| Age | Commit message (Collapse) | Author |
|---|
|
This is the DPCM based machine driver with MAX98090
Signed-off-by: Koro Chen <koro.chen@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
This is the DPCM based platform driver of AFE (Audio Front End) unit.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: Koro Chen <koro.chen@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Each Renesas sound mod (= SSI/SRC/DVC) might be called from many path
if it supports MIXer. In such case, mod <-> io is no longer 1:1
relationship. rsnd_mod_to_io() is no longer needed. Let's remove it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Each Renesas sound mod (= SSI/SRC/DVC) might be called from many paths
if it supports MIXer. In such case, mod <-> io is no longer 1:1
relationship.
This patch checks module working status via io instead of mod
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Each Renesas sound mod (= SSI/SRC/DVC) might be called from many paths
if it supports MIXer. In such case, mod <-> io is no longer 1:1
relationship. This patch removes rsnd_mod_to_io() from snd_kcontrol
and related function.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Each Renesas sound mod (= SSI/SRC/DVC) might be called from many paths
if it supports MIXer. In such case, mod <-> io is no longer 1:1
relationship. This patch removes rsnd_mod_to_io() from rsnd_src_xxx()
and related function.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Each Renesas sound mod (= SSI/SRC/DVC) might be called from many paths
if it supports MIXer. In such case, mod <-> io is no longer 1:1
relationship. This patch removes rsnd_mod_to_io() from rsnd_ssi_xxx()
and related function.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Each Renesas sound mod (= SSI/SRC/DVC) might be called from many paths
if it supports MIXer. In such case, mod <-> io is no longer 1:1
relationship. This patch removes rsnd_mod_to_io() from rsnd_dma_xxx()
and related function
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Each Renesas sound mod (= SSI/SRC/DVC) might be called from many paths
if it supports MIXer. In such case, mod <-> io is no longer 1:1
relationship. This patch removes rsnd_mod_to_io() from rsnd_get_adinr()
and its related function
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Each Renesas sound mod (= SSI/SRC/DVC) might be called from many paths
if it supports MIXer. In such case, mod <-> io is no longer 1:1
relationship. Then, interrupt handler can't use rsnd_mod_to_io().
This patch adds SSI/SRC/DMA common interrupt handler frame
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Each Renesas sound mod (= SSI/SRC/DVC) might be called from many paths
if it supports MIXer. In such case, mod <-> io is no longer 1:1
relationship. This means we can't call rsnd_mod_to_io() any more.
This patch adds struct rsnd_dai_stream to each function as parameter.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Each Renesas sound mod (= SSI/SRC/DVC) might be called from many paths
if it supports MIXer. In such case, mod <-> io is no longer 1:1
relationship. This means we can't use rsnd_mod_to_io() in SSI/SRC/DMA
interrupt handler. In such case, we need to check all io in interrupt
handler, and then, "priv" is needed.
This patch adds rsnd_priv pointer in rsnd_mod for prepare it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Each Renesas sound mod (= SSI/SRC/DVC) might be called from many paths
if it supports MIXer. Then, we don't need to re-call each mod function
that had been called. This patch count each mod status.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
rsrc-card driver is based on simple-card driver which is caring about
CPU / Codec connection. OTOH, rsrc-card is used for DPCM system.
FE portion is constituted by CPU and dummy Codec, and BE is constituted
by dummy CPU and Codec in DPCM system.
Because of this, current rsrc-card is doing pointless method. It works well
if FE/BE was 1:1, but not good for multi FE/BE.
This patch cleanups rsrc-card driver for DPCM. and this is prepare for
MIX support for Renesas sound driver.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
This is prepare for DPCM cleanup
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Current rsrc-card is assuming 1 FE (= CPU), 1 BE (= codec) on card.
But, it will support multi FE/BE card. This is prepare for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Current dai_link name is using "cpu_dai_name + codec_dai_name",
but one of them is always "snd-soc-dummy-dai" when DPCM.
This patch uses "fe.xxx" for cpu, "be.xxx" for codec.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
'a9e1ac1a9e4585b5("ASoC: rsnd: spin lock for interrupt handler")'
added spin lock under interrupt handler to solve HW restart issue.
OTOH, current rsnd driver calls snd_pcm_period_elapsed() from
rsnd_dai_pointer_update(). but, it will be called under spin lock
if SSI was PIO mode.
If it was called under spin lock, it will call
snd_pcm_update_state() -> snd_pcm_drain_done().
Then, it calls rsnd_soc_dai_trigger() and will be dead-lock.
This patch doesn't call rsnd_dai_pointer_update() under spin lock
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
PIO is used only for checking data path / codec settings. And underrun
is very normal when PIO mode. Let's don't care about under/over run
error when PIO case. Otherwise, 1) too many HW restart happens, 2) some
sounds which need much data transfer can't play since it falls into
error detection method which was created for DMA transfer
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Yet another non-trivial conflicts for HDA legacy stuff.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
When CONFIG_SND_HDA_I915=n, we get a compile warning:
sound/pci/hda/hda_intel.c: In function ‘azx_probe_continue’:
sound/pci/hda/hda_intel.c:1882:2: warning: label ‘skip_i915’ defined but not used [-Wunused-label]
Fix it by putting again ifdef to it. Sigh.
Fixes: bf06848bdbe5 ('ALSA: hda - Continue probing even if i915 binding fails')
Reported-by: Borislav Petkov <bp@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Developing a driver for an Asus X205TA laptop I get these dmesg
errors:
rt5645 i2c-10EC5648:00: ASoC: mux RT5650 IF1 ADC1 Swap Mux has no paths
rt5645 i2c-10EC5648:00: ASoC: mux RT5650 IF1 ADC2 Swap Mux has no paths
rt5645 i2c-10EC5648:00: ASoC: mux RT5650 IF1 ADC3 Swap Mux has no paths
rt5645 i2c-10EC5648:00: ASoC: mux RT5650 IF1 ADC Mux has no paths
rt5645 i2c-10EC5648:00: ASoC: mux RT5650 IF1 DAC1 L Mux has no paths
rt5645 i2c-10EC5648:00: ASoC: mux RT5650 IF1 DAC1 R Mux has no paths
rt5645 i2c-10EC5648:00: ASoC: mux RT5650 IF1 DAC2 L Mux has no paths
rt5645 i2c-10EC5648:00: ASoC: mux RT5650 IF1 DAC2 R Mux has no paths
so, move these muxes to the rt5650_specific_dapm_widgets[] list.
Signed-off-by: Michele Curti <michele.curti@gmail.com>
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The new Dell XPS13 also requires the similar quirk for fixing the
noisy outputs. (But, as the codec was changed, now the fixup for
Latitude is used instead.)
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=99851
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
They're based on DM1500 (ArchWave produced), and BeBoB version 3 is
installed.
$ cat /proc/asound/FCA610/firewire/firmware
Manufacturer: bridgeCo
Protocol Ver: 3
Build Ver: 0
GUID: 0x001564000002AD73
Model ID: 0x03
Model Rev: 0
Firmware Date: 20121102
Firmware Time: 153431
Firmware ID: 0x610
Firmware Ver: 8348
Base Addr: 0x400C0080
Max Size: 1422624
Loader Date: 20121015
Loader Time: 104710
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Behringer FCA610 transmits packets with periodic noisy PCM samples
when receiving no streams, and generates a bit noisy sound.
ALSA BeBoB driver is programmed to establish both in/out connections
when starting streaming, then transfers packets as userspace applications
requested. This means that there's a case that one of incoming/outgoing
streams is running, to save CPU and bandwidth usage. Although, it's natural
to start transferring packets in both direction.
This commit makes this driver to keeps duplex streams always.
Tested-by: Kim Tore Jensen <kim@incendio.no>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
PrismSound Orpheus, Behringer UFX1604 and FCA610 work with BeBoB v3, and
they're confirmed to transmit discontinuous packets in the beginning of
streaming.
payload CIP headers
8 0x00070000 0x9002FFFF
8 0x00070000 0x9002FFFF
8 0x00070000 0x9002FFFF
8 0x00070008 0x9002FFFF <-
8 0x00070008 0x9002FFFF
8 0x00070008 0x9002FFFF
8 0x00070008 0x9002FFFF
8 0x00070008 0x9002FFFF
8 0x00070008 0x9002FFFF
232 0x00070000 0x9002E798 <-
232 0x00070008 0x9002FB99
232 0x00070010 0x90021398
8 0x00070018 0x9002FFFF
(This sample was got with Behringer FCA610 and FFADO library.)
This commit sets CIP_EMPTY_HAS_WRONG_DBC and CIP_SKIP_DBC_ZERO_CHECK to
ignore these discontinuities.
Tested-by: Kim Tore Jensen <kim@incendio.no>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Behringer FCA610 and UFX1604 is confirmed to require more time till
transmitting packets after establishing connections. This seems to
be a quirk of DM1500 ASIC which ArchWave produced.
For this quirk, this commit extends the time to wait up to 2 seconds.
As a result, in worst cases, below userspace functions require 2 seconds
to return.
- snd_pcm_prepare()
- snd_pcm_hw_params()
- snd_pcm_recover()
Tested-by: Kim Tore Jensen <kim@incendio.no>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
BeBoB installed devices have BeBoB register area. This area stores
basic information about its firmware. A register has its protocol
version.
This commit adds 'version' member and store the device's protocol
version to handle v3 quirks in following commits.
Tested-by: Kim Tore Jensen <kim@incendio.no>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
In previous commits, this driver can detect the source of clock as mush
as possible. SYT-Match mode is also available.
This commit purge the restriction.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
The old string literals were completely replaced by new normalized
representation.
This commit obsoletes it.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
This commit changes function prototype and its processing. As a result,
function caller can execute additional processing according to detected
clock source.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
representation for model-dependent structures
Previous commit adds a enumerator as a normalized representation of
clock source, while model-dependent structures still use string literals
for this purpose.
This commit is a preparation for replacement.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Previous commit allows this driver to detect several types of clock
source, while there's no normalized expression for it.
This commit adds a new enumerator for this purpose.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
With BeBoB version 3, current ALSA BeBoB driver detects the type of
current clock signal source wrongly. This is due to a lack of proper
implementation to parse the information.
This commit renews the parser. As a result, this driver detects
SYT-Match clock signal, thus it can start streams with two modes;
SYT-Match mode and the others. SYT-Match mode will be supported in future
commits.
There's a constrain about detected internal/external clock source.
When detecting external clock source, this driver allows userspace
applications to use current sampling rate only. This is due to consider
abour synchronization to external clock sources such as S/PDIF, ADAT or
word-clock.
According to several information from some devices, I guesss that the
internal clock of most devices synchronize to IEEE 1394 cycle start
packet. In this case, by a usual way, it's detect as 'Sync type
of output Music Sub-Unit' connected to 'Sync type of PCR output Unit
(oPCR)', and this driver judges it as internal clock. Therefore,
userspace applications is allowed to request arbitrary supported sampling
rates.
On the other hand, several devices based on BeBoB version 3 have
additional internal clock. In this case, by a usual way, it's detect as
'Sync/Additional type of External input Unit'. Unfortunately, there's no
way to distinguish this sync type from the other external clock sources
such as word-clock. In this case, this driver handles it as external and
userspace applications is forced to use current sampling rate.
I note that when the source of clock is detected as 'Isochronous stream
type of input PCR[0]', it's under 'SYT-Match' mode. In this mode, the
synchronization clock is generated according to SYT-series in received
packets. In this case, this driver generates the series by myself. I
experienced this mode often make the device silent suddenly during
playbacking. This means that the mode is easy to lost synchronization.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Fix the missing dependency on PCM stuff.
[Add the same fix for HAL2, too -- tiwai]
Signed-off-by: Nicholas Mc Guire <hofrat@osadl.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Conflicts:
sound/pci/hda/patch_realtek.c
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Constify the ACPI device ID array, it doesn't need to be writable at
runtime.
Signed-off-by: Mathias Krause <minipli@googlemail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Constify the ACPI device ID array and the register map, no need to have
them writable at runtime. Also drop the unneeded RT5670_INIT_REG_LEN
define.
Signed-off-by: Mathias Krause <minipli@googlemail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Constify the ACPI device ID array and the register map, no need to have
them writable at runtime.
Signed-off-by: Mathias Krause <minipli@googlemail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Constify the ACPI device ID array and the register map, no need to have
them writable at runtime. Also drop the unneeded RT5640_INIT_REG_LEN
define.
Signed-off-by: Mathias Krause <minipli@googlemail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Current code has duplicate code for 16000, 32000 and 48000 sample rates.
get_srate() returns negative error code for unsupported rate, so we can
remove the duplicate code in the swith cases by calling get_srate() first.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
We still got a report that the audio crackles and noises occur with
the recent 4.1 kernels on Dell machines. These machines seem to need
similar workarounds that have been applied to the recent Dell XPS 13
models. Since the codec of these machines (Dell Latitute E7240 and
E7440) is different from XPS 13's one, we need a new fixup entry.
Also, it was confirmed that the previous workaround to disable the
widget power-save (commit [219f47e4f964: ALSA: hda - Disable widget
power-saving for ALC292 & co]) is no longer needed after this fix.
So, this patch includes the partial revert of the commit, too.
Reported-and-tested-by: Mihai Donțu <mihai.dontu@gmail.com>
Tested-by: Jonathan McDowell <noodles@earth.li>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
In order to make TI button interrupt working max98090 codec
Need provide mic bias all the time as long as mic is present
so SHDN and micbias pin are forced on.we also need set max98090
codec bias close or lower than TI bias.We set them in bios/coreboot
kernel reads them from device property
Signed-off-by: Fang, Yang A <yang.a.fang@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Some tiny improvements, cutting 180 bytes off the generated code.
- use strchr() for single-character needle
- compute index using pointer subtraction instead of two strlen()
calls
- factor out the common check for whether the initial part of
kctl->id.name (before the space) is identical to w->name.
Signed-off-by: Rasmus Villemoes <linux@rasmusvillemoes.dk>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Tested-by: Fang, Yang A <yang.a.fang@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
On a HP Envy TouchSmart laptop, there are 2 speakers (main speaker
and subwoofer speaker), 1 headphone and 2 DACs, without this fixup,
the headphone will be assigned to a DAC and the 2 speakers will be
assigned to another DAC, this assignment makes the surround-2.1
channels invalid.
To fix it, here using a DAC/pin preference map to bind the main
speaker to 1 DAC and the subwoofer speaker will be assigned to another
DAC.
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Signed-off-by: Matt Ranostay <mranostay@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
