Commit Graph

18901 Commits

Author SHA1 Message Date
Takashi Sakamoto
326b9cacf4 ALSA: bebob: Add support for Terratec PHASE, EWS series and Aureon
This commit allows this driver to support all of models which Terratec produced
with DM1000/BeBoB. They are:
 - PHASE 24 FW
 - PHASE X24 FW
 - PHASE 88 Rack FW
 - EWS MIC2
 - EWS MIC4
 - Aureon 7.1 Firewire

For Phase series, this commit adds a Terratec specific operation. To get source
of clock. AV/C Audio Subunit command is used.

For EWS series and Aureon, this module uses normal operations.

Tested-by: Maximilian Engelhardt <maxi@daemonizer.de> (PHASE 24 FW)
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:31:25 +02:00
Takashi Sakamoto
1fc9522a08 ALSA: bebob: Prepare for device specific operations
This commit is for some devices which have its own operations or quirks.

Many functionality should be implemented in user land. Then this commit adds
functionality related to stream such as sampling frequency or clock source. For
help to debug, this commit adds the functionality to get metering information
if it's available.

To help these functionalities, this commit adds some AV/C commands defined in
'AV/C Audio Subunit Specification (1394TA).

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:31:15 +02:00
Takashi Sakamoto
618eabeae7 ALSA: bebob: Add hwdep interface
This interface is designed for mixer/control application. By using hwdep
interface, the application can get information about firewire node, can
lock/unlock kernel streaming and can get notification at starting/stopping
kernel streaming.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:31:03 +02:00
Takashi Sakamoto
fbbebd2c40 ALSA: bebob: Add PCM interface
This commit adds a functionality to capture/playback PCM samples.

When AMDTP stream is already running for PCM or the source of clock is not
internal, available sampling rate is limited at current one.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:30:46 +02:00
Takashi Sakamoto
248b78027d ALSA: bebob: Add MIDI interface
This commit adds a functionality to capture/playback MIDI messages.

When no AMDTP streams are running, this module starts AMDTP stream at current
sampling rate for MIDI substream.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:30:16 +02:00
Takashi Sakamoto
ad9697bad7 ALSA: bebob: Add proc interface for debugging purpose
This commit adds proc interface to get these information for debugging:
 - firmware information
 - stream formation
 - current clock source and sampling rate

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:30:00 +02:00
Takashi Sakamoto
b6bc812327 ALSA: bebob/firewire-lib: Add a quirk for discontinuity at bus reset
Normal BeBoB firmware has a quirk. When receiving bus reset, it transmits
packets with discontinuous value in dbc field.

This causes two situation, one is to abort streaming by firewire-lib as a
result of detecting the discontinuity. Another is to call driver's .update()
because of bus reset. These two is generated independently. (The former
depends on isochronous stream and the latter depends on IEEE1394 bus driver.)

When BeBoB driver works with XRUN-recoverable applications, this situation
looks like stream_start_duplex() call followed by stream_update_duplex() call
because applications will call snd_pcm_prepare() immediately at XRUN.

To update connections and streams at first, this commit use completion. When
queueing error occurs, stream_start_duplex() is forced to wait maximum
1000msec. During this, when .update() is called, the completion is waken and
stream_start_duplex() is processed without breaking connections.

At bus reset, stream_start_duplex() shouldn't break/establish connections and
stream_update_duplex() should update connections because a caller of
fw_iso_resources_allocate() is responsible for calling
fw_iso_resources_update() on bus reset.

This commit also adds a flag, which has an effect to skip checking continuity
for first packet. This flag is useful for BeBoB quirk to start handling packets
during streaming.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:29:44 +02:00
Takashi Sakamoto
eb7b3a056c ALSA: bebob: Add commands and connections/streams management
This commit adds management functionality for connections and streams.
BeBoB uses CMP to manage connections and uses AMDTP for streams.

This commit also adds some BridgeCo's AV/C extension commands. There are some
BridgeCo's AV/C extension commands but this commit just uses below commands
to get device's capability and status:

 1.Extended Plug Info commands
  - Plug Channel Position Specific Data
  - Plug Type Specific Data
  - Cluster(Section) Info Specific Data
  - Plug Input Specific Data
 2.Extended Stream Format Information commands
  - Extended Stream Format Information Command - List Request

For Extended Plug Info commands for Cluster Info Specific Data, I pick up
'section' instead of 'cluster' from document to prevent from misunderstanding
because 'cluster' is also used in IEC 61883-6.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:29:29 +02:00
Takashi Sakamoto
fd6f4b0dc1 ALSA: bebob: Add skelton for BeBoB based devices
This commit adds a new driver for BeBoB based devices with no specific
operations. Currently this driver just create/remove card instance according
to callbacks.

BeBoB is 'BridgeCo enhanced Breakout Box'. This is installed to firewire
devices with DM1000/DM1100/DM1500 chipset. It gives common way for host
system to handle BeBoB based devices.

Current supported devices:
 - Edirol FA-66/FA-101
 - PreSonus FIREBOX/FIREPOD/FP10/Inspire1394
 - BridgeCo RDAudio1/Audio5
 - Mackie Onyx 1220/1620/1640 (Firewire I/O Card)
 - Mackie d.2 (Firewire Option)
 - Stanton FinalScratch 2 (ScratchAmp)
 - Tascam IF-FW DM
 - Behringer XENIX UFX 1204/1604
 - Behringer Digital Mixer X32 series (X-UF Card)
 - Apogee Rosetta 200/Rosetta 400 (X-FireWire card)
 - Apogee DA-16X/AD-16X/DD-16X (X-FireWire card)
 - Apogee Ensemble
 - ESI Quotafire610
 - AcousticReality eARMasterOne
 - CME MatrixKFW
 - Phonix Helix Board 12 MkII/18 MkII/24 MkII
 - Phonic Helix Board 12 Universal/18 Universal/24 Universal
 - Lynx Aurora 8/16 (LT-FW)
 - ICON FireXon
 - PrismSound Orpheus/ADA-8XR

Devices possible to be supported if identifying IDs:
 - Apogee Mini-Me Firewire/Mini-DAC Firewire
 - Behringer F-Control Audio 610/1616
 - Cakewalk Sonar Power Studio 66
 - CME UF400e
 - ESI Quotafire XL
 - Infrasonic DewX/Windy6
 - Mackie Digital X Bus x.200/400
 - Phonic Helix Board 12/18/24
 - Phonic FireFly 202/302
 - Rolf Spuler Firewire Guitar

Tested-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:29:12 +02:00
Takashi Sakamoto
555e8a8f7f ALSA: fireworks: Add command/response functionality into hwdep interface
This commit adds two functionality for hwdep interface, adds two parameters for
this driver, add a node for proc interface.

To receive responses from devices, this driver already allocate own callback
into initial memory space in host controller. This means no one can allocate
its own callback to the address. So this driver must give a way for user
applications to receive responses.

This commit adds a functionality to receive responses via hwdep interface. The
application can receive responses to read from this interface. To achieve this,
this commit adds a buffer to queue responses. The default size of this buffer is
1024 bytes. This size can be changed to give preferrable size to
'resp_buf_size' parameter for this driver. The application should notice rest
of space in this buffer because this driver don't push responses when this
buffer has no space.

Additionaly, this commit adds a functionality to transmit commands via hwdep
interface. The application can transmit commands to write into this interface.
I note that the application can transmit one command at once, but can receive
as many responses as possible untill the user-buffer is full.

When using these interfaces, the application must keep maximum number of
sequence number in command within the number in firewire.h because this driver
uses this number to distinguish the response is against the command by the
application or this driver.

Usually responses against commands which the application transmits are pushed
into this buffer. But to enable 'resp_buf_debug' parameter for this driver, all
responses are pushed into the buffer. When using this mode, I reccomend to
expand the size of buffer.

Finally this commit adds a new node into proc interface to output status of the
buffer.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:28:58 +02:00
Takashi Sakamoto
594ddced82 ALSA: fireworks: Add hwdep interface
This interface is designed for mixer/control application. To use hwdep
interface, the application can get information about firewire node, can
lock/unlock kernel streaming and can get notification at starting/stopping
kernel streaming.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:28:41 +02:00
Takashi Sakamoto
aa02bb6e60 ALSA: fireworks: Add PCM interface
This commit adds a functionality to capture/playback PCM samples.

When AMDTP stream is already running for PCM or the source of clock is not
internal, available sampling rate is limited at current one.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:28:27 +02:00
Takashi Sakamoto
53111cdc53 ALSA: fireworks/firewire-lib: Add a quirk of data blocks for MIDI in out-stream
Fireworks has a quirk to ignore MIDI messages in data blocks more than 8.
This commit adds a flag for this quirk and codes to skip 8 or more data
blocks to transfer MIDI messages.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:28:14 +02:00
Takashi Sakamoto
a63d3ff105 ALSA: fireworks: Add MIDI interface
This commit adds a functionality to capture/playback MIDI messages.

When no AMDTP streams are running, this driver starts AMDTP stream for MIDI
stream at current sampling rate.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:28:01 +02:00
Takashi Sakamoto
6a22683e89 ALSA: fireworks: Add proc interface for debugging purpose
This commit adds proc interface to output infomation for debugging.
 - firmware information
 - sampling rate and clock source
 - physical metering (linear value)

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:27:47 +02:00
Takashi Sakamoto
b84b1a27b4 ALSA: fireworks/firewire-lib: Add a quirk to reset data block counter at bus reset
Fireworks has a quirk to reset data block counter at bus reset.

This commit adds a flag of CIP_SKIP_DBC_ZERO_CHECK. This flag has an effect
to skip checking dbc continuity when dbc is zero.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:26:44 +02:00
Takashi Sakamoto
d9cd0065c8 ALSA: fireworks/firewire-lib: Add a quirk for fixed interval of reported dbc
Fireworks firmware version 5.5 reports fix interval for dbc in each packet.

For example, AudioFire4:
CIP0     CIP1     Payload
00070000 900484FF 72
00070008 9004A8FF 72
00070008 90FFFFFF 02
00070010 9004D0FF 72
00070018 9004C4FF 72
00070020 9004E8FF 72
00070020 90FFFFFF 02
00070028 900410FE 72

The interval of each dbc should be 16 except for empty packet but it's still 8.

This commit adds a flag for this quirk and codes to refer to a fixed value.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:25:15 +02:00
Takashi Sakamoto
697022391e ALSA: fireworks/firewire-lib: Add a quirk for wrong dbs in tx packets
One of Fireworks firmware, named  as 'AudioFire9', seems to transmit
packets with wrong value of dbs. It's always 0x11 but actual size of
data block is different.

This commit adds a flag for this quirk and some codes to calculate
correct size.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:25:00 +02:00
Takashi Sakamoto
c8bdf49b99 ALSA: fireworks/firewire-lib: Add a quirk for the meaning of dbc
Fireworks has a quirk for the value of dbc field in transmitted packets.
For Fireworks, dbc means the end of events in current packet. This is out
of specification.

For example, AudioFire4:
CIP0        CIP1    Payload
01070092 90FFFFFF 02
0107009A 9001E17B 3A <-
010700A2 9001F6E5 3A
010700A2 90FFFFFF 02
010700AA 9001104F 3A <-
010700B2 900125B9 3A
010700BA 90013B23 3A
010700BA 90FFFFFF 02
010700C2 9001548E 3A <-
010700CA 900169F8 3A
010700CA 90FFFFFF 02
010700D2 90018362 3A <-
010700DA 900198CC 3A

According to IEC 61883-1/6, a packet following to empty packet has the same
value for its dbc. But for Fireworks, it's incremented and empty packet has
the same value as previous packet in dbc field.

This commit adds a flag for Fireworks and some codes to checking dbc continuity.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:24:47 +02:00
Takashi Sakamoto
7ab566453f ALSA: fireworks/firewire-lib: Add a quirk for empty packet with TAG0
Fireworks has a quirk to transmit empty packets with TAG0. This commit
adds handling this quirk for full duplex stream synchronization.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:24:33 +02:00
Takashi Sakamoto
315fd41fe9 ALSA: fireworks: Add connection and stream management
Fireworks manages connections by CMP and can transmit/receive AMDTP streams
with a few quirks. This commit adds functionality to start/stop the streams.

Major Fireworks products don't support 'SYT-Match' clock source mode, except
for AudioFire12/8(till 2009 July) with firmware version 1.0. Already in
previous commit, this driver don't support such old firmwares. So this commit
adds support for non 'SYT-Match' clock source modes.

I note that this driver has a short gap for MIDI streams when starting PCM
stream. When AMDTP streams are running only for MIDI data and PCM data is
going to be joined at different sampling rate, then AMDTP streams are
stopped once and started again after changing sampling rate.

Unfortunately, Fireworks is not fully compliant to IEC 61883-1/6. Some commits
following to this commit add these quirks.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:24:19 +02:00
Takashi Sakamoto
bde8a8f23b ALSA: fireworks: Add transaction and some commands
Fireworks uses own command and response. This commit adds functionality to
transact and adds some commands required for sound card instance and kernel
streaming.

There are two ways to deliver substance of this transaction:
1.AV/C vendor dependent command for command/response
2.Async transaction to specific addresses for command/response

By way 1, I confirm AudioFire12 cannot correctly response to some commands with
firmware version 5.0 or later. This is also confirmed by FFADO. So this driver
implement way 2.

The address for response gives an issue. When this driver allocate own callback
function into the address, then no one can allocate its own callback function.
This situation is not good for applications in user-land. This issue is solved
in later commit.

I note there is a command to change the address for response if the device
supports. But this driver uses default value. So users should not execute this
command as long as hoping this driver works correctly.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:24:03 +02:00
Takashi Sakamoto
b5b0433601 ALSA: fireworks: Add skelton for Fireworks based devices
This commit adds a new driver for devices based on Fireworks. This driver
just creates/removes card instance according to callbacks.

Fireworks is a board module which Echo Audio produced. This module
consists of three chipsets:
 - Communication chipset for IEEE1394 PHY/Link and IEC 61883-1/6
 - DSP or/and FPGA for signal processing
 - Flash Memory to store firmwares

Current supported devices:
 - Mackie Onyx 400F/1200F
 - Echo AudioFire12/8(until 2009 July)
 - Echo AudioFire2/4/Pre8/8(since 2009 July)
 - Echo Fireworks 8/HDMI
 - Gibson Robot Interface pack/GoldTop

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:23:36 +02:00
Takashi Sakamoto
1017abed18 ALSA: firewire-lib: Add some AV/C general commands
This commit adds three commands, which may be used by some firewire device
drivers. These commands are defined in 'AV/C Digital Interface Command Set
General Specification Version 4.2 (2004006, 1394TA)'.

1. PLUG INFO command (clause 10.1)
2. INPUT PLUG SIGNAL FORMAT command (clause 10.10)
3. OUTPUT PLUG SIGNAL FORMAT command (clause 10.11)

By the command 1, the drivers can get the number of plugs for AV/C unit or
subunit.
By the command 2 and 3, the drivers can get/set sampling frequency.

The 'firewire-speakers' already uses INPUT PLUG SIGNAL FORMAT command to set
sampling rate. So this commit also affects the driver.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:23:13 +02:00
Takashi Sakamoto
00a7bb81c2 ALSA: firewire-lib: Add support for deferred transaction
Some devices based on BeBoB use this type of AV/C transaction.

'Deferred Transaction' is defined in 'AV/C Digital Interface Command Set
General Specification' and is used by targets to make a response deferred
during processing it.

If a target may not be able to complete a command within 100msec since
receiving the command, then the target shall return INTERIM response,
to which final response will follow later. CONTROL/NOTIFY commands are
allowed for deferred transaction.

In the specification, devices allow to send INTERIM response just one time.
But this commit allows to handle several INTERIM response with two reasons.
One reason is to simplify codes, and another reason is to prepare for
devices which is out of specification.

There is an issue. In the specification, the interval between INTERIM
response and final response is 'Unspecified interval'. The specification
depends on each subunit specification for this interval.

But we promise to finish this function for caller. In this reason, I use
FCP_TIMEOUT_MS for this interval. Currently it's 125msec. When we find
devices which needs more time for this interval, then let us add some codes
to apply more interval for 'Unspecified interval'.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:22:56 +02:00
Takashi Sakamoto
b04479fb85 ALSA: firewire-lib: Add a new function to check others' connection
Plug Control Registers have two fields related to the number of established
connections, one is 'Broadcast connection counter' and another is
'Point-to-point connection counter'. The driver can know there are established
connections or not to check these fields.

This commit is for considering about JACK/FFADO streaming. Currently, when
JACK/FFADO starts its streaming to the device, cmp_connection_establish() is
failed expectedly. This seems to be enough but there are some devices which
needs to change sampling frequency before trying to establish connections.
For such devices, this functionality is needed.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:22:46 +02:00
Takashi Sakamoto
44aff6980a ALSA: firewire-lib: Add handling output connection by CMP
This patch adds some macros, codes with condition of direction and new functions
to handle output connection. Once cmp_connection_init() is executed with its
direction, CMP input and output connection can be handled by the same way.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:22:37 +02:00
Takashi Sakamoto
c68a1c6584 ALSA: firewire-lib: Add 'direction' member to 'cmp_connection' structure
This patch adds 'direction' member to 'cmp_connection' structure to indicate
the direction of connection. This patch also adds 'direction' argument to
cmp_connection_init() function to determine the direction.

The cmp_connection_init() function is exported and used in snd-firewire-speakers
so this patch also affect it.

This patch just add them. Actual implementation will be done by followed
patches.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:22:14 +02:00
Takashi Sakamoto
a7fa0d047f ALSA: firewire-lib: Rename macros, variables and functions for CMP
Referring to IEC 61883-1, oMPR and iMPR, oPCR and iPCR have some fields with
the same role in the same position. This patch renames some macros, variables
and function arguments with "i" in its prefix to reuse them between oMPR and
iMPR, oPCR and iPCR.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:21:57 +02:00
Takashi Sakamoto
c8de6dbbbb ALSA: firewire-lib: Restrict calling flush_context_completion() when context exists
Currently, drivers can bring XRUN state for PCM substreams when error to
queue packets or detecting discontinuity of packet. The application may try to
recover this state by calling snd_pcm_prepare().

Depending on each driver, .prepare() includes restart streaming. Then there
is a state that PCM substreams are running but isochronous contexts are
stopped. In this case, when .pointer() is called, it refers to error pointer.

This commit is for a prevention of this bug.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:21:56 +02:00
Takashi Sakamoto
7b2d99fa6b ALSA: firewire-lib/dice/speakers: Add common PCM constraints for AMDTP streams
This commit adds common PCM constraints according to current firewire-lib
implementation.

1.Maximum width for each sample is limited by 24.
AM824 in IEC 61883-6 can deliver 24bit data.

2. Minimum time for period is 5msec.
Apply the old value. For shorter latency, further works are needed.

3. In blocking mode, frames per period/buffer is aligned to 32.
Each packet can include some frames depending on its sampling rate. In
blocking mode, the number equals to SYT_INTERVAL. Currently firewire-lib
can schedule snd_pcm_period_elapsed() for each packet. So, for accurate
PCM interrupt, the number of frames per period/buffer should be aligned
to SYT_INTERVAL.
Currently firewire-lib is lack of better rules to achieve this. So LCM of
each value of SYT_INTERVALs (=32) is applied. This can be improved for
further work.

[Fixed the compile error due to the missing "&" by tiwai]

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:21:46 +02:00
Takashi Sakamoto
10550bea44 ALSA: dice/firewire-lib: Keep dualwire mode but obsolete CIP_HI_DUALWIRE
In previous commit, AMDTP functionality in firewire-lib supports mapping
for PCM data channels. With this mapping, firewire-lib can obsolete
a flag, CIP_HI_DUALWIRE, but Dice driver still keeps dual wire mode.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:15:10 +02:00
Takashi Sakamoto
77d2a8a4f6 ALSA: firewire-lib: Add support for channel mapping
Some devices arrange the position of PCM/MIDI data channel in AMDTP packet.
This commit allows drivers to set channel mapping.

To be simple, the mapping table is an array with fixed length. Then the number
of channels for PCM is restricted by 64 channels.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:14:41 +02:00
Takashi Sakamoto
7b3b0d8583 ALSA: firewire-lib: Add support for duplex streams synchronization in blocking mode
Generally, the devices can synchronize to handle 'presentation timestamp'
in CIP packets. This commit adds functionality to pick up this timestamp from
in-packets transmitted by the device, then use it for out packets.

In current implementation, this module generated the timestamp by itself. This
is 'SYT Match' mode. Then drivers with this module acts as synchronization
master. This commit allows this module to act as synchronization slave.

This commit restricts this mechanism is only available in blocking mode because
handling the timestamp in non-blocking mode is more complicated than in
blocking mode.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:13:59 +02:00
Takashi Sakamoto
ccccad8646 ALSA: firewire-lib: Give syt value as parameter to handle_out_packet()
For duplex streams with synchronization, drivers should pick up
'presentation timestamp' from in-packets and use the timestamp for
out-packets. This commit is preparation for this.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:13:44 +02:00
Takashi Sakamoto
83d8d72dff ALSA: firewire-lib: Add support for MIDI capture/playback
For capturing/playbacking MIDI messages, this commit adds one MIDI conformant
data channel. This data channel has multiplexed 8 MIDI data streams. So this
data channel can transfer messages from/to 8 MIDI ports.

And this commit allows to set PCM format even if AMDTP streams already start.
I suppose the case that PCM substreams are going to be joined into AMDTP
streams when AMDTP streams are already started for MIDI substreams. Each
driver must count how many PCM/MIDI substreams use AMDTP streams to stop
AMDTP streams.

There are differences between specifications about MIDI conformant data.

About the multiplexing, IEC 61883-6:2002, itself, has no information. It
describes labels and bytes for MIDI messages and refers to MMA/AMEI RP-027
for 'successfull implementation'. MMA/AMEI RP-027 describes 8 MPX-MIDI data
streams for one MIDI conformant data channel. IEC 61883-6:2005 adds
'sequence multiplexing' and apply this way and describe incompatibility
between 2002 and 2005.

So this commit applies IEC 61883-6:2005. When we find some devices compliant
to IEC 61883-6:2002, then this difference should be handles as device quirk
in additional work.

About the number of bytes in an MIDI conformant data, IEC 61883-6:2002 describe
0,1,2,3 bytes. MMA/AMEI RP-027 describes 'MIDI1.0-1x-SPEED', 'MIDI1.0-2x-SPEED',
'MIDI1.0-3x-SPEED' modes and the maximum bytes for each mode corresponds to 1,
2, 3 bytes. The 'MIDI1.0-2x/3x-SPEED' modes are accompanied with 'negotiation
procedure' and 'encapsulation details' but there is no specifications for them.

So this commit implements 'MIDI1.0-1x-SPEED' mode for playback, but allows
to pick up 1-3 bytes for capturing.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:12:44 +02:00
Takashi Sakamoto
2b3fc456fe ALSA: firewire-lib: Add support for AMDTP in-stream and PCM capture
For capturing PCM, this commit adds the functionality to handle in-stream.
This is also applied for dual-wire mode.

Currently, capturing 32bit samples are supported.

When the sequence of in-packet has discontinuity of dbc, in-stream isn't handled
and amdtp_streaming_error() returns true.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:12:35 +02:00
Takashi Sakamoto
4b7da117e5 ALSA: firewire-lib: Split some codes into functions to reuse for both streams
Some codes can be reused to handle in-stream. This commit adds new functions.
This commit also renames some functions to keep naming consistency.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:11:57 +02:00
Takashi Sakamoto
3ff7e8f0d4 ALSA: firewire-lib: Add 'direction' member to 'amdtp_stream' structure
This patch adds 'direction' member to amdtp_stream structure to indicate its
direction. This patch also adds 'direction' argument to amdtp_stream_init()
function to determine its direction.

The amdtp_stream_init() function is exported and used by firewire-speakers and
dice so this patch also affects them.

This patch just add them. Actual implementation will be done by followed
patches.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:11:42 +02:00
Takashi Sakamoto
b445db440c ALSA: firewire-lib: Add macros instead of fixed value for AMDTP
This patch adds some macros instead of fixed value for AMDTP according to
IEC 61883-1/6. These macros will also be used by followed patches.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:11:22 +02:00
Takashi Sakamoto
be4a28940a ALSA: firewire-lib: Rename functions, structure, member for AMDTP
This patch renames some functions, a structure and its member to reuse them
in both AMDTP in/out stream.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:11:10 +02:00
Hui Wang
e191893830 ALSA: hda - add an instance to use snd_hda_pick_pin_fixup
Just two members in the alc269_pin_fixup_tbl[] can cover more than
10 Dell laptop models.

Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 11:06:22 +02:00
Hui Wang
c687200b9d ALSA: hda - drop def association and sequence from pinconf comparing
A lot a machine have the same codec, but they have different default
pinconf setting just because the def association and sequence is
different, as a result they can't share a hda_pintbl[], to overcome
it, we don't compare def association and sequence in the pinconf
matching.

Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 11:05:55 +02:00
Hui Wang
621b5a047e ALSA: hda - get subvendor from codec rather than pci_dev
It is safer for non-pci situation.

Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 11:05:26 +02:00
David Henningsson
20531415ad ALSA: hda - Add a new quirk match based on default pin configuration
Normally, we match on pci ssid only. This works but needs new code
for every machine. To catch more machines in the same quirk, let's add
a new type of quirk, where we match on
 1) PCI Subvendor ID (i e, not device, just vendor)
 2) Codec ID
 3) Pin configuration default

If all these three match, we could be reasonably certain that the
quirk should apply to the machine even though it might not be the
exact same device.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 11:03:53 +02:00
David Henningsson
c21c8cf77f ALSA: hda - Add fixup_forced flag
The "fixup_forced" flag will indicate whether a specific fixup
(or nofixup) has been set by the user, to override the driver's
default.
This flag will help future patches.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 11:03:38 +02:00
Daniel Mack
a860d95f74 ALSA: snd-usb: mixer: remove error messages on failed kmalloc()
If kmalloc() fails, warnings will be loud enough. We can safely just
return -ENOMEM in such cases.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-25 09:09:01 +02:00
Daniel Mack
6bc170e4e8 ALSA: snd-usb: mixer: coding style fixups
Shorten some over-long lines, multi-line comments, spurious whitespaces,
curly brakets etc.  No functional change.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-25 09:08:46 +02:00
Takashi Iwai
77f07800cb ALSA: hda - Fix onboard audio on Intel H97/Z97 chipsets
The recent Intel H97/Z97 chipsets need the similar setups like other
Intel chipsets for snooping, etc.  Especially without snooping, the
audio playback stutters or gets corrupted.  This fix patch just adds
the corresponding PCI ID entry with the proper flags.

Reported-and-tested-by: Arthur Borsboom <arthurborsboom@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-23 09:09:26 +02:00
Sylwester Nawrocki
a6aba536ab ASoC: samsung: Handle errors when getting the op_clk clock
Ensure i2s->op_clk is not used when clk_get() for this clock fails.
This prevents working with an incorrectly configured clock in some
conditions.

Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-22 17:57:27 +01:00
Takashi Iwai
0c1d121016 ASoC: Updates for v3.16
Lots of cleanup work going on in the core this release but very little
 visible to external users except for the new drivers that have been
 added.
 
  - Support for specifying aux CODECs in DT.
  - Removal of the deprecated mux and enum macros.
  - More moves towards full componentisation.
  - Removal of some unused I/O code.
  - Lots of cleanups, fixes and enhancements to the davinci, Freescale,
    Haswell and Realtek drivers.
  - Several drivers exposed directly in Kconfig for use with simple-card.
  - New drivers for Cirrus CS42L56, Realtek RT5639, RT5642 and RT5651 and
    ST STA350.
 -----BEGIN PGP SIGNATURE-----
 Version: GnuPG v1
 
 iQIcBAABAgAGBQJTfhiIAAoJELSic+t+oim9jy0P/jcbZQkwBD23UWDbtUph5hRr
 O7tQ832exPbpFAKsxybFSt67H2yUb4iaGFy9hwM8GSKWzay23FAO7bw7Hxtd9tUC
 F7xPc7iqbk///aGpviHQAFV9wCtUB0jxz/fyUNPYFi9ZvhyYXzxa0xi7OfhCwuY/
 mb2LoaZ0EcpESyxwMQbVQhh22ZUXlf7BXEZuybSWFyH4DiyX/F6atSRWzki3mEMR
 1x7GnDr0jsCxQHJMfLYzSzYQXlj7F99iXqXCNUmV+sVuJdMHV8rFdsKt4x0YnfEp
 kQH1XE6pwyxgL6fGdN1FbJTE2pDqfsUh5IiJ5fIWVfKwY0dtJzn9CMsCVr7XLIsc
 RDa/ATzeeXWc/KQEVraHJd6BEr/1XfDRGoJIUA9qSsEg+xhHOmG0OiaHjr+/ci8R
 KcmIj3ETWh61V48sJeG3U1EArUqUyupufmYZdrzPGmhgFjxmgyKQB/8E1k5IPJCh
 TlFko05Xy+A1XTv54nrhJyM+zsFtbrmhkdFqUOSZINHzRAWv6/Y/55Q6HfVuuyPO
 7p0hBapp4EqyKT4gch7qd2yLGyVn/SYG9lusp1I8wcPWOX7g5iHJzs3VLc/4gLZs
 FJKsLxeUy7AWVWLVBMnJqgf1WBMQs+lkeOwB4MwZsJ0slYMezkI/EO8AxyMULr2w
 ilpbOkn3TZvqmAaRTPxG
 =OKes
 -----END PGP SIGNATURE-----

Merge tag 'asoc-v3.16' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next

ASoC: Updates for v3.16

Lots of cleanup work going on in the core this release but very little
visible to external users except for the new drivers that have been
added.

 - Support for specifying aux CODECs in DT.
 - Removal of the deprecated mux and enum macros.
 - More moves towards full componentisation.
 - Removal of some unused I/O code.
 - Lots of cleanups, fixes and enhancements to the davinci, Freescale,
   Haswell and Realtek drivers.
 - Several drivers exposed directly in Kconfig for use with simple-card.
 - New drivers for Cirrus CS42L56, Realtek RT5639, RT5642 and RT5651 and
   ST STA350.
2014-05-22 17:50:00 +02:00
Benoit Taine
6f51f6cf68 ALSA: Replace DEFINE_PCI_DEVICE_TABLE macro use
We should prefer `const struct pci_device_id` over
`DEFINE_PCI_DEVICE_TABLE` to meet kernel coding style guidelines.
This issue was reported by checkpatch.

A simplified version of the semantic patch that makes this change is as
follows (http://coccinelle.lip6.fr/):

// <smpl>

@@
identifier i;
declarer name DEFINE_PCI_DEVICE_TABLE;
initializer z;
@@

- DEFINE_PCI_DEVICE_TABLE(i)
+ const struct pci_device_id i[]
= z;

// </smpl>

It has been tested by compilation.

Signed-off-by: Benoit Taine <benoit.taine@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-22 17:46:56 +02:00
Mark Brown
cee429e5c5 Merge remote-tracking branches 'asoc/topic/ux500', 'asoc/topic/wm8731', 'asoc/topic/wm8804', 'asoc/topic/wm8955' and 'asoc/topic/wm8985' into asoc-next 2014-05-22 00:24:04 +01:00
Mark Brown
04f87446c2 Merge remote-tracking branches 'asoc/topic/rt5651', 'asoc/topic/samsung', 'asoc/topic/sgtl5000', 'asoc/topic/sh', 'asoc/topic/simple', 'asoc/topic/sirf', 'asoc/topic/sta350' and 'asoc/topic/tlv320dac33' into asoc-next 2014-05-22 00:24:00 +01:00
Mark Brown
6f821c6449 Merge remote-tracking branches 'asoc/topic/nuc900', 'asoc/topic/omap', 'asoc/topic/pxa', 'asoc/topic/rcar', 'asoc/topic/rt5640' and 'asoc/topic/rt5645' into asoc-next 2014-05-22 00:23:57 +01:00
Mark Brown
6630f30ed5 Merge remote-tracking branches 'asoc/topic/headers', 'asoc/topic/intel', 'asoc/topic/jz4740', 'asoc/topic/max98090', 'asoc/topic/max98095', 'asoc/topic/mc13783' and 'asoc/topic/multicodec' into asoc-next 2014-05-22 00:23:54 +01:00
Mark Brown
3a6a489fd8 Merge remote-tracking branches 'asoc/topic/devm', 'asoc/topic/fsl', 'asoc/topic/fsl-esai', 'asoc/topic/fsl-sai', 'asoc/topic/fsl-spdif' and 'asoc/topic/fsl-ssi' into asoc-next 2014-05-22 00:23:51 +01:00
Mark Brown
0c5dacf2ca Merge remote-tracking branches 'asoc/topic/cs42l56', 'asoc/topic/cs42xx8' and 'asoc/topic/davinci' into asoc-next 2014-05-22 00:23:49 +01:00
Mark Brown
b03a1c7029 Merge remote-tracking branches 'asoc/topic/ad1980', 'asoc/topic/adsp', 'asoc/topic/ak4104', 'asoc/topic/ak4642', 'asoc/topic/alc5623', 'asoc/topic/arizona', 'asoc/topic/atmel' and 'asoc/topic/cache' into asoc-next 2014-05-22 00:23:45 +01:00
Mark Brown
497c11a946 Merge remote-tracking branch 'asoc/topic/pcm512x' into asoc-next 2014-05-22 00:23:45 +01:00
Mark Brown
b79e16cb4a Merge remote-tracking branch 'asoc/topic/pcm' into asoc-next 2014-05-22 00:23:44 +01:00
Mark Brown
e3ac3f2510 Merge remote-tracking branch 'asoc/topic/enum' into asoc-next 2014-05-22 00:23:44 +01:00
Mark Brown
566d4eeff8 Merge remote-tracking branch 'asoc/topic/dt' into asoc-next 2014-05-22 00:23:43 +01:00
Mark Brown
8e8fbd8f58 Merge remote-tracking branch 'asoc/topic/dapm-init' into asoc-next 2014-05-22 00:23:42 +01:00
Mark Brown
6bf88ab2ec Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next 2014-05-22 00:23:42 +01:00
Mark Brown
1450da3cf6 Merge remote-tracking branch 'asoc/topic/core' into asoc-next 2014-05-22 00:23:41 +01:00
Mark Brown
0f4019e6f4 Merge remote-tracking branch 'asoc/topic/component' into asoc-next 2014-05-22 00:23:41 +01:00
Mark Brown
228704bbdd Merge remote-tracking branch 'asoc/fix/max98090' into asoc-linus 2014-05-22 00:23:37 +01:00
Mark Brown
95b9cff321 ASoC: Intel fixes for v3.15
This is a relatively large batch of fixes for the newly added
 Haswell/Baytrail drivers from Intel.  It's a bit larger than is good for
 this point in the cycle but it's all for a newly added driver so not so
 worrying as it might otherwise be.  Some of it's integration problems,
 some of it's the sort of problem usually turned up in stress tests.
 -----BEGIN PGP SIGNATURE-----
 Version: GnuPG v1
 
 iQIcBAABAgAGBQJTc1YrAAoJELSic+t+oim9losP/0I50EKB6hrEugvFrNT+vBkR
 OQxhcj+WGgIguVQNqaHPBr3rQ1iGG54C5Yf1KDSi2jNIjkHSTIJ1bT2QB40rmclT
 14V6LZJL481TadcKcjLux1jgmaMvNO5Fa1gqYdLLVZFUYP/yTCLhqYrVIDO7NC6M
 VCKtZ965l4u6TMJACTdz9MzCbAwlhb7OTpgxIiyVQlE4SPq2D7M8W9IceLjiDzGg
 rviGK7MejMxc2b4i2vJGi4msaqK8aazDmMhoqrI+HZr/6pZgWkvKJ2S/zZf8AdEH
 6KfEQ8vPU/ag8M4UoH72JJtn1gsjphkEY8GyNWCvP7fIXnpcYB34c/cyaaMQ6lGN
 fGPfwQpbNobx3sJsPMRj21kFgy61rXM2PcbA4QEhPniHd6UlVPUgjkxBNE2YVM3d
 0+tWtgzWVT10F10fcKIkw00/gDohBK+4rViAu0VK5Ml90F0eYLeITWYFVyjmU52f
 7Q+0Duwm65LsA4hqFKcH5lRbk6IM29yRte/wGYY8mLWODADO0+cU6WmVRDHTlRFo
 35HidtfY8EGcU+rS24XyGd0+GRoO+nghzKTckoD3z9OCPilePkMb/dD5vC7NvmMO
 5Q15VbxmJgsus0aTDOPD634XoVTlQ/ESBA6bj5APylZorNKyANorJBphn4rBfZV+
 GXtzzjFrllDCOpKtjDzB
 =ROtF
 -----END PGP SIGNATURE-----

Merge tag 'asoc-v3.15-rc5-intel' into asoc-linus

ASoC: Intel fixes for v3.15

This is a relatively large batch of fixes for the newly added
Haswell/Baytrail drivers from Intel.  It's a bit larger than is good for
this point in the cycle but it's all for a newly added driver so not so
worrying as it might otherwise be.  Some of it's integration problems,
some of it's the sort of problem usually turned up in stress tests.

# gpg: Signature made Wed 14 May 2014 12:40:27 BST using RSA key ID 7EA229BD
# gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>"
# gpg:                 aka "Mark Brown <broonie@debian.org>"
# gpg:                 aka "Mark Brown <broonie@kernel.org>"
# gpg:                 aka "Mark Brown <broonie@tardis.ed.ac.uk>"
# gpg:                 aka "Mark Brown <broonie@linaro.org>"
# gpg:                 aka "Mark Brown <Mark.Brown@linaro.org>"
2014-05-22 00:23:36 +01:00
Mark Brown
dd97254f5c ASoC: Driver fixes for v3.15
A small set of driver fixes, nothing remarkable in itself or of any
 relevance outside of the driver.
 -----BEGIN PGP SIGNATURE-----
 Version: GnuPG v1
 
 iQIcBAABAgAGBQJTc1hlAAoJELSic+t+oim9H/sP/A6r/9/qBesBiBvK7q+jXKNo
 FW6M6zM2lZpLuAnUlSLNC6nb6e6nQYTvvCV4VIXfuj8r8NPMmxwdreg5Vp2fJdAk
 0Qlr3l025B4VmPxqik5wA04jE74G8BW8ttXHpWxxOXdceaUJ4uDzryJBuCpV3vJk
 +2gprKcnR4/wgqpzkGBAwNfurkfXOxpqB8DfPj+FcfRdWyxTqFtvZi+9dSwm2bLv
 T7z8alaMcG/ZpkH7nF+Q4Vu/0tJhgyHIVRTFsJvBKNYVr8cdGi+VDq++AE/62YOE
 v6EGpsrjn243AA8UDBnCznVVZcnwhSf33OpBg44DHb0/1J343x5r4bsGlQkb+19V
 FkBBnhiozWRwVjTBU/E6ss5eua+ESNqL6/EFLpCrD7ykduNOxC/qyKJxcWmmYq9K
 B7Z3tWjX/EzdQ/tMEt26MTnXWSmtk/Yq8dg2UDkdF5r4zBsGyNgHwpi1nbYDCNcm
 alBF7TKkhCV5woJcF+ygG7oFarnQnjSL+J+LTuKRo1gn6k4/NWaVmb/8K9pkN5wS
 hZhL12rjySVXcijrbwVJC9HfFYCtGDjJeTC0ifGQ9jTvEan/Hwi/vwk5XocTywbF
 PglDm9ygwyLAC3A602RWhd/WdgLdmfCYuyKJEcnlnVlkEFmpa5LyJ7wSi6qYYmAk
 j+xPZI2ROuKtebPAP6Es
 =0trh
 -----END PGP SIGNATURE-----

Merge tag 'asoc-v3.15-rc5-drivers' into asoc-linus

ASoC: Driver fixes for v3.15

A small set of driver fixes, nothing remarkable in itself or of any
relevance outside of the driver.

# gpg: Signature made Wed 14 May 2014 12:49:57 BST using RSA key ID 7EA229BD
# gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>"
# gpg:                 aka "Mark Brown <broonie@debian.org>"
# gpg:                 aka "Mark Brown <broonie@kernel.org>"
# gpg:                 aka "Mark Brown <broonie@tardis.ed.ac.uk>"
# gpg:                 aka "Mark Brown <broonie@linaro.org>"
# gpg:                 aka "Mark Brown <Mark.Brown@linaro.org>"
2014-05-22 00:23:31 +01:00
Mark Brown
266bd275b9 ASoC: Core fixes for v3.15
A few things here:
 
  - Fix the creation of spurious CODEC<->CODEC links which caused DAPM to
    have audio paths which shouldn't be present causing spurious powerups
    and potential audible issues for users.
  - Ensure the suspend->off transition doesn't have spurious transitions
    to prepare added to the sequence.
  - Fix incorrect skipping of PCM suspension for active audio streams.
  - Remove Timur Tabi from the CS4270 maintainers, Cirrus are now doing
    this and Timur no longer has the boards that he was using.
 -----BEGIN PGP SIGNATURE-----
 Version: GnuPG v1
 
 iQIcBAABAgAGBQJTc1qXAAoJELSic+t+oim9rQsP/0plEA4KmnSUAD5l+LnFLZVI
 l4PrbTM9BrjdNpGk/dn0yeF9xAmKvlEa6hcclhPMSF5QIVWUxK5wiNZ4ZN1W64as
 VQsh9XHnD0AHSCD1v7ugz3vh4bvDk2cQkQUsJb9CS4Eh4Dgt1T2bjk74K3AnUBCV
 3yDkGW+15Yumo7WW8TKB1Qd7fIsuo95qua+caC1btnFz2VLWkdWixZ5D/t7tp4G2
 SriITNMqUF6gT0RWCue9sDKyfMkCN8tIOh5mvckHEYWLC5/pSgi7zKabDulUayS9
 GgG8mQIto49LL6NmzeyzBsDlf8Gk0O50BZOrEHvHQWw4dMiQ6ml8NTTxuz6oynzE
 k/b1aBUlnOf7wFHA1ILIgAHq3rMah9+/XVkxnHqPBxmP9IgIZoL//rc3DwCtqMbs
 CiIgHOPONdfiHtxMPJwCiBNqfQKDCerYVq4dmZTwU3m99Zn1keFKSZ2dcEWJK02S
 s3kQlYQ6sStpGjPrlrDbS7UcNtX+pqSI2c46GXHHRsLZbAAhHe2kwQ2y/Iry0ntc
 eh1ztL8FLZylEiXmYWjC2Sx5azKOhWVMJGdlKBbX3CNtxFTKzG0NXwHjXJXH72Y3
 Zm8SS9lK4uuJz4IKr7k2RHXETxFb1vdFlijYN5VvkGmYBoGo69G/dFDv6QAg0Ow/
 ANnqLBtrgKir3q49T2YQ
 =s9u3
 -----END PGP SIGNATURE-----

Merge tag 'asoc-v3.15-rc5-core' into asoc-linus

ASoC: Core fixes for v3.15

A few things here:

 - Fix the creation of spurious CODEC<->CODEC links which caused DAPM to
   have audio paths which shouldn't be present causing spurious powerups
   and potential audible issues for users.
 - Ensure the suspend->off transition doesn't have spurious transitions
   to prepare added to the sequence.
 - Fix incorrect skipping of PCM suspension for active audio streams.
 - Remove Timur Tabi from the CS4270 maintainers, Cirrus are now doing
   this and Timur no longer has the boards that he was using.

# gpg: Signature made Wed 14 May 2014 12:59:19 BST using RSA key ID 7EA229BD
# gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>"
# gpg:                 aka "Mark Brown <broonie@debian.org>"
# gpg:                 aka "Mark Brown <broonie@kernel.org>"
# gpg:                 aka "Mark Brown <broonie@tardis.ed.ac.uk>"
# gpg:                 aka "Mark Brown <broonie@linaro.org>"
# gpg:                 aka "Mark Brown <Mark.Brown@linaro.org>"
2014-05-22 00:23:30 +01:00
Tushar Behera
1d55417e12 ASoC: samsung: Add devm_clk_get to pcm.c
clk_get in probe function can be safely replaced with devm_clk_get.

Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-22 00:20:11 +01:00
Tushar Behera
7253e354e7 ASoC: samsung: Use devm_snd_soc_register_component
Replaced snd_soc_register_component with its devres equivalent,
devm_snd_soc_register_component.

Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-22 00:20:11 +01:00
Tushar Behera
55313bd3b0 ASoC: samsung: Use devm_snd_soc_register_platform
Replaced snd_soc_register_platform with devm_snd_soc_register_platform
in samsung_asoc_dma_platform_register(). This makes the function
samsung_asoc_dma_platform_unregister() redundant. This is removed and
all its users are updated.

Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-22 00:20:11 +01:00
Tushar Behera
c583883ecd ASoC: samsung: Use devm_snd_soc_register_card
Replace snd_soc_register_card with devm_snd_soc_register_card.
With this change, we can delete the empty remove functions.

Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-22 00:20:11 +01:00
Kailang Yang
13fd08a339 ALSA: hda/realtek - Add support headset mode for ALC233
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-21 11:13:17 +02:00
Toralf Förster
2d3a277822 ALSA: lola: fix format type mismatch in sound/pci/lola/lola_proc.c
Signed-off-by: Toralf Förster <toralf.foerster@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-21 11:12:15 +02:00
Toralf Förster
e7fc496066 ALSA: hda - fix format type mismatch in sound/pci/hda/patch_sigmatel.c
Signed-off-by: Toralf Förster <toralf.foerster@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-21 11:11:50 +02:00
Takashi Iwai
e9bd7d5ce8 ALSA: hda - Disable AA-mix on Sony Vaio S13
The analog-loopback causes the speaker noises even if it's set to zero
volume.  As a simple workaround, just get rid of the loopback mixer.

Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=873704
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-21 11:06:49 +02:00
Gabriele Mazzotta
5e6db6699b ALSA: hda - White noise fix for XPS13 9333
Disable the AA-loopback path to get rid of the constant white noise
that can be heard when headphones are used.

Signed-off-by: Gabriele Mazzotta <gabriele.mzt@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-21 11:00:06 +02:00
Lars-Peter Clausen
fbfad49076 ASoC: neo1973_wm8753: Automatically disconnected non-connected pins
The DAPM routes for this board are complete, hence we can let the core take care
of disconnecting non-connected pins rather than doing it manually.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-20 23:29:22 +01:00
Sylwester Nawrocki
c86d50f9dc ASoC: samsung: Allow setting OP_CLK of the IIS Multi Audio Interface
This patch adds support for setting source clock of the "Core CLK"
of the IIS Multi Audio Interface.

Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-20 23:20:57 +01:00
Arnd Bergmann
b45281412a ASoC: pxa: remove mach header dependency
As we are moving the mmp platform towards multiplatform support,
we have to stop including platform header files.

This changes the pxa-ssp sound driver file to no longer depend
on mach/hardware.h and mach/dma.h. The code using the definitions
from those headers is actually gone already, the only thing
that was still being used was the pxa_dma_desc typedef, which
we can easily work around by using the normal 'struct pxa_dma_desc'
name.

The pxa2xx-dma driver still uses this header, so we include it
explicitly there, which is ok because that is only used on pxa,
not on mmp.

Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Xia Kaixu <kaixu.xia@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-20 23:14:49 +01:00
Andrew Lunn
7d6d478f38 ASoC: alc5623: Add device tree binding
Let the ALC5623 codec be instantiated from DT. Add a simple binding
for the additional control register and the jack detect register.

Also, add a prompt to the Kconfig entry for this CODEC, so that it can
be selected. Since kirkwood-t5325.c will no longer be used, we need to
be able to enable the CODEC in the mvebu_v5_defconfig etc.

Signed-off-by: Andrew Lunn <andrew@lunn.ch>
Acked-by: Jason Cooper <jason@lakedaemon.net>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-20 23:12:23 +01:00
Sascha Hauer
ee9daad495 ASoC: fsl-ssi: Move fsl_ssi_set_dai_sysclk above fsl_ssi_hw_params
fsl_ssi_set_dai_sysclk will be called from fsl_ssi_hw_params in the
next patch. Move up to avoid forward declaration and to keep the next patch
more readable. No functional change.

Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-20 23:05:03 +01:00
Markus Pargmann
504894799f ASoC: fsl-ssi: Transmit enable synchronization
When the fsl-ssi unit is used in i2s slave mode, it is possible that the
SSI unit starts transmitting data on the wrong channel. This happens
because the SSI does not synchronize with the left-right-clock by
default.

This patch enables transmit enable synchronization.

Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-20 23:04:11 +01:00
Markus Pargmann
171d683d2a ASoC: fsl-ssi: Remove unnecessary variables from ssi_private
There are some variables defined in struct fsl_ssi_private that describe
states that are also described by other variables.

This patch adds some helper functions that return exactly the same
information based on available variables. This helps to clean up struct
fsl_ssi_private and remove them from the probe function.

It also removes some not really used variables (new_binding, name).

Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-20 23:02:46 +01:00
Markus Pargmann
4d9b7926f2 ASoC: fsl-ssi: Cleanup probe function
Reorder the probe function to be able to move the second imx-specific
block to the seperate imx probe function. The patch also removes some
comments/variables/code that are not used anymore or could be simply
replaced by other variables.

Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-20 23:02:42 +01:00
Markus Pargmann
ed0f1604e9 ASoC: fsl-ssi: Remove useless DMA code
Simplify dma DT property handling. fsl,ssi-dma-events is not used
anymore. It passes invalid data to imx_pcm_dma_params_init_data() which
copies some data into an imx dma struct. This struct is never used in
imx-dma or imx-sdma because of generic OF DMA handling. The
"fsl,ssi-dma-events" is not used anywhere in dts files.

Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-20 23:02:38 +01:00
Markus Pargmann
49da09e265 ASoC: fsl-ssi: Move imx-specific probe to seperate function
Move imx specific probe code to a seperate function. It reduces the
size of the probe() function and makes the code and error handling
easier to understand.

Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-20 23:02:34 +01:00
Markus Pargmann
2a1d102de4 ASoC: fsl-ssi: Use dev_name for DAI driver struct
Instead of creating a name using string manipulation functions, we can
simply use the device name for the DAI driver struct.

Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-20 23:02:31 +01:00
Markus Pargmann
f138e62124 ASoC: fsl-ssi: Move debugging to seperate file
Move all code that is only used for debugging to a seperate file. This
makes it easier to see what functions are used for debugging only.

Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-20 23:02:26 +01:00
Markus Pargmann
65c961cc59 ASoC: fsl-ssi: Fix register values when disabling
The bits we have to clear when disabling are different when the other
stream is still active.

This patch fixes the calculation of new register values after disabling
one stream. It also adds a more detailed description of the new register
value calculation.

Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-20 23:02:22 +01:00
Lars-Peter Clausen
55bc825369 ASoC: mop500_ab8500: Replace instances of rtd->codec->card with rtd->card
No need to go via the CODEC to get a pointer to the card. This will help to
eventually remove the card field from the snd_soc_codec struct.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-20 22:55:39 +01:00
Lars-Peter Clausen
0596f70069 ASoC: omap: Replace instances of rtd->codec->card with rtd->card
No need to go via the CODEC to get a pointer to the card. This will help to
eventually remove the card field from the snd_soc_codec struct.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-20 22:54:54 +01:00
Lars-Peter Clausen
cf7b71f46b ASoC: ad1980: Replace goto loop with do-while loop
Using a proper do-while loop here instead of a open-coded goto loop is both
cleaner and shorter.

Also fixes the following warnings from smatch:
	sound/soc/codecs/ad1980.c:213 ad1980_reset() info: loop could be replaced with if statement.
	sound/soc/codecs/ad1980.c:212 ad1980_reset() info: ignoring unreachable code.
	sound/soc/codecs/ad1980.c:215 ad1980_reset() info: ignoring unreachable code.

While we are at it also change retry_cnt to unsigned int, using u16 for a
on-stack loop counter doesn't make that much sense.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-20 22:53:36 +01:00
Dylan Reid
f73387cb6b ALSA: hda/tegra - Fix MODULE_DEVICE_TABLE typo.
I missed a rename during the review process.  Fix the
MODULE_DEVICE_TABLE to match the structure.

Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-20 20:56:49 +02:00
Dylan Reid
3c320f3f56 ALSA: hda - Add driver for Tegra SoC HDA
This adds a driver for the HDA block in Tegra SoCs.  The HDA bus is
used to communicate with the HDMI codec on Tegra124.

Most of the code is re-used from the Intel/PCI HDA driver.  It brings
over only two of the module params, power_save and probe_mask.

Signed-off-by: Dylan Reid <dgreid@chromium.org>
Reviewed-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-20 09:38:38 +02:00
Sumit Bhattacharya
9674678633 ALSA: hda/hdmi - Add Nvidia Tegra124 HDMI support
Add the Tegra12x HDA codec id to patch_hdmi.

Signed-off-by: Sumit Bhattacharya <sumitb@nvidia.com>
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Acked-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-20 09:38:27 +02:00
Kevin Strasser
2fa190ce33 ASoC: Intel: Fix pcm stream context restore crash
In some cases the pcm stream is closed while context has been
scheduled to be restored, causing a null pointer deref panic.
Cancel work to ensure stream does not get freed while work is
still active/pending.

Also, restoring the pcm context can be safely skipped after the
stream has been stopped. Check if pcm stream is still running
before restoring stream context to help pending work finish
more quickly in stream close path.

Signed-off-by: Kevin Strasser <kevin.strasser@intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-19 17:30:56 +01:00
Axel Lin
8c32570441 ASoC: rt5645: Fix updating wrong register for T5645_AIF2 case
This looks like a copy-paste bug, fix it.

Signed-off-by: Axel Lin <axel.lin@ingics.com>
Tested-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-19 17:23:14 +01:00
Jarkko Nikula
d77a14b579 ASoC: Remove needless snd_soc_dapm_enable_pin() from machine driver inits
ALSA SoC core marks widgets as connected by default when they are
initialized in snd_soc_dapm_new_control() so there is no need to call
snd_soc_dapm_enable_pin() from machine driver init functions.

Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-19 17:19:18 +01:00
Jarkko Nikula
831ffa45e7 ASoC: Remove needless snd_soc_dapm_sync() from machine driver inits
ALSA SoC core takes care of calling snd_soc_dapm_sync() at the end
snd_soc_instantiate_card() so there is no need to call it from machine
driver init functions.

Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-19 17:19:18 +01:00
Lars-Peter Clausen
c1406846e4 ASoC: rt5651: Do not use rtd->codec
rtd->codec does not necessarily point to the CODEC instance for which the
callback was called (e.g. for CODEC<->CODEC or multi-CODEC links). Use
dai->codec instead.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-19 17:16:04 +01:00
Lars-Peter Clausen
5958de23ed ASoC: cs42xx8: Do not use rtd->codec
rtd->codec does not necessarily point to the CODEC instance for which the
callback was called (e.g. for CODEC<->CODEC or multi-CODEC links). Use
dai->codec instead.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-19 17:14:02 +01:00
Andy Shevchenko
052c233e98 ALSA: fm801: convert struct description to kernel-doc
Just move field descriptions to the struct description in the kernel-doc
format. There is no functional change.

Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-19 14:33:36 +02:00
Tushar Behera
02fb05a598 ALSA: pcm_dmaengine: Add check during device suspend
Currently snd_dmaengine_pcm_trigger() calls dmaengine_pause()
unconditinally during device suspend. In case where DMA controller
doesn't support PAUSE/RESUME functionality, this call is not able
to stop the DMA controller. In this scenario, audio playback doesn't
resume after device resume.

Calling dmaengine_pause/dmaengine_terminate_all conditionally fixes
the issue.

It has been tested with audio playback on Samsung platform having
PL330 DMA controller which doesn't support PAUSE/RESUME.

Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-19 11:31:24 +02:00
Julia Lawall
47c9807425 sound: mpu401.c: make return of 0 explicit
Delete unnecessary local variable whose value is always 0 and that hides
the fact that the result is always 0.

A simplified version of the semantic patch that fixes this problem is as
follows: (http://coccinelle.lip6.fr/)

// <smpl>
@r exists@
local idexpression ret;
expression e;
position p;
@@

-ret = 0;
... when != ret = e
return
- ret
+ 0
  ;
// </smpl>

Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-19 10:08:43 +02:00
Jarkko Nikula
a735d992c2 ASoC: max98090: Move microphone bias voltage setting to probe function
Microphone bias level configuration register can configure voltage between
2.2 V and 2.8 V but doesn't manage is voltage on or off. Microphone bias
on/off state is controlled by "MICBIAS" DAPM widget.

Therefore there is no need to update bias voltage conditionally depending on
jack state each time when codec goes to SND_SOC_BIAS_ON state and setting
can be moved to max98090_probe() as driver currently doesn't support other
levels than 2.8 V.

Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-16 19:59:26 +01:00
Liam Girdwood
541423dde4 ASoC: max98090: Make sure we configure BCLK in one place
BCL is being configured in two places producing a warning message.
Make sure we only configure BCLK once and when we are master.

Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-16 19:59:25 +01:00
Jarkko Nikula
70f29d3889 ASoC: max98090: Add ACPI probing support
Add ACPI ID for MAX98090 and ACPI 5 I2C device probing support.

Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-16 19:59:25 +01:00
Liam Girdwood
f1c0bc9145 ASoC: max98090: Mark cache as dirty prior to restoring
Make sure the cache is fully flushed at resume time.

Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-16 19:57:15 +01:00
Liam Girdwood
46b0e97dcf ASoC: max98090: Reset codec on resume
Make sure we reset codec and clear any IRQs on resume. This matches
the init sequence in probe.

Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-16 19:57:15 +01:00
Liam Girdwood
25b4ab430f ASoC: max98090: Fix reset at resume time
Reset needs to wait 20ms before other codec IO is performed. This wait
was not being performed. Fix this by making sure the reset register is not
restored with the cache, but use the manual reset method in resume with
the wait.

Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
2014-05-16 19:56:23 +01:00
Liam Girdwood
729af1ce6c ASoC: max98090: Fix digital sidetone gain TLV
TLV for digital sidetone volume is wrong, this fix matches it to the
datasheet.

Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-16 19:56:20 +01:00
Vinod Koul
d7b54c3083 ASoC: Intel: remove codec memeber from codec structs
As we already have a memeber struct snd_sst_params.codec to fill this.
so removing duplicate instance

Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-16 19:46:06 +01:00
Vinod Koul
bd17aa45cd ASoC: Intel: add drain_notify support
This patch adds the support to implement drain_notify in Intels mfld driver

Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-16 19:46:06 +01:00
Vinod Koul
5106f5a17e ASoC: Intel: Revert "rename pcm dias to media dai"
This reverts commit 0cac6fc3eb.
This comiit was dropped from rev2 and would not be required as it renames the
platform ops as well which is not required.

Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-16 19:44:49 +01:00
Jarkko Nikula
8c44b2b1ae ASoC: Intel: Fix simultaneous Baytrail SST capture and playback
I managed to drop a change to stream ID setting from commit 49fee17816
("ASoC: Intel: Only export one Baytrail DAI") leading to non-working
simultaneous capture-playback since after one DAI conversion
rtd->cpu_dai->id + 1 will be the same for both playback and capture.

Use substream->stream + 1 like it was in original Liam's patch.

Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-15 16:53:07 +01:00
Laurent Pinchart
e6b0d896ab ASoC: rsnd: Fix warnings due to improper printk formats
Use the %pap printk specifier to print resource_size_t variables. This
fixes warnings on platforms where resource_size_t has a different size
than int.

Signed-off-by: Laurent Pinchart <laurent.pinchart+renesas@ideasonboard.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-15 11:13:17 +01:00
Liam Girdwood
49fee17816 ASoC: Intel: Only export one Baytrail DAI
We don't need more than one DAI for Baytrail SST. Usage becomes also more
straightforward by grouping playback and capture streams under the same PCM
device.

[Jarkko: I made Liam's sst-baytrail-pcm.c change a few lines smaller and
squashed together with my byt-rt5640.c change]

Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-14 16:57:27 +01:00
Liam Girdwood
3a46c7b7cc ASoC: Intel: Make Baytrail PCM data per stream rather than per DAI device
Prepare for single Baytrail DAI playback/capture link by accessing PCM data
using stream ID instead of rtd->dev. Now rtd->dev is unique for playback
and capture since they are exported as separate DAIs but not once converted
to single DAI.

[Jarkko: Separated from another commit with updated commit log]

Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-14 16:57:26 +01:00
Dan Carpenter
15b8e94f74 ASoC: compress: indent an if statement
The return statement was not indented correctly.  I lined up the
condition a bit as well.

Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-14 16:15:03 +01:00
Dan Carpenter
d576422eda ALSA: hda - if statement not indented
The "break;" should be indented.

Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-14 16:47:27 +02:00
Dan Carpenter
665ebe926e ALSA: sb_mixer: missing return statement
The if condition here was supposed to return on error but the return
statement is missing.  The effect is that the ->mixername is set to
"???" instead of "DT019X".

Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-14 16:46:48 +02:00
Takashi Iwai
ff2354bc6e ASoC: Intel fixes for v3.15
This is a relatively large batch of fixes for the newly added
 Haswell/Baytrail drivers from Intel.  It's a bit larger than is good for
 this point in the cycle but it's all for a newly added driver so not so
 worrying as it might otherwise be.  Some of it's integration problems,
 some of it's the sort of problem usually turned up in stress tests.
 -----BEGIN PGP SIGNATURE-----
 Version: GnuPG v1
 
 iQIcBAABAgAGBQJTc1YrAAoJELSic+t+oim9losP/0I50EKB6hrEugvFrNT+vBkR
 OQxhcj+WGgIguVQNqaHPBr3rQ1iGG54C5Yf1KDSi2jNIjkHSTIJ1bT2QB40rmclT
 14V6LZJL481TadcKcjLux1jgmaMvNO5Fa1gqYdLLVZFUYP/yTCLhqYrVIDO7NC6M
 VCKtZ965l4u6TMJACTdz9MzCbAwlhb7OTpgxIiyVQlE4SPq2D7M8W9IceLjiDzGg
 rviGK7MejMxc2b4i2vJGi4msaqK8aazDmMhoqrI+HZr/6pZgWkvKJ2S/zZf8AdEH
 6KfEQ8vPU/ag8M4UoH72JJtn1gsjphkEY8GyNWCvP7fIXnpcYB34c/cyaaMQ6lGN
 fGPfwQpbNobx3sJsPMRj21kFgy61rXM2PcbA4QEhPniHd6UlVPUgjkxBNE2YVM3d
 0+tWtgzWVT10F10fcKIkw00/gDohBK+4rViAu0VK5Ml90F0eYLeITWYFVyjmU52f
 7Q+0Duwm65LsA4hqFKcH5lRbk6IM29yRte/wGYY8mLWODADO0+cU6WmVRDHTlRFo
 35HidtfY8EGcU+rS24XyGd0+GRoO+nghzKTckoD3z9OCPilePkMb/dD5vC7NvmMO
 5Q15VbxmJgsus0aTDOPD634XoVTlQ/ESBA6bj5APylZorNKyANorJBphn4rBfZV+
 GXtzzjFrllDCOpKtjDzB
 =ROtF
 -----END PGP SIGNATURE-----

Merge tag 'asoc-v3.15-rc5-intel' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Intel fixes for v3.15

This is a relatively large batch of fixes for the newly added
Haswell/Baytrail drivers from Intel.  It's a bit larger than is good for
this point in the cycle but it's all for a newly added driver so not so
worrying as it might otherwise be.  Some of it's integration problems,
some of it's the sort of problem usually turned up in stress tests.
2014-05-14 14:27:12 +02:00
Takashi Iwai
7ca33c7a1d ASoC: Driver fixes for v3.15
A small set of driver fixes, nothing remarkable in itself or of any
 relevance outside of the driver.
 -----BEGIN PGP SIGNATURE-----
 Version: GnuPG v1
 
 iQIcBAABAgAGBQJTc1hlAAoJELSic+t+oim9H/sP/A6r/9/qBesBiBvK7q+jXKNo
 FW6M6zM2lZpLuAnUlSLNC6nb6e6nQYTvvCV4VIXfuj8r8NPMmxwdreg5Vp2fJdAk
 0Qlr3l025B4VmPxqik5wA04jE74G8BW8ttXHpWxxOXdceaUJ4uDzryJBuCpV3vJk
 +2gprKcnR4/wgqpzkGBAwNfurkfXOxpqB8DfPj+FcfRdWyxTqFtvZi+9dSwm2bLv
 T7z8alaMcG/ZpkH7nF+Q4Vu/0tJhgyHIVRTFsJvBKNYVr8cdGi+VDq++AE/62YOE
 v6EGpsrjn243AA8UDBnCznVVZcnwhSf33OpBg44DHb0/1J343x5r4bsGlQkb+19V
 FkBBnhiozWRwVjTBU/E6ss5eua+ESNqL6/EFLpCrD7ykduNOxC/qyKJxcWmmYq9K
 B7Z3tWjX/EzdQ/tMEt26MTnXWSmtk/Yq8dg2UDkdF5r4zBsGyNgHwpi1nbYDCNcm
 alBF7TKkhCV5woJcF+ygG7oFarnQnjSL+J+LTuKRo1gn6k4/NWaVmb/8K9pkN5wS
 hZhL12rjySVXcijrbwVJC9HfFYCtGDjJeTC0ifGQ9jTvEan/Hwi/vwk5XocTywbF
 PglDm9ygwyLAC3A602RWhd/WdgLdmfCYuyKJEcnlnVlkEFmpa5LyJ7wSi6qYYmAk
 j+xPZI2ROuKtebPAP6Es
 =0trh
 -----END PGP SIGNATURE-----

Merge tag 'asoc-v3.15-rc5-drivers' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Driver fixes for v3.15

A small set of driver fixes, nothing remarkable in itself or of any
relevance outside of the driver.
2014-05-14 14:24:09 +02:00
Takashi Iwai
927cdab3b6 ASoC: Core fixes for v3.15
A few things here:
 
  - Fix the creation of spurious CODEC<->CODEC links which caused DAPM to
    have audio paths which shouldn't be present causing spurious powerups
    and potential audible issues for users.
  - Ensure the suspend->off transition doesn't have spurious transitions
    to prepare added to the sequence.
  - Fix incorrect skipping of PCM suspension for active audio streams.
  - Remove Timur Tabi from the CS4270 maintainers, Cirrus are now doing
    this and Timur no longer has the boards that he was using.
 -----BEGIN PGP SIGNATURE-----
 Version: GnuPG v1
 
 iQIcBAABAgAGBQJTc1qXAAoJELSic+t+oim9rQsP/0plEA4KmnSUAD5l+LnFLZVI
 l4PrbTM9BrjdNpGk/dn0yeF9xAmKvlEa6hcclhPMSF5QIVWUxK5wiNZ4ZN1W64as
 VQsh9XHnD0AHSCD1v7ugz3vh4bvDk2cQkQUsJb9CS4Eh4Dgt1T2bjk74K3AnUBCV
 3yDkGW+15Yumo7WW8TKB1Qd7fIsuo95qua+caC1btnFz2VLWkdWixZ5D/t7tp4G2
 SriITNMqUF6gT0RWCue9sDKyfMkCN8tIOh5mvckHEYWLC5/pSgi7zKabDulUayS9
 GgG8mQIto49LL6NmzeyzBsDlf8Gk0O50BZOrEHvHQWw4dMiQ6ml8NTTxuz6oynzE
 k/b1aBUlnOf7wFHA1ILIgAHq3rMah9+/XVkxnHqPBxmP9IgIZoL//rc3DwCtqMbs
 CiIgHOPONdfiHtxMPJwCiBNqfQKDCerYVq4dmZTwU3m99Zn1keFKSZ2dcEWJK02S
 s3kQlYQ6sStpGjPrlrDbS7UcNtX+pqSI2c46GXHHRsLZbAAhHe2kwQ2y/Iry0ntc
 eh1ztL8FLZylEiXmYWjC2Sx5azKOhWVMJGdlKBbX3CNtxFTKzG0NXwHjXJXH72Y3
 Zm8SS9lK4uuJz4IKr7k2RHXETxFb1vdFlijYN5VvkGmYBoGo69G/dFDv6QAg0Ow/
 ANnqLBtrgKir3q49T2YQ
 =s9u3
 -----END PGP SIGNATURE-----

Merge tag 'asoc-v3.15-rc5-core' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Core fixes for v3.15

A few things here:

 - Fix the creation of spurious CODEC<->CODEC links which caused DAPM to
   have audio paths which shouldn't be present causing spurious powerups
   and potential audible issues for users.
 - Ensure the suspend->off transition doesn't have spurious transitions
   to prepare added to the sequence.
 - Fix incorrect skipping of PCM suspension for active audio streams.
 - Remove Timur Tabi from the CS4270 maintainers, Cirrus are now doing
   this and Timur no longer has the boards that he was using.
2014-05-14 14:23:48 +02:00
Mark Brown
cf86197ec5 Merge remote-tracking branch 'asoc/fix/pcm' into asoc-linus 2014-05-14 12:52:41 +01:00
Mark Brown
f9a405961e Merge remote-tracking branches 'asoc/fix/audmux', 'asoc/fix/cs42l52', 'asoc/fix/fsl-esai', 'asoc/fix/fsl-spdif', 'asoc/fix/rcar', 'asoc/fix/tlv320aic31xx' and 'asoc/fix/wm8962' into asoc-linus 2014-05-14 12:49:10 +01:00
Tushar Behera
deeaa686b9 ASoC: samsung: Add missing pm ops for Snow sound card driver
Adding missing pm ops so that audio playback works across
suspend and resume cycle.

Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-14 12:34:50 +01:00
Sascha Hauer
5cd15e29a4 ASoC: ak4642: Add support for extended sysclk frequencies of the ak4648
Additionally to the ak4642 pll frequencies the ak4648 also supports 13MHz,
19.2MHz and 26MHz. This adds support for these frequencies.

Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-14 12:32:04 +01:00
Sascha Hauer
d815c703ce ASoC: ak4642: Add driver data and driver private struct
Currently unused, this is done to let the driver distinguish between
the different supported codec types in later patches.

Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-14 12:32:03 +01:00
Sascha Hauer
370f83a156 ASoC: ak4642: Add ALC controls
ALC and ALC Zero crossing detection has been enabled unconditionally.
Add controls for this.

Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-14 12:32:03 +01:00
Sascha Hauer
da731845d5 ASoC: ak4642: Fix typo zoro -> zero
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-14 12:32:03 +01:00
Kuninori Morimoto
bff58ea4f4 ASoC: rsnd: add DVC support
This patch adds DVC (Digital Volume Controller)
support which is member of CMD unit.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-13 19:06:16 +01:00
Kuninori Morimoto
68b6af3656 ASoC: rsnd: enable to use multi parameter on rsnd_dai_call/rsnd_mod_call
rsnd_mod_ops would like to come to use multi parameter.
modify macro to enable it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-13 19:06:16 +01:00
Kuninori Morimoto
b42fccf69c ASoC: rsnd: remove duplicate parameter from rsnd_mod_ops
Now, it can get rsnd_dai_stream pointer from rsnd_mod.
Remove duplicate parameter from rsnd_mod_ops

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-13 19:06:16 +01:00
Kuninori Morimoto
d7bdbc5d9e ASoC: rsnd: add rsnd_get_adinr()
SRC module needs ADINR register settings,
but, it has many similar xxx_ADINR register,
and needs same settings.
This patch adds rsnd_get_adinr() to sharing code.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-13 19:06:16 +01:00
Kuninori Morimoto
739f9502fd ASoC: rsnd: add rsnd_path_parse() macro
Current R-Car sound supports only SRC/SSI,
but, other module will be supported.
This patch adds rsnd_path_parse() macro to share code

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-13 19:06:15 +01:00
Charles Keepax
44330ab516 ASoC: wm8962: Update register CLASS_D_CONTROL_1 to be non-volatile
The register CLASS_D_CONTROL_1 is marked as volatile because it contains
a bit, DAC_MUTE, which is also mirrored in the ADC_DAC_CONTROL_1
register. This causes problems for the "Speaker Switch" control, which
will report an error if the CODEC is suspended because it relies on a
volatile register.

To resolve this issue mark CLASS_D_CONTROL_1 as non-volatile and
manually keep the register cache in sync by updating both bits when
changing the mute status.

Reported-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
2014-05-13 19:02:30 +01:00
Mark Brown
8bee1fd482 Merge branch 'fix/intel' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-intel
Conflicts:
	sound/soc/intel/sst-baytrail-dsp.c
2014-05-13 18:23:56 +01:00
Jarkko Nikula
cffd6665f5 ASoC: Intel: Fix Baytrail SST DSP firmware loading
Commit 10df350977 ("ASoC: Intel: Fix Audio DSP usage when IOMMU is
enabled.") caused following regression in Baytrail SST:

baytrail-pcm-audio baytrail-pcm-audio: error: DMA alloc failed
baytrail-pcm-audio baytrail-pcm-audio: error: failed to load firmware

Fix this by calling dma_coerce_mask_and_coherent() in sst_byt_init() with
the same dma_dev device what is now used in sst_fw_new() when allocating the
DMA buffer.

Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-13 18:21:02 +01:00
Jarkko Nikula
dfe1951b0c ASoC: Intel: Use ACPI device for Baytrail PCM buffer allocation
This follows the same idea than commit 10df350977
("ASoC: Intel: Fix Audio DSP usage when IOMMU is enabled.") by using only
ACPI device for all DMA allocations. Since DMA masking is already done in
firmware loading it can be removed from here.

Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-13 11:54:11 +01:00
Mengdong Lin
7189eb9b8f ALSA: hda - mask buggy stream DMA0 for Broadwell display controller
Broadwell display controller has 3 stream DMA engines. DMA0 cannot update DMA
postion buffer properly while DMA1 and DMA2 can work well. So this patch masks
the buggy DMA0 by keeping it as opened.

This is a tentative workaround, so keep the change small as Takashi suggested.

Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-13 12:11:58 +02:00
Aaron Plattner
ec5fe98886 ALSA: hda - Add new GPU codec ID to snd-hda
Vendor ID 0x10de0071 is used by a yet-to-be-named GPU chip.

Signed-off-by: Aaron Plattner <aplattner@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-13 09:14:13 +02:00
Nicolin Chen
f975ca46f6 ASoC: fsl_esai: Bypass divider settings if clock requirement is not changed
We don't need to change those dividers if bclk and mclk remains the same
directions and values.

Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-12 23:15:25 +01:00
Nicolin Chen
4f8210f66e ASoC: fsl_esai: Set PCRC and PRRC registers at the end of hw_params()
According to Reference Manual -- ESAI Initialization chapter, as the
standard procedure of ESAI personal reset, the PCRC and PRRC registers
should be remained in its reset value and then configured after T/RCCR
and T/RCR configurations's done but before TE/RE's enabling.

So this patch moves PCRC and PRRC settings to the end of hw_params().

Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-12 23:13:13 +01:00
Nicolin Chen
57ebbcafab ASoC: fsl_esai: Only bypass sck_div for EXTAL source
ESAI can only output EXTAL clock source directly. But for FSYS clock source,
ESAI can not output it without getting through PSR PM dividers.

So this patch adds an extra check in the code.

Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-12 23:13:13 +01:00
Nicolin Chen
89e47f62cf ASoC: fsl_esai: Fix incorrect condition within ratio range check for FP
The range here from 1 to 16 is confined to FP divider only while the
sck_div indicates if the calculation contains PSR and PM dividers. So
for the case using PSR and PM since the sck_div is true, the range of
ratio would simply become bigger than 16.

So this patch fixes the condition here and adds one line comments to
make the purpose here clear.

Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-12 23:13:13 +01:00
Bard Liao
57f174f47e ASoC: rt5640: add default case for unexpected ID
We may read an unexpected value when detemining which codec is attached.
In that case, either a unsupported codec is attached or something wrong
with I2C. The driver will not work properly on both cases. So we return
an error for that.

Signed-off-by: Bard Liao <bardliao@realtek.com>
Reviewed-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-12 22:09:30 +01:00
Lars-Peter Clausen
797f283b61 ASoC: Remove runtime field from DAI
This was initially removed in commit 6423c1875 ("ASoC: Remove runtime field from
DAI"), but was, presumably by accident, brought back in commit f0fba2ad1 ("ASoC:
multi-component - ASoC Multi-Component Support"). But has never been
initialized to anything but NULL ever since. This commit removes it again.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-12 22:08:36 +01:00
Lars-Peter Clausen
b74f7be90f ASoC: atmel-pcm-pdc: Remove broken suspend/resume code
Suspend/resume support for the atmel-pcm-pdc driver was broken in commit
f0fba2ad1 ("ASoC: multi-component - ASoC Multi-Component Support"). It
essentially reverted the modifications done in commit 10cab262 ("ASoC: Change
how suspend and resume obtain the PCM runtime"). The suspend and resume handlers
at the beginning check if dai->runtime is not NULL, but dai->runtime is always
NULL, hence the code never runs. Considering that nobody noticed any problems in
the last 4 years since the code was broken and that the driver does not set
SNDRV_PCM_INFO_RESUME, which means applications are expected to stop and restart
the audio stream during suspend/resume, it is probably safe to assume that his
code is not needed and can be removed.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-12 22:08:36 +01:00
Lars-Peter Clausen
ce85a4d726 ASoC: dapm: Fix SUSPEND -> OFF bias sequence
Currently when the DAPM context bias level is SUSPEND and the target bias level
is OFF dapm_pre_sequence_async() will first transition to PREPARE and
dapm_post_sequence_async() will then transition back from PREPARE to STANDBY and
then to OFF.

This patch makes sure that dapm_pre_sequence_async() only transitions to PREPARE
when either going to ON or away from ON. This avoids the extra unnecessary
transitions.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-12 22:06:34 +01:00
Jarkko Nikula
6fb8b02b4b ASoC: Intel: Allow byt-5640 machine driver and SST core go to suspend
Since there is no support for compressed audio in Baytrail ADSP firmware
there is no need to leave it on during suspend since ALSA PCM buffers are
too small for leaving ADSP on for playing or recording.

Implement PM callbacks to Baytrail byt-rt5640.c machine driver that call
snd_soc_suspend and snd_soc_resume functions and unset the ignore_suspend
fields in DAI links.

This makes soc-core and ALSA core gracefully suspend and resume active
stream and call sst_byt_pcm_trigger() during suspend-resume cycle.

Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-12 22:02:18 +01:00
Liam Girdwood
af94aa558b ASoC: Intel: Add Baytrail suspend/resume support
Add suspend and resume support to Baytrail SST DSP. This is implemented by
unloading firmware modules and putting DSP into reset prior suspend and
restarting DSP again in normal boot state after resume.

Context restore for running streams is implemented by scheduling a work from
sst_byt_pcm_trigger() that will allocate a stream with existing parameters
and start it from last known buffer position before suspend.

[Jarkko: Squashed together 5 WIP patches from Liam and 1 from me]

Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-12 22:02:18 +01:00
Liam Girdwood
609a13e5c9 ASoC: Intel: Allow Rx/Tx message list can be cleared prior to suspend
Suspend/resume requires reloading FW to boot state so we need to also make
sure that the driver matches the FW state at boot.

Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-12 22:02:17 +01:00
Jarkko Nikula
800be5900b ASoC: Intel: Move Baytrail extended fw address saving to sst_byt_boot()
We have to save the physical address of extended firmware block in the
beginning of mailbox every time when we boot the DSP firmware since that
mailbox address is re-used after DSP firmware is running. Otherwise DSP
firmware will get bogus extended firmware block address during next DSP
boot.

Currently this is not problem but becomes when DSP runtime rebooting is
implemented. Prepare for that by moving extended firmware address saving
from sst_byt_init() to sst_byt_boot().

Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-12 22:02:17 +01:00
Jarkko Nikula
a6686ed553 ASoC: Intel: Pass stream start position to sst_byt_stream_start()
Stream start position will be needed in resume code. Prepare for it by
adding start offset argument to sst_byt_stream_start().

Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-12 22:02:17 +01:00
Jarkko Nikula
65ee9e8fb6 ASoC: Intel: Simplify Baytrail stream control IPC construction
Baytrail ADSP stream IPC simplifies a little by moving IPC_IA_START_STREAM
construction and sending directly into sst_byt_stream_start() from
sst_byt_stream_operations(). This is because IPC_IA_START_STREAM is only
stream IPC with extra message data so this move saves a few code lines.

Main motivation for this is to prepare for passing stream start position
to sst_byt_stream_start() which will be needed in resume code.

Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-12 22:02:17 +01:00
Jarkko Nikula
c83649e3cd ASoC: Intel: Sample Baytrail DSP DMA pointer only after each period
This is for preparing suspend/resume support but can give also more
safeguard against concurrent timestamp structure access between DSP firmware
and host.

Now DSP DMA pointer is sampled in each pcm pointer callback in
sst_byt_pcm_pointer() but that is unneeded since DSP updates the timestamp
period basis and can potentially be racy if sst_byt_pcm_pointer() is called
when DSP is updating the timestamp.

By taking DSP DMA pointer only after period elapsed IPC messages in
byt_notify_pointer() and returning stored hw pointer in
sst_byt_pcm_pointer() there is less risk for concurrent access.

The same stored hw pointer can be also used in suspend/resume code for
restarting the stream at the same position.

Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-12 22:02:17 +01:00
Lars-Peter Clausen
94986198f5 ASoC: dapm: Handle SND_SOC_DAPM_REG() generically
Commit commit de9ba98b6d ("ASoC: dapm: Make widget power register settings more
flexible") added generic support for on_val/off_val in the DAPM core. With this
in place there is no need anymore for having a special event callback for
SND_SOC_DAPM_REG() widgets.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-12 21:48:08 +01:00
Lars-Peter Clausen
0f9bd7b194 ASoC: dapm: Simplify snd_soc_dapm_link_dai_widgets()
If we find a widget who's stream name matches the name of a DAI widget then
thats the one it should be connected to. Based on the widget id we can say in
which direction the path should be. No need to go back to the DAI and check the
stream names.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-12 21:37:17 +01:00
Lars-Peter Clausen
fe83897fc5 ASoC: dapm: Use snd_soc_dapm_add_path() in snd_soc_dapm_new_pcm()
We already know the widgets we want to connect, so use snd_soc_dapm_add_path()
instead of snd_soc_dapm_add_route() in snd_soc_dapm_new_pcm().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-12 21:34:49 +01:00
Lars-Peter Clausen
9887c20b9f ASoC: dapm: Use snd_soc_dapm_add_path() in connect_dai_link_widgets()
We already know which two widgets should be connected, so use
snd_soc_dapm_add_path() instead of snd_soc_dapm_add_route() in
snd_soc_dapm_connect_dai_link_widgets().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-12 21:34:48 +01:00
Lars-Peter Clausen
a4e9154c42 ASoC: dapm: Revert "ASoC: dapm: Fix double prefix addition"
This reverts commit bd23c5b661.

The patch claims that the patch is necessary to avoid double prefix addition
when calling snd_soc_dapm_add_route() from snd_soc_dapm_connect_dai_link_widgets().
But snd_soc_dapm_add_route() is called with the card's DAPM context, which does
not have a prefix, which means there is no prefix that could be added a second
time.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-12 21:34:43 +01:00
Lars-Peter Clausen
ca5106ae3d ASoC: dapm: Skip CODEC<->CODEC links in connect_dai_link_widgets()
For CODEC to CODEC DAI links the paths are created in snd_soc_dapm_new_pcm().
Also for CODEC to CODEC links the widgets are connected cross-over via a DAI
link widget, meaning that the capture widget of one CODEC will be connected to
the playback widget of the other and vice versa. Whereas
snd_soc_dapm_connect_dai_link_widgets() directly connects the playback widget of
the CPU DAI to the playback widget of the CODEC DAI and the capture widget of
the CPU DAI to the capture widget of the CODEC DAI. So not skipping
CODEC<->CODEC links in snd_soc_dapm_connect_dai_link_widgets() will create
incorrect connections between the two CODECs which will cause DAPM to detect
active paths where there are none and unnecessarily power up widgets.

Fixes: b893ea5 ("ASoC: sapm: Automatically connect DAI link widgets in DAPM graph.")
Cc: <stable@vger.kernel.org> (for 3.14+)
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-12 21:33:36 +01:00
Nicolin Chen
868a6ca84e ASoC: pcm: Fix incorrect condition check for case SNDRV_PCM_TRIGGER_SUSPEND
The regular state before we execute SNDRV_PCM_TRIGGER_SUSPEND should be
SNDRV_PCM_TRIGGER_START, not SNDRV_PCM_TRIGGER_STOP. Thus fix it.

Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-12 21:16:06 +01:00
Mark Brown
b9d4cf74b9 ASoC: Intel: Build Medfield compressed ops
Since commit 4b68b4e1c5 (ASoC: Intel: split the pcm and compress to
different files) the compressed ops haven't been built causing link
failures on allyesconfig and making the driver unbuildable.  Add the
object to the Makefile to fix that.

Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by Vinod Koul <vinod.koul@intel.com>
2014-05-09 10:28:42 +01:00
Hui Wang
a1f3b5fa11 ALSA: hda - add headset mic detect quirks for three Dell laptops
When we plug a 3-ring headset on the Dell machines (VID: 0x10ec0255,
SID: 0x1028065c; VID: 0x10ec0255, SID: 0x10280680; VID: 0x10ec0292,
SID: 0x10280684), the headset mic can't be detected, after apply this
patch, the headset mic can work well.

And on the machine with SID 0x10280684, and the Lineout and external
microphone should be routed to docking, this patch also fix this
problem.

BugLink: https://bugs.launchpad.net/bugs/1297581
Cc: David Henningsson <david.henningsson@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-09 07:25:44 +02:00
Vinod Koul
0cac6fc3eb ASoC: Intel: rename pcm dias to media dai
this is for further updates to driver which supports DPCM :)

Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-08 18:25:05 +01:00
Vinod Koul
6f46c0d33e ASoC: Intel: remove unused sst-mfld platform dais
With DPCM we have media dai used and no seperate headset and speaker dai so
remove the speaker dai
The vibra is no longer supported thru audio, so remove

Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-08 18:25:05 +01:00
Vinod Koul
4b68b4e1c5 ASoC: Intel: split the pcm and compress to different files
For manging them and adding support for more platforms
Code move only

Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-08 18:25:05 +01:00
Vinod Koul
4496ffab7d ASoC: Intel: mark sst_set_stream_status as non static
as this will be used in compressed split file in subsequent patch

Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-08 18:25:05 +01:00
Vinod Koul
e11fd7c3ac ASoc: Intel: rename sst-mfld-platform.c
to sst-mfld-platform-pcm.c so that we can split pcm and compress to different
files for upcoming changes to support more platforms

Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-08 18:25:05 +01:00
Vinod Koul
300f53bf19 ASoC: Intel: remove FSF snail mail address
As this address can move

Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-08 18:25:05 +01:00
Vinod Koul
2b4c78df05 ASoC: Intel: move component registration blob
to the place near it is used

Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-08 18:24:54 +01:00
Liam Girdwood
555f8a80c3 ASoC: Intel: Add support to unload/reload firmware modules.
Add some SST API calls to unload and reload firmware modules. This can be used
by PM code to restore state and also allow modular FW to unload and release
memory blocks.

Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-08 18:20:58 +01:00
Kuninori Morimoto
29e69fd2cd ASoC: rsnd: remove compatibility code
Now, all platform is using new style rsnd_dai_platform_info.
Keeping compatibility is no longer needed.
We can cleanup code.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-08 12:17:59 +01:00
Kuninori Morimoto
5e392ea0da ASoC: rsnd: remove old clock style support
All platform which used old style was
switched to new style.
R-Car sound can remove old style clock support,
use device dependent clock now.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-08 12:17:59 +01:00
Oder Chiou
71bfa9b4d6 ASoC: rt5645: fix coccinelle warnings
Return statements in functions returning bool should use
true/false instead of 1/0.

Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-08 09:02:41 +01:00
Oder Chiou
0f776efd86 ASoC: rt5645: Correct the cache sync function
The patch corrects the cache sync function

Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-08 09:02:41 +01:00
Oder Chiou
4809b96ebb ASoC: rt5645: Move settings from probe() to reg_default struct
The patch moves the private register settings from probe() to reg_default
struct.

Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-08 09:02:41 +01:00
Oder Chiou
9e22f7826a ASoC: rt5645: Staticise non-exported symbols
The patch is for staticising non-exported symbols

Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-08 08:00:43 +01:00
Oder Chiou
92e160ddf6 ASoC: rt5645: Remove the unused variable
The patch is for removing the unused variable.

Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-08 08:00:43 +01:00
Nicolas Ferre
15fb63a08b ASoC: sam9g20_wm8731: remove useless mach/gpio.h
This include file is about to disapear. In addition it is
useless for this code. So it is time to remove it.

Signed-off-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Acked-by: Mark Brown <broonie@linaro.org>
2014-05-07 18:27:20 +02:00
Takashi Iwai
1c37c22332 ALSA: hda - Add dock pin setups for Thinkpad T440
The headphone and mic jacks on Thinkpad T440 are assigned to pins NID
0x16 and 0x19, respectively.  These need to be set up manually by a
fixup.

Reported-and-tested-by: Joschi Brauchle <joschi.brauchle@tum.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-07 11:40:27 +02:00
Lars-Peter Clausen
db88a8e3ca ASoC: Remove unused num_dai field from CODEC
Commit d191bd8de8 ("ASoC: snd_soc_codec includes snd_soc_component") removed the
last user of the num_dai field. Also remove the field itself.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-07 10:21:26 +01:00
Lars-Peter Clausen
af0881ffbd ASoC: Remove unused 'list' field form card
The global card list was removed in commit b19e6e7b7 ("ASoC: core: Use driver
core probe deferral"). The 'list' field of the snd_soc_card struct has been
unused since then. This patch removes the field.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-07 10:21:25 +01:00
Lars-Peter Clausen
24faf76568 ASoC: Remove card's DAI list
Commit f0fba2ad1 ("ASoC: multi-component - ASoC Multi-Component Support") added
a per card list that keeps track of all the DAIs that have been registered with
the card, but the list has never been used. This patch removes it again.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-07 10:21:25 +01:00
Mark Brown
387f837b3d Merge branch 'topic/component' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-core 2014-05-07 10:21:22 +01:00
Liam Girdwood
2b39aab18a ASoC: Intel: Fix block offset calculations.
Block offset calculations are done in the contiguous allocator so
are not required here.

Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-07 09:38:29 +01:00
Brian Austin
272b5edd3b ASoC: Add support for CS42L56 CODEC
This patch adds support for the Cirrus Logic Low Power Stereo I2C CODEC

Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-05 18:20:22 -07:00
Daniel Mack
7c2fcccc32 ASoC: sta350: add support for bits in miscellaneous registers
Add support for RPDNEN, NSHHPEN, BRIDGOFF, CPWMEN and PNDLSL, and add DT
bindings to access them.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-05 12:52:59 -07:00
Liam Girdwood
e9024f0ba3 ASoC: Intel: Fix check for pdata usage before dereference.
This patch fixes the following dereference check ordering.

 sound/soc/intel/sst-haswell-pcm.c:749 hsw_pcm_probe() warn: variable dereferenced before check 'pdata' (see line 746)

 git remote add asoc git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
 git remote update asoc
 git checkout 0b708c87f6
 vim +/pdata +749 sound/soc/intel/sst-haswell-pcm.c

 a4b12990 Mark Brown    2014-03-12  740  };
 a4b12990 Mark Brown    2014-03-12  741
 a4b12990 Mark Brown    2014-03-12  742  static int hsw_pcm_probe(struct snd_soc_platform *platform)
 a4b12990 Mark Brown    2014-03-12  743  {
 a4b12990 Mark Brown    2014-03-12  744  	struct sst_pdata *pdata = dev_get_platdata(platform->dev);
 a4b12990 Mark Brown    2014-03-12  745  	struct hsw_priv_data *priv_data;
 0b708c87 Liam Girdwood 2014-05-02 @746  	struct device *dma_dev = pdata->dma_dev;
 0b708c87 Liam Girdwood 2014-05-02  747  	int i, ret = 0;
 a4b12990 Mark Brown    2014-03-12  748
 a4b12990 Mark Brown    2014-03-12 @749  	if (!pdata)
 a4b12990 Mark Brown    2014-03-12  750  		return -ENODEV;
 a4b12990 Mark Brown    2014-03-12  751
 a4b12990 Mark Brown    2014-03-12  752  	priv_data = devm_kzalloc(platform->dev, sizeof(*priv_data), GFP_KERNEL);

Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-05 12:42:00 -07:00
Lars-Peter Clausen
c9e065c27f ASoC: dapm: Make sure to always update the DAPM graph in _put_volsw()
When using auto-muted controls it may happen that the register value will not
change when changing a control from enabled to disabled (since the control might
be physically disabled due to the auto-muting). We have to make sure to still
update the DAPM graph and disconnect the mixer input.

Fixes: commit 5729507 ("ASoC: dapm: Implement mixer input auto-disable")
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-05 12:31:14 -07:00
Lars-Peter Clausen
6b0a0b3b4e ASoC: Make soc_find_matching_codec() static
The function is only used locally, make it static.

Fixes the following warning from sparse:
	sound/soc/soc-core.c:1644:22: warning: symbol 'soc_find_matching_codec' was not declared. Should it be static?

Fixes: 3ca041ed ("ASoC: dt: Allow Aux Codecs to be specified using DT")
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-By: Sebastian Reichel <sre@kernel.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-05 12:29:25 -07:00
Nicolin Chen
b8a832a0b6 ASoc: fsl_spdif: Add descriptions for fsl_spdif_priv
Other people would clearly understand each member and improve if they want.

Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-05 12:27:40 -07:00
Nicolin Chen
527cda78eb ASoC: fsl_spdif: Print actual sample rate for debug
People would simply know what the driver gets the best for the current
sample rate playback.

Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-05 12:27:39 -07:00
Nicolin Chen
27c647bff2 ASoC: fsl_spdif: Add sysclk df support to derive txclk from sysclk
The sysclk is one the clock sources that could be selected to derive
tx clock. But the route for sysclk is a bit different that it does
not only contain txclk df divider but also have an extra sysclk df.

So this patch mainly adds syclk df configuration support so as to
let the driver be able to get clock from sysclk.

Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-05 12:27:39 -07:00