Commit Graph

23062 Commits

Author SHA1 Message Date
Takashi Iwai
094435d41d ALSA: pcm: Avoid double hw_free calls at releasing a stream
snd_pcm_release_substream() always calls hw_free op when the stream
was opened.  This is superfluous in most cases because it's been
already released via explicit hw_free ioctl.  Although this double
call is usually OK as this callback should be written to be called
multiple times, it's better to avoid superfluous calls.

Reported-by: Vinod Koul <vinod.koul@intel.com>
Tested-by: Jeeja Kp <jeeja.kp@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-29 12:57:42 +02:00
Takashi Iwai
d8dfacf8b5 Merge branch 'topic/firewire-update' into for-next 2015-09-29 12:52:07 +02:00
Takashi Sakamoto
df075feefb ALSA: firewire-lib: complete AM824 data block processing layer
This commit moves the codes related to data block processing from packet
streaming layer to AM824 layer.

Each driver initializes amdtp stream structure for AM824 data block by
calling amdtp_am824_init(). Then, a memory block is allocated for AM824
specific structure. This memory block is released by calling
amdtp_stream_destroy().

When setting streaming parameters, it calls amdtp_am824_set_parameters().
When starting packet streaming, it calls amdtp_stream_start(). When
stopping packet streaming, it calls amdtp_stream_stop().

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-29 12:51:20 +02:00
Takashi Sakamoto
49c7b3fcd9 ALSA: firewire-lib: rename macros with AM824 prefix
This commit renames some macros just related to AM824 format. In later
commit, they're moved to AM824 layer.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-29 12:50:23 +02:00
Takashi Sakamoto
85130cb43e ALSA: firewire-lib: rename PCM format helper function
Setting the format of PCM substream to AMDTP stream structure is important
to set a handler to copy actual PCM samples between buffers. The
processing should be in data block processing layer because essentially
it has no relationship to packet streaming.

This commit renames PCM format setting function to prepare for integrating
AM824 layer.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-29 12:49:04 +02:00
Takashi Sakamoto
03e2a67eed ALSA: firewire-lib: move MIDI trigger helper function to AM824 layer
In IEC 61883-6, MIDI messages are transferred in MIDI conformant data
channel. Essentially, packet streaming layer is not responsible for MIDI
functionality.

This commit moves MIDI trigger helper function from the layer to AM824
layer. The rest of codes related to MIDI functionality will be moved in
later commits.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-29 12:48:49 +02:00
Takashi Sakamoto
f65be911c0 ALSA: firewire-lib: add helper functions to set positions of data channels
In IEC 61883-6, several types of data are available in AM824 format. The
data is transferred in each data channel. The position of data channel in
data block differs depending on model.

Current implementation has an array to map the index of data channel in an
data block to the position of actual data channel. The implementation
allows each driver to access the mapping directly.

In later commit, the mapping is in specific structure pushed into an
opaque pointer. Helper functions are required.

This commit adds the helper functions for this purpose. In IEC 61883-6,
AM824 format supports many data types, while this specification easily
causes over-engineering. Current AM824 implementation is allowed to handle
two types of data, Multi Bit Linear Audio data (=PCM samples) and MIDI
conformant data (=MIDI messages).

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-29 12:48:36 +02:00
Takashi Sakamoto
bc8500da3e ALSA: firewire-lib: move PCM substream constraint to AM824 layer
In IEC 61883-6, PCM frames are transferred in Multi Bit Linear Audio data
channel. The data channel transfers 16/20/24 bit PCM samples. Thus, PCM
substream has a constrain about it.

This commit moves codes related to the constraint from packet streaming
layer to AM824 data block processing layer.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-29 12:48:19 +02:00
Takashi Sakamoto
51c29fd213 ALSA: firewire-lib: rename parameter setting function for AM824 with FDF field
The value of FDF field in CIP header is protocol-dependent. Thus, it's
better to allow data block processing layer to decide the value in any
timing.

In AM824 data format, the value of FDF field in CIP header indicates
N-flag and Nominal Sampling Frequency Code (sfc). The N-flag is for
switching 'Clock-based rate control mode' and 'Command-based rate control
mode'. In our implementation, 'Clock-based rate control mode' is just
supported. Therefore, When sampling transfer frequency is decided, then
the FDF can be set.

This commit replaces 'amdtp_stream_set_parameters' with
'amdtp_am824_set_parameters' to set the FDF. This is the same timing
to decide the ration between the number of data blocks and the number of
PCM frames.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-29 12:48:02 +02:00
Takashi Sakamoto
5955815e71 ALSA: firewire-lib: add data block processing layer for AM824 format
This commit adds data block processing layer for AM824 format. The new
layer initializes streaming layer with its value for fmt field.

Currently, most implementation of data block processing still remains
streaming layer. In later commits, these codes will be moved to the layer.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-29 12:47:45 +02:00
Takashi Sakamoto
d67c46b953 ALSA: firewire-lib: rename 'amdtp' to 'amdtp-stream' to prepare for functional separation
In later commit, data block processing layer will be newly added. This
layer will be named as 'amdtp-am824'.

This commit renames current amdtp file to amdtp-stream, to distinguish it
from the new layer.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-29 12:47:11 +02:00
Takashi Sakamoto
414ba022a5 ALSA: firewire-lib: add support arbitrary value for fmt/fdf fields in CIP header
Some vendor specific protocol uses its own value for fmt/fdf fields in
CIP header.

This commit support to set arbitrary values for the fields.

In IEC 61883-6, NO-DATA code is defined for FDF field. A packet with this
code includes no data even if it includes some data blocks. This commit
still leaves a condition to handle this special packet.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-29 12:35:48 +02:00
Takashi Sakamoto
20e445771e ALSA: firewire-lib: add helper functions as interfaces between packet streaming layer and data block processing layer
ALSA PCM framework uses PCM buffer with a concept of 'period' to
synchronize userspace operations to hardware for nearly-realtime
processing. Each driver implements snd_pcm_period_elapsed() to tell across
of the period boundary to ALSA PCM middleware. To call the function, some
drivers utilize hardware interrupt handlers, the others count handled PCM
frames.

Drivers for sound units on IEEE 1394 bus are the latter. They use two
buffers; PCM buffer and DMA buffer for IEEE 1394 isochronous packet. PCM
frames are copied between these two buffers and 'amdtp_stream' structure
counts the handled PCM frames. Then, snd_pcm_period_elapsed() is called if
required.

Essentially, packet streaming layer should not be responsible for PCM
frame processing. The PCM frame processing should be handled in each data
block processing layer as a result of handling data blocks. Although, PCM
frame counting is a common work for all of protocols which ALSA firewire
stack is going to support.

This commit adds two new helper functions as interfaces between packet
streaming layer to data block processing layer. In future, each data block
processing layer implements these functions. The packet streaming layer
calls data block processing layer per packet by calling the functions. The
data block processing layer processes data blocks and PCM frames, and
returns the number of processed PCM frames. Then the packet streaming layer
calculates handled PCM frames and calls snd_pcm_period_elapsed().

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-29 12:35:47 +02:00
Takashi Sakamoto
6a4e89ff0f ALSA: firewire-lib: add a member of frame_multiplier instead of double_pcm_frames
In future commit, interface between data block processing layer and packet
stream processing layer is defined. These two layers communicate the
number of data blocks and the number of PCM frames.

The data block processing layer has a responsibility for calculating the
number of PCM frames. Therefore, 'dual wire' of Dice quirk should be
handled in data block processing layer.

This commit adds a member of 'frame_multiplier'. This member represents
the ratio of the number of PCM frames against the number of data blocks.
Usually, the value of this member is 1, while it's 2 in Dice's 'dual wire'.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-29 12:35:46 +02:00
Takashi Sakamoto
27ec83b5c6 ALSA: firewire-lib: add an argument for Dice's dual wire mode
In IEC 61883-6, one data block represents one event. In ALSA, the event is
one PCM frame. Therefore, when processing one data block, current
implementation counts one PCM frame.

On the other hand, Dice platform has a quirk called as 'dual wire' at
higher sampling rate. In detail, see comment of commit 6eb6c81eee
("ALSA: dice: Split stream functionality into a file").

Currently, to handle this quirk, AMDTP stream structure has a
'double_pcm_frames' member. When this is enabled, two PCM frames are
counted. Each driver set this flag by accessing the structure member
directly.

In future commit, some members related to AM824 data block will be moved
to specific structure, to separate packet streaming layer and data block
processing layer. The access will be limited by opaque pointer.

For this reason, this commit adds an argument into
amdtp_stream_set_parameter() to set the flag.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-29 12:35:46 +02:00
Takashi Sakamoto
547e631ce3 ALSA: firewire-lib: return error code when amdtp_stream_set_parameters() detects error
Currently, amdtp_stream_set_parameters() returns no error even if wrong
arguments are given. This is not good for streaming layer because drivers
can continue processing ignoring capability of streaming layer.

This commit changes this function to return error code.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-29 12:35:45 +02:00
Takashi Sakamoto
10b2b6dc1a ALSA: firewire-lib: arrange structure for AMDTP stream
In later commit, some members related to AM824 data format will be moved
from AMDTP stream structure to data block structure. This commit is a
preparation for it. Additionally, current layout of AMDTP stream structure
is a bit mess by several extensions. This commit also arranges the layout.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-29 12:35:44 +02:00
Dan Carpenter
e87359efca ALSA: usb-audio: harmless underflow in snd_audigy2nx_led_put()
We want to verify that "value" is either zero or one, so we test if it
is greater than one.  Unfortunately, this is a signed int so it could
also be negative.  I think this is harmless but it introduces a static
checker warning.  Let's make "value" unsigned.

Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-28 14:33:03 +02:00
Takashi Iwai
010cf269a4 Merge branch 'topic/drm-sync-audio-rate' into for-next 2015-09-25 10:07:55 +02:00
Libin Yang
ddd621fbba ALSA: hda - display audio call sync_audio_rate callback
For display audio, call the sync_audio_rate callback function
to do the synchronization between gfx driver and audio driver.

Signed-off-by: Libin Yang <libin.yang@intel.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-25 10:04:59 +02:00
Takashi Iwai
0b2c8c12f9 ASoC: Updates for v4.4
A first batch of updates targetted at v4.4.  There are no substantial
 core fixes here, the biggest block of changes is updates to the rcar
 drivers and the addition of a CODEC driver for the AK4613.
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Merge tag 'asoc-v4.3-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next

ASoC: Updates for v4.4

A first batch of updates targetted at v4.4.  There are no substantial
core fixes here, the biggest block of changes is updates to the rcar
drivers and the addition of a CODEC driver for the AK4613.
2015-09-25 08:33:11 +02:00
Takashi Iwai
7f57d803ee ALSA: hda - Disable power_save_node for Thinkpads
Lenovo Thinkpads with recent Realtek codecs seem suffering from click
noises at power transition since the introduction of widget power
saving in 4.1 kernel.  Although this might be solved by some delays in
appropriate points, as a quick workaround, just disable the
power_save_node feature for now.  The gain it gives is relatively
small, and this makes the situation back to pre 4.1 time.

This patch ended up with a bit more code changes than usual because
the existing fixup for Thinkpads is highly chained.  Instead of adding
yet another chain, combine a few of them into a single fixup entry, as
a gratis cleanup.

Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=943982
Cc: <stable@vger.kernel.org> # v4.1+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-24 20:49:00 +02:00
Takashi Iwai
1ce3cbe2ab ASoC: Fixes for v4.3
A disappointingly large set of fixes, though none of them very big and
 very widely spread over many different drivers.  Nothing especially
 stands out, it's mostly all device specific and relatively minor.
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Merge tag 'asoc-fix-v4.3-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Fixes for v4.3

A disappointingly large set of fixes, though none of them very big and
very widely spread over many different drivers.  Nothing especially
stands out, it's mostly all device specific and relatively minor.
2015-09-24 20:48:01 +02:00
Takashi Iwai
db25f440f4 Merge branch 'for-linus' into for-next 2015-09-24 14:02:48 +02:00
Takashi Iwai
83510441bc ALSA: hda/tegra - async probe for avoiding module loading deadlock
The Tegra HD-audio controller driver causes deadlocks when loaded as a
module since the driver invokes request_module() at binding with the
codec driver.  This patch works around it by deferring the probe in a
work like Intel HD-audio controller driver does.  Although hovering
the codec probe stuff into udev would be a better solution, it may
cause other regressions, so let's try this band-aid fix until the more
proper solution gets landed.

Reported-by: Thierry Reding <treding@nvidia.com>
Tested-by: Thierry Reding <treding@nvidia.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-24 14:01:46 +02:00
Dan Carpenter
145d92e77e ALSA: core: check for underflow in snd_pcm_sw_params()
As far as I can see, having an invalid ->tstamp_mode is harmless, but
adding a check silences a static checker warning.

Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-24 11:46:25 +02:00
Takashi Sakamoto
13f3a46d2a ALSA: oxfw: add Mackie Onyx Satellite quirk to inform wrong value in 'dbs' field in tx CIP header
Mackie Onyx Satellite has two usage; standalone and with base station.

These two modes has different stream formats. In standalone mode, rx data
block includes 2 Multi Bit Linear Audio (MBLA) data channels and tx data
block includes 2. With base station, they're 6 and 2.

Although, with base station, the actual tx packet include wrong value in
'dbs' field in its CIP header. This quirk causes packet streaming layer to
detect packet discontinuity and to stop PCM substream.

This is a response of 'single' subfunction 'extended stream format
information' command in AV/C Stream Format Information Specification 1.1.
It means that a data block in tx packets includes 2 MBLA data channels.

$ ./firewire-request /dev/fw1 fcp 0x01ffbfc001000000ffffffff
response: 000: 0c ff bf c0 01 00 00 00 ff 00 90 40 03 02 01 02
response: 010: 06

Current OXFW driver parses the response and detects stream formats
correctly.

$ cat /proc/asound/card1/firewire/formation
...
Output Stream from device:
	Rate	PCM	MIDI
* 	48000	2	0
 	44100	2	0
 	88200	2	0
	96000	2	0

On the other hand, in actual tx CIP, the 'dbs' field has 6. But the number
of quadlets in CIP payload is not multiple of 6. Seeing the subtraction of
two successive payload quadlets, it should be multiple of 2.

payload  CIP      CIP
quadlets header0  header1
      24 00060052 9002ffff
      24 0006005e 9002ffff
      26 0006006a 9002ffff
      24 00060077 9002ffff
      24 00060083 9002ffff
      26 0006008f 9002ffff
      24 0006009c 9002ffff
      24 000600a8 9002ffff
      26 000600b4 9002ffff
      24 000600c1 9002ffff

This commit adds support for this quirk to OXFW driver, by using
CIP_WRONG_DBS flag. When this flag is set, packet streaming layer uses
the value of its 'data_block_quadlets' member instead of the value in CIP
header. This value is already set by OXFW driver and no discontinuity is
detected.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-24 11:44:25 +02:00
Mark Brown
4922e7a1b7 Merge remote-tracking branch 'asoc/topic/ux500' into asoc-next 2015-09-23 11:01:30 -07:00
Mark Brown
727e53a12f Merge remote-tracking branches 'asoc/topic/sh', 'asoc/topic/simple', 'asoc/topic/spear', 'asoc/topic/sunxi' and 'asoc/topic/tlv320aic3x' into asoc-next 2015-09-23 11:01:27 -07:00
Mark Brown
e32efed3ea Merge remote-tracking branches 'asoc/topic/pxa', 'asoc/topic/qcom', 'asoc/topic/rcar' and 'asoc/topic/rockchip' into asoc-next 2015-09-23 11:01:25 -07:00
Mark Brown
9abbe73e7d Merge remote-tracking branches 'asoc/topic/jz4740', 'asoc/topic/kirkwood', 'asoc/topic/mtk', 'asoc/topic/multi' and 'asoc/topic/mxs' into asoc-next 2015-09-23 11:01:22 -07:00
Mark Brown
f7b93159ab Merge remote-tracking branches 'asoc/topic/blackfin', 'asoc/topic/davinci', 'asoc/topic/fsl', 'asoc/topic/hdmi' and 'asoc/topic/intel' into asoc-next 2015-09-23 11:01:19 -07:00
Mark Brown
97e15d9a8b Merge remote-tracking branches 'asoc/topic/ak4613', 'asoc/topic/ak4642', 'asoc/topic/arizona', 'asoc/topic/atmel' and 'asoc/topic/au1x' into asoc-next 2015-09-23 11:01:16 -07:00
Mark Brown
f950aa96d1 Merge remote-tracking branch 'asoc/topic/rt5645' into asoc-next 2015-09-23 11:01:15 -07:00
Mark Brown
ed14ee0eea Merge remote-tracking branches 'asoc/fix/wm8960' and 'asoc/fix/wm8962' into asoc-linus 2015-09-23 11:01:12 -07:00
Mark Brown
d86a2fe4dc Merge remote-tracking branches 'asoc/fix/mtk', 'asoc/fix/psc', 'asoc/fix/pxa', 'asoc/fix/spear', 'asoc/fix/sti' and 'asoc/fix/wm0010' into asoc-linus 2015-09-23 11:01:08 -07:00
Mark Brown
312e0bce7f Merge remote-tracking branches 'asoc/fix/davinci', 'asoc/fix/doc', 'asoc/fix/fsl-card', 'asoc/fix/fsl-ssi', 'asoc/fix/intel' and 'asoc/fix/maintainers' into asoc-linus 2015-09-23 11:01:03 -07:00
Mark Brown
fcffa0dbb6 Merge remote-tracking branch 'asoc/fix/rt5645' into asoc-linus 2015-09-23 11:01:02 -07:00
Mark Brown
5ee84ba8c7 Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus 2015-09-23 11:01:00 -07:00
Oder Chiou
fce97b4d70 ASoC: rt5645: Prevent the pop sound in case of playback and the jack is plugging
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-23 10:59:13 -07:00
Oder Chiou
4f4794124e ASoC: rt5645: Increase the delay time to remove the pop sound
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-23 10:59:10 -07:00
Oder Chiou
21cb13e72b ASoC: rt5645: Use the type SOC_DAPM_SINGLE_AUTODISABLE to prevent the weird sound in runtime of power up
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-23 10:59:07 -07:00
Oder Chiou
de3f8fdf73 ASoC: rt5645: Remove the repeated definitions
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-23 10:57:52 -07:00
Robert Jarzmik
8811191fdf ASoC: pxa: pxa2xx-ac97: fix dma requestor lines
PCM receive and transmit DMA requestor lines were reverted, breaking the
PCM playback interface for PXA platforms using the sound/soc/ variant
instead of the sound/arm variant.

The commit below shows the inversion in the requestor lines.

Fixes: d65a14587a ("ASoC: pxa: use snd_dmaengine_dai_dma_data")
Signed-off-by: Robert Jarzmik <robert.jarzmik@free.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
2015-09-22 14:12:46 -07:00
John Lin
917536aeb8 ASoC: rt5645: Add jd_invert for Broadwell
Broadwell can not triger the IRQ falling and rising simultaneously,
so it can not detect jack-in and jack-out simultaneously.
We add a flag "jd_invert" to platform data. If this flag is set,
codec IRQ will be set to invert that forces IRQ as pulse when jack-in
and jack-out.

Signed-off-by: John Lin <john.lin@realtek.com>
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-22 09:46:33 -07:00
John Lin
e9159e7577 ASoC: rt5645: Add dmi for Broadwell
Add DMI data for Buddy project.

Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: John Lin <john.lin@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-22 09:46:32 -07:00
Mark Brown
c44506997c Merge branch 'fix/rt5645' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-rt5645 2015-09-22 09:45:46 -07:00
Adam Thomson
5b64c173cd ASoC: fsl_ssi: Fix checking of dai format for AC97 mode
Current code incorrectly treats dai format for AC97 as bit mask
whereas it's actually an integer value. This causes DAI formats
other than AC97 (e.g. DSP_B) to trigger AC97 related code,
which is incorrect and breaks functionality. This patch fixes
the code to correctly compare values to determine AC97 or not.

Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Acked-by: Timur Tabi <timur@tabi.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-19 11:07:42 -07:00
Ricard Wanderlof
cde79035c6 ASoC: Handle multiple codecs with split playback / capture
Add the capability to use multiple codecs on the same DAI linke where
one codec is used for playback and another one is used for capture.

Tested on a setup using an SSM2518 for playback and an ICS43432 I2S MEMS
microphone for capture.

No verification is made that the playback and capture codec setups are
compatible in terms of number of TDM slots (where applicable).

Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-19 10:58:04 -07:00
Jyri Sarha
9529138276 ASoC: hdmi: Remove obsolete dummy HDMI codec
The hdmi stub codec has not been used since refactoring of OMAP HDMI
audio support.

Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-19 10:44:39 -07:00