verb conf file are allowed to be

in the folder named of card_long_name, so when finding the verb conf

file, we need to check if it is in the folder of card_long_name or

card_name.

Configuration

- conf/ucm: bytcr-rt5640: Improve human readable input/output names

The Comment field is displayed tot the end user in various UIs as such

names like MonoSpeaker and DigitalMics without any spaces are no good.

Also the names themselves as well as how they get displayed in the

typical UI (in separate input / output tabs) makes the adding of

playback and capture to the comment superfluous and this looks weird

in the UI, so drop it.

Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>

- conf/ucm: bytcr-rt5651: Add bytcr-rt5651-mono-spk-in2-mic-hp-swapped config

Add a longname profile for devices with a mono speaker, the Internal Mic

hooked up to IN2 and the left and right channels of their headphones

output swapped.

Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>

- conf/ucm: bytcr-rt5651: Add long-name UCM profiles

After recent kernel work, the kernel now sets a long-name for bytcr-rt5651

boards which indicates if a single (mono) speaker or stereo speakers are

used and if in1, in2, or in1 and 2 are used for the internal mic(s) (the

headset mic sofar is always on in3).

This commit adds UCM profiles for bytcr-rt5651 boards using these new

long-names, based on the generic bytcr-rt5651 profile.

Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>

- conf/ucm: bytcr-rt5651: Add mono speaker output profile

Many rt5651 devices only have a single speaker and even though there is

some external mixing done on the PCB, the quality of that mixing is quite

poor and various sounds come out garbled when relying on the on PCB mixing.

Using the codecs builtin mixer to mix left + right to the left output works

much better. This commits adds a new MonoSpeaker.conf output profile which

allows this.

Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>

- conf/ucm: bytcr-rt5651: Split into 1 .conf file per input / output

Split the bytcr-rt5651 config into 1 .conf file per input / output as

has already been done for the bytcr-rt5640 and the chtnau8824 profiles.

This allows easy creation of long-name profiles with the specific input /

output combinations found on a board without needing to copy and paste

things.

Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>

- conf/ucm: bytcr-rt5651: Add support for mic input on various pins

Note this commit replaces the pre-existing "Handset Microphone" and

"Main Microphone" options, these come from the first commit of the

bytcr-rt5651 UCM profile and were based on wrong assumptions about the

input mappings. None of the existing devices has the Hand/Headset mic

on IN1 as these options assumed.

The rt5651 is used in various configurations with the Internal Mic(s)

hooked up to IN1, IN2, or to IN1 and IN2 and the Headset Mic hooked up

to IN3.

Add support for all these to the generic bytcr-rt5651 profile and name

them accotding to their input + functions.

A follow up commit will add specialized longname configs which

will only expose the inputs actually used on the board with that

longname.

Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>

- conf/ucm: bytcr-rt5651: Fix ADC and Mic capture volumes

Fix ADC and Mic capture volumes, so that the microphone inputs actually

work.

Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>

- conf/ucm: bytcr-rt5651: Use generic SSP enable + disable sequence, support SSP0

Use the generic SSP enable sequence from bytcr/PlatformEnableSeq.conf,

for boards using SSP2 this is identical the code it replaces and this

adds support for boards using SSP0.

This fixes sound not working on Bay Trail CR tablets with a rt5651 codec.

This commit also calls the generic disable sequence on shutdown

(this is new).

Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>

- conf/ucm: bytcr-rt5651: Change Speaker/Headphone en/disable sequences to fix switching

pulseaudio will run the DisableSequence of the current playback device

before running the EnableSequence of the new playback device.

This causes the Platform Clock and BIAS to temporarily get turned off which

on the rt5651 breaks audio-streams which are playing when switching.

This commit moves the disabling to the EnableSequence of the other device

fixing this.

Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>

- conf/ucm: bytcr-rt5651: Start with all outputs and inputs disabled

Start with all switches disabled, so that e.g. the

LOUT L/R Playback Switches are not left enabled when starting with

headphones plugged in.

This fixes the platform clock being kept on by these in some cases.

While at also move the IN? Boost and IF1 ASRC Switch lines around

a bit to match the order from https://github.com/plbossart/UCM so

the profiles can be more easily compared.

Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>

- conf/ucm: bytcr-rt5651: Configure all volumes at 0dB

The volumes are taken from this commit:

https://github.com/plbossart/UCM/commit/753e2430cd0e6cafbf10c6b29b689dbd32ac5250

That commit also adds line-in support, so it has not been

taken in its entirety.

Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>

- conf/ucm: bytcr-rt5651: Do not use both DAC1 and HPVOL inputs for HP

The headphones can either be driven directly from DAC1, or through

the HP volume mixer chain to allow volume control, both can be enabled

at the same time, but this should not be done.

Mix only DAC1 to the headphones and not the HP volume path, there

are 2 reasons to choice the DAC1 path;

1) It is the power-on-reset default

2) We don't expose the volume control to e.g. pulseaudio anyways so it

is not useful

While at it also move the "HPO MIX DAC1" and "HPO MIX HPVOL" entries up a

bit so that they are no longer inbetween the "HPO L Playback Switch" and

"HPO R Playback Switch" entries.

Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>

- conf/ucm: bytcr-rt5645: Use the generic bytcr/PlatformEnableSeq.conf

Use the generic Intel SSP bytcr/PlatformEnableSeq.conf file, it is

identical to all the cset statements this commit removes.

Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>

- conf/ucm: VEYRON-I2S: Add internal speakers and mic, other fixes

Other fixes include output/input names (comments) for UIs (pavucontrol)

to display, and Playback/CapturePCM entries so pulseaudio initializes

correctly on this hardware.

- conf/ucm: chtnau8824: Add Cube iWork8 Air and Pipo W2S specific profiles

The Cube iWork8 Air and Pipo W2S tablets both only have a single speaker.

Add long-name profiles for them which are identical to the default

chtnau8824 profile, except that they include the nau8824/MonoSpeaker.conf

snippet instead of the nau8824/Speaker.conf one.

- conf/ucm: chtnau8824: Add UCM profile for chtnau8824 boards

Add UCM profile for chtnau8824 boards based on:

https://github.com/plbossart/UCM/blob/master/chtnau8824

Split into multiple files in the same way as this was done for the

bytcr-rt5640 support, re-using the existing ucm/PlatformEnableSeq.conf

and ucm/PlatformDisableSeq.conf files for the SST mixer settings.

- conf/ucm: bytcr-rt5640: Add default DisableSequence

Add a disable sequence powering off the SST mixer elements, loosely

based on the default DisableSequence from:

https://github.com/plbossart/UCM/blob/master/chtnau8824/HiFi.conf

- conf/ucm: bytcr-rt5640: Add long-name UCM profiles

With a recently merged kernel commit, the kernel now sets a long-name for

bytcr-rt5640 boards which indicates if a single (mono) speaker or stereo

speakers are used and wether dmic1, in1 or in3 is used for the internal

mic (the headset mic sofar is always in2).

This commit adds UCM profiles for bytcr-rt5640 boards using these new

long-names, based on the generic bytcr-rt5640 profile.

The added profiles have the unnecessary input / output options from the

generic profile removed leaving only 2 input and 2 output options, which

are automatically switched between by e.g. pulse based on jack-detect.

- conf/ucm: bytcr-rt5640: Add generic bytcr-rt5640 UCM profile

This commit adds the generic UCM profile for bytcr-rt5640 boards from:

https://github.com/plbossart/UCM, plus the fixes from this pull-req:

https://github.com/plbossart/UCM/pull/31

The profile has been split up into separate per input / output files to

allow for creation of long-name profiles with the specific input / output

combinations found on a board without needing to copy and paste things.

Note this profile exports all inputs and both stereo/mono speaker setups

even though a typical device will not use all. Ideally a long-name based

device specific profile made up of the various parts should be used

instead.

- conf: USB-audio: Fix for Xonar U7 SPDIF device

Add the entry for Xonar U7 to make SPDIF working on it.

Reported-by: Steve Banks <eassbank@gmail.com>

- conf/ucm: Add a UCM profile for Dell WD15 Dock USB-audio

USB-audio device on Dell WD15 docking station provides two individual

PCM streams, one for headphone and another for line out. A UCM

profile gives the proper roles for these.

- conf/ucm: chtrt5645: At config for the Lenovo Ideapad Miix 320

The Lenovo Ideapad Miix 320 uses a digital mic connected to the DMIC2 input

(unlike the Asus T100HA which has it connected to the DMIC1 input), add a

long-name config specific for the Miix 320, which is a copy of the standard

chtrt5645 config with the internal analog mic section replaced with one

for a digital mic connected to the DMIC2 input.

- conf/ucm: chtrt5645: At config for the Asus T100HA

The Asus T100HA uses a digital mic rather then an analog one, add

long-name config specific for the T100HA, which is a copy of the standard

chtrt5645 config with the internal analog mic section replaced with one

for the digital mic found on the Asus T100HA.

- conf/ucm: chtrt5645: Fix recording from internal analog microphone

The internal analog mic switch is called 'Int Analog Mic Switch'

(not 'Int Mic Switch') and is connected to BST2 not BST1.

Also change the analog mic volume levels so that we get better

audio / less noise.

- conf/ucm: chtrt5645: Microphone recording fixes

Apply microphone recording changes from:

https://github.com/plbossart/UCM.git

[hdegoede@redhat.com: Drop non generic DMIC changes]

- conf/ucm: chtrt5645: Cleanup and playback fixes

Apply cleanup and playback fixes changes from:

https://github.com/plbossart/UCM.git

[hdegoede@redhat.com: Modify commit msg and paths for merging into alsa-lib]

- conf/ucm: increase the input volume for LineIn

Otherwise, the boost value is 0, and the sound captured from that

LineIn jack is too weak for users.

- ASoC: topology: Fix bclk and fsync inversion in set_link_hw_format()

The values of bclk and fsync are inverted WRT the codec. But the existing

solution already works for Broadwell, see the alsa-lib config:

`alsa-lib/src/conf/topology/broadwell/broadwell.conf`

This commit provides the backwards-compatible solution to fix this misuse.

This commit goes in pair with the corresponding patch for linux.

Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>

Tested-by: Pan Xiuli <xiuli.pan@linux.intel.com>

Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>

- conf: remove alsa.conf.d from the datadir

- alsa.conf: change the location for add-on configs to /etc/alsa/conf.d

The add-on configuration files should be placed to a volatile place.

External PCM I/O Plugin SDK

- ioplug: Fix comment in struct snd_pcm_ioplug

This patch fixes the comment about 'pcm' field in 'struct

snd_pcm_ioplug' which wrongly refers to snd_pcm_extplug_create().

Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>

- pcm: ioplug: Provide avail helper function for plugins

This function can be called without calling snd_pcm_avail_update().

The call to snd_pcm_avail_update() can take some time.

Therefore some developers would not like to call it from a real-time

context (e.g. from JACK client context).

Kernel Headers

- timer: fix wrong comment to refer to 'SNDRV_TIMER_PSFLG_*'

ALSA timer core has a comment referring to 'SNDRV_MIXER_PSFLG_*' in

a definition of 'struct snd_timer_params' of UAPI header. I can see

this in initial state of ALSA timer core, at least in

'alsa-driver-0.4.0.tar.gz'.

This commit fixes the comment.

- control: add a series of macro for offset of several types of TLV

In development period for Linux v4.18, a series of SNDRV_CTL_TLVO_XXX

macro was introduced to kernel stuffs for position offset of TLV data.

This commit adds these macros to backport header in this library.

- ASoC: topology: Add definitions for mclk_direction values

Current comment makes not clear the direction of mclk. Previously, similar

description caused a misunderstanding for bclk_master and fsync_master.

This commit solves the potential confusion the same way it is solved for

bclk_master and fsync_master.

- ASoC: topology: Add missing clock gating parameter when parsing hw_configs

Clock gating parameter is a part of `dai_fmt`. It is supported by

`alsa-lib` when creating a topology binary file, but ignored by kernel

when loading this topology file.

After applying this commit, the clock gating parameter is not ignored any

more. This solution is backwards compatible. The existing behaviour is

not broken, because by default the parameter value is 0 and is ignored.

snd_soc_tplg_hw_config.clock_gated = 0 => no effect

snd_soc_tplg_hw_config.clock_gated = 1 => SND_SOC_DAIFMT_GATED

snd_soc_tplg_hw_config.clock_gated = 2 => SND_SOC_DAIFMT_CONT

For example, the following config, based on

alsa-lib/src/conf/topology/broadwell/broadwell.conf, is now supported:

~~~~

SectionHWConfig."CodecHWConfig" {

id "1"

format "I2S" # physical audio format.

pm_gate_clocks "true" # clock can be gated

}

SectionLink."Codec" {

# used for binding to the physical link

id "0"

hw_configs [

"CodecHWConfig"

]

default_hw_conf_id "1"

}

~~~~

Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>

- ASoC: topology: Fix bclk and fsync inversion in set_link_hw_format()

The values of bclk and fsync are inverted WRT the codec. But the existing

solution already works for Broadwell, see the alsa-lib config:

`alsa-lib/src/conf/topology/broadwell/broadwell.conf`

This commit provides the backwards-compatible solution to fix this misuse.

This commit goes in pair with the corresponding patch for linux.

Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>

Tested-by: Pan Xiuli <xiuli.pan@linux.intel.com>

Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>

Test/Example code

- test: use position offset macro of TLV data

A series of SNDRV_CTL_TLVO_XXX macro was introduced for position offset

of TLV data. This commit applies a code optimization.

alsa-utils

Core

- Release v1.1.7

- Release v1.1.6

ALSA Control (alsactl)

- alsactl: lock - fix the array size (gcc warning)

- alsactl: use signalfd to catch UNIX signal

In a mode of 'monitor, event loop runs to dispatch asynchronous event

emitted by control node. In this case, UNIX signal is used to terminate

the event loop.

This commit uses signalfd to catch the UNIX signal.

- alsactl: handle detection of new sound card

At present, plug-and-play is not supported in a mode of 'monitor',

thus new sound card is not handled during runtime. This is not happy.

This commit uses Linux-specific inotify(7) to monitor '/dev/snd'

directory. When some files are newly added to the directory,

event dispatcher is suspended. Event sources are scanned again and the

dispatcher continue to run.

- alsactl: handle disconnection of sound card

Once sound card becomes disconnection state, corresponding control node

becomes to emit error event for listeners. When catching this type of

event, event dispatcher should stop observation of the node. However,

at present, a mode of monitor can't handle this correctly. As a result,

poll(2) is executed quite frequently in loop with no wait. This results

100% consumption of CPU time.

This commit takes the dispatcher to remove the node from observation

list when detecting the disconnection state.

Reported-by: Thomas Gläßle <thomas@coldfix.de>

- alsactl: obsolete array for maintenance of handlers

In former commits, handlers of control node are maintained by link list,

instead of one-dimensional array.

This commit obsoletes the array and split source preparation to a

function.

- alsactl: use a list of source for event dispatcher instead of an array of source

In a previous commit, handlers of control nodes are maintained by link

list.

This commit uses the list to register/unregister event sources to

dispatcher.

- alsactl: use link list to maintain source of events

At present, handlers for control nodes are maintained by one-dimensional

array. This is not necessarily useful to maintain handlers with

associated information.

This commit adds link-list for the maintenance.

- alsactl: use epoll(7) instead of poll(2)

Linux kernel supports unique system call; epoll(7). This allows

applications to make associations for descriptor-unique data in a

easy way.

This commit uses epoll(7) instead of poll(2) for this point.

- alsactl: add an iterator of registered instances of sound card

In a mode of 'monitor', when given no argument, all of available control

node is observed for their events. At present, discovering the nodes is

done according to sound card number, instead of listing nodes in

configuration space of alsa-lib.

This commit adds a structure to discover sound cards with a simple

interface.

- alsactl: split event loop code to a function

In a mode of 'monitor', an event loop runs.

This commit applies a small refactoring to splits the loop into a

function for readability.

Speaker Test

- speaker-test: Allow sampling rates up to 768000

There are audio devices around that support up to 768kHz playback, allow

testing them by increasing the maximum supported sampling rate.

- speaker-test: Remove unused variable

- speaker-test: Support S24_3LE sample format

Implement support signed 24 bit samples, packed in 3 bytes.

alsaloop

- alsaloop: pcmjob - fix few warnings

alsamixer

- alsamixer: fix gcc warnings

alsatplg (topology)

- alsatplg: add man file

alsaucm

- alsatplg: add man file

- alsaucm: add alsa-ucm udev rules for PAZ00 (Toshiba AC100/Dynabook AZ).

aplay/arecord

- aplay: add missing block brackets

- aplay: Fix invalid file size check for non-regular files

aplay tries to check the file size via fstat() at parsing the format

headers and avoids parsing when the size is shorter than the given

size. This works fine for regular files, but when a special file like

pipe is passed, it fails, eventually leading to the fallback mode

wrongly.

A proper fix is to do this sanity check only for a regular file.

Reported-by: Jay Foster <jay@systech.com>

bat (basic audio tester)

- alsabat: Allow custom sample format for round trip latency test

Setting the format to BAT_PCM_FORMAT_S16_LE in the round trip latency

test initialization is redundant as it is already set by default to

BAT_PCM_FORMAT_S16_LE unless a sample format is specified on the command

line.

- bat: alsa.c - move the thread cleanup pop before goto exit3

alsa-tools

Core

- Release v1.1.7

hwmixvolume

- hwmixvolume: mention the new dependencies in the README

Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>

- hwmixvolume: add my copyright

Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>

- hwmixvolume: switch the shebang to python

This signifies that this code is now compatible with both Python 2 and

Python 3, as per PEP-0394[1].

[1] https://www.python.org/dev/peps/pep-0394/

Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>

- hwmixvolume: use a with context to open files

This feature has been added in Python 2.5 and automatically closes an

open file once the context exits.

Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>

- hwmixvolume: switch to GTK+ 3.0 and GLib 2.0

This replaces VBox and HBox with Grid (using Gtk.Orientation), HScale

with Scale, creates labels with mnemonics, set hexpand and vexpand

properly, use the correct enum container classes, use the correct getter

for size request, and finally update to the correct GLib watch function.

Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>

- hwmixvolume: replace PyGTK with gobject-introspection

This doesn’t work yet, we require Gtk 3.0 rather than 2.0 and the API

changed quite a lot, so this is but a preparatory patch.

This is done so that we can get rid of GTK+ 2 which has been EOL for

many years already, and to add Python 3 support because Python 2 will

very soon be EOL as well.

Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>

- hwmixvolume: use four spaces instead of one tab for indent

This is the recommended coding style for all Python programs, as

specified in PEP-0008[1].

[1] https://www.python.org/dev/peps/pep-0008/

Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>

ld10k1 (EMU10K1+ DSP Code Loader)

- ld10k1: define _GNU_SOURCE and use sighandler_t

__sighandler_t is a glibc internal type which doesn't exist in musl. By using

AC_USE_SYSTEM_EXTENSIONS to get _GNU_SOURCE defined, both glibc and musl

then expose sighandler_t.

us428control

- us428control: Fix clang -Wreserved-user-defined-literal warnings

| us428control.cc:66:18: error: invalid suffix on literal; C++11 requires a space between literal and identifier [-Wreserved-user-defined-literal]

| printf("usage: "PROGNAME" [-v verbosity_level 0..2] [-c card] [-D device] [-u usb-device] [-m mode]\n");

| ^

alsa-plugins

Core

- Release v1.1.7

- Fix typo in configure.ac (--disable-libav)

- configure: Require alsa-lib 1.1.6 and newer

JACK plugin requires the recently added API function.

- configure: change --with-alsaaddondir to --with-alsagconfdir and --with-alsalconfdir

The local add-on configuration directory (/etc/alsa/conf.d) contains

links to the global configuration directory (/usr/share/alsa/alsa.conf.d) now.

- Move rate-lavc to rate-lav subdirectory and update to use libavresample

- --disable-avcodec renamed to --disable-libav

- --avcodec-includedir renamed to --libav-includedir

- --avcodec-libdir renamed to --libav-libdir

- --disable-lavcrate renamed to --disable-lavrate

The .c changes are from Anton Khirnov. The rest is from Jaroslav Kysela.

- config/Makefile: make everything modular

- configure: add --with-alsaaddondir, add default config files for plugins

/Makefile.am

- Move rate-lavc to rate-lav subdirectory and update to use libavresample

- --disable-avcodec renamed to --disable-libav

- --avcodec-includedir renamed to --libav-includedir

- --avcodec-libdir renamed to --libav-libdir

- --disable-lavcrate renamed to --disable-lavrate

The .c changes are from Anton Khirnov. The rest is from Jaroslav Kysela.

- config/Makefile: make everything modular

A52 Output plugin

- a52: Fix leaks when a52_set_hw_constraint() fails

If a52_set_hw_constraint() returns error, we leak 'rec' and

'rec->slave'. This patch fixes the issue by jumping to 'error' label

where the proper clean up is already done.

Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>

- configure: change --with-alsaaddondir to --with-alsagconfdir and --with-alsalconfdir

The local add-on configuration directory (/etc/alsa/conf.d) contains

links to the global configuration directory (/usr/share/alsa/alsa.conf.d) now.

- Move rate-lavc to rate-lav subdirectory and update to use libavresample

- --disable-avcodec renamed to --disable-libav

- --avcodec-includedir renamed to --libav-includedir

- --avcodec-libdir renamed to --libav-libdir

- --disable-lavcrate renamed to --disable-lavrate

The .c changes are from Anton Khirnov. The rest is from Jaroslav Kysela.

- configure: add --with-alsaaddondir, add default config files for plugins

Arcam AV control plugin

- arcam-av: Add explicit -lpthread to *_LIBADD

The pthread library has to be linked for this module. In some tool

chains, it results in a build error.

- configure: change --with-alsaaddondir to --with-alsagconfdir and --with-alsalconfdir

The local add-on configuration directory (/etc/alsa/conf.d) contains

links to the global configuration directory (/usr/share/alsa/alsa.conf.d) now.

- configure: add --with-alsaaddondir, add default config files for plugins

Documentation

- doc: fix makefile, the files should be included to the distribution package unconditionally

- Move rate-lavc to rate-lav subdirectory and update to use libavresample

- --disable-avcodec renamed to --disable-libav

- --avcodec-includedir renamed to --libav-includedir

- --avcodec-libdir renamed to --libav-libdir

- --disable-lavcrate renamed to --disable-lavrate

The .c changes are from Anton Khirnov. The rest is from Jaroslav Kysela.

- config/Makefile: make everything modular

Jack PCM plugin

- jack: Fix leaks when jack_set_hw_constraint() fails

If jack_set_hw_constraint() returns error, we leak the 'jack' object and

all the resources referenced by it. This patch fixes the issue by

calling snd_pcm_jack_free() before returning.

Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>

- jack: Only allow ALSA periods multiple of JACKd period

There is a higher Xrun probability whenever the ALSA period is not a

multiple of the JACKd period and the ALSA buffer is only twice the period.

Allowing ALSA buffers of min. 3x period is not a good solution because this

would increase the latency.

As an example in case of ALSA is using a period of 11 frames and JACK a

period of 12 frames and a buffer of 22 frames (2x ALSA period) is used:

- The buffer is filled and contains 22 frames

- JACK is reading 12 frames. Buffer is only containing 10 frames

- Now, ALSA has to be scheduled exactly after the last JACK read and

before the next read. Otherwise we will get an Xrun

In case of ALSA is using a multiple period of JACK, JACK can always read

2 periods before an Xrun if the buffer was full.

In case of ALSA is using a period of 12 frames and JACK a period of 11

frames and a buffer of 24 frames (2x ALSA period) is used:

- The buffer is filled and contains 24 frames

- JACK is reading 11 frames. Buffer is now containing 13 frames. But ALSA

can still not write to the buffer

- JACK reads 11 additional frames. Buffer contains 2 frames

- Now, ALSA has to be scheduled exactly after the last JACK read and

before the next read. Otherwise we will get an Xrun

In case of ALSA is using a multiple period of JACK, ALSA can always write

to the buffer as long as the buffer is filled with less or equal of half

of buffer.

Therefore this patch enforces an ALSA period which is always a multiple

of the JACKd period.

- jack: Allow build with older alsa-lib

The new API snd_pcm_ioplug_avail() doesn't appear prior to alsa-lib

1.1.7. Add an own code for allowing the build with alsa-lib 1.1.6.

It has a very slight overhead than the original snd_pcm_ioplug_avail()

but should be OK as a workaround.

- jack: Update poll_fd also in draining state

... to support snd_pcm_drain for the JACK IO plugin.

With this changes there will be an poll_fd event in DRAINING state even if

the min_avail was not yet reached. Otherwise the application would never

recognize that all samples were processed by JACK.

In addition the JACK real-time thread is also processing when in DRAINING

state and not only when in RUNNING or PREPARE state.

- jack: Avoid call to snd_pcm_avail_update() from JACK thread

snd_pcm_avail_update() can call snd_pcm_jack_stop() but

snd_pcm_jack_stop() should not be called by the JACK thread.

It should only be called by the thread how has called

snd_pcm_jack_start().

In addition the execution of snd_pcm_avail_update() can take a while.

Therefore it should not be called by the JACK thread to not block this

thread.

- configure: change --with-alsaaddondir to --with-alsagconfdir and --with-alsalconfdir

The local add-on configuration directory (/etc/alsa/conf.d) contains

links to the global configuration directory (/usr/share/alsa/alsa.conf.d) now.

- configure: add --with-alsaaddondir, add default config files for plugins

OSS Mixer -> ALSA Control plugin

- oss: Fix leaks when oss_hw_constraint() fails

If oss_hw_constraint() returns error, we leak 'oss' object and all the

resources referenced by it. This patch fixes the issue by jumping to

'error' label where the proper clean up is already done.

Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>

- oss/Makefile.am: fix typo

Rate Converter plugins

- samplerate: fix unused variable warning

USB stream plugin

- usb_stream: Fix leaks when us_set_hw_constraint() fails

If us_set_hw_constraint() returns error, we leak the 'us' object and all

the resources referenced by it. This patch fixes the issue by calling

us_free() before returning.

Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>

- usb_stream: Add explicit -lpthread to *_LIBADD

The pthread library has to be linked for this module. In some tool

chains, it results in a build error.

tinycompress

Core

- Release v1.1.7

Library

- tinycompress: Update email address for Vinod

- compress: remove dead code _is_codec_supported()

The _is_codec_supported() was dead and none using it, so remove this and

eliminate unused function warning

compress.c:145:13: warning: ‘_is_codec_supported’ defined but not used [-Wunused-function]

We can take from git if user appears

Utilities

- tinycompress: Update email address for Vinod

- crecord: fix incorrect print specifier warning

We get a warning in crecord for incorrect print specifier, fix it by using

right one %u for unsigned int

crecord.c:380:17: warning: format ‘%lu’ expects argument of type ‘long unsigned int’, but argument 6 has type ‘unsigned int’ [-Wformat=]

fprintf(finfo, "Recording file %s On Card %u device %u, with buffer of %lu bytes\n",

- cplay: fix incorrect print specifier warning

We get a warning in cplay for incorrect print specifier, fix it by using

right one %ld for long unsigned int.

cplay.c:327:19: warning: format ‘%d’ expects argument of type ‘int’, but argument 3 has type ‘long unsigned int’ [-Wformat=]

fprintf(stderr, "codec ID %d is not supported\n", codec_id);

- cplay: remove dead code check_codec_format_supported()

The check_codec_format_supported() was added but never used so remove

it and eliminate unused function warning

cplay.c:103:20: warning: ‘codec_name_from_id’ defined but not used [-Wunused-function]

We can take from git if user appears

- cplay: remove dead code codec_name_from_id()

The codec_name_from_id() was added but never used so remove it and

eliminate unused function warning

cplay.c:103:20: warning: ‘codec_name_from_id’ defined but not used [-Wunused-function]

We can take from git if user appears

- crecord: Use whole buffer requested by user

Currently, crecord will only actually use 1 fragment worth of its

internal buffer no matter what the user requests. Correct this so that

the whole buffer requested by the user is used for copying data.