Update D0 <-> D3 sequence to correctly transition hardware and DSP core
from and to D3. On top of that, set SHIM registers to their recommended
defaults during D0 and D3 proceduces as HW does not reset registers for
us.
Connected to:
[alsa-devel][BUG] bdw-rt5650 DSP boot timeout
https://mailman.alsa-project.org/pipermail/alsa-devel/2019-July/153098.html
Github issue ticket reference:
https://github.com/thesofproject/linux/pull/1842
Tested on:
- BDW-Y RVP with rt286
- SAMUS with rt5677
Proposed solution (both in July 2019 and on github):
'Revert "ASoC: Intel: Work around to fix HW d3 potential crash issue"'
is NAKed as it only covers the problem up and actually brings back the
undefined behavior: some registers (e.g.: APLLSE) are describing LPT
offsets rather than WPT ones. In consequence, during power-transitions
driver issues incorrect writes and leaves the regs of interest alone.
Existing patch - the non-revert - does not resolve the HW D3 issue at
all as it ignores the recommended sequence and does not initialize
hardware registers as expected. And thus, leaving things as are is also
unacceptable.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Tested-by: Ross Zwisler <zwisler@google.com>
Link: https://lore.kernel.org/r/20200330194520.13253-1-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Long series made of a relatively small changes from multiple SOF
contributors. I didn't find a good way to split this series since it
tracks SOF minor ABI changes (backwards-compatible with older firmware
files) and needs to be kept in-order. Future series should be much
shorter.
The main addition is support for an extended firmware manifest, which
helps retrieve capabilities directly from the firmware file instead of
the current IPC mechanism (still supported but will be deprecated).
The IPC is realigned with the firmware, along with type cleanups, and
the DMIC interface is simplified.
The topology changes are mainly about a multi-cpu DAI fix, a new DC
blocking component, better parsing of tuples and new parameters for
ALH (SoundWire) and HDaudio DAIs. New tokens are also added to clarify
the firmware behavior in the case of dependent pipelines, e.g. for
echo reference generation.
Artur Kloniecki (1):
ASoC: SOF: Add XRUN flags field to struct sof_ipc_buffer.
Bard Liao (5):
ASoC: SOF: topology: fix: handle DAI widget connections properly with
multiple CPU DAI's
ASoC: SOF: align sof_ipc_dai_alh_params with FW
ASoC: SOF: topology: Get ALH rate amd channels from topology
ASoC: SOF: topology: fix: parse hda_tokens to &config->hda
ASoC: SOF: topology: Get HDA rate and channels from topology
Jaska Uimonen (2):
ASoC: SOF: topology: stop parsing when all tokens have been found
ASoC: SOF: topology: handle multiple sets of tuple arrays
Karol Trzcinski (6):
ASoC: SOF: Mark get_ext* function ext_hdr arguments as const
ASoC: SOF: Introduce offset in firmware data
ASoC: SOF: Introduce extended manifest
ASoC: SOF: ext_manifest: parse firmware version
ASoC: SOF: ext_manifest: parse windows
ASoC: SOF: ext_manifest: parse compiler version
Pan Xiuli (6):
ASoC: SOF: add probe support extend data
ASoC: SOF: add debug ABI version
ASoC: SOF: change type char to uint8_t in info.h
ASoC: SOF: change type char to uint8_t in trace.h
ASoC: SOF: change type char to uint8_t in topology.h
ASoC: SOF: make sof_ipc_cc_version to fixed length
Sebastiano Carlucci (1):
ASoC: SOF: topology: Add support for DC Blocker
Seppo Ingalsuo (3):
ASoC: SOF: Intel: Fix typo in header file comment text
ASoC: SOF: Intel: Change DMIC load IPC to fixed length
ASoC: SOF: Intel: Rename deprecated DMIC IPC struct field
include/sound/sof.h | 3 +
include/sound/sof/dai-intel.h | 20 +-
include/sound/sof/info.h | 26 ++-
include/sound/sof/topology.h | 16 +-
include/sound/sof/trace.h | 2 +-
include/uapi/sound/sof/abi.h | 2 +-
include/uapi/sound/sof/ext_manifest.h | 91 ++++++++
include/uapi/sound/sof/tokens.h | 8 +
sound/soc/sof/intel/hda-loader.c | 9 +-
sound/soc/sof/loader.c | 226 ++++++++++++++++--
sound/soc/sof/topology.c | 323 ++++++++++++++++----------
11 files changed, 568 insertions(+), 158 deletions(-)
create mode 100644 include/uapi/sound/sof/ext_manifest.h
base-commit: 83b35f4586
--
2.20.1
Daniel Baluta <daniel.baluta@nxp.com>:
From: Daniel Baluta <daniel.baluta@nxp.com>
This patch series adds support for SOF on i.MX8M family. First board
from this family that has a DSP is i.MX8MP.
First 2 patches are trying to fix some compilation issues, the next two
are adding the imx8m support and the last one adds the devicetree
binding.
Changes since v2:
- add reviewed by from Rob to DT patch
- fix ownership for patch 2
Daniel Baluta (3):
ASoC: SOF: imx: Add i.MX8M HW support
ASoC: SOF: Add i.MX8MP device descriptor
dt-bindings: dsp: fsl: Add fsl,imx8mp-dsp entry
Pierre-Louis Bossart (1):
ASoC: SOF: imx: fix undefined reference issue
YueHaibing (1):
ASoC: SOF: imx8: Fix randbuild error
.../devicetree/bindings/dsp/fsl,dsp.yaml | 2 +
sound/soc/sof/imx/Kconfig | 32 +-
sound/soc/sof/imx/Makefile | 2 +
sound/soc/sof/imx/imx8m.c | 279 ++++++++++++++++++
sound/soc/sof/sof-of-dev.c | 14 +
5 files changed, 325 insertions(+), 4 deletions(-)
create mode 100644 sound/soc/sof/imx/imx8m.c
--
2.17.1
Widget's parameters are set in topology and they usually consist of
several different types of tuple arrays like strings, words and bytes.
Here this kind of combination is called a "set".
Lately we've seen more complex widget definitions with multiple
identical sets of tuple arrays. One example is the dmic pdm
configuration, which is currently handled as a special case in token
parsing. This is not scalable for other components with multiple sets.
So add a new function sof_parse_token_sets, which can be used to parse
multiple sets. This function defines the number of sets and an offset to
copy the tokens to correct positions in the destination ipc struct. Old
sof_parse_token function will be a special case of calling
sof_parse_token_sets to parse 1 set with offset 0.
Finally modify the dmic dai link loading to use the new
sof_parse_array_sets to load multiple pdm configs.
Signed-off-by: Jaska Uimonen <jaska.uimonen@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200415202816.934-25-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently if a component source buffer underruns or a component sink
buffer overruns the pipeline will enter an XRUN status and attempt
recovery. This is desired in most pipelines but some topologies need to
support use cases where we expect buffers to underrun or overrun.
Host ---> Proc----> Selector0 --> Buf0 ---- > DAI Playback
|
v
Buf1
|
v
Host <---------------Selector1 <----- Buf2 <----- Echo Ref DAI
In the example above we two host PCMs that can be independently
started/stopped thereby causing buf1 to either underrun or overrun
(and stop the pipelines). Buf1 should be permitted to underrun or overrun
without invoking pipeline XRUN logic and should over write oldest data
(for overrun) and readback 0s (for underrun).
2 flags have been added for use during buffer instantiation:
SOF_BUF_OVERRUN_PERMITTED and SOF_BUF_UNDERRUN_PERMITTED,
along with struct sof_ipc_buffer member fields: flags and reserved.
Flags field is supposed to hold the above-mentioned flags to allow
some control over XRUN behaviour.
Also added reserved field to the structure in case it comes in handy
some time in the future.
This is an incremental ABI change as the new fields are ignored by older
versions of the firmware.
Signed-off-by: Artur Kloniecki <arturx.kloniecki@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Link: https://lore.kernel.org/r/20200415202816.934-16-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Align struct sof_ipc_cc_version to firmware definition in SOF ABI 3.15.0.
The struct definition was changed due to errors in FW build.
The Cadence XCC compiler produces incorrect linkage section sizes, when a
variable length array is used in the compiler version struct. The firmware
definition was changed to a fixed 32 byte compiler description string.
This length covers all released firmware binaries and thus only a minor
ABI change is needed.
As the same structure is used in IPC messages between driver and firmware,
the kernel needs to be aligned to firmware change.
Signed-off-by: Pan Xiuli <xiuli.pan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200415202816.934-15-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This adds skeleton support for the audio DSP hardware found on NXP i.MX8M
platform.
There is one notable difference between i.MX8M and i.MX8, which doesn't
allow us to reuse HW support from imx8.c file designed for i.MX8:
On i.MX8M resources (clocks, power, pinctrl, etc) are managed by the
Linux kernel while on i.MX8 resources are managed by a separate
System Controller Firmware. This makes the interface to those resources
completely different.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Link: https://lore.kernel.org/r/20200409071832.2039-4-daniel.baluta@oss.nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
when do randconfig like this:
CONFIG_SND_SOC_SOF_IMX8_SUPPORT=y
CONFIG_SND_SOC_SOF_IMX8=y
CONFIG_SND_SOC_SOF_OF=y
CONFIG_IMX_DSP=m
CONFIG_IMX_SCU=y
there is a link error:
sound/soc/sof/imx/imx8.o: In function 'imx8_send_msg':
imx8.c:(.text+0x380): undefined reference to 'imx_dsp_ring_doorbell'
Select IMX_DSP in SND_SOC_SOF_IMX8_SUPPORT to fix this
Fixes: f9ad754684 ("ASoC: SOF: imx: fix reverse CONFIG_SND_SOC_SOF_OF dependency")
Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Link: https://lore.kernel.org/r/20200409071832.2039-2-daniel.baluta@oss.nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
On Baytrail/Cherrytrail, the Atom/SST driver fails miserably:
[ 9.741953] intel_sst_acpi 80860F28:00: FW Version 01.0c.00.01
[ 9.832992] intel_sst_acpi 80860F28:00: FW sent error response 0x40034
[ 9.833019] intel_sst_acpi 80860F28:00: FW alloc failed ret -4
[ 9.833028] intel_sst_acpi 80860F28:00: sst_get_stream returned err -5
[ 9.833033] sst-mfld-platform sst-mfld-platform: ASoC: DAI prepare error: -5
[ 9.833037] Baytrail Audio Port: ASoC: prepare FE Baytrail Audio Port failed
[ 9.853942] intel_sst_acpi 80860F28:00: FW sent error response 0x40034
[ 9.853974] intel_sst_acpi 80860F28:00: FW alloc failed ret -4
[ 9.853984] intel_sst_acpi 80860F28:00: sst_get_stream returned err -5
[ 9.853990] sst-mfld-platform sst-mfld-platform: ASoC: DAI prepare error: -5
[ 9.853994] Baytrail Audio Port: ASoC: prepare FE Baytrail Audio Port failed
Commit b56be800f1 ("ASoC: soc-pcm: call
snd_soc_dai_startup()/shutdown() once") was the initial problematic
commit.
Commit 1ba616bd1a ("ASoC: soc-dai: fix DAI startup/shutdown sequence")
was an attempt to fix things but it does not work on Baytrail,
reverting all changes seems necessary for now.
Fixes: 1ba616bd1a ("ASoC: soc-dai: fix DAI startup/shutdown sequence")
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Tested-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20200415030437.23803-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
For some reason, the MI2S DAIs do not have channels_min/max defined.
This means that snd_soc_dai_stream_valid() returns false,
i.e. the DAIs have neither valid playback nor capture stream.
It's quite surprising that this ever worked correctly,
but in 5.7-rc1 this is now failing badly: :)
Commit 0e9cf4c452 ("ASoC: pcm: check if cpu-dai supports a given stream")
introduced a check for snd_soc_dai_stream_valid() before calling
hw_params(), which means that the q6i2s_hw_params() function
was never called, eventually resulting in:
qcom-q6afe aprsvc:q6afe:4:4: no line is assigned
... even though "qcom,sd-lines" is set in the device tree.
Commit 9b5db05936 ("ASoC: soc-pcm: dpcm: Only allow playback/capture if supported")
now even avoids creating PCM devices if the stream is not supported,
which means that it is failing even earlier with e.g.:
Primary MI2S: ASoC: no backend playback stream
Avoid all that trouble by adding channels_min/max for the MI2S DAIs.
Fixes: 24c4cbcfac ("ASoC: qdsp6: q6afe: Add q6afe dai driver")
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200415150050.616392-1-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
At the moment, PCM devices for DPCM are only created based on the
dpcm_playback/capture parameters of the DAI link, without considering
if the CPU/FE DAI is actually capable of playback/capture.
Normally the dpcm_playback/capture parameter should match the
capabilities of the CPU DAI. However, there is no way to set that
parameter from the device tree (e.g. with simple-audio-card or
qcom sound cards). dpcm_playback/capture are always both set to 1.
This causes problems when the CPU DAI does only support playback
or capture. Attemting to open that PCM device with an unsupported
stream type then results in a null pointer dereference:
Unable to handle kernel NULL pointer dereference at virtual address 0000000000000128
Internal error: Oops: 96000044 [#1] PREEMPT SMP
CPU: 3 PID: 1582 Comm: arecord Not tainted 5.7.0-rc1
pc : invalidate_paths_ep+0x30/0xe0
lr : snd_soc_dapm_dai_get_connected_widgets+0x170/0x1a8
Call trace:
invalidate_paths_ep+0x30/0xe0
snd_soc_dapm_dai_get_connected_widgets+0x170/0x1a8
dpcm_path_get+0x38/0xd0
dpcm_fe_dai_open+0x70/0x920
snd_pcm_open_substream+0x564/0x840
snd_pcm_open+0xfc/0x228
snd_pcm_capture_open+0x4c/0x78
snd_open+0xac/0x1a8
...
... because the DAI playback/capture_widget is not set in that case.
We could add checks there to fix the problem (maybe we should
anyway), but much easier is to not expose the device as
playback/capture in the first place. Attemting to use that
device would always fail later anyway.
Add checks for snd_soc_dai_stream_valid() to the DPCM case
to avoid exposing playback/capture if it is not supported.
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Link: https://lore.kernel.org/r/20200415104928.86091-1-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
As mentioned slightly out of patch context in the code, there
is no reset routine for the chip. On boards where the chip is
supplied by a fixed regulator, it might not even be resetted
during (e.g. watchdog) reboot and can be in any state.
If the device is probed with VAG enabled, the driver's probe
routine will generate a loud pop sound when ANA_POWER is
being programmed. Avoid this by properly disabling just the
VAG bit and waiting the required power down time.
Signed-off-by: Sebastian Reichel <sebastian.reichel@collabora.com>
Reviewed-by: Fabio Estevam <festivem@gmail.com>
Link: https://lore.kernel.org/r/20200414181140.145825-1-sebastian.reichel@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Decrease the dmesg verbosity to remove unnecessary logs on SoundWire
platforms, and conversely add more information to help the community
and downstream distros with HDaudio/SOF support (DMIC detection and card
instanciation are the most prevalent issues on GitHub).
Pierre-Louis Bossart (3):
ASoC: codecs: rt1308-sdw: reduce verbosity
ASoC: SOF: Intel: hda: reduce verbosity on SoundWire detection
ASoC: SOF: Intel: hda: log number of microphones detected in NHLT
tables
Ranjani Sridharan (1):
ASoC: soc-core: Add dynamic debug logs in soc_dai_link_sanity_check()
sound/soc/codecs/rt1308-sdw.c | 4 ++--
sound/soc/soc-core.c | 18 +++++++++++++++---
sound/soc/sof/intel/hda.c | 10 ++++++----
3 files changed, 23 insertions(+), 9 deletions(-)
base-commit: dd8e871d4e
--
2.20.1
Hello,
Some devices have a 4-pin jack instead of a 3-pin and currently the
WM8903 configuration is hardcoded to the case of 3-pin jack in the
Tegra's ASoC driver. A new device-tree property is required in order
to convey that hardware has a 4-pin jack, and thus, microphone's
detection needs to be done in a different way.
In particular this is needed for Acer A500 tablet device that has
a 4-pin headset jack, otherwise userspace sees headset instead of
headphones and internal microphone isn't enabled by ALSA UCM rule
when it should be. Please review and apply, thanks in advance.
Dmitry Osipenko (2):
dt-bindings: sound: tegra-wm8903: Document new nvidia,headset property
ASoC: tegra: tegra_wm8903: Support nvidia,headset property
.../devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt | 1 +
sound/soc/tegra/tegra_wm8903.c | 6 +++++-
2 files changed, 6 insertions(+), 1 deletion(-)
--
2.25.1
