Failure to terminate this match table can lead to boot failures
depending on where the compiler places the match table.
Signed-off-by: Stephen Boyd <sboyd@codeaurora.org>
Acked-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The DMAC src/dst addr needs to be set from driver when DT case.
(It was set from SoC/DMAEngine code when non-DT case)
This patch adds rsnd_gen_dma_addr() to set DMAC src/dst addr.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Renesas sound driver is supporting to use DMAEngine.
But, DMA slave channel name "tx", "rx" is not enough
in DT case.
Becuase, it has many ports and path combination.
This patch adds rsnd_dma_of_name() to find
DMA channel name, for example
memory to SSI0 is "mem_ssi0",
SSI0 to memory is "ssi0_mem",
SSI0 to SRC0 is "ssi0_src0",
SRC0 to SSI0 is "src0_ssi0",
SRC0 to DVC0 is "src0_dvc0"...
Renesas sound want to use PIO transfer mode for some reasons.
It will be PIO tranfer mode if device node doesn't have
DMA settings.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Renesas sound driver uses many modules (= SSI/SRC/DVC),
and each module had own name.
But, each module name can be used as several purpose,
like clock name, DMA name etc...
This patch uses common name for each module.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Renesas sound driver is supporting Gen1/Gen2.
SRC probe can return error if it was unknown
generation.
Now, rsnd_src_non_ops is not needed.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
DT DMA support needs struct platform_device pointer,
and it can get struct device pointer from platform_device.
Save platform_device instead of device.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Driver needs to call of_node_put() after of_get_chile_by_name()
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
snd_soc_jack_add_gpios() schedules a work queue item to poll the GPIO to
generate an initial jack status report. If sound card initialization
fails, that work item needs to be cancelled, so it doesn't run after the
card has been freed. Specifically, freeing the card calls
snd_jack_dev_free() which calls snd_jack_dev_disconnect() which sets
jack->input_dev = NULL, and input_dev is used by snd_jack_report(), which
is called from the work queue item.
snd_soc_jack_free_gpios() cancels the work item. The Tegra ASoC machine
drivers do call this function in the platform driver remove() callback.
However, this happens after the sound card is freed, at least when the
card is freed due to errors late during snd_soc_instantiate_card(). This
leaves a window where the work item can execute after the card is freed.
In next-20140522, sound card initialization does fail for unrelated
reasons, and hits the problem described above.
To solve this, fix the Tegra ASoC machine drivers to clean up the Jack
GPIOs during the snd_soc_card's .remove() callback, which is executed
before the overall card object is freed. also, gGuard the cleanup call
based on whether we actually setup up the GPIOs in the first place.
Ideally, we'd do the cleanup in a struct snd_soc_dai_link .fini/remove
function to match where the GPIOs get set up. However, there is no such
callback.
This change fixes all Tegra machine drivers. By code inspection, I
believe some non-Tegra machine drivers have the same issue. I'll send a
patch for that separately, once this is reviewed.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This makes sure a format string can never get processed into the worker
thread name from the device name.
Signed-off-by: Kees Cook <keescook@chromium.org>
Acked-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Some platforms require that the codecs mclk is a fixed multiplication
factor of the audio stream rate. Add a optional property to the
binding to hold this factor and implement a hw_params() function to
make use of it.
Signed-off-by: Andrew Lunn <andrew@lunn.ch>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds the clock divisor and multiplier NI, MI values for audio
sampling frequencies 44100 and 48000 Hz and PCLK 19.2 MHz. This is useful
for the Odroid X2/U2 boards when the codec works in master mode and its
MCLK clock is fed from the I2S CDCLK output.
Signed-off-by: Chen Zhen <zhen1.chen@samsung.com>
[s.nawrocki@samsung.com: edited the commit description]
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
In post commit, a quirk of this firmware about transactions is reported.
This commit apply a workaround for this quirk.
They often fail transactions due to gap_count mismatch. This state is changed
by generating bus reset.
The fw_schedule_bus_reset() is an exported symbol in firewire-core. But there
are no header for public. This commit moves its prototype from
drivers/firewire/core.h to include/linux/firewire.h.
This mismatch still affects bus management before generating this bus reset.
It still takes a time to call driver's probe() because transactions are still
often failed.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just powering on, these devices below wait to download firmware.
- Firewire Audiophile
- Firewire 410
- Firewire 1814
- ProjectMix I/O
But firmware version 5058 or later, flash memory in the device stores the
firmware. So this driver can enable these devices by sending a certain cue to
load the firmware.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The firmwares for M-Audio Firewire 410/1814 and ProjectMix I/O has a quirk to
ignore MIDI messages in data blocks more than 8. This commit uses a flag which
Fireworks uses for a similar quirk.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
M-Audio Firewire 1814 has a quirk, ProjectMix I/O also has. They transmit
empty packet with wrong value of dbc incremented by 8 at high sampling rate.
According to IEC 61883-1, this value should be the same as the one in
previous packet.
This commit adds a flag named as CIP_EMPTY_HAS_WRONG_DBC. With flag, the value
of dbc in empty packet is overwittern by an expected value.
This is an example of this quirk:
CIP Header 0 CIP Header 1 Payload size
010D0000 9004F759 210
010D0010 90040B59 210
010D0020 90042359 210
01020028 9004FFFF 2 <-
010D0030 90043759 210
010D0040 90044B59 210
010D0050 90046359 210
01020058 9004FFFF 2 <-
010D0060 90047759 210
010D0070 90048B59 210
010D0080 9004A359 210
01020088 9004FFFF 2 <-
010D0090 9004B759 210
010D00A0 9004CB59 210
010D00B0 9004E359 210
010200B8 9004FFFF 2 <-
010D00C0 9004F759 210
010D00D0 90040B59 210
010D00E0 90042359 210
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit allows this driver to support some models which M-Audio produces
with DM1000 but its firmware is special. They are:
- Firewire 1814
- ProjectMix I/O
They have heavily customized firmware. The usual operations can't be applied to
them. For this reason, this commit adds a model specific member to 'struct
snd_bebob' and some model specific functions. Some parameters are write-only so
this commit also adds control interface for applications to set them.
M-Audio special firmware quirks:
- Just after powering on, they wait to download firmware. This state is
changed when receiving cue. Then bus reset is generated and the device is
recognized as a different model with the uploaded firmware.
- They don't respond against BridgeCo AV/C extension commands. So drivers
can't get their stream formations and so on.
- They do not start to transmit packets only by establishing connection but
also by receiving SIGNAL FORMAT command.
- After booting up, they often fail to send response against driver's request
due to mismatch of gap_count.
This module don't support to upload firmware.
Tested-by: Darren Anderson <darrena092@gmail.com> (ProjectMix I/O)
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit allows this driver to support some models which M-Audio produces
with DM1000/DM1000E with usual firmware. They are:
- Firewire 410
- Firewire AudioPhile
- Firewire Solo
- Ozonic
- NRV10
- FirewireLightBridge
According to a person who worked in BridgeCo, some models are produced with
'Pre-BeBoB'. This means that these products were released before BeBoB was
officially produced, and later BeBoB specification was formed. So these models
have some quirks.
M-Audio usual firmware quirks:
- Just after powering on, 'Firewire 410' waits to download firmware. This
state is changed when receiving cue. Then bus reset is generated and the
device is recognized as a different model with the uploaded firmware.
- 'Firewire Audiophile' also waits to download firmware but its
vendor id/model id is the same as the one after loading firmware.
- The information of channel mapping for MIDI conformant data channel is
invalid against BridgeCo specification.
This commit adds some codes for these quirks but don't support to upload
firmware.
This commit also adds specific operations to get metering information. The
metering information also includes status of clock synchronization if the model
supports to switch source of clock.
The specification of FirewireLightBridge is unknown. So in this time, normal
operations are applied for this model.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit allows this driver to support all of models which Focusrite
produces with DM1000/BeBoB. They are:
- Saffire
- Saffire LE
- SaffirePro 10 I/O
- SaffirePro 26 I/O
This commit adds Focusrite specific operations:
1. Get source of clock
2. Get/Set sampling frequency
3. Get metering information
The driver uses these functionalities to read/write specific address by async
transaction.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit allows this driver to support all of models which Yamaha produced
with DM1000/BeBoB. They are:
- GO44
- GO46
This commit adds Yamaha specific operations. To get source of clock, AV/C Audio
Subunit command is used.
I note that their appearances are similar to some models of TerraTec; 'Go44' is
similar to 'PHASE 24 FW' and 'GO46' is similar to 'PHASE X24 FW'. But their
combination of Audio/Music subunits is a bit different.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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.
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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.
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>
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.
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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>"
A small set of driver fixes, nothing remarkable in itself or of any
relevance outside of the driver.
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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>"
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.
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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>"
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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.
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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.
A small set of driver fixes, nothing remarkable in itself or of any
relevance outside of the driver.
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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.
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.
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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.
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>
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>
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>
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>
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>
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>
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>
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>
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>
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
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
Stephen Boyd