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Each module maintains connected status & a lock to protect it.
Using codec->lock we can safely serialize ASoC specific callbacks
(in response to mixer_ctl update or dai_ops) and gb module
disconnect. Thus is_connected field can be removed.
Signed-off-by: Vaibhav Agarwal <vaibhav.agarwal@linaro.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@google.com>
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Device type info shared to above HAL is currently hard coded
to SPK only. Actual device type is identifed while parsing
widget types from topology FW shared by codec module.
Signed-off-by: Vaibhav Agarwal <vaibhav.agarwal@linaro.org>
Reviewed-by: Mark Greer <mgreer@animalcreek.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@google.com>
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Register jack with ASoC sound card in case audio module
populates it via codec FW. Currently, only a single jack
with 4 buttons can be registered for each module.
Signed-off-by: Vaibhav Agarwal <vaibhav.agarwal@linaro.org>
Reviewed-by: Mark Greer <mgreer@animalcreek.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@google.com>
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We have single I2S port via APB1 for communication with all
audio modules. Thus, we should register single codec driver
and manage all individual audio modules internally within
this driver.
Signed-off-by: Vaibhav Agarwal <vaibhav.agarwal@linaro.org>
Reviewed-by: Mark Greer <mark.greer@animalcreek.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@google.com>
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Provide finer-grained control of the audio streaming on APB1 by
splitting the transmit/receive start and stop requests into prepare,
start, stop, and shutdown.
CC: Vaibhav Agarwal <vaibhav.agarwal@linaro.org>
Signed-off-by: Mark Greer <mgreer@animalcreek.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@google.com>
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Currently, it is assumed that all audio data CPorts registered on
APB1 are used for transmitting audio data. That may not always be
true like when a microphone is connected but no speakers. Also,
the current special protocol lacks a way to tell APB1 whether the CPort
being registered is for transmitting, receiving, or both.
Fix by adding a 'direction' field to the register and unregister CPort
requests and define bits indicating which direction (or both) audio
data will go on that CPort.
Signed-off-by: Mark Greer <mgreer@animalcreek.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@google.com>
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In response to codec module removal, user space is reported about
the event. In response to this, ALSA layer will update DAPM route
and cleanup DAPM states.
As a fallback mechanism, kernel can cleanup the DAPM state for codec
module. But, this would cause immediate playback (first trial) to fail,
since DSP is still in inconsistent state.
To avoid such situation, a workqueue is scheduled for codec cleanup
with timeout=50ms.
Thus, normally it is expected from above layers to update routes and
perform cleanup. However, fallback mechanism still holds good after
50ms.
Signed-off-by: Vaibhav Agarwal <vaibhav.agarwal@linaro.org>
Reviewed-by: Mark Greer <mgreer@animalcreek.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@google.com>
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there is race condition between _disconnect() request &
stop_trigger() in case of abrupt module removal.
And sometimes this can lead to deadlock while acquiring
codec_info->lock.
To avoid such situation, atomic variable is used to maintain
codec connected state.
During dai operations (trigger, shutdown, etc.), 'is_connected'
variable is validated to avoid unnecessary lock acquire in
case module already removed.
Signed-off-by: Vaibhav Agarwal <vaibhav.agarwal@linaro.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@google.com>
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audio codec driver is now moved to bundle driver approach.
This resolved many race conditions related to audio mgmt &
data connection init/exit sequence.
Thus, a lot of helper functions can now be safely removed.
Signed-off-by: Vaibhav Agarwal <vaibhav.agarwal@linaro.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@google.com>
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Convert the legacy audio management and data protocol drivers to a
bundle driver.
The Audio bundle driver can support a single management and any number
of data cports, and so we expect multiple data cports to be present for
the bundle during initialization.
Signed-off-by: Viresh Kumar <viresh.kumar@linaro.org>
Reviewed-by: Johan Hovold <johan@hovoldconsulting.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@google.com>
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We need to clean up GB protocl connections, otherwise successive
codec insertions fails repeatedly.
NOTE: As per suggestion, since codec is already removed, one should
not trigger any GB command. It'll cause a delay of atleast TIMEOUT
value.
HOwever, failing to cleanup GB protocol, causes successive module
insertion to fail
Signed-off-by: Vaibhav Agarwal <vaibhav.agarwal@linaro.org>
Signed-off-by: Mark Greer <mgreer@animalcreek.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@google.com>
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GB-Audio-manager module is currently used to report uevent
to above layer in response to any codec module inserted or
removed.
Signed-off-by: Vaibhav Agarwal <vaibhav.agarwal@linaro.org>
Signed-off-by: Mark Greer <mgreer@animalcreek.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@google.com>
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For each GB codec module inserted, DAPM widgets, kcontrols, routes
and DAIs can be fetched through greybus in a binary chunk and parsed
locally to create & populate DAPM graph for the specific module.
It is required by each codec module to populate a minimum set of
kcontrols with fixed names to support basic audio usecase.
To support advanced features of codec module, the same can be polpulated
with existing topology parser. However, to use them for different usecase
separate mechanism (may be via MSP) is required to inform userspace about
their configuration value & enable/disable sequence.
ToDos:
Currently, support for enumerated kcontrol/dapm control is hardcoded.
Need to add complete logic within the parser.
Signed-off-by: Vaibhav Agarwal <vaibhav.agarwal@linaro.org>
Signed-off-by: Mark Greer <mgreer@animalcreek.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@google.com>
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Use GB Audio mgmt, data protocol ids to register codec module with
GB protocol. And in response to mgmt->connection_init(), register
GB codec driver with ASoC.
Now, using msm8994 machine to register DAI link dynamically on
codec insertion.
ToDos:
- snd_soc_register_codec() uses driver->name to identify device id.
However, for GB device, .driver{} is not yet populated by GB core.
Thus, defining dummy structure within codec driver. This should
come from GB core itself.
Even existing .driver{} may cause problem in case of multiple
modules inserted or inserted at a different slot.
- Fix logic for gbcodec->dais & gbcodec->dailinks. Current
implementation contains some hard coded data with assumption of
count=1.
- Evaluate definition of 'gbaudio_dailink.be_id' in case of multiple
DAI links.
Signed-off-by: Vaibhav Agarwal <vaibhav.agarwal@linaro.org>
Signed-off-by: Mark Greer <mgreer@animalcreek.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@google.com>
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Added CONFIG_XXX flag check before compiling audio module.
Once we add dynamic DAI link registration from audio driver,
this check wil be required to avoid compilation failures with
other kernel revisions.
Also, renamed header file to better align with .c file name.
Signed-off-by: Vaibhav Agarwal <vaibhav.agarwal@linaro.org>
Signed-off-by: Mark Greer <mgreer@animalcreek.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@google.com>
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