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[alsa-devel] [PATCH v3 0/5] ASoC: mediatek: add support for mt6797 SoC
by KaiChieh Chuang 25 Apr '18

25 Apr '18

This patch adds basic support for Mediatek AFE for MT6797 SoC. Change since the previous patch set: * mt6797, return IRQ_NONE when receive unexpected irq signal. * mt6351, gain control use tlv * mt6351, set regmap to component by snd_soc_component_init_regmap KaiChieh Chuang (5): ASoC: add mt6351 codec driver ASoC: mt6797: add structure define and clock control function for 6797 ASoC: mt6797: add mt6797 platform driver ASoC: add mt6797-mt6351 driver and config option ASoC: mediatek: add documents for mt6797 Documentation/devicetree/bindings/sound/mt6351.txt | 16 + .../devicetree/bindings/sound/mt6797-afe-pcm.txt | 42 + .../devicetree/bindings/sound/mt6797-mt6351.txt | 14 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/mt6351.c | 1516 ++++++++++++++++++++ sound/soc/codecs/mt6351.h | 108 ++ sound/soc/mediatek/Kconfig | 20 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/mt6797/Makefile | 19 + sound/soc/mediatek/mt6797/mt6797-afe-clk.c | 132 ++ sound/soc/mediatek/mt6797/mt6797-afe-clk.h | 25 + sound/soc/mediatek/mt6797/mt6797-afe-common.h | 57 + sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 1241 ++++++++++++++++ sound/soc/mediatek/mt6797/mt6797-interconnection.h | 41 + sound/soc/mediatek/mt6797/mt6797-mt6351.c | 186 +++ sound/soc/mediatek/mt6797/mt6797-reg.h | 846 +++++++++++ 17 files changed, 4270 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mt6351.txt create mode 100644 Documentation/devicetree/bindings/sound/mt6797-afe-pcm.txt create mode 100644 Documentation/devicetree/bindings/sound/mt6797-mt6351.txt create mode 100644 sound/soc/codecs/mt6351.c create mode 100644 sound/soc/codecs/mt6351.h create mode 100644 sound/soc/mediatek/mt6797/Makefile create mode 100644 sound/soc/mediatek/mt6797/mt6797-afe-clk.c create mode 100644 sound/soc/mediatek/mt6797/mt6797-afe-clk.h create mode 100644 sound/soc/mediatek/mt6797/mt6797-afe-common.h create mode 100644 sound/soc/mediatek/mt6797/mt6797-afe-pcm.c create mode 100644 sound/soc/mediatek/mt6797/mt6797-interconnection.h create mode 100644 sound/soc/mediatek/mt6797/mt6797-mt6351.c create mode 100644 sound/soc/mediatek/mt6797/mt6797-reg.h -- 2.12.5

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[alsa-devel] [PATCH v4 0/5] ASoC: mediatek: add support for mt6797 SoC
by KaiChieh Chuang 25 Apr '18

25 Apr '18

This patch adds basic support for Mediatek AFE for MT6797 SoC. Change since the previous patch set: * mt6797, return IRQ_NONE when receive unexpected irq signal. * mt6351, gain control use tlv * mt6351, set regmap to component by snd_soc_component_init_regmap * mt6351, playback_tlv, starts from -10 dB KaiChieh Chuang (5): ASoC: add mt6351 codec driver ASoC: mt6797: add structure define and clock control function for 6797 ASoC: mt6797: add mt6797 platform driver ASoC: add mt6797-mt6351 driver and config option ASoC: mediatek: add documents for mt6797 Documentation/devicetree/bindings/sound/mt6351.txt | 16 + .../devicetree/bindings/sound/mt6797-afe-pcm.txt | 42 + .../devicetree/bindings/sound/mt6797-mt6351.txt | 14 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/mt6351.c | 1516 ++++++++++++++++++++ sound/soc/codecs/mt6351.h | 108 ++ sound/soc/mediatek/Kconfig | 20 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/mt6797/Makefile | 19 + sound/soc/mediatek/mt6797/mt6797-afe-clk.c | 132 ++ sound/soc/mediatek/mt6797/mt6797-afe-clk.h | 25 + sound/soc/mediatek/mt6797/mt6797-afe-common.h | 57 + sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 1241 ++++++++++++++++ sound/soc/mediatek/mt6797/mt6797-interconnection.h | 41 + sound/soc/mediatek/mt6797/mt6797-mt6351.c | 186 +++ sound/soc/mediatek/mt6797/mt6797-reg.h | 846 +++++++++++ 17 files changed, 4270 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mt6351.txt create mode 100644 Documentation/devicetree/bindings/sound/mt6797-afe-pcm.txt create mode 100644 Documentation/devicetree/bindings/sound/mt6797-mt6351.txt create mode 100644 sound/soc/codecs/mt6351.c create mode 100644 sound/soc/codecs/mt6351.h create mode 100644 sound/soc/mediatek/mt6797/Makefile create mode 100644 sound/soc/mediatek/mt6797/mt6797-afe-clk.c create mode 100644 sound/soc/mediatek/mt6797/mt6797-afe-clk.h create mode 100644 sound/soc/mediatek/mt6797/mt6797-afe-common.h create mode 100644 sound/soc/mediatek/mt6797/mt6797-afe-pcm.c create mode 100644 sound/soc/mediatek/mt6797/mt6797-interconnection.h create mode 100644 sound/soc/mediatek/mt6797/mt6797-mt6351.c create mode 100644 sound/soc/mediatek/mt6797/mt6797-reg.h -- 2.12.5

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[alsa-devel] [PATCH v5 00/23] ASoC: qcom: Add support to QDSP based Audio
by srinivas.kandagatla＠linaro.org 25 Apr '18

25 Apr '18

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Thankyou everyone for providing feedback on v4 patchset. This patchset aims to provide a basic version of QCOM DSP based audio support which is available in downstream andriod kernels. This patchset support audio playback on HDMI-RX, MI2S, SLIMBus and will add support to other features as we move on. QDSP has both static and dynamic modules. static modules like AFE (Audio FrontEnd), ADM (Audio Device Manager), ASM(Audio Stream Manager) and CORE to provide this audio services. All these services use APR (Asynchronous Packet Router) protocol via smd/glink transport to communicate with Application processor. More details on each module is availble in there respective patch. This patchset is tested on DB820c, with HDMI audio playback, MI2S on DB410c on top of mainline, Also tested SLIMBus analog audio using wcd9355 with an additional patches. Previous version of these patches are also tested on SDM845 by Rohit. Here is my test branch incase someone want to try these patches https://git.linaro.org/people/srinivas.kandagatla/linux.git/log/?h=v4.16-qd… Here is block diagram to give a quick overview of the components +---------+ +---------+ +---------+ | q6asm | |q6routing| | q6afe | | dais | <------> | mixers | <-----> | dais | +---------+ +---------+ +---------+ ^ ^ ^ | | | | +------------------+----------------+ | | | | | | v v v v v +---------+ +---------+ +---------+ | q6ASM | | q6ADM | | q6AFE | +---------+ +---------+ +---------+ ^ ^ ^ ^ | | | CPU Side | ------+---------------------+-------------------+-------- | | | | |APR(smd/glink) | | | | | +------------------+----------------+ | | | | | | +-----+--+-----------------------------------+--+------- | | | | | QDSP Side | v v v v v v +---------+ +---------+ +---------+ | ASM | <------> | ADM | <-----> | AFE | +---------+ +---------+ +---------+ ^ | +-------------------+ | ---------------------------+-------------------------- | Audio I/O | v v +--------------------------------------------------+ | Audio devices | | CODEC | HDMI-TX | PCM | SLIMBUS | I2S |MI2S |...| | | +--------------------------------------------------+ Changes since v4 (https://lkml.org/lkml/2018/3/9/1207) - Updated dt bindings as suggested by Rob H. - removed dt patch as db820c requires some more display dt patches to get hdmi audio. - remove static flags in q6core as suggested by Rohit. - add Rohits tested and reviewed tag - fixed few static checker warnings. - use NULL topology flag as suggested by Rohit. Thanks, srini Srinivas Kandagatla (23): soc: qcom dt-bindings: Add APR bus bindings soc: qcom: Add APR bus driver ASoC: qdsp6: q6common: Add qdsp6 helper functions ASoC: qdsp6: dt-bindings: Add q6afe dt bindings ASoC: qdsp6: dt-bindings: Add q6adm dt bindings ASoC: qdsp6: dt-bindings: Add q6asm dt bindings ASoC: qdsp6: q6afe: Add q6afe driver ASoC: qdsp6: qdafe: Add SLIMBus port Support ASoC: qdsp6: q6afe: Add support to MI2S ports ASoC: qdsp6: q6afe: Add support to MI2S sysclks ASoC: qdsp6: q6adm: Add q6adm driver ASoC: qdsp6: q6asm: Add q6asm driver ASoC: qdsp6: q6asm: Add support to memory map and unmap ASoC: qdsp6: q6asm: Add support to audio stream apis ASoC: qdsp6: q6core: Add q6core driver ASoC: qdsp6: q6routing: Add q6routing driver ASoC: qdsp6: q6routing: Add support to all SLIMBus Mixers ASoC: qdsp6: q6routing: Add support to MI2S Mixers ASoC: qdsp6: q6afe: Add q6afe dai driver ASoC: qdsp6: q6asm: Add q6asm dai driver ASoC: qdsp6: dt-bindings: Add apq8096 machine bindings ASoC: qcom: apq8096: Add db820c machine driver MAINTAINERS: Add myself as co-maintainer of qcom audio .../devicetree/bindings/soc/qcom/qcom,apr.txt | 85 ++ .../devicetree/bindings/sound/qcom,apq8096.txt | 76 ++ .../devicetree/bindings/sound/qcom,q6adm.txt | 33 + .../devicetree/bindings/sound/qcom,q6afe.txt | 88 ++ .../devicetree/bindings/sound/qcom,q6asm.txt | 33 + MAINTAINERS | 1 + drivers/soc/qcom/Kconfig | 9 + drivers/soc/qcom/Makefile | 1 + drivers/soc/qcom/apr.c | 385 ++++++ include/dt-bindings/soc/qcom,apr.h | 27 + include/dt-bindings/sound/qcom,q6afe.h | 31 + include/dt-bindings/sound/qcom,q6asm.h | 22 + include/linux/mod_devicetable.h | 11 + include/linux/soc/qcom/apr.h | 132 ++ sound/soc/qcom/Kconfig | 50 + sound/soc/qcom/Makefile | 5 + sound/soc/qcom/apq8096.c | 238 ++++ sound/soc/qcom/qdsp6/Makefile | 8 + sound/soc/qcom/qdsp6/q6adm.c | 633 ++++++++++ sound/soc/qcom/qdsp6/q6adm.h | 25 + sound/soc/qcom/qdsp6/q6afe-dai.c | 752 +++++++++++ sound/soc/qcom/qdsp6/q6afe.c | 1068 ++++++++++++++++ sound/soc/qcom/qdsp6/q6afe.h | 193 +++ sound/soc/qcom/qdsp6/q6asm-dai.c | 632 ++++++++++ sound/soc/qcom/qdsp6/q6asm.c | 1310 ++++++++++++++++++++ sound/soc/qcom/qdsp6/q6asm.h | 69 ++ sound/soc/qcom/qdsp6/q6core.c | 380 ++++++ sound/soc/qcom/qdsp6/q6core.h | 15 + sound/soc/qcom/qdsp6/q6dsp-common.c | 66 + sound/soc/qcom/qdsp6/q6dsp-common.h | 24 + sound/soc/qcom/qdsp6/q6dsp-errno.h | 51 + sound/soc/qcom/qdsp6/q6routing.c | 982 +++++++++++++++ sound/soc/qcom/qdsp6/q6routing.h | 9 + 33 files changed, 7444 insertions(+) create mode 100644 Documentation/devicetree/bindings/soc/qcom/qcom,apr.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,apq8096.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6adm.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6afe.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6asm.txt create mode 100644 drivers/soc/qcom/apr.c create mode 100644 include/dt-bindings/soc/qcom,apr.h create mode 100644 include/dt-bindings/sound/qcom,q6afe.h create mode 100644 include/dt-bindings/sound/qcom,q6asm.h create mode 100644 include/linux/soc/qcom/apr.h create mode 100644 sound/soc/qcom/apq8096.c create mode 100644 sound/soc/qcom/qdsp6/Makefile create mode 100644 sound/soc/qcom/qdsp6/q6adm.c create mode 100644 sound/soc/qcom/qdsp6/q6adm.h create mode 100644 sound/soc/qcom/qdsp6/q6afe-dai.c create mode 100644 sound/soc/qcom/qdsp6/q6afe.c create mode 100644 sound/soc/qcom/qdsp6/q6afe.h create mode 100644 sound/soc/qcom/qdsp6/q6asm-dai.c create mode 100644 sound/soc/qcom/qdsp6/q6asm.c create mode 100644 sound/soc/qcom/qdsp6/q6asm.h create mode 100644 sound/soc/qcom/qdsp6/q6core.c create mode 100644 sound/soc/qcom/qdsp6/q6core.h create mode 100644 sound/soc/qcom/qdsp6/q6dsp-common.c create mode 100644 sound/soc/qcom/qdsp6/q6dsp-common.h create mode 100644 sound/soc/qcom/qdsp6/q6dsp-errno.h create mode 100644 sound/soc/qcom/qdsp6/q6routing.c create mode 100644 sound/soc/qcom/qdsp6/q6routing.h -- 2.16.2

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[alsa-devel] [PATCH v4 00/13] soundwire: Add stream support
by Vinod Koul 25 Apr '18

25 Apr '18

This series adds support in SoundWire subsystem for: - Documentation for stream support - stream management - data port management - DAI ops in cadence and Intel drivers - ASoC API to propagate SDW stream Updates in v4: - Add more comments for sdw_release_master_stream() - Make sure ports are freed upon sdw_stream_remove_slave() - Clarify snd_soc_dai_set_sdw_stream() and setting of NULL upon release. Updates in v3: - Remove stream state transition checks and redefine stream states - Update bansk switch routines - Other nitpicks pointed out by Pierre. Updates in v2: - Make ASoC API inlined - Make stream states as states and not action - Update the direction enum - Fix some typos and comment updates Sanyog Kale (7): Documentation: soundwire: Add more documentation soundwire: Add support for SoundWire stream management soundwire: Add support for port management soundwire: Add Master and Slave port programming soundwire: Add helpers for ports operations soundwire: Add bank switch routine soundwire: Add stream configuration APIs Shreyas NC (2): ASoC: Add SoundWire stream programming interface soundwire: Remove cdns_master_ops Vinod Koul (4): soundwire: cdns: Add port routines soundwire: cdns: Add stream routines soundwire: intel: Add stream initialization soundwire: intel: Add audio DAI ops .../driver-api/soundwire/error_handling.rst | 65 + Documentation/driver-api/soundwire/index.rst | 3 + Documentation/driver-api/soundwire/locking.rst | 106 ++ Documentation/driver-api/soundwire/stream.rst | 372 +++++ drivers/soundwire/Kconfig | 2 +- drivers/soundwire/Makefile | 2 +- drivers/soundwire/bus.c | 43 + drivers/soundwire/bus.h | 72 + drivers/soundwire/cadence_master.c | 453 +++++- drivers/soundwire/cadence_master.h | 151 ++ drivers/soundwire/intel.c | 528 ++++++- drivers/soundwire/intel.h | 4 + drivers/soundwire/intel_init.c | 3 + drivers/soundwire/stream.c | 1480 ++++++++++++++++++++ include/linux/soundwire/sdw.h | 332 ++++- include/linux/soundwire/sdw_intel.h | 14 + include/sound/soc-dai.h | 23 + 17 files changed, 3639 insertions(+), 14 deletions(-) create mode 100644 Documentation/driver-api/soundwire/error_handling.rst create mode 100644 Documentation/driver-api/soundwire/locking.rst create mode 100644 Documentation/driver-api/soundwire/stream.rst create mode 100644 drivers/soundwire/stream.c -- 2.7.4

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[alsa-devel] [PATCH] ALSA: hda - Skip jack and others for non-existing PCM streams
by Takashi Iwai 25 Apr '18

25 Apr '18

When CONFIG_SND_DYNAMIC_MINORS isn't set, there are only limited number of devices available, and HD-audio, especially with HDMI/DP codec, will fail to create more than two devices. The driver warns about the lack of such devices and skips the PCM device creations, but the HDMI driver still tries to create the corresponding JACK, SPDIF and ELD controls even for the non-existing PCM substreams. This results in confusion on user-space, and even may break the operation. Similarly, Intel HDMI/DP codec builds the ELD notification from i915 graphics driver, and this may be broken if a notification is sent for the non-existing PCM stream. This patch adds the check of the existence of the assigned PCM substream in the both scenarios above, and skips the further operation if the PCM substream is not assigned. Fixes: 9152085defb6 ("ALSA: hda - add DP MST audio support") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_hdmi.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index b4f1b6e88305..7d7eb1354eee 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1383,6 +1383,8 @@ static void hdmi_pcm_setup_pin(struct hdmi_spec *spec, pcm = get_pcm_rec(spec, per_pin->pcm_idx); else return; + if (!pcm->pcm) + return; if (!test_bit(per_pin->pcm_idx, &spec->pcm_in_use)) return; @@ -2151,8 +2153,13 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) int dev, err; int pin_idx, pcm_idx; - for (pcm_idx = 0; pcm_idx < spec->pcm_used; pcm_idx++) { + if (!get_pcm_rec(spec, pcm_idx)->pcm) { + /* no PCM: mark this for skipping permanently */ + set_bit(pcm_idx, &spec->pcm_bitmap); + continue; + } + err = generic_hdmi_build_jack(codec, pcm_idx); if (err < 0) return err; -- 2.16.3

1 0

[alsa-devel] ASoC: BCM2835-I2S: TDM slots / slot mask ignored
by Steven Eckhoff 25 Apr '18

25 Apr '18

Hi, I am testing a codec driver interfaced to the Raspberry Pi in TDM mode and on the logic analyzer I see that the I2S controller is shifting out data in slots 2 - 7 when playing 2 channel audio. The codec is master and only works at 256 x fs, so after the frame sync slots 0 and 1 transfer as they should but after that data continues to be shifted out. It is my understanding that the remaining bits should be zero padded. On a similar note I see that the I2S bcm2835-i2s driver only supports 2 slots, but it seems it can also do 8. For example if I take an 8 channel audio source file(NOTE: I had to hack the file to look like 2 channels so hw_params didn't return an error) and play it I hear the first two channels fine and in the part that I am writing the dirver for I can easily route the other slots to additional DACs and get those to play as well. So I guess I am wondering about two things: 1. In the first case why does my 2 channel audio get shifted out on all 8 slots? It seems if configured for 2 it should only shift data in the first two and wait for the next fsync before starting to shift again. 2. Why does the driver obly support 2 channels when it seems the hardware is capable of more? I figure there is something about this controller that made this tricky or impossible, but I am still curious about it. Regards, Steven

1 0

[alsa-devel] azx_probe() called twice
by Paul Menzel 25 Apr '18

25 Apr '18

Dear Linux folks, With the attached debug patch, `azx_probe()` seems to have been called twice on the ASRock E350M1. Is that expected? It also differs, that it’s only called once during some boots, and twice during others. [ 0.000000] Linux version 4.17.0-rc2+ (root@35489178f57d) (gcc version 7.3.0 (Debian 7.3.0-15)) #12 SMP Mon Apr 23 06:57:55 UTC 2018 […] [ 2.746962] calling azx_driver_init+0x0/0xfe4 [snd_hda_intel] @ 214 [ 2.777881] random: get_random_u32 called from arch_rnd.part.2+0x18/0x40 with crng_init=1 [ 2.777900] random: get_random_u32 called from load_elf_binary+0x76a/0x1d20 with crng_init=1 [ 2.777907] random: get_random_u32 called from arch_align_stack+0x45/0x70 with crng_init=1 [ 2.777938] random: get_random_u32 called from arch_rnd.part.2+0x18/0x40 with crng_init=1 [ 2.778822] random: get_random_u32 called from arch_align_stack+0x45/0x70 with crng_init=1 [ 2.778846] random: get_random_u32 called from randomize_page+0x42/0x57 with crng_init=1 [ 2.781026] calling shpcd_init+0x0/0x1000 [shpchp] @ 219 [ 2.781154] random: get_random_u32 called from cache_alloc_refill+0x5bb/0x13d0 with crng_init=1 [ 2.781477] calling k10temp_driver_init+0x0/0x1000 [k10temp] @ 215 [ 2.796426] usb usb3: New USB device found, idVendor=1d6b, idProduct=0001, bcdDevice= 4.17 [ 2.796434] usb usb3: New USB device strings: Mfr=3, Product=2, SerialNumber=1 [ 2.796438] usb usb3: Product: OHCI PCI host controller [ 2.796441] usb usb3: Manufacturer: Linux 4.17.0-rc2+ ohci_hcd [ 2.796445] usb usb3: SerialNumber: 0000:00:12.0 [ 2.796922] hub 3-0:1.0: USB hub found [ 2.796968] hub 3-0:1.0: 5 ports detected [ 2.798039] ohci-pci 0000:00:13.0: OHCI PCI host controller [ 2.798056] ohci-pci 0000:00:13.0: new USB bus registered, assigned bus number 4 [ 2.798123] ohci-pci 0000:00:13.0: irq 18, io mem 0xf0149000 [ 2.804555] random: get_random_u32 called from cache_random_seq_create+0xa3/0x1f0 with crng_init=1 [ 2.804604] random: get_random_u32 called from copy_process.part.33+0x186/0x2100 with crng_init=1 [ 2.804625] random: get_random_u32 called from cache_random_seq_create+0xa3/0x1f0 with crng_init=1 [ 2.804669] random: get_random_u32 called from cache_alloc_refill+0x5bb/0x13d0 with crng_init=1 [ 2.860446] usb usb4: New USB device found, idVendor=1d6b, idProduct=0001, bcdDevice= 4.17 [ 2.860455] usb usb4: New USB device strings: Mfr=3, Product=2, SerialNumber=1 [ 2.860459] usb usb4: Product: OHCI PCI host controller [ 2.860463] usb usb4: Manufacturer: Linux 4.17.0-rc2+ ohci_hcd [ 2.860466] usb usb4: SerialNumber: 0000:00:13.0 [ 2.860899] hub 4-0:1.0: USB hub found [ 2.860941] hub 4-0:1.0: 5 ports detected [ 2.861606] 1 [ 2.861632] 2 [ 2.861991] 3 [ 2.861992] 4 [ 2.861993] 5 [ 2.861995] 6 [ 2.861996] 7 [ 2.862004] 8 [ 2.862005] 9 [ 2.862290] 1 [ 2.862307] 2 [ 2.862642] 3 [ 2.862643] 4 [ 2.862645] 5 [ 2.862646] 6 [ 2.862647] 7 [ 2.862654] 8 [ 2.862655] 9 [ 2.862934] initcall azx_driver_init+0x0/0xfe4 [snd_hda_intel] returned 0 after 79525 usecs Please find the full log attached. Kind regards, Paul

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[alsa-devel] [GIT PULL] ASoC fixes for v4.17
by Mark Brown 25 Apr '18

25 Apr '18

The following changes since commit 445bb423f6a3c50788529a33b82e45148001e855: Merge remote-tracking branch 'asoc/topic/zx_aud96p22' into asoc-next (2018-03-28 10:32:03 +0800) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-4.17-rc2 for you to fetch changes up to a8419a0cd98ddf628a9e38a92110af7cc650dde7: ASoC: msm8916-wcd-analog: use threaded context for mbhc events (2018-04-19 12:30:42 +0100) ---------------------------------------------------------------- ASoC: Fixes for v4.17 A small batch of fixes collected since the merge window, none of which are particularly large or remarkable. They've all been cooking in -next for a while. ---------------------------------------------------------------- Arnd Bergmann (1): ASoC: rsnd: mark PM functions __maybe_unused Dan Carpenter (1): ASoC: topology: fix some tiny memory leaks Danny Smith (1): ASoC: adau17x1: Handling of DSP_RUN register during fw setup Liam Girdwood (1): ASoC: topology: Check widget kcontrols before deref Nicolin Chen (2): ASoC: fsl_esai: Fix divisor calculation failure at lower ratio ASoC: fsl_ssi: Fix mode setting when changing channel number Pierre-Louis Bossart (1): ASoC: Intel: atom: fix ACPI/PCI Kconfig Srinivas Kandagatla (1): ASoC: msm8916-wcd-analog: use threaded context for mbhc events Tero Kristo (1): ASoC: dmic: Fix clock parenting Wei Yongjun (1): ASoC: amd: acp-da7219-max98357: Make symbol da7219_dai_clk static Yan Wang (1): ASoC: topology: Fix bugs of freeing soc topology oder_chiou(a)realtek.com (1): ASoC: rt5514: Add the missing register in the readable table sound/soc/amd/acp-da7219-max98357a.c | 2 +- sound/soc/codecs/adau17x1.c | 26 ++++++++++++++++++++------ sound/soc/codecs/adau17x1.h | 3 ++- sound/soc/codecs/msm8916-wcd-analog.c | 9 ++++++--- sound/soc/codecs/rt5514.c | 3 +++ sound/soc/fsl/fsl_esai.c | 7 +++++++ sound/soc/fsl/fsl_ssi.c | 14 +++++++++++--- sound/soc/intel/Kconfig | 22 +++++++++++++--------- sound/soc/omap/omap-dmic.c | 14 +++++++++++--- sound/soc/sh/rcar/core.c | 4 ++-- sound/soc/soc-topology.c | 14 +++++++++----- 11 files changed, 85 insertions(+), 33 deletions(-)

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[alsa-devel] change the location for one of two front mics for Lenovo ThinkCentre
by Kailang 25 Apr '18

25 Apr '18

Hi Takashi, Lenovo ThinkCentre machine had two front mics. we change the location for one of them. Relation commit: f33f79f3d0e5caf04dd889cd7cf636261970f009 BR, Kailang

2 1

[alsa-devel] [PATCH 0/8] arm: renesas: Change platform dependency to ARCH_RENESAS
by Geert Uytterhoeven 25 Apr '18

25 Apr '18

Hi all, Commit 9b5ba0df4ea4f940 ("ARM: shmobile: Introduce ARCH_RENESAS") started the conversion from ARCH_SHMOBILE to ARCH_RENESAS for Renesas ARM SoCs. This patch series completes the conversion, by: 1. Updating dependencies for drivers that weren't converted yet, 2. Removing the ARCH_SHMOBILE Kconfig symbols on ARM and ARM64. The first 6 patches can be applied independently by subsystem maintainers. The last two patches depend on the first 6 patches, and are thus marked RFC. Thanks for your comments! Geert Uytterhoeven (8): arm: shmobile: Change platform dependency to ARCH_RENESAS dmaengine: shdmac: Change platform check to CONFIG_ARCH_RENESAS [media] v4l: rcar_fdp1: Change platform dependency to ARCH_RENESAS sh_eth: Change platform check to CONFIG_ARCH_RENESAS staging: emxx_udc: Change platform dependency to ARCH_RENESAS ASoC: sh: Update menu title and platform dependency [RFC] ARM: shmobile: Remove the ARCH_SHMOBILE Kconfig symbol [RFC] arm64: renesas: Remove the ARCH_SHMOBILE Kconfig symbol arch/arm/Kconfig | 2 +- arch/arm/Makefile | 2 +- arch/arm/mach-shmobile/Kconfig | 4 --- arch/arm64/Kconfig.platforms | 42 +++++++++++++---------------- drivers/dma/sh/shdmac.c | 50 +++++++++++++++-------------------- drivers/media/platform/Kconfig | 2 +- drivers/net/ethernet/renesas/sh_eth.h | 2 +- drivers/staging/emxx_udc/Kconfig | 2 +- sound/soc/sh/Kconfig | 4 +-- 9 files changed, 47 insertions(+), 63 deletions(-) -- 2.7.4 Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

7 22

[alsa-devel] Fix ALC233 default init
by Kailang 25 Apr '18

25 Apr '18

Hi Takashi, Attach patch was for fix ALC233 for default initial. BR, Kailang

2 3

[alsa-devel] Update ALC255 depop optimize
by Kailang 25 Apr '18

25 Apr '18

Hi Takashi, Attach patch was update ALC255 depop optimize. BR, Kailang

2 1

[alsa-devel] [PATCH] sound: core: Change return type to vm_fault_t
by Souptick Joarder 25 Apr '18

25 Apr '18

Use new return type vm_fault_t for fault handler. For now, this is just documenting that the function returns a VM_FAULT value rather than an errno. Once all instances are converted, vm_fault_t will become a distinct type. Commit 1c8f422059ae ("mm: change return type to vm_fault_t") Signed-off-by: Souptick Joarder <jrdr.linux(a)gmail.com> Reviewed-by: Matthew Wilcox <mawilcox(a)microsoft.com> --- sound/core/pcm_native.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 77ba50d..2d331f4 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3229,7 +3229,7 @@ static __poll_t snd_pcm_capture_poll(struct file *file, poll_table * wait) /* * mmap status record */ -static int snd_pcm_mmap_status_fault(struct vm_fault *vmf) +static vm_fault_t snd_pcm_mmap_status_fault(struct vm_fault *vmf) { struct snd_pcm_substream *substream = vmf->vma->vm_private_data; struct snd_pcm_runtime *runtime; @@ -3265,7 +3265,7 @@ static int snd_pcm_mmap_status(struct snd_pcm_substream *substream, struct file /* * mmap control record */ -static int snd_pcm_mmap_control_fault(struct vm_fault *vmf) +static vm_fault_t snd_pcm_mmap_control_fault(struct vm_fault *vmf) { struct snd_pcm_substream *substream = vmf->vma->vm_private_data; struct snd_pcm_runtime *runtime; @@ -3354,7 +3354,7 @@ static int snd_pcm_mmap_control(struct snd_pcm_substream *substream, struct file /* * fault callback for mmapping a RAM page */ -static int snd_pcm_mmap_data_fault(struct vm_fault *vmf) +static vm_fault_t snd_pcm_mmap_data_fault(struct vm_fault *vmf) { struct snd_pcm_substream *substream = vmf->vma->vm_private_data; struct snd_pcm_runtime *runtime; -- 1.9.1

2 1

[alsa-devel] [PATCH] sound: usb: usx2y: Change return type to vm_fault_t
by Souptick Joarder 25 Apr '18

25 Apr '18

Use new return type vm_fault_t for fault handler. For now, this is just documenting that the function returns a VM_FAULT value rather than an errno. Once all instances are converted, vm_fault_t will become a distinct type. Commit 1c8f422059ae ("mm: change return type to vm_fault_t") Signed-off-by: Souptick Joarder <jrdr.linux(a)gmail.com> Reviewed-by: Matthew Wilcox <mawilcox(a)microsoft.com> --- sound/usb/usx2y/us122l.c | 2 +- sound/usb/usx2y/usX2Yhwdep.c | 2 +- sound/usb/usx2y/usx2yhwdeppcm.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c index ebcab5c..8082f7b 100644 --- a/sound/usb/usx2y/us122l.c +++ b/sound/usb/usx2y/us122l.c @@ -139,7 +139,7 @@ static void usb_stream_hwdep_vm_open(struct vm_area_struct *area) snd_printdd(KERN_DEBUG "%i\n", atomic_read(&us122l->mmap_count)); } -static int usb_stream_hwdep_vm_fault(struct vm_fault *vmf) +static vm_fault_t usb_stream_hwdep_vm_fault(struct vm_fault *vmf) { unsigned long offset; struct page *page; diff --git a/sound/usb/usx2y/usX2Yhwdep.c b/sound/usb/usx2y/usX2Yhwdep.c index d8bd7c9..c1dd9a7 100644 --- a/sound/usb/usx2y/usX2Yhwdep.c +++ b/sound/usb/usx2y/usX2Yhwdep.c @@ -31,7 +31,7 @@ #include "usbusx2y.h" #include "usX2Yhwdep.h" -static int snd_us428ctls_vm_fault(struct vm_fault *vmf) +static vm_fault_t snd_us428ctls_vm_fault(struct vm_fault *vmf) { unsigned long offset; struct page * page; diff --git a/sound/usb/usx2y/usx2yhwdeppcm.c b/sound/usb/usx2y/usx2yhwdeppcm.c index 0d05052..4fd9276 100644 --- a/sound/usb/usx2y/usx2yhwdeppcm.c +++ b/sound/usb/usx2y/usx2yhwdeppcm.c @@ -652,7 +652,7 @@ static void snd_usX2Y_hwdep_pcm_vm_close(struct vm_area_struct *area) } -static int snd_usX2Y_hwdep_pcm_vm_fault(struct vm_fault *vmf) +static vm_fault_t snd_usX2Y_hwdep_pcm_vm_fault(struct vm_fault *vmf) { unsigned long offset; void *vaddr; -- 1.9.1

2 1

[alsa-devel] [PATCH 1/3] ASoC: amd: acp dma driver code cleanup
by Vijendar Mukunda 25 Apr '18

25 Apr '18

Added dma configuration parameters in audio_substream_data structure. Moved dma configuration parameters initialization to dma hw params callback. Removed separate byte count variables for playback and capture. Added variables to store ACP register offsets in audio_substream_data structure. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/acp-pcm-dma.c | 241 ++++++++++++++++++-------------------------- sound/soc/amd/acp.h | 35 +++++-- 2 files changed, 126 insertions(+), 150 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 5ffe2ef..4a4bbdf 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -317,54 +317,21 @@ static void acp_pte_config(void __iomem *acp_mmio, struct page *pg, } static void config_acp_dma(void __iomem *acp_mmio, - struct audio_substream_data *audio_config, + struct audio_substream_data *rtd, u32 asic_type) { - u32 pte_offset, sram_bank; - u16 ch1, ch2, destination, dma_dscr_idx; - - if (audio_config->direction == SNDRV_PCM_STREAM_PLAYBACK) { - pte_offset = ACP_PLAYBACK_PTE_OFFSET; - ch1 = SYSRAM_TO_ACP_CH_NUM; - ch2 = ACP_TO_I2S_DMA_CH_NUM; - sram_bank = ACP_SHARED_RAM_BANK_1_ADDRESS; - destination = TO_ACP_I2S_1; - - } else { - pte_offset = ACP_CAPTURE_PTE_OFFSET; - ch1 = SYSRAM_TO_ACP_CH_NUM; - ch2 = ACP_TO_I2S_DMA_CH_NUM; - switch (asic_type) { - case CHIP_STONEY: - sram_bank = ACP_SHARED_RAM_BANK_3_ADDRESS; - break; - default: - sram_bank = ACP_SHARED_RAM_BANK_5_ADDRESS; - } - destination = FROM_ACP_I2S_1; - } - - acp_pte_config(acp_mmio, audio_config->pg, audio_config->num_of_pages, - pte_offset); - if (audio_config->direction == SNDRV_PCM_STREAM_PLAYBACK) - dma_dscr_idx = PLAYBACK_START_DMA_DESCR_CH12; - else - dma_dscr_idx = CAPTURE_START_DMA_DESCR_CH14; - + acp_pte_config(acp_mmio, rtd->pg, rtd->num_of_pages, + rtd->pte_offset); /* Configure System memory <-> ACP SRAM DMA descriptors */ - set_acp_sysmem_dma_descriptors(acp_mmio, audio_config->size, - audio_config->direction, pte_offset, ch1, - sram_bank, dma_dscr_idx, asic_type); - - if (audio_config->direction == SNDRV_PCM_STREAM_PLAYBACK) - dma_dscr_idx = PLAYBACK_START_DMA_DESCR_CH13; - else - dma_dscr_idx = CAPTURE_START_DMA_DESCR_CH15; + set_acp_sysmem_dma_descriptors(acp_mmio, rtd->size, + rtd->direction, rtd->pte_offset, + rtd->ch1, rtd->sram_bank, + rtd->dma_dscr_idx_1, asic_type); /* Configure ACP SRAM <-> I2S DMA descriptors */ - set_acp_to_i2s_dma_descriptors(acp_mmio, audio_config->size, - audio_config->direction, sram_bank, - destination, ch2, dma_dscr_idx, - asic_type); + set_acp_to_i2s_dma_descriptors(acp_mmio, rtd->size, + rtd->direction, rtd->sram_bank, + rtd->destination, rtd->ch2, + rtd->dma_dscr_idx_2, asic_type); } /* Start a given DMA channel transfer */ @@ -700,7 +667,6 @@ static irqreturn_t dma_irq_handler(int irq, void *arg) static int acp_dma_open(struct snd_pcm_substream *substream) { - u16 bank; int ret = 0; struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *prtd = substream->private_data; @@ -720,6 +686,7 @@ static int acp_dma_open(struct snd_pcm_substream *substream) default: runtime->hw = acp_pcm_hardware_playback; } + adata->bytescount = 0; } else { switch (intr_data->asic_type) { case CHIP_STONEY: @@ -728,6 +695,7 @@ static int acp_dma_open(struct snd_pcm_substream *substream) default: runtime->hw = acp_pcm_hardware_capture; } + adata->bytescount = 0; } ret = snd_pcm_hw_constraint_integer(runtime, @@ -749,28 +717,6 @@ static int acp_dma_open(struct snd_pcm_substream *substream) */ if (!intr_data->play_i2ssp_stream && !intr_data->capture_i2ssp_stream) acp_reg_write(1, adata->acp_mmio, mmACP_EXTERNAL_INTR_ENB); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - intr_data->play_i2ssp_stream = substream; - /* - * For Stoney, Memory gating is disabled,i.e SRAM Banks - * won't be turned off. The default state for SRAM banks is ON. - * Setting SRAM bank state code skipped for STONEY platform. - */ - if (intr_data->asic_type != CHIP_STONEY) { - for (bank = 1; bank <= 4; bank++) - acp_set_sram_bank_state(intr_data->acp_mmio, - bank, true); - } - } else { - intr_data->capture_i2ssp_stream = substream; - if (intr_data->asic_type != CHIP_STONEY) { - for (bank = 5; bank <= 8; bank++) - acp_set_sram_bank_state(intr_data->acp_mmio, - bank, true); - } - } - return 0; } @@ -779,6 +725,7 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, { int status; uint64_t size; + u16 bank; u32 val = 0; struct page *pg; struct snd_pcm_runtime *runtime; @@ -804,6 +751,60 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, acp_reg_write(val, adata->acp_mmio, mmACP_I2S_16BIT_RESOLUTION_EN); } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + switch (adata->asic_type) { + case CHIP_STONEY: + rtd->pte_offset = ACP_ST_PLAYBACK_PTE_OFFSET; + break; + default: + rtd->pte_offset = ACP_PLAYBACK_PTE_OFFSET; + } + rtd->ch1 = SYSRAM_TO_ACP_CH_NUM; + rtd->ch2 = ACP_TO_I2S_DMA_CH_NUM; + rtd->sram_bank = ACP_SRAM_BANK_1_ADDRESS; + rtd->destination = TO_ACP_I2S_1; + rtd->dma_dscr_idx_1 = PLAYBACK_START_DMA_DESCR_CH12; + rtd->dma_dscr_idx_2 = PLAYBACK_START_DMA_DESCR_CH13; + rtd->byte_cnt_high_reg_offset = + mmACP_I2S_TRANSMIT_BYTE_CNT_HIGH; + rtd->byte_cnt_low_reg_offset = mmACP_I2S_TRANSMIT_BYTE_CNT_LOW; + adata->play_i2ssp_stream = substream; + /* + * For Stoney, Memory gating is disabled,i.e SRAM Banks + * won't be turned off. The default state for SRAM banks is ON. + * Setting SRAM bank state code skipped for STONEY platform. + */ + if (adata->asic_type != CHIP_STONEY) { + for (bank = 1; bank <= 4; bank++) + acp_set_sram_bank_state(adata->acp_mmio, + bank, true); + } + } else { + rtd->pte_offset = ACP_CAPTURE_PTE_OFFSET; + rtd->ch1 = SYSRAM_TO_ACP_CH_NUM; + rtd->ch2 = ACP_TO_I2S_DMA_CH_NUM; + switch (adata->asic_type) { + case CHIP_STONEY: + rtd->sram_bank = ACP_SRAM_BANK_2_ADDRESS; + break; + default: + rtd->sram_bank = ACP_SRAM_BANK_5_ADDRESS; + } + rtd->destination = FROM_ACP_I2S_1; + rtd->dma_dscr_idx_1 = CAPTURE_START_DMA_DESCR_CH14; + rtd->dma_dscr_idx_2 = CAPTURE_START_DMA_DESCR_CH15; + rtd->byte_cnt_high_reg_offset = + mmACP_I2S_RECEIVED_BYTE_CNT_HIGH; + rtd->byte_cnt_low_reg_offset = mmACP_I2S_RECEIVED_BYTE_CNT_LOW; + adata->capture_i2ssp_stream = substream; + if (adata->asic_type != CHIP_STONEY) { + for (bank = 5; bank <= 8; bank++) + acp_set_sram_bank_state(adata->acp_mmio, + bank, true); + } + } + size = params_buffer_bytes(params); status = snd_pcm_lib_malloc_pages(substream, size); if (status < 0) @@ -837,26 +838,15 @@ static int acp_dma_hw_free(struct snd_pcm_substream *substream) return snd_pcm_lib_free_pages(substream); } -static u64 acp_get_byte_count(void __iomem *acp_mmio, int stream) +static u64 acp_get_byte_count(struct audio_substream_data *rtd) { - union acp_dma_count playback_dma_count; - union acp_dma_count capture_dma_count; - u64 bytescount = 0; + union acp_dma_count byte_count; - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - playback_dma_count.bcount.high = acp_reg_read(acp_mmio, - mmACP_I2S_TRANSMIT_BYTE_CNT_HIGH); - playback_dma_count.bcount.low = acp_reg_read(acp_mmio, - mmACP_I2S_TRANSMIT_BYTE_CNT_LOW); - bytescount = playback_dma_count.bytescount; - } else { - capture_dma_count.bcount.high = acp_reg_read(acp_mmio, - mmACP_I2S_RECEIVED_BYTE_CNT_HIGH); - capture_dma_count.bcount.low = acp_reg_read(acp_mmio, - mmACP_I2S_RECEIVED_BYTE_CNT_LOW); - bytescount = capture_dma_count.bytescount; - } - return bytescount; + byte_count.bcount.high = acp_reg_read(rtd->acp_mmio, + rtd->byte_cnt_high_reg_offset); + byte_count.bcount.low = acp_reg_read(rtd->acp_mmio, + rtd->byte_cnt_low_reg_offset); + return byte_count.bytescount; } static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) @@ -872,15 +862,10 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) return -EINVAL; buffersize = frames_to_bytes(runtime, runtime->buffer_size); - bytescount = acp_get_byte_count(rtd->acp_mmio, substream->stream); + bytescount = acp_get_byte_count(rtd); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (bytescount > rtd->i2ssp_renderbytescount) - bytescount = bytescount - rtd->i2ssp_renderbytescount; - } else { - if (bytescount > rtd->i2ssp_capturebytescount) - bytescount = bytescount - rtd->i2ssp_capturebytescount; - } + if (bytescount > rtd->bytescount) + bytescount = bytescount - rtd->bytescount; pos = do_div(bytescount, buffersize); return bytes_to_frames(runtime, pos); } @@ -898,21 +883,15 @@ static int acp_dma_prepare(struct snd_pcm_substream *substream) if (!rtd) return -EINVAL; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - config_acp_dma_channel(rtd->acp_mmio, SYSRAM_TO_ACP_CH_NUM, - PLAYBACK_START_DMA_DESCR_CH12, - NUM_DSCRS_PER_CHANNEL, 0); - config_acp_dma_channel(rtd->acp_mmio, ACP_TO_I2S_DMA_CH_NUM, - PLAYBACK_START_DMA_DESCR_CH13, - NUM_DSCRS_PER_CHANNEL, 0); - } else { - config_acp_dma_channel(rtd->acp_mmio, ACP_TO_SYSRAM_CH_NUM, - CAPTURE_START_DMA_DESCR_CH14, - NUM_DSCRS_PER_CHANNEL, 0); - config_acp_dma_channel(rtd->acp_mmio, I2S_TO_ACP_DMA_CH_NUM, - CAPTURE_START_DMA_DESCR_CH15, - NUM_DSCRS_PER_CHANNEL, 0); - } + + config_acp_dma_channel(rtd->acp_mmio, + rtd->ch1, + rtd->dma_dscr_idx_1, + NUM_DSCRS_PER_CHANNEL, 0); + config_acp_dma_channel(rtd->acp_mmio, + rtd->ch2, + rtd->dma_dscr_idx_2, + NUM_DSCRS_PER_CHANNEL, 0); return 0; } @@ -934,15 +913,13 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: - bytescount = acp_get_byte_count(rtd->acp_mmio, - substream->stream); + bytescount = acp_get_byte_count(rtd); + if (rtd->bytescount == 0) + rtd->bytescount = bytescount; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (rtd->i2ssp_renderbytescount == 0) - rtd->i2ssp_renderbytescount = bytescount; - acp_dma_start(rtd->acp_mmio, - SYSRAM_TO_ACP_CH_NUM, false); + acp_dma_start(rtd->acp_mmio, rtd->ch1, false); while (acp_reg_read(rtd->acp_mmio, mmACP_DMA_CH_STS) & - BIT(SYSRAM_TO_ACP_CH_NUM)) { + BIT(rtd->ch1)) { if (!loops--) { dev_err(component->dev, "acp dma start timeout\n"); @@ -950,40 +927,21 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) } cpu_relax(); } - - acp_dma_start(rtd->acp_mmio, - ACP_TO_I2S_DMA_CH_NUM, true); - - } else { - if (rtd->i2ssp_capturebytescount == 0) - rtd->i2ssp_capturebytescount = bytescount; - acp_dma_start(rtd->acp_mmio, - I2S_TO_ACP_DMA_CH_NUM, true); } + acp_dma_start(rtd->acp_mmio, rtd->ch2, true); ret = 0; break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: - /* - * Need to stop only circular DMA channels : - * ACP_TO_I2S_DMA_CH_NUM / I2S_TO_ACP_DMA_CH_NUM. Non-circular - * channels will stopped automatically after its transfer - * completes : SYSRAM_TO_ACP_CH_NUM / ACP_TO_SYSRAM_CH_NUM - */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - ret = acp_dma_stop(rtd->acp_mmio, - SYSRAM_TO_ACP_CH_NUM); - ret = acp_dma_stop(rtd->acp_mmio, - ACP_TO_I2S_DMA_CH_NUM); - rtd->i2ssp_renderbytescount = 0; + acp_dma_stop(rtd->acp_mmio, rtd->ch1); + ret = acp_dma_stop(rtd->acp_mmio, rtd->ch2); } else { - ret = acp_dma_stop(rtd->acp_mmio, - I2S_TO_ACP_DMA_CH_NUM); - ret = acp_dma_stop(rtd->acp_mmio, - ACP_TO_SYSRAM_CH_NUM); - rtd->i2ssp_capturebytescount = 0; + acp_dma_stop(rtd->acp_mmio, rtd->ch2); + ret = acp_dma_stop(rtd->acp_mmio, rtd->ch1); } + rtd->bytescount = 0; break; default: ret = -EINVAL; @@ -1028,8 +986,6 @@ static int acp_dma_close(struct snd_pcm_substream *substream) DRV_NAME); struct audio_drv_data *adata = dev_get_drvdata(component->dev); - kfree(rtd); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { adata->play_i2ssp_stream = NULL; /* @@ -1059,6 +1015,7 @@ static int acp_dma_close(struct snd_pcm_substream *substream) if (!adata->play_i2ssp_stream && !adata->capture_i2ssp_stream) acp_reg_write(0, adata->acp_mmio, mmACP_EXTERNAL_INTR_ENB); + kfree(rtd); return 0; } diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index 0e6089b..62695ed 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -10,17 +10,28 @@ #define ACP_PLAYBACK_PTE_OFFSET 10 #define ACP_CAPTURE_PTE_OFFSET 0 +/* Playback and Capture Offset for Stoney */ +#define ACP_ST_PLAYBACK_PTE_OFFSET 0x04 +#define ACP_ST_CAPTURE_PTE_OFFSET 0x00 + #define ACP_GARLIC_CNTL_DEFAULT 0x00000FB4 #define ACP_ONION_CNTL_DEFAULT 0x00000FB4 #define ACP_PHYSICAL_BASE 0x14000 -/* Playback SRAM address (as a destination in dma descriptor) */ -#define ACP_SHARED_RAM_BANK_1_ADDRESS 0x4002000 - -/* Capture SRAM address (as a source in dma descriptor) */ -#define ACP_SHARED_RAM_BANK_5_ADDRESS 0x400A000 -#define ACP_SHARED_RAM_BANK_3_ADDRESS 0x4006000 +/* + * In case of I2S SP controller instance, Stoney uses SRAM bank 1 for + * playback and SRAM Bank 2 for capture where as in case of BT I2S + * Instance, Stoney uses SRAM Bank 3 for playback & SRAM Bank 4 will + * be used for capture. Carrizo uses I2S SP controller instance. SRAM Banks + * 1, 2, 3, 4 will be used for playback & SRAM Banks 5, 6, 7, 8 will be used + * for capture scenario. + */ +#define ACP_SRAM_BANK_1_ADDRESS 0x4002000 +#define ACP_SRAM_BANK_2_ADDRESS 0x4004000 +#define ACP_SRAM_BANK_3_ADDRESS 0x4006000 +#define ACP_SRAM_BANK_4_ADDRESS 0x4008000 +#define ACP_SRAM_BANK_5_ADDRESS 0x400A000 #define ACP_DMA_RESET_TIME 10000 #define ACP_CLOCK_EN_TIME_OUT_VALUE 0x000000FF @@ -85,9 +96,17 @@ struct audio_substream_data { unsigned int order; u16 num_of_pages; u16 direction; + u16 ch1; + u16 ch2; + u16 destination; + u16 dma_dscr_idx_1; + u16 dma_dscr_idx_2; + u32 pte_offset; + u32 sram_bank; + u32 byte_cnt_high_reg_offset; + u32 byte_cnt_low_reg_offset; uint64_t size; - u64 i2ssp_renderbytescount; - u64 i2ssp_capturebytescount; + u64 bytescount; void __iomem *acp_mmio; }; -- 2.7.4

4 6

[alsa-devel] [PATCH v3][RESEND] media: i2c: tda1997: replace codec to component
by Kuninori Morimoto 25 Apr '18

25 Apr '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Now we can replace Codec to Component. Let's do it. Note: xxx_codec_xxx() -> xxx_component_xxx() .idle_bias_off = 0 -> .idle_bias_on = 1 .ignore_pmdown_time = 0 -> .use_pmdown_time = 1 - -> .endianness = 1 - -> .non_legacy_dai_naming = 1 Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Tim, Mauro 2 weeks passed. I re-send. This replace patch is needed for ALSA SoC, otherwise it can't probe anymore. v2 -> v3 - fixup driver name (= tda1997) drivers/media/i2c/tda1997x.c | 25 +++++++++++++------------ 1 file changed, 13 insertions(+), 12 deletions(-) diff --git a/drivers/media/i2c/tda1997x.c b/drivers/media/i2c/tda1997x.c index 3021913..33d7fcf 100644 --- a/drivers/media/i2c/tda1997x.c +++ b/drivers/media/i2c/tda1997x.c @@ -2444,7 +2444,7 @@ static int tda1997x_pcm_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct tda1997x_state *state = snd_soc_dai_get_drvdata(dai); - struct snd_soc_codec *codec = dai->codec; + struct snd_soc_component *component = dai->component; struct snd_pcm_runtime *rtd = substream->runtime; int rate, err; @@ -2452,11 +2452,11 @@ static int tda1997x_pcm_startup(struct snd_pcm_substream *substream, err = snd_pcm_hw_constraint_minmax(rtd, SNDRV_PCM_HW_PARAM_RATE, rate, rate); if (err < 0) { - dev_err(codec->dev, "failed to constrain samplerate to %dHz\n", + dev_err(component->dev, "failed to constrain samplerate to %dHz\n", rate); return err; } - dev_info(codec->dev, "set samplerate constraint to %dHz\n", rate); + dev_info(component->dev, "set samplerate constraint to %dHz\n", rate); return 0; } @@ -2479,20 +2479,22 @@ static int tda1997x_pcm_startup(struct snd_pcm_substream *substream, .ops = &tda1997x_dai_ops, }; -static int tda1997x_codec_probe(struct snd_soc_codec *codec) +static int tda1997x_codec_probe(struct snd_soc_component *component) { return 0; } -static int tda1997x_codec_remove(struct snd_soc_codec *codec) +static void tda1997x_codec_remove(struct snd_soc_component *component) { - return 0; } -static struct snd_soc_codec_driver tda1997x_codec_driver = { - .probe = tda1997x_codec_probe, - .remove = tda1997x_codec_remove, - .reg_word_size = sizeof(u16), +static struct snd_soc_component_driver tda1997x_codec_driver = { + .probe = tda1997x_codec_probe, + .remove = tda1997x_codec_remove, + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, }; static int tda1997x_probe(struct i2c_client *client, @@ -2737,7 +2739,7 @@ static int tda1997x_probe(struct i2c_client *client, else formats = SNDRV_PCM_FMTBIT_S16_LE; tda1997x_audio_dai.capture.formats = formats; - ret = snd_soc_register_codec(&state->client->dev, + ret = devm_snd_soc_register_component(&state->client->dev, &tda1997x_codec_driver, &tda1997x_audio_dai, 1); if (ret) { @@ -2782,7 +2784,6 @@ static int tda1997x_remove(struct i2c_client *client) struct tda1997x_platform_data *pdata = &state->pdata; if (pdata->audout_format) { - snd_soc_unregister_codec(&client->dev); mutex_destroy(&state->audio_lock); } -- 1.9.1 _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

3 4

[alsa-devel] [alsa-tools][PATCH] Fix clang -Wreserved-user-defined-literal warnings
by Khem Raj 24 Apr '18

24 Apr '18

| us428control.cc:66:18: error: invalid suffix on literal; C++11 requires a space between literal and identifier [-Wreserved-user-defined-literal] | printf("usage: "PROGNAME" [-v verbosity_level 0..2] [-c card] [-D device] [-u usb-device] [-m mode]\n"); | ^ Signed-off-by: Khem Raj <raj.khem(a)gmail.com> --- us428control/us428control.cc | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/us428control/us428control.cc b/us428control/us428control.cc index e839bf4..8cb3c42 100644 --- a/us428control/us428control.cc +++ b/us428control/us428control.cc @@ -63,7 +63,7 @@ static void usage(void) { printf("Tascam US-428 Control\n"); printf("version %s\n", VERSION); - printf("usage: "PROGNAME" [-v verbosity_level 0..2] [-c card] [-D device] [-u usb-device] [-m mode]\n"); + printf("usage: " PROGNAME " [-v verbosity_level 0..2] [-c card] [-D device] [-u usb-device] [-m mode]\n"); printf("mode is one of (us224, us428, mixxx)\n"); } /* -- 2.17.0

2 1

[alsa-devel] [PATCH] ASoC: wm_adsp: Account for name prefixes when toggling preloader
by Charles Keepax 24 Apr '18

24 Apr '18

Use the correct functions to allow a name prefix assigned through codec_conf to be taken into consideration whilst enabling and disabling the preloader widget. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 82b0927e6ed7..b7b914963c62 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -2666,9 +2666,9 @@ int wm_adsp2_preloader_put(struct snd_kcontrol *kcontrol, dsp->preloaded = ucontrol->value.integer.value[0]; if (ucontrol->value.integer.value[0]) - snd_soc_dapm_force_enable_pin(dapm, preload); + snd_soc_component_force_enable_pin(component, preload); else - snd_soc_dapm_disable_pin(dapm, preload); + snd_soc_component_disable_pin(component, preload); snd_soc_dapm_sync(dapm); @@ -2852,11 +2852,11 @@ EXPORT_SYMBOL_GPL(wm_adsp2_event); int wm_adsp2_component_probe(struct wm_adsp *dsp, struct snd_soc_component *component) { - struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); char preload[32]; snprintf(preload, ARRAY_SIZE(preload), "DSP%d Preload", dsp->num); - snd_soc_dapm_disable_pin(dapm, preload); + + snd_soc_component_disable_pin(component, preload); wm_adsp2_init_debugfs(dsp, component); -- 2.11.0

2 1

[alsa-devel] [PATCH v2] ALSA: dice: improve support for ancient firmware for DICE
by Takashi Sakamoto 24 Apr '18

24 Apr '18

In early stage of firmware SDK, DICE seems to lose its backward compatibility due to some registers on global address section. I found this with Alesis Multimix 12 FireWire with ancient firmware (approx. shipped version). According to retrieved log from the unit, global section has 96 byte space. On the other hand, current version of ALSA dice driver assumes that all of supported unit has at least 100 byte space. $ ./firewire-request /dev/fw1 read 0xffffe0000000 28 result: 000: 00 00 00 0a 00 00 00 18 00 00 00 22 00 00 00 8a result: 010: 00 00 00 ac 00 00 01 12 00 00 00 00 00 00 00 00 result: 020: 00 00 00 00 00 00 00 00 This commit adds support for the ancient firmware. Check of global section is loosened to accept the smaller space. The lack of information is already compensated by hard-coded parameters. I experienced that the latest version of Windows driver for this model can't handle this unit, too. This means that TCAT releases firmware SDK without backward compatibility for the ancient firmware. Below list is a early history of driver/firmware package released by Alesis. I investigated on wayback machine on Internet Archive: * Unknown: PAL v1.0.41.2, firmware v1.0.3 * Mar 2006: PAL v1.54.0, firmware v1.0.4 * Dec 2006: PAL v2.0.0.2, firmware v2.0 * Jun 2007: PAL v3.0.41.5, firmware v2.0 * Jul 2007: PAL v3.0.56.2. firmware v2.0 * Jan 2008: PAL v3.0.81.1080, firmware v2.0 If I can assume that firmware version is the same as DICE version, DICE version for the issued firmware may be v1.0.3. According to code base of userspace driver project (FFADO), I can read DICE v1.0.4 supports global space larger than 100 byte. I guess the smaller space of global section is a feature of DICE v1.0.3. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/dice/dice-interface.h | 9 +++++-- sound/firewire/dice/dice-proc.c | 10 +++---- sound/firewire/dice/dice-transaction.c | 49 +++++++++++++++++++--------------- 3 files changed, 40 insertions(+), 28 deletions(-) diff --git a/sound/firewire/dice/dice-interface.h b/sound/firewire/dice/dice-interface.h index 15a484b05298..9cad3d608229 100644 --- a/sound/firewire/dice/dice-interface.h +++ b/sound/firewire/dice/dice-interface.h @@ -174,14 +174,19 @@ */ #define GLOBAL_SAMPLE_RATE 0x05c +/* + * Some old firmware versions do not have the following global registers. + * Windows drivers produced by TCAT lost backward compatibility in its + * early release because they can handle firmware only which supports the + * following registers. + */ + /* * The version of the DICE driver specification that this device conforms to; * read-only. */ #define GLOBAL_VERSION 0x060 -/* Some old firmware versions do not have the following global registers: */ - /* * Supported sample rates and clock sources; read-only. */ diff --git a/sound/firewire/dice/dice-proc.c b/sound/firewire/dice/dice-proc.c index f5c1d1bced59..cc079323ed30 100644 --- a/sound/firewire/dice/dice-proc.c +++ b/sound/firewire/dice/dice-proc.c @@ -148,12 +148,12 @@ static void dice_proc_read(struct snd_info_entry *entry, >> CLOCK_RATE_SHIFT)); snd_iprintf(buffer, " ext status: %08x\n", buf.global.extended_status); snd_iprintf(buffer, " sample rate: %u\n", buf.global.sample_rate); - snd_iprintf(buffer, " version: %u.%u.%u.%u\n", - (buf.global.version >> 24) & 0xff, - (buf.global.version >> 16) & 0xff, - (buf.global.version >> 8) & 0xff, - (buf.global.version >> 0) & 0xff); if (quadlets >= 90) { + snd_iprintf(buffer, " version: %u.%u.%u.%u\n", + (buf.global.version >> 24) & 0xff, + (buf.global.version >> 16) & 0xff, + (buf.global.version >> 8) & 0xff, + (buf.global.version >> 0) & 0xff); snd_iprintf(buffer, " clock caps:"); for (i = 0; i <= 6; ++i) if (buf.global.clock_caps & (1 << i)) diff --git a/sound/firewire/dice/dice-transaction.c b/sound/firewire/dice/dice-transaction.c index 0f0350320ae8..b7e138b5abcf 100644 --- a/sound/firewire/dice/dice-transaction.c +++ b/sound/firewire/dice/dice-transaction.c @@ -265,7 +265,7 @@ int snd_dice_transaction_reinit(struct snd_dice *dice) static int get_subaddrs(struct snd_dice *dice) { static const int min_values[10] = { - 10, 0x64 / 4, + 10, 0x60 / 4, 10, 0x18 / 4, 10, 0x18 / 4, 0, 0, @@ -301,33 +301,40 @@ static int get_subaddrs(struct snd_dice *dice) } } - /* - * Check that the implemented DICE driver specification major version - * number matches. - */ - err = snd_fw_transaction(dice->unit, TCODE_READ_QUADLET_REQUEST, - DICE_PRIVATE_SPACE + - be32_to_cpu(pointers[0]) * 4 + GLOBAL_VERSION, - &version, sizeof(version), 0); - if (err < 0) - goto end; + if (be32_to_cpu(pointers[1]) > 0x18) { + /* + * Check that the implemented DICE driver specification major + * version number matches. + */ + err = snd_fw_transaction(dice->unit, TCODE_READ_QUADLET_REQUEST, + DICE_PRIVATE_SPACE + + be32_to_cpu(pointers[0]) * 4 + GLOBAL_VERSION, + &version, sizeof(version), 0); + if (err < 0) + goto end; - if ((version & cpu_to_be32(0xff000000)) != cpu_to_be32(0x01000000)) { - dev_err(&dice->unit->device, - "unknown DICE version: 0x%08x\n", be32_to_cpu(version)); - err = -ENODEV; - goto end; + if ((version & cpu_to_be32(0xff000000)) != + cpu_to_be32(0x01000000)) { + dev_err(&dice->unit->device, + "unknown DICE version: 0x%08x\n", + be32_to_cpu(version)); + err = -ENODEV; + goto end; + } + + /* Set up later. */ + dice->clock_caps = 1; } dice->global_offset = be32_to_cpu(pointers[0]) * 4; dice->tx_offset = be32_to_cpu(pointers[2]) * 4; dice->rx_offset = be32_to_cpu(pointers[4]) * 4; - dice->sync_offset = be32_to_cpu(pointers[6]) * 4; - dice->rsrv_offset = be32_to_cpu(pointers[8]) * 4; - /* Set up later. */ - if (be32_to_cpu(pointers[1]) * 4 >= GLOBAL_CLOCK_CAPABILITIES + 4) - dice->clock_caps = 1; + /* Old firmware doesn't support these fields. */ + if (pointers[7]) + dice->sync_offset = be32_to_cpu(pointers[6]) * 4; + if (pointers[9]) + dice->rsrv_offset = be32_to_cpu(pointers[8]) * 4; end: kfree(pointers); return err; -- 2.14.1

2 1

[alsa-devel] [PATCH] ALSA: hda - Sanity check of access to SPDIF controls array
by Takashi Iwai 24 Apr '18

24 Apr '18

Put WARN_ON() and bail out if the given index is over the allocated array of the given SPDIF controls. It's merely a sanity check to catch any potential issues (if any). Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 0aa923d129f5..63f177d975fd 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2175,6 +2175,8 @@ static int snd_hda_spdif_default_get(struct snd_kcontrol *kcontrol, int idx = kcontrol->private_value; struct hda_spdif_out *spdif; + if (WARN_ON(codec->spdif_out.used <= idx)) + return -EINVAL; mutex_lock(&codec->spdif_mutex); spdif = snd_array_elem(&codec->spdif_out, idx); ucontrol->value.iec958.status[0] = spdif->status & 0xff; @@ -2282,6 +2284,8 @@ static int snd_hda_spdif_default_put(struct snd_kcontrol *kcontrol, unsigned short val; int change; + if (WARN_ON(codec->spdif_out.used <= idx)) + return -EINVAL; mutex_lock(&codec->spdif_mutex); spdif = snd_array_elem(&codec->spdif_out, idx); nid = spdif->nid; @@ -2308,6 +2312,8 @@ static int snd_hda_spdif_out_switch_get(struct snd_kcontrol *kcontrol, int idx = kcontrol->private_value; struct hda_spdif_out *spdif; + if (WARN_ON(codec->spdif_out.used <= idx)) + return -EINVAL; mutex_lock(&codec->spdif_mutex); spdif = snd_array_elem(&codec->spdif_out, idx); ucontrol->value.integer.value[0] = spdif->ctls & AC_DIG1_ENABLE; @@ -2336,6 +2342,8 @@ static int snd_hda_spdif_out_switch_put(struct snd_kcontrol *kcontrol, unsigned short val; int change; + if (WARN_ON(codec->spdif_out.used <= idx)) + return -EINVAL; mutex_lock(&codec->spdif_mutex); spdif = snd_array_elem(&codec->spdif_out, idx); nid = spdif->nid; @@ -2483,6 +2491,8 @@ void snd_hda_spdif_ctls_unassign(struct hda_codec *codec, int idx) { struct hda_spdif_out *spdif; + if (WARN_ON(codec->spdif_out.used <= idx)) + return; mutex_lock(&codec->spdif_mutex); spdif = snd_array_elem(&codec->spdif_out, idx); spdif->nid = (u16)-1; @@ -2503,6 +2513,8 @@ void snd_hda_spdif_ctls_assign(struct hda_codec *codec, int idx, hda_nid_t nid) struct hda_spdif_out *spdif; unsigned short val; + if (WARN_ON(codec->spdif_out.used <= idx)) + return; mutex_lock(&codec->spdif_mutex); spdif = snd_array_elem(&codec->spdif_out, idx); if (spdif->nid != nid) { -- 2.16.3

1 0

[alsa-devel] [PATCH] ALSA: hda: Request driver probe from an async task
by Paul Menzel 24 Apr '18

24 Apr '18

From: Paul Menzel <pmenzel(a)molgen.mpg.de> Date: Sat, 24 Mar 2018 09:28:43 +0100 On an ASRock E350M1, with Linux 4.17-rc1 according to `initcall_debug` calling `azx_driver_init` takes sometimes more than a few milliseconds, and up to 200 ms. ``` [ 2.892598] calling azx_driver_init+0x0/0xfe4 [snd_hda_intel] @ 218 [ 2.943002] initcall azx_driver_init+0x0/0xfe4 [snd_hda_intel] returned 0 after 49195 usecs ``` Trying to execute the Linux kernel in less than 500 ms, this is quite a hold-up, and therefore request the probe from an async task. With this change, the test shows, that the function returns earlier. ``` [ 3.254800] calling azx_driver_init+0x0/0xfe4 [snd_hda_intel] @ 227 [ 3.254887] initcall azx_driver_init+0x0/0xfe4 [snd_hda_intel] returned 0 after 66 usecs ``` The same behavior is visible on a Dell OptiPlex 7010. The longer times seem to happen, when the module *e1000e* is probed during the same time. Signed-off-by: Paul Menzel <pmenzel(a)molgen.mpg.de> --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 6347256bc017..827834b9c07d 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2671,6 +2671,7 @@ static struct pci_driver azx_driver = { .driver = { .pm = AZX_PM_OPS, }, + .driver.probe_type = PROBE_PREFER_ASYNCHRONOUS, }; module_pci_driver(azx_driver); -- 2.16.2

2 7

[alsa-devel] [PATCH] ALSA: dice: improve support for ancient firmware for DICE
by Takashi Sakamoto 24 Apr '18

24 Apr '18

In early stage of firmware SDK, DICE seems to lose its backward compatibility due to some registers on global address section. I found this with Alesis Multimix 12 FireWire with ancient firmware (approx. shipped version). According to retrieved log from the unit, global section has 96 byte space. On the other hand, current version of ALSA dice driver assumes that all of supported unit has at least 100 byte space. $ ./firewire-request /dev/fw1 read 0xffffe0000000 28 result: 000: 00 00 00 0a 00 00 00 18 00 00 00 22 00 00 00 8a result: 010: 00 00 00 ac 00 00 01 12 00 00 00 00 00 00 00 00 result: 020: 00 00 00 00 00 00 00 00 This commit adds support for the ancient firmware. Check of global section is loosened to accept the smaller space. The lack of information is already compensated by hard-coded parameters. I experienced that the latest version of Windows driver for this model can't handle this unit, too. This means that TCAT releases firmware SDK without backward compatibility for the ancient firmware. Below list is a early history of driver/firmware package released by Alesis. I investigated on wayback machine on Internet Archive: * Unknown: PAL v1.0.41.2, firmware v1.0.3 * Mar 2006: PAL v1.54.0, firmware v1.0.4 * Dec 2006: PAL v2.0.0.2, firmware v2.0 * Jun 2007: PAL v3.0.41.5, firmware v2.0 * Jul 2007: PAL v3.0.56.2. firmware v2.0 * Jan 2008: PAL v3.0.81.1080, firmware v2.0 If I can assume that firmware version is the same as DICE version, DICE version for the issued firmware may be v1.0.3. According to code base of userspace driver project (FFADO), I can read DICE v1.0.4 supports global space larger than 100 byte. I guess the smaller space of global section is a feature of DICE v1.0.3 but no way to clear it. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/dice/dice-interface.h | 9 +++++-- sound/firewire/dice/dice-proc.c | 10 +++---- sound/firewire/dice/dice-transaction.c | 49 +++++++++++++++++++--------------- 3 files changed, 40 insertions(+), 28 deletions(-) diff --git a/sound/firewire/dice/dice-interface.h b/sound/firewire/dice/dice-interface.h index 15a484b05298..9cad3d608229 100644 --- a/sound/firewire/dice/dice-interface.h +++ b/sound/firewire/dice/dice-interface.h @@ -174,14 +174,19 @@ */ #define GLOBAL_SAMPLE_RATE 0x05c +/* + * Some old firmware versions do not have the following global registers. + * Windows drivers produced by TCAT lost backward compatibility in its + * early release because they can handle firmware only which supports the + * following registers. + */ + /* * The version of the DICE driver specification that this device conforms to; * read-only. */ #define GLOBAL_VERSION 0x060 -/* Some old firmware versions do not have the following global registers: */ - /* * Supported sample rates and clock sources; read-only. */ diff --git a/sound/firewire/dice/dice-proc.c b/sound/firewire/dice/dice-proc.c index f5c1d1bced59..cc079323ed30 100644 --- a/sound/firewire/dice/dice-proc.c +++ b/sound/firewire/dice/dice-proc.c @@ -148,12 +148,12 @@ static void dice_proc_read(struct snd_info_entry *entry, >> CLOCK_RATE_SHIFT)); snd_iprintf(buffer, " ext status: %08x\n", buf.global.extended_status); snd_iprintf(buffer, " sample rate: %u\n", buf.global.sample_rate); - snd_iprintf(buffer, " version: %u.%u.%u.%u\n", - (buf.global.version >> 24) & 0xff, - (buf.global.version >> 16) & 0xff, - (buf.global.version >> 8) & 0xff, - (buf.global.version >> 0) & 0xff); if (quadlets >= 90) { + snd_iprintf(buffer, " version: %u.%u.%u.%u\n", + (buf.global.version >> 24) & 0xff, + (buf.global.version >> 16) & 0xff, + (buf.global.version >> 8) & 0xff, + (buf.global.version >> 0) & 0xff); snd_iprintf(buffer, " clock caps:"); for (i = 0; i <= 6; ++i) if (buf.global.clock_caps & (1 << i)) diff --git a/sound/firewire/dice/dice-transaction.c b/sound/firewire/dice/dice-transaction.c index 0f0350320ae8..2410c082bf8e 100644 --- a/sound/firewire/dice/dice-transaction.c +++ b/sound/firewire/dice/dice-transaction.c @@ -265,7 +265,7 @@ int snd_dice_transaction_reinit(struct snd_dice *dice) static int get_subaddrs(struct snd_dice *dice) { static const int min_values[10] = { - 10, 0x64 / 4, + 10, 0x60 / 4, 10, 0x18 / 4, 10, 0x18 / 4, 0, 0, @@ -301,33 +301,40 @@ static int get_subaddrs(struct snd_dice *dice) } } - /* - * Check that the implemented DICE driver specification major version - * number matches. - */ - err = snd_fw_transaction(dice->unit, TCODE_READ_QUADLET_REQUEST, - DICE_PRIVATE_SPACE + - be32_to_cpu(pointers[0]) * 4 + GLOBAL_VERSION, - &version, sizeof(version), 0); - if (err < 0) - goto end; + if (be32_to_cpu(pointers[1]) > 0x18) { + /* + * Check that the implemented DICE driver specification major + * version number matches. + */ + err = snd_fw_transaction(dice->unit, TCODE_READ_QUADLET_REQUEST, + DICE_PRIVATE_SPACE + + be32_to_cpu(pointers[0]) * 4 + GLOBAL_VERSION, + &version, sizeof(version), 0); + if (err < 0) + goto end; - if ((version & cpu_to_be32(0xff000000)) != cpu_to_be32(0x01000000)) { - dev_err(&dice->unit->device, - "unknown DICE version: 0x%08x\n", be32_to_cpu(version)); - err = -ENODEV; - goto end; + if ((version & cpu_to_be32(0xff000000)) != + cpu_to_be32(0x01000000)) { + dev_err(&dice->unit->device, + "unknown DICE version: 0x%08x\n", + be32_to_cpu(version)); + err = -ENODEV; + goto end; + } + + /* Set up later. */ + dice->clock_caps = 1; } dice->global_offset = be32_to_cpu(pointers[0]) * 4; dice->tx_offset = be32_to_cpu(pointers[2]) * 4; dice->rx_offset = be32_to_cpu(pointers[4]) * 4; - dice->sync_offset = be32_to_cpu(pointers[6]) * 4; - dice->rsrv_offset = be32_to_cpu(pointers[8]) * 4; - /* Set up later. */ - if (be32_to_cpu(pointers[1]) * 4 >= GLOBAL_CLOCK_CAPABILITIES + 4) - dice->clock_caps = 1; + /* Old firmware doesn't support these fields. */ + if (pointers[7] > 0) + dice->sync_offset = be32_to_cpu(pointers[6]) * 4; + if (pointers[9] > 0) + dice->rsrv_offset = be32_to_cpu(pointers[8]) * 4; end: kfree(pointers); return err; -- 2.14.1

3 4

[alsa-devel] ALC892 - Line-out not working properly
by Xristos 24 Apr '18

24 Apr '18

Hi everyone, I'm having a problem with my line-out port not being working properly. Specifically it can't play audio when the 'alsamixer' auto-mute mode is enabled, so sound is coming only from the headphones.However, when auto-mute is disabled the sound is coming from both ports simultaneously. I'm using the Asus Z170-A mobo with ALC892 codec, OS: Arch I ran alsa-info.sh on both cases and i saw that the Pin-Complex 0x14 had 'Pin-ctls: OUT' when auto-mute was disabled whereas when auto-mute was enabled it was 'Pin-ctls: 0x00' (as it currenly is). So i ran HDA-analyzer tool and changed that widget setting to 'OUT' and obviously i produced the same behaviour like when i disabled 'auto-mute' (again sound coming from both ports) and also in all cases PulseAudio lists the 'Line-out' port as 'unplugged'. Also 2 more things: 1st: In BIOS when i changed the driver for the Front panel from 'HD-Audio' to 'AC97' and the ports worked correctly, however no 'auto-mute' mode was available so i had to mute the ports manually when i plugged/unplugged the headphones for e.x I tried uninstalling PulseAudio to see if it was the case but the problem persisted so i think it really is some codec problem since i also have a similar problem in Windows OS. 2nd: When i unplugged the Front-panel connection from the mobo the port played correctly. So is it maybe a mixer problem or something related? Finally i tried every 'model' parameter from HD-Audio-Models.txt for ALC892 but got nothing. In any case i'm attaching the current alsa-info.sh. If you would like i could also send alsa-info.sh for all those cases above. I don't think i'm alone in this issue but i would greatly appreciate the help! Thanks in advance

1 0

[alsa-devel] [PATCH] ALSA: dice: fix OUI for TC group
by Takashi Sakamoto 24 Apr '18

24 Apr '18

OUI for TC Electronic is 0x000166, for TC GROUP A/S. 0x001486 is for Echo Digital Audio Corporation. Fixes: 7cafc65b3aa1 ('ALSA: dice: force to add two pcm devices for listed models') Cc: <stable(a)vger.kernel.org> # v4.6+ Reference: http://standards-oui.ieee.org/oui/oui.txt Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/dice/dice.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index 4ddb4cdd054b..96bb01b6b751 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -14,7 +14,7 @@ MODULE_LICENSE("GPL v2"); #define OUI_WEISS 0x001c6a #define OUI_LOUD 0x000ff2 #define OUI_FOCUSRITE 0x00130e -#define OUI_TCELECTRONIC 0x001486 +#define OUI_TCELECTRONIC 0x000166 #define DICE_CATEGORY_ID 0x04 #define WEISS_CATEGORY_ID 0x00 -- 2.14.1

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: Skip broken EU on Dell dock USB-audio
by Takashi Iwai 24 Apr '18

24 Apr '18

The Dell Dock USB-audio device with 0bda:4014 is behaving notoriously bad, and we have already applied some workaround to avoid the firmware hiccup. Yet we still need to skip one thing, the Extension Unit at ID 4, which doesn't react correctly to the mixer ctl access. Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=1090658 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer_maps.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 9038b2e7df73..eaa03acd4686 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -353,8 +353,11 @@ static struct usbmix_name_map bose_companion5_map[] = { /* * Dell usb dock with ALC4020 codec had a firmware problem where it got * screwed up when zero volume is passed; just skip it as a workaround + * + * Also the extension unit gives an access error, so skip it as well. */ static const struct usbmix_name_map dell_alc4020_map[] = { + { 4, NULL }, /* extension unit */ { 16, NULL }, { 19, NULL }, { 0 } -- 2.16.3

1 0

[alsa-devel] [PATCH] ALSA: usb-audio: Fix missing endian conversion
by Takashi Iwai 24 Apr '18

24 Apr '18

The UAC2 jack detection support introduced the bmControls checks in a couple of places, but they forgot the endian conversion; the bmControls of UAC2 terminal descriptor is __le16, not a byte like in UAC1. Fixes: 5a222e849452 ("ALSA: usb-audio: UAC2 jack detection") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 301ad61ed426..f78b500b3c0e 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1859,7 +1859,7 @@ static int parse_audio_input_terminal(struct mixer_build *state, int unitid, check_input_term(state, d->bTerminalID, &iterm); if (state->mixer->protocol == UAC_VERSION_2) { /* Check for jack detection. */ - if (uac_v2v3_control_is_readable(d->bmControls, + if (uac_v2v3_control_is_readable(le16_to_cpu(d->bmControls), UAC2_TE_CONNECTOR)) { build_connector_control(state, &iterm, true); } @@ -2561,7 +2561,7 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) if (err < 0 && err != -EINVAL) return err; - if (uac_v2v3_control_is_readable(desc->bmControls, + if (uac_v2v3_control_is_readable(le16_to_cpu(desc->bmControls), UAC2_TE_CONNECTOR)) { build_connector_control(&state, &state.oterm, false); -- 2.16.3

2 1

[alsa-devel] [PATCH] ASoC: Intel: board: add BE DAI link for WoV
by mac.chiang＠intel.com 24 Apr '18

24 Apr '18

From: Mac Chiang <mac.chiang(a)intel.com> create dai link in kbl_da7219_max98357a driver for wake on voice functionality. changes picked from broonie's tree commit 0c7941a63a0f ("ASoC: Intel: Skylake: Use refcap device for mono recording") commit 2154be362c90 ("ASoc: Intel: boards: Add WOV as sink for nau88l25_ssm4567 machine") Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> --- sound/soc/intel/boards/kbl_da7219_max98357a.c | 35 +++++++++++++++++++++++++++ 1 file changed, 35 insertions(+) diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c index e84baaf..c332a32 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98357a.c +++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c @@ -45,6 +45,7 @@ struct kbl_codec_private { enum { KBL_DPCM_AUDIO_PB = 0, KBL_DPCM_AUDIO_CP, + KBL_DPCM_AUDIO_REF_CP, KBL_DPCM_AUDIO_DMIC_CP, KBL_DPCM_AUDIO_HDMI1_PB, KBL_DPCM_AUDIO_HDMI2_PB, @@ -349,6 +350,27 @@ static const unsigned int ch_mono[] = { 1, }; +static const struct snd_pcm_hw_constraint_list constraints_refcap = { + .count = ARRAY_SIZE(ch_mono), + .list = ch_mono, +}; + +static int kabylake_refcap_startup(struct snd_pcm_substream *substream) +{ + substream->runtime->hw.channels_max = 1; + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_refcap); + + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_16000); +} + +static struct snd_soc_ops skylaye_refcap_ops = { + .startup = kabylake_refcap_startup, +}; + /* kabylake digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link kabylake_dais[] = { /* Front End DAI links */ @@ -380,6 +402,19 @@ static struct snd_soc_dai_link kabylake_dais[] = { SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, .dpcm_capture = 1, }, + [KBL_DPCM_AUDIO_REF_CP] = { + .name = "Kbl Audio Reference cap", + .stream_name = "Wake on Voice", + .cpu_dai_name = "Reference Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .init = NULL, + .dpcm_capture = 1, + .nonatomic = 1, + .dynamic = 1, + .ops = &skylaye_refcap_ops, + }, [KBL_DPCM_AUDIO_DMIC_CP] = { .name = "Kbl Audio DMIC cap", .stream_name = "dmiccap", -- 2.7.4

1 0

[alsa-devel] [PATCH 0/3] ASoc: TAS6424: Add support for mute, standby, and faster power on
by Jean-Jacques Hiblot 24 Apr '18

24 Apr '18

mute and standby pins are available on the codec. If they are connected, they should be managed by the driver, instead of relying on gpio hogs or on the initial state of the GPIOs. This series also includes a patch to improve the start-up time of the channels by disabling built-in DC diagnostics. Those diagnosdtics basically serve to detect : - wires shorted together - wire shorted to ground or vbat - wire disconnected This is not useful for all platforms and the addition to the startup time is quite noticeable (230ms). Jean-Jacques Hiblot (3): ASoC: tas6424: Allow disabling auto diagnostics for faster power-on ASoC: tas6424: Add support for the standby pin ASoC: tas6424: Add support for the mute pin .../devicetree/bindings/sound/ti,tas6424.txt | 4 ++ sound/soc/codecs/tas6424.c | 74 +++++++++++++++++++++- sound/soc/codecs/tas6424.h | 3 + 3 files changed, 79 insertions(+), 2 deletions(-) -- 2.7.4

2 8

[alsa-devel] [PATCH] sound, isapnp: allow building more drivers with COMPILE_TEST
by Mauro Carvalho Chehab 24 Apr '18

24 Apr '18

Drivers that depend on ISAPNP currently can't be built with COMPILE_TEST. However, looking at isapnp.h, there are already stubs there to allow drivers to include it even when isa PNP is not supported. So, remove such dependencies when COMPILE_TEST. Signed-off-by: Mauro Carvalho Chehab <mchehab(a)s-opensource.com> --- drivers/pnp/isapnp/Kconfig | 2 +- sound/isa/Kconfig | 6 +++--- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/drivers/pnp/isapnp/Kconfig b/drivers/pnp/isapnp/Kconfig index f1ef36673ad4..a1af146d2d90 100644 --- a/drivers/pnp/isapnp/Kconfig +++ b/drivers/pnp/isapnp/Kconfig @@ -3,7 +3,7 @@ # config ISAPNP bool "ISA Plug and Play support" - depends on ISA + depends on ISA || COMPILE_TEST help Say Y here if you would like support for ISA Plug and Play devices. Some information is in <file:Documentation/isapnp.txt>. diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index d2a6cdd0395c..43b35a873d78 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -39,7 +39,7 @@ config SND_ADLIB config SND_AD1816A tristate "Analog Devices SoundPort AD1816A" - depends on PNP && ISA + depends on PNP select ISAPNP select SND_OPL3_LIB select SND_MPU401_UART @@ -67,7 +67,7 @@ config SND_AD1848 config SND_ALS100 tristate "Diamond Tech. DT-019x and Avance Logic ALSxxx" - depends on PNP && ISA + depends on PNP select ISAPNP select SND_OPL3_LIB select SND_MPU401_UART @@ -108,7 +108,7 @@ config SND_AZT2316 config SND_AZT2320 tristate "Aztech Systems AZT2320" - depends on PNP && ISA + depends on PNP select ISAPNP select SND_OPL3_LIB select SND_MPU401_UART -- 2.14.3

3 2

[alsa-devel] [PATCH 0/8] ALSA: Hardening for potential Spectre v1
by Takashi Iwai 24 Apr '18

24 Apr '18

Hi, this is a patchset to cover the potential issues wrt Spectre v1 that were spotted out recently by Smatch: https://marc.info/?l=linux-kernel&m=152411496503418&w=2 Although all reports for sound/* seem like no serious issues, it's better to cover potential usages than a too late sorry. The first patch is basically irrelevant for Spectre but a preliminary fix for the following fix. Most of the rest patches are straightforward, but the second one, the fix for OSS emulation, became a fair amount of code refactoring in the end. An open question is whether we should put Cc-to-stable to these fixes. Maybe yes, although we've never proven that the problem would actually exist, the issue is still "hot", so people may want to see them more-or-less covered... thanks, Takashi === Takashi Iwai (8): ALSA: seq: oss: Fix unbalanced use lock for synth MIDI device ALSA: seq: oss: Hardening for potential Spectre v1 ALSA: control: Hardening for potential Spectre v1 ALSA: hda: Hardening for potential Spectre v1 ALSA: opl3: Hardening for potential Spectre v1 ALSA: asihpi: Hardening for potential Spectre v1 ALSA: hdspm: Hardening for potential Spectre v1 ALSA: rme9652: Hardening for potential Spectre v1 include/sound/control.h | 7 +++- sound/core/seq/oss/seq_oss_event.c | 15 ++++--- sound/core/seq/oss/seq_oss_midi.c | 2 + sound/core/seq/oss/seq_oss_synth.c | 85 ++++++++++++++++++++++---------------- sound/core/seq/oss/seq_oss_synth.h | 3 +- sound/drivers/opl3/opl3_synth.c | 7 +++- sound/pci/asihpi/hpimsginit.c | 13 ++++-- sound/pci/asihpi/hpioctl.c | 4 +- sound/pci/hda/hda_hwdep.c | 12 +++++- sound/pci/rme9652/hdspm.c | 24 ++++++----- sound/pci/rme9652/rme9652.c | 6 ++- 11 files changed, 113 insertions(+), 65 deletions(-) -- 2.16.3

1 8

[alsa-devel] [PATCH 0/4] USB Audio Device Class 3.0 BADD profiles support
by Ruslan Bilovol 24 Apr '18

24 Apr '18

This patchset adds BADD profiles support from the USB Audio Device Class 3.0 spec [1]. BADD profile support is defined as mandatory feature of UAC3-compliant device, it should be implemented as a separate USB configuration. Notable issue with BADD configuration is that it misses class-specific descriptors (and it's mandatory as per spec), so host should guess them from BADD profile number and parameters of endpoints (type, number of endpoints and max packet size) This patchset adds support of all known/existing BADD profiles from the UAC3 specification. It's an alternative implementation comparing to [2], and doesn't build usb descriptors on the host but instead initializes alsa-usb structures with known parameters, so we don't need to keep whole class-specific descriptors in the driver since we anyway need to have BADD-specific logic. I've picked one Jorge's UAC1 patch-improvement and updated it to v4.17 wich contais recently introduced header's sanity checks. Remaining part is to add interrupt endpoint support so we will be able to detect jack insertion in the Headset Adapter profile. This has been tested on ARM and x86-64 machines with custom UAC3 gadget which I'll post later to linux-usb Comments and testing are welcome. [1] http://www.usb.org/developers/docs/devclass_docs/USB_Audio_v3.0.zip [2] https://www.spinics.net/lists/alsa-devel/msg71614.html Jorge Sanjuan (1): ALSA: usb: Only get AudioControl header for UAC1 class. Ruslan Bilovol (3): ALSA: usb: stream: refactor audio interface parsing include: usb: audio-v3: add BADD-specific values ALSA: usb: add UAC3 BADD profiles support include/linux/usb/audio-v3.h | 26 ++ sound/usb/card.c | 53 ++-- sound/usb/clock.c | 9 +- sound/usb/mixer.c | 313 +++++++++++++++++++- sound/usb/mixer_maps.c | 65 +++++ sound/usb/stream.c | 678 +++++++++++++++++++++++++------------------ sound/usb/usbaudio.h | 2 + 7 files changed, 836 insertions(+), 310 deletions(-) -- 1.9.1

5 13

[alsa-devel] [PATCH v4 0/3] ASoC: Add multiple CPU DAI support
by Vinod Koul 24 Apr '18

24 Apr '18

As discussed in [1], DAI link supports multiple codec DAIs however it does not do so for CPU DAIs. So, add support for multiple CPU DAIs on a DAI Link by adding multiple CPU DAI in Card instantiation, suspend and resume functions, PCM ops, stream handling functions and DAPM. [1]: https://www.spinics.net/lists/alsa-devel/msg71369.html changes in v4: - rebase on asoc-next - update changelogs to add more description of changes done Shreyas NC (3): ASoC: Add initial support for multiple CPU DAIs ASoC: Add multiple CPU DAI support for PCM ops ASoC: Add multiple CPU DAI support in DAPM include/sound/soc.h | 6 + sound/soc/soc-core.c | 283 ++++++++++++++++++++---------- sound/soc/soc-dapm.c | 71 +++++--- sound/soc/soc-pcm.c | 483 +++++++++++++++++++++++++++++++++------------------ 4 files changed, 565 insertions(+), 278 deletions(-) -- 2.7.4

1 3

[alsa-devel] Linux-Kernel: System freezes after hibernate or suspend on a Mac mini 6, 2
by Johannes Geiss 23 Apr '18

23 Apr '18

Hello ALSA-Development, I bisected a problem in the Linux kernel 4.14. (This may also concern newer kernel versions.) Using a Mac mini 6,2 computer the kernel freezes after a resume from hibernation or suspend. The bisection resulted, that the patch 1f7f51a63114bab3a05920f4b1343154e95e2cb6 may be the culprit. Please have a look at the description and comments of the bug https://bugzilla.kernel.org/show_bug.cgi?id=198233 and tell me if our assumption is correct or not. Thank you very much for your cooperation. Johannes Geiss -- ------//-------------------------------------------------------------- // Johannes R. Geiss Mac mini server and \\ // OpenPandora user --\X/-----------------------------------------------------------------

3 3

[alsa-devel] [PATCH] ALSA: hda - Use a macro for snd_array iteration loops
by Takashi Iwai 23 Apr '18

23 Apr '18

Introduce a new helper macro, snd_array_for_each(), to iterate for each snd_array element. It slightly improves the readability than lengthy open codes at each place. Along with it, add const prefix to some obvious places. There should be no functional changes by this. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/hdaudio.h | 5 +++++ sound/hda/hdac_regmap.c | 4 ++-- sound/pci/hda/hda_auto_parser.c | 10 +++++----- sound/pci/hda/hda_codec.c | 36 ++++++++++++++++++------------------ sound/pci/hda/hda_generic.c | 27 +++++++++++++-------------- sound/pci/hda/hda_sysfs.c | 20 ++++++++++---------- sound/pci/hda/patch_conexant.c | 5 ++--- sound/pci/hda/patch_realtek.c | 4 ++-- 8 files changed, 57 insertions(+), 54 deletions(-) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index 06536e01ed94..c052afc27547 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -571,4 +571,9 @@ static inline unsigned int snd_array_index(struct snd_array *array, void *ptr) return (unsigned long)(ptr - array->list) / array->elem_size; } +/* a helper macro to iterate for each snd_array element */ +#define snd_array_for_each(array, idx, ptr) \ + for ((idx) = 0, (ptr) = (array)->list; (idx) < (array)->used; \ + (ptr) = snd_array_elem(array, ++(idx))) + #endif /* __SOUND_HDAUDIO_H */ diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index 47a358fab132..419e285e0226 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -65,10 +65,10 @@ static bool hda_writeable_reg(struct device *dev, unsigned int reg) { struct hdac_device *codec = dev_to_hdac_dev(dev); unsigned int verb = get_verb(reg); + const unsigned int *v; int i; - for (i = 0; i < codec->vendor_verbs.used; i++) { - unsigned int *v = snd_array_elem(&codec->vendor_verbs, i); + snd_array_for_each(&codec->vendor_verbs, i, v) { if (verb == *v) return true; } diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index d3ea73171a3d..b9a6b66aeb0e 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -793,11 +793,11 @@ EXPORT_SYMBOL_GPL(snd_hda_add_verbs); */ void snd_hda_apply_verbs(struct hda_codec *codec) { + const struct hda_verb **v; int i; - for (i = 0; i < codec->verbs.used; i++) { - struct hda_verb **v = snd_array_elem(&codec->verbs, i); + + snd_array_for_each(&codec->verbs, i, v) snd_hda_sequence_write(codec, *v); - } } EXPORT_SYMBOL_GPL(snd_hda_apply_verbs); @@ -890,10 +890,10 @@ EXPORT_SYMBOL_GPL(snd_hda_apply_fixup); static bool pin_config_match(struct hda_codec *codec, const struct hda_pintbl *pins) { + const struct hda_pincfg *pin; int i; - for (i = 0; i < codec->init_pins.used; i++) { - struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i); + snd_array_for_each(&codec->init_pins, i, pin) { hda_nid_t nid = pin->nid; u32 cfg = pin->cfg; const struct hda_pintbl *t_pins; diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 5bc3a7468e17..0aa923d129f5 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -481,9 +481,10 @@ static struct hda_pincfg *look_up_pincfg(struct hda_codec *codec, struct snd_array *array, hda_nid_t nid) { + struct hda_pincfg *pin; int i; - for (i = 0; i < array->used; i++) { - struct hda_pincfg *pin = snd_array_elem(array, i); + + snd_array_for_each(array, i, pin) { if (pin->nid == nid) return pin; } @@ -618,14 +619,15 @@ EXPORT_SYMBOL_GPL(snd_hda_codec_get_pin_target); */ void snd_hda_shutup_pins(struct hda_codec *codec) { + const struct hda_pincfg *pin; int i; + /* don't shut up pins when unloading the driver; otherwise it breaks * the default pin setup at the next load of the driver */ if (codec->bus->shutdown) return; - for (i = 0; i < codec->init_pins.used; i++) { - struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i); + snd_array_for_each(&codec->init_pins, i, pin) { /* use read here for syncing after issuing each verb */ snd_hda_codec_read(codec, pin->nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0); @@ -638,13 +640,14 @@ EXPORT_SYMBOL_GPL(snd_hda_shutup_pins); /* Restore the pin controls cleared previously via snd_hda_shutup_pins() */ static void restore_shutup_pins(struct hda_codec *codec) { + const struct hda_pincfg *pin; int i; + if (!codec->pins_shutup) return; if (codec->bus->shutdown) return; - for (i = 0; i < codec->init_pins.used; i++) { - struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i); + snd_array_for_each(&codec->init_pins, i, pin) { snd_hda_codec_write(codec, pin->nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pin->ctrl); @@ -697,8 +700,7 @@ get_hda_cvt_setup(struct hda_codec *codec, hda_nid_t nid) struct hda_cvt_setup *p; int i; - for (i = 0; i < codec->cvt_setups.used; i++) { - p = snd_array_elem(&codec->cvt_setups, i); + snd_array_for_each(&codec->cvt_setups, i, p) { if (p->nid == nid) return p; } @@ -1076,8 +1078,7 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, /* make other inactive cvts with the same stream-tag dirty */ type = get_wcaps_type(get_wcaps(codec, nid)); list_for_each_codec(c, codec->bus) { - for (i = 0; i < c->cvt_setups.used; i++) { - p = snd_array_elem(&c->cvt_setups, i); + snd_array_for_each(&c->cvt_setups, i, p) { if (!p->active && p->stream_tag == stream_tag && get_wcaps_type(get_wcaps(c, p->nid)) == type) p->dirty = 1; @@ -1140,12 +1141,11 @@ static void really_cleanup_stream(struct hda_codec *codec, static void purify_inactive_streams(struct hda_codec *codec) { struct hda_codec *c; + struct hda_cvt_setup *p; int i; list_for_each_codec(c, codec->bus) { - for (i = 0; i < c->cvt_setups.used; i++) { - struct hda_cvt_setup *p; - p = snd_array_elem(&c->cvt_setups, i); + snd_array_for_each(&c->cvt_setups, i, p) { if (p->dirty) really_cleanup_stream(c, p); } @@ -1156,10 +1156,10 @@ static void purify_inactive_streams(struct hda_codec *codec) /* clean up all streams; called from suspend */ static void hda_cleanup_all_streams(struct hda_codec *codec) { + struct hda_cvt_setup *p; int i; - for (i = 0; i < codec->cvt_setups.used; i++) { - struct hda_cvt_setup *p = snd_array_elem(&codec->cvt_setups, i); + snd_array_for_each(&codec->cvt_setups, i, p) { if (p->stream_tag) really_cleanup_stream(codec, p); } @@ -2461,10 +2461,10 @@ EXPORT_SYMBOL_GPL(snd_hda_create_dig_out_ctls); struct hda_spdif_out *snd_hda_spdif_out_of_nid(struct hda_codec *codec, hda_nid_t nid) { + struct hda_spdif_out *spdif; int i; - for (i = 0; i < codec->spdif_out.used; i++) { - struct hda_spdif_out *spdif = - snd_array_elem(&codec->spdif_out, i); + + snd_array_for_each(&codec->spdif_out, i, spdif) { if (spdif->nid == nid) return spdif; } diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 5cc65093d941..51030f040745 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -264,10 +264,10 @@ static struct nid_path *get_nid_path(struct hda_codec *codec, int anchor_nid) { struct hda_gen_spec *spec = codec->spec; + struct nid_path *path; int i; - for (i = 0; i < spec->paths.used; i++) { - struct nid_path *path = snd_array_elem(&spec->paths, i); + snd_array_for_each(&spec->paths, i, path) { if (path->depth <= 0) continue; if ((!from_nid || path->path[0] == from_nid) && @@ -325,10 +325,10 @@ EXPORT_SYMBOL_GPL(snd_hda_get_path_from_idx); static bool is_dac_already_used(struct hda_codec *codec, hda_nid_t nid) { struct hda_gen_spec *spec = codec->spec; + const struct nid_path *path; int i; - for (i = 0; i < spec->paths.used; i++) { - struct nid_path *path = snd_array_elem(&spec->paths, i); + snd_array_for_each(&spec->paths, i, path) { if (path->path[0] == nid) return true; } @@ -351,11 +351,11 @@ static bool is_reachable_path(struct hda_codec *codec, static bool is_ctl_used(struct hda_codec *codec, unsigned int val, int type) { struct hda_gen_spec *spec = codec->spec; + const struct nid_path *path; int i; val &= AMP_VAL_COMPARE_MASK; - for (i = 0; i < spec->paths.used; i++) { - struct nid_path *path = snd_array_elem(&spec->paths, i); + snd_array_for_each(&spec->paths, i, path) { if ((path->ctls[type] & AMP_VAL_COMPARE_MASK) == val) return true; } @@ -638,13 +638,13 @@ static bool is_active_nid(struct hda_codec *codec, hda_nid_t nid, { struct hda_gen_spec *spec = codec->spec; int type = get_wcaps_type(get_wcaps(codec, nid)); + const struct nid_path *path; int i, n; if (nid == codec->core.afg) return true; - for (n = 0; n < spec->paths.used; n++) { - struct nid_path *path = snd_array_elem(&spec->paths, n); + snd_array_for_each(&spec->paths, n, path) { if (!path->active) continue; if (codec->power_save_node) { @@ -2696,10 +2696,10 @@ static const struct snd_kcontrol_new out_jack_mode_enum = { static bool find_kctl_name(struct hda_codec *codec, const char *name, int idx) { struct hda_gen_spec *spec = codec->spec; + const struct snd_kcontrol_new *kctl; int i; - for (i = 0; i < spec->kctls.used; i++) { - struct snd_kcontrol_new *kctl = snd_array_elem(&spec->kctls, i); + snd_array_for_each(&spec->kctls, i, kctl) { if (!strcmp(kctl->name, name) && kctl->index == idx) return true; } @@ -4021,8 +4021,7 @@ static hda_nid_t set_path_power(struct hda_codec *codec, hda_nid_t nid, struct nid_path *path; int n; - for (n = 0; n < spec->paths.used; n++) { - path = snd_array_elem(&spec->paths, n); + snd_array_for_each(&spec->paths, n, path) { if (!path->depth) continue; if (path->path[0] == nid || @@ -5831,10 +5830,10 @@ static void init_digital(struct hda_codec *codec) */ static void clear_unsol_on_unused_pins(struct hda_codec *codec) { + const struct hda_pincfg *pin; int i; - for (i = 0; i < codec->init_pins.used; i++) { - struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i); + snd_array_for_each(&codec->init_pins, i, pin) { hda_nid_t nid = pin->nid; if (is_jack_detectable(codec, nid) && !snd_hda_jack_tbl_get(codec, nid)) diff --git a/sound/pci/hda/hda_sysfs.c b/sound/pci/hda/hda_sysfs.c index 9b7efece4484..6ec79c58d48d 100644 --- a/sound/pci/hda/hda_sysfs.c +++ b/sound/pci/hda/hda_sysfs.c @@ -80,10 +80,10 @@ static ssize_t pin_configs_show(struct hda_codec *codec, struct snd_array *list, char *buf) { + const struct hda_pincfg *pin; int i, len = 0; mutex_lock(&codec->user_mutex); - for (i = 0; i < list->used; i++) { - struct hda_pincfg *pin = snd_array_elem(list, i); + snd_array_for_each(list, i, pin) { len += sprintf(buf + len, "0x%02x 0x%08x\n", pin->nid, pin->cfg); } @@ -217,10 +217,10 @@ static ssize_t init_verbs_show(struct device *dev, char *buf) { struct hda_codec *codec = dev_get_drvdata(dev); + const struct hda_verb *v; int i, len = 0; mutex_lock(&codec->user_mutex); - for (i = 0; i < codec->init_verbs.used; i++) { - struct hda_verb *v = snd_array_elem(&codec->init_verbs, i); + snd_array_for_each(&codec->init_verbs, i, v) { len += snprintf(buf + len, PAGE_SIZE - len, "0x%02x 0x%03x 0x%04x\n", v->nid, v->verb, v->param); @@ -267,10 +267,10 @@ static ssize_t hints_show(struct device *dev, char *buf) { struct hda_codec *codec = dev_get_drvdata(dev); + const struct hda_hint *hint; int i, len = 0; mutex_lock(&codec->user_mutex); - for (i = 0; i < codec->hints.used; i++) { - struct hda_hint *hint = snd_array_elem(&codec->hints, i); + snd_array_for_each(&codec->hints, i, hint) { len += snprintf(buf + len, PAGE_SIZE - len, "%s = %s\n", hint->key, hint->val); } @@ -280,10 +280,10 @@ static ssize_t hints_show(struct device *dev, static struct hda_hint *get_hint(struct hda_codec *codec, const char *key) { + struct hda_hint *hint; int i; - for (i = 0; i < codec->hints.used; i++) { - struct hda_hint *hint = snd_array_elem(&codec->hints, i); + snd_array_for_each(&codec->hints, i, hint) { if (!strcmp(hint->key, key)) return hint; } @@ -783,13 +783,13 @@ void snd_hda_sysfs_init(struct hda_codec *codec) void snd_hda_sysfs_clear(struct hda_codec *codec) { #ifdef CONFIG_SND_HDA_RECONFIG + struct hda_hint *hint; int i; /* clear init verbs */ snd_array_free(&codec->init_verbs); /* clear hints */ - for (i = 0; i < codec->hints.used; i++) { - struct hda_hint *hint = snd_array_elem(&codec->hints, i); + snd_array_for_each(&codec->hints, i, hint) { kfree(hint->key); /* we don't need to free hint->val */ } snd_array_free(&codec->hints); diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 5b4dbcec6de8..093d2a9ece85 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -588,6 +588,7 @@ static void cxt_fixup_olpc_xo(struct hda_codec *codec, const struct hda_fixup *fix, int action) { struct conexant_spec *spec = codec->spec; + struct snd_kcontrol_new *kctl; int i; if (action != HDA_FIXUP_ACT_PROBE) @@ -606,9 +607,7 @@ static void cxt_fixup_olpc_xo(struct hda_codec *codec, snd_hda_codec_set_pin_target(codec, 0x1a, PIN_VREF50); /* override mic boost control */ - for (i = 0; i < spec->gen.kctls.used; i++) { - struct snd_kcontrol_new *kctl = - snd_array_elem(&spec->gen.kctls, i); + snd_array_for_each(&spec->gen.kctls, i, kctl) { if (!strcmp(kctl->name, "Mic Boost Volume")) { kctl->put = olpc_xo_mic_boost_put; break; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fc77bf7a1544..c0c40430b663 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2828,6 +2828,7 @@ static int find_ext_mic_pin(struct hda_codec *codec); static void alc286_shutup(struct hda_codec *codec) { + const struct hda_pincfg *pin; int i; int mic_pin = find_ext_mic_pin(codec); /* don't shut up pins when unloading the driver; otherwise it breaks @@ -2835,8 +2836,7 @@ static void alc286_shutup(struct hda_codec *codec) */ if (codec->bus->shutdown) return; - for (i = 0; i < codec->init_pins.used; i++) { - struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i); + snd_array_for_each(&codec->init_pins, i, pin) { /* use read here for syncing after issuing each verb */ if (pin->nid != mic_pin) snd_hda_codec_read(codec, pin->nid, 0, -- 2.16.3

1 0

[alsa-devel] [PATCH 1/2] ASoC: sgtl5000: Fix the spelling of 'exceed'
by Fabio Estevam 23 Apr '18

23 Apr '18

From: Fabio Estevam <fabio.estevam(a)nxp.com> Fix the spelling of 'exceed' in two comments. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/codecs/sgtl5000.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 7c1d658..ce0d0d7 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -457,7 +457,7 @@ static int dac_put_volsw(struct snd_kcontrol *kcontrol, * avc_put_threshold function: register_value = 10^(dB/20) * 0.636 * 2^15 ==> * dB = ( fls(register_value) - 14.347 ) * 6.02 * - * As this calculation is expensive and the threshold dB values may not exeed + * As this calculation is expensive and the threshold dB values may not exceed * 0 to 96 we use pre-calculated values. */ static int avc_get_threshold(struct snd_kcontrol *kcontrol, @@ -490,7 +490,7 @@ static int avc_get_threshold(struct snd_kcontrol *kcontrol, * * The register value is calculated by following formula: * register_value = 10^(dB/20) * 0.636 * 2^15 - * As this calculation is expensive and the threshold dB values may not exeed + * As this calculation is expensive and the threshold dB values may not exceed * 0 to 96 we use pre-calculated values. */ static int avc_put_threshold(struct snd_kcontrol *kcontrol, -- 2.7.4

2 2

[alsa-devel] [PATCH v2 1/2] ASoC: sgtl5000: Fix the spelling of 'exceed'
by Fabio Estevam 23 Apr '18

23 Apr '18

From: Fabio Estevam <fabio.estevam(a)nxp.com> Fix the spelling of 'exceed' in two comments. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Changes since v1: - None sound/soc/codecs/sgtl5000.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 7c1d658..ce0d0d7 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -457,7 +457,7 @@ static int dac_put_volsw(struct snd_kcontrol *kcontrol, * avc_put_threshold function: register_value = 10^(dB/20) * 0.636 * 2^15 ==> * dB = ( fls(register_value) - 14.347 ) * 6.02 * - * As this calculation is expensive and the threshold dB values may not exeed + * As this calculation is expensive and the threshold dB values may not exceed * 0 to 96 we use pre-calculated values. */ static int avc_get_threshold(struct snd_kcontrol *kcontrol, @@ -490,7 +490,7 @@ static int avc_get_threshold(struct snd_kcontrol *kcontrol, * * The register value is calculated by following formula: * register_value = 10^(dB/20) * 0.636 * 2^15 - * As this calculation is expensive and the threshold dB values may not exeed + * As this calculation is expensive and the threshold dB values may not exceed * 0 to 96 we use pre-calculated values. */ static int avc_put_threshold(struct snd_kcontrol *kcontrol, -- 2.7.4

2 2

[alsa-devel] [PATCH] ASoC: codecs: msm8916-wcd: use threaded IRQ handlers
by Daniel Mack 23 Apr '18

23 Apr '18

The handler for the mbhc switch irq calls into snd_soc_jack_report() which in turn triggers a notifier chain that is blocking, which means it allows its callbacks to sleep. This leads to a "scheduling while atomic" Ooops when the jack notifier callback uses snd_soc_dapm_sync(), for instance. Fix this by switching from atomic to threaded IRQ handlers. Fixes: de66b3455023e ("ASoC: codecs: msm8916-wcd-analog: add MBHC support") Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- sound/soc/codecs/msm8916-wcd-analog.c | 29 +++++++++++++++++------------ 1 file changed, 17 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index 12ee83d52405..52b4b6d836a4 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -1187,10 +1187,12 @@ static int pm8916_wcd_analog_spmi_probe(struct platform_device *pdev) return irq; } - ret = devm_request_irq(dev, irq, pm8916_mbhc_switch_irq_handler, - IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | - IRQF_ONESHOT, - "mbhc switch irq", priv); + ret = devm_request_threaded_irq(dev, irq, NULL, + pm8916_mbhc_switch_irq_handler, + IRQF_TRIGGER_RISING | + IRQF_TRIGGER_FALLING | + IRQF_ONESHOT, + "mbhc switch irq", priv); if (ret) dev_err(dev, "cannot request mbhc switch irq\n"); @@ -1201,10 +1203,12 @@ static int pm8916_wcd_analog_spmi_probe(struct platform_device *pdev) return irq; } - ret = devm_request_irq(dev, irq, mbhc_btn_press_irq_handler, - IRQF_TRIGGER_RISING | - IRQF_TRIGGER_FALLING | IRQF_ONESHOT, - "mbhc btn press irq", priv); + ret = devm_request_threaded_irq(dev, irq, NULL, + mbhc_btn_press_irq_handler, + IRQF_TRIGGER_RISING | + IRQF_TRIGGER_FALLING | + IRQF_ONESHOT, + "mbhc btn press irq", priv); if (ret) dev_err(dev, "cannot request mbhc button press irq\n"); @@ -1214,10 +1218,11 @@ static int pm8916_wcd_analog_spmi_probe(struct platform_device *pdev) return irq; } - ret = devm_request_irq(dev, irq, mbhc_btn_release_irq_handler, - IRQF_TRIGGER_RISING | - IRQF_TRIGGER_FALLING | IRQF_ONESHOT, - "mbhc btn release irq", priv); + ret = devm_request_threaded_irq(dev, irq, NULL, + mbhc_btn_release_irq_handler, + IRQF_TRIGGER_RISING | + IRQF_TRIGGER_FALLING | IRQF_ONESHOT, + "mbhc btn release irq", priv); if (ret) dev_err(dev, "cannot request mbhc button release irq\n"); -- 2.14.3

2 3

[alsa-devel] [PATCH] ALSA: control: Fix missing __user annotation
by Takashi Iwai 23 Apr '18

23 Apr '18

There is one place missing __user annotation to the pointer used by the recent code refactoring. Reported by sparse. Fixes: 450296f305f1 ("ALSA: control: code refactoring TLV ioctl handler") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/control.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/control.c b/sound/core/control.c index 69734b0eafd0..9aa15bfc7936 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1492,7 +1492,7 @@ static int snd_ctl_tlv_ioctl(struct snd_ctl_file *file, int op_flag) { struct snd_ctl_tlv header; - unsigned int *container; + unsigned int __user *container; unsigned int container_size; struct snd_kcontrol *kctl; struct snd_ctl_elem_id id; -- 2.16.3

2 1

[alsa-devel] [PATCH v2 0/5] ASoC: mediatek: add support for mt6797 SoC
by KaiChieh Chuang 23 Apr '18

23 Apr '18

This patch adds basic support for Mediatek AFE for MT6797 SoC. Change since the previous patch set: * fix build warning in sound/soc/codecs/mt6351.c KaiChieh Chuang (5): ASoC: add mt6351 codec driver ASoC: mt6797: add structure define and clock control function for 6797 ASoC: mt6797: add mt6797 platform driver ASoC: add mt6797-mt6351 driver and config option ASoC: mediatek: add documents for mt6797 Documentation/devicetree/bindings/sound/mt6351.txt | 16 + .../devicetree/bindings/sound/mt6797-afe-pcm.txt | 42 + .../devicetree/bindings/sound/mt6797-mt6351.txt | 14 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/mt6351.c | 1639 ++++++++++++++++++++ sound/soc/codecs/mt6351.h | 108 ++ sound/soc/mediatek/Kconfig | 20 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/mt6797/Makefile | 19 + sound/soc/mediatek/mt6797/mt6797-afe-clk.c | 132 ++ sound/soc/mediatek/mt6797/mt6797-afe-clk.h | 25 + sound/soc/mediatek/mt6797/mt6797-afe-common.h | 57 + sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 1234 +++++++++++++++ sound/soc/mediatek/mt6797/mt6797-interconnection.h | 41 + sound/soc/mediatek/mt6797/mt6797-mt6351.c | 186 +++ sound/soc/mediatek/mt6797/mt6797-reg.h | 846 ++++++++++ 17 files changed, 4386 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mt6351.txt create mode 100644 Documentation/devicetree/bindings/sound/mt6797-afe-pcm.txt create mode 100644 Documentation/devicetree/bindings/sound/mt6797-mt6351.txt create mode 100644 sound/soc/codecs/mt6351.c create mode 100644 sound/soc/codecs/mt6351.h create mode 100644 sound/soc/mediatek/mt6797/Makefile create mode 100644 sound/soc/mediatek/mt6797/mt6797-afe-clk.c create mode 100644 sound/soc/mediatek/mt6797/mt6797-afe-clk.h create mode 100644 sound/soc/mediatek/mt6797/mt6797-afe-common.h create mode 100644 sound/soc/mediatek/mt6797/mt6797-afe-pcm.c create mode 100644 sound/soc/mediatek/mt6797/mt6797-interconnection.h create mode 100644 sound/soc/mediatek/mt6797/mt6797-mt6351.c create mode 100644 sound/soc/mediatek/mt6797/mt6797-reg.h -- 2.12.5

2 15

[alsa-devel] [PATCH] ALSA: usb-audio: Fix forgotten conversion of control query functions
by Takashi Iwai 23 Apr '18

23 Apr '18

The recent code refactoring made the argument for some helper functions to be the explicit UAC_CS_* and UAC2_CS_* value instead of 0-based offset. However, there was one place left forgotten, and it caused a regression on some devices appearing as the inconsistent mixer setup. This patch corrects the forgotten conversion. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=199449 Fixes: 21e9b3e931f7 ("ALSA: usb-audio: fix uac control query argument") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 301ad61ed426..3387483310b1 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1776,7 +1776,8 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, build_feature_ctl(state, _ftr, ch_bits, control, &iterm, unitid, ch_read_only); if (uac_v2v3_control_is_readable(master_bits, control)) - build_feature_ctl(state, _ftr, 0, i, &iterm, unitid, + build_feature_ctl(state, _ftr, 0, control, + &iterm, unitid, !uac_v2v3_control_is_writeable(master_bits, control)); } -- 2.16.3

3 3

[alsa-devel] [PATCH] [alsa-lib] pcm: softvol: Allow up to 90 dB of gain
by Ricard Wanderlof 23 Apr '18

23 Apr '18

The gain algorithm used in softvol can handle gain factors of up to 32767 which is slightly more than 90 dB, so allow a max_dB of 90 dB. This doesn't affect existing asound.conf files, but does allow a max_dB of up to 90 dB when needed. Tested using Audacity that there is no undue distorsion or other artefacts when 90 dB of gain is applied to a suitable signal (i.e. a signal quiet enough not be clipped whan applying 90 dB of gain). Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- I discussed this with Takashi and Jaroslav back in January, but the discussion petered out and didn't reach a conclusion on which maximum gain would be acceptable, so I thought I'd send an actual patch and we can take it from here. We have a couple of usecases where we are using digital microphones, which have 24 bit resolution, and with an acoustic overload point of about 120 dB SPL, the output level can be quite low when the sound source is a fair distance form the microphone. The current softvol limit of 50 dB is therefore a bit restrictive. When studying the code, it becomes apparent that the limit of the actual gain function is a gain factor of 32767, which corresponds to 90.3 dB. So it makes sense to set the MAX_DB_UPPER_LIMIT (whose only function is to cause _snd_pcm_softvol_open() to bail out and emit an error message if the gain is set higher than this limit) to 90 dB as it corresponds to the technical limitations of the gain algorithm. Granted, 90 dB is probably more than most people need, but there's no real harm in allowing a large gain; the user gain limit in a given application is still set by the max_dB parameter in asound.conf . The worst that can happen is that the signal becomes clipped as would happen with any signal if the applied gain drives it over 0 dB. If in fact a signal is close to 0 dB to start with, then the resulting clipping when increased by 90 dB is not much worse than when increased by 50 dB. src/pcm/pcm_softvol.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/src/pcm/pcm_softvol.c b/src/pcm/pcm_softvol.c index 8bb4a39..0eaeace 100644 --- a/src/pcm/pcm_softvol.c +++ b/src/pcm/pcm_softvol.c @@ -59,7 +59,11 @@ typedef struct { #define PRESET_RESOLUTION 256 #define PRESET_MIN_DB -51.0 #define ZERO_DB 0.0 -#define MAX_DB_UPPER_LIMIT 50 +/* + * The gain algorithm as it stands supports gain factors up to 32767, which + * is a fraction more than 90 dB, so set 90 dB as the maximum possible gain. + */ +#define MAX_DB_UPPER_LIMIT 90 static const unsigned int preset_dB_value[PRESET_RESOLUTION] = { 0x00b8, 0x00bd, 0x00c1, 0x00c5, 0x00ca, 0x00cf, 0x00d4, 0x00d9, -- 2.1.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

2 2

[alsa-devel] [PATCH] patch_ca0132.c: Handle SBZ and R3Di sound cards
by Connor McAdams 23 Apr '18

23 Apr '18

This patch adds support for the Sound Blaster Z and the Recon3Di. With the current version, both cards only had one output supported, and none of the inputs were working. This patch adds support for all outputs and inputs, digital and analog. It also adds support for the onboard DSP effects for both input and output. This patch changes the control names for both cards to accommodate surround sound controls. The main effect switches, "PlayEnhancement" for output, and "CrystalVoice" for input, have been changed to "Enable OutFX" for output, and "Enable InFX" for input to make them more clear. Also, all DSP related effects have the prefix "FX:" attached to them, to make it more clear that "Enable *dir*FX" enables or disables them. Controls have also been added to change the effect levels of the individual effects, through a volume slider. This patch also adds support for alternative firmware for both cards, under the names "ctefx-sbz.bin" for the Sound Blaster Z, and "ctefx-r3di.bin" for the Recon3Di. These firmwares are not necessary however, and there is a switch to use the original ctefx.bin file if neither alternative firmwares are detected. This patch also adds the ability to set the DSP volume, which is set with a decibel value sent in floating point. The only issue I'm aware of at this point is an "Unexpected TLV callback for slave Front Playback Volume", which relates to the vmaster control. For some reason, the TLV data doesn't match the rest of the controls, even though they seem to share the same tlv function. Signed-off-by: Connor McAdams <conmanx360(a)gmail.com> --- sound/pci/hda/patch_ca0132.c | 3313 ++++++++++++++++++++++++++++++++++++++---- 1 file changed, 3064 insertions(+), 249 deletions(-) mode change 100644 => 100755 sound/pci/hda/patch_ca0132.c diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c old mode 100644 new mode 100755 index 768ea86..ddbb658 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -22,13 +22,14 @@ */ #include <linux/init.h> +#include <linux/types.h> #include <linux/delay.h> #include <linux/slab.h> #include <linux/mutex.h> #include <linux/module.h> #include <linux/firmware.h> -#include <linux/kernel.h> -#include <sound/core.h> +#include <linux/io.h> +#include <linux/pci.h> #include "hda_codec.h" #include "hda_local.h" #include "hda_auto_parser.h" @@ -39,10 +40,11 @@ /* Enable this to see controls for tuning purpose. */ /*#define ENABLE_TUNING_CONTROLS*/ -#define FLOAT_ZERO 0x00000000 -#define FLOAT_ONE 0x3f800000 -#define FLOAT_TWO 0x40000000 -#define FLOAT_MINUS_5 0xc0a00000 +#define FLOAT_0 0x00000000 +#define FLOAT_1 0x3F800000 +#define FLOAT_2 0x40000000 +#define FLOAT_3 0x40400000 +#define FLOAT_8 0x41000000 #define UNSOL_TAG_DSP 0x16 @@ -64,29 +66,44 @@ #define MEM_CONNID_MICIN2 5 #define MEM_CONNID_MICOUT1 12 #define MEM_CONNID_MICOUT2 14 -#define MEM_CONNID_WUH 10 -#define MEM_CONNID_DSP 16 +#define MEM_CONNID_WUH 10 +#define MEM_CONNID_DSP 16 #define MEM_CONNID_DMIC 100 #define SCP_SET 0 #define SCP_GET 1 -#define EFX_FILE "ctefx.bin" +#define EFX_FILE "ctefx.bin" +#define SBZ_EFX_FILE "ctefx-sbz.bin" +#define R3DI_EFX_FILE "ctefx-r3di.bin" #ifdef CONFIG_SND_HDA_CODEC_CA0132_DSP MODULE_FIRMWARE(EFX_FILE); +MODULE_FIRMWARE(SBZ_EFX_FILE); +MODULE_FIRMWARE(R3DI_EFX_FILE); #endif static char *dirstr[2] = { "Playback", "Capture" }; +#define SBZ_NUM_OF_OUTPUTS 3 enum { SPEAKER_OUT, - HEADPHONE_OUT + HEADPHONE_OUT, + SURROUND_OUT }; enum { DIGITAL_MIC, - LINE_MIC_IN + LINE_MIC_IN, +}; + +/* Strings for Input Source Enum Control */ +static char *in_src_str[3] = {"Rear Mic", "Line", "Front Mic" }; +#define IN_SRC_NUM_OF_INPUTS 3 +enum { + REAR_MIC, + REAR_LINE_IN, + FRONT_MIC, }; enum { @@ -101,7 +118,7 @@ enum { #define VNODES_COUNT (VNODE_END_NID - VNODE_START_NID) #define EFFECT_START_NID 0x90 -#define OUT_EFFECT_START_NID EFFECT_START_NID +#define OUT_EFFECT_START_NID EFFECT_START_NID SURROUND = OUT_EFFECT_START_NID, CRYSTALIZER, DIALOG_PLUS, @@ -122,19 +139,45 @@ enum { VOICEFX = IN_EFFECT_END_NID, PLAY_ENHANCEMENT, CRYSTAL_VOICE, - EFFECT_END_NID + EFFECT_END_NID, + OUTPUT_SOURCE_ENUM, + INPUT_SOURCE_ENUM, + XBASS_XOVER, + EQ_PRESET_ENUM, + SMART_VOLUME_ENUM, + MIC_BOOST_ENUM #define EFFECTS_COUNT (EFFECT_END_NID - EFFECT_START_NID) }; +/* Strings for Smart Volume Setting enum */ +#define NUM_OF_SVM_SETTINGS 3 +static char *out_svm_set_enum_str[3] = {"Normal", "Loud", "Night" }; + /* Effects values size*/ #define EFFECT_VALS_MAX_COUNT 12 +/* + * Maximum amount of commands used for the ca0132_alt_out_select + * SCP command struct. + */ +#define ALT_OUT_SET_MAX_COMMANDS 9 + +/* Amount of effect level sliders */ +#define EFFECT_LEVEL_SLIDERS 5 + +/* + * Default values for the effect slider controls, they are in order of their + * effect NID's. Surround, Crystalizer, Dialog Plus, Smart Volume, and then + * X-bass. + */ +static unsigned int effect_slider_defaults[] = {67, 65, 50, 74, 50}; + /* Latency introduced by DSP blocks in milliseconds. */ -#define DSP_CAPTURE_INIT_LATENCY 0 -#define DSP_CRYSTAL_VOICE_LATENCY 124 -#define DSP_PLAYBACK_INIT_LATENCY 13 -#define DSP_PLAY_ENHANCEMENT_LATENCY 30 -#define DSP_SPEAKER_OUT_LATENCY 7 +#define DSP_CAPTURE_INIT_LATENCY 0 +#define DSP_CRYSTAL_VOICE_LATENCY 124 +#define DSP_PLAYBACK_INIT_LATENCY 13 +#define DSP_PLAY_ENHANCEMENT_LATENCY 30 +#define DSP_SPEAKER_OUT_LATENCY 7 struct ct_effect { char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; @@ -472,63 +515,213 @@ static struct ct_voicefx_preset ca0132_voicefx_presets[] = { } }; +/* ca0132 EQ presets, taken from Windows Sound Blaster Z Driver */ + +#define EQ_PRESET_MAX_PARAM_COUNT 11 + +struct ct_eq { + char *name; + hda_nid_t nid; + int mid; + int reqs[EQ_PRESET_MAX_PARAM_COUNT]; /*effect module request*/ +}; + +struct ct_eq_preset { + char *name; /*preset name*/ + unsigned int vals[EQ_PRESET_MAX_PARAM_COUNT]; +}; + +static struct ct_eq ca0132_alt_eq_enum = { + .name = "FX: Equalizer Preset Switch", + .nid = EQ_PRESET_ENUM, + .mid = 0x96, + .reqs = {10, 11, 12, 13, 14, 15, 16, 17, 18, 19, 20} +}; + + +static struct ct_eq_preset ca0132_alt_eq_presets[] = { + { .name = "Flat", + .vals = { 0x00000000, 0x00000000, 0x00000000, + 0x00000000, 0x00000000, 0x00000000, + 0x00000000, 0x00000000, 0x00000000, + 0x00000000, 0x00000000 } + }, + { .name = "Acoustic", + .vals = { 0x00000000, 0x00000000, 0x3F8CCCCD, + 0x40000000, 0x00000000, 0x00000000, + 0x00000000, 0x00000000, 0x40000000, + 0x40000000, 0x40000000 } + }, + { .name = "Classical", + .vals = { 0x00000000, 0x00000000, 0x40C00000, + 0x40C00000, 0x40466666, 0x00000000, + 0x00000000, 0x00000000, 0x00000000, + 0x40466666, 0x40466666 } + }, + { .name = "Country", + .vals = { 0x00000000, 0xBF99999A, 0x00000000, + 0x3FA66666, 0x3FA66666, 0x3F8CCCCD, + 0x00000000, 0x00000000, 0x40000000, + 0x40466666, 0x40800000 } + }, + { .name = "Dance", + .vals = { 0x00000000, 0xBF99999A, 0x40000000, + 0x40466666, 0x40866666, 0xBF99999A, + 0xBF99999A, 0x00000000, 0x00000000, + 0x40800000, 0x40800000 } + }, + { .name = "Jazz", + .vals = { 0x00000000, 0x00000000, 0x00000000, + 0x3F8CCCCD, 0x40800000, 0x40800000, + 0x40800000, 0x00000000, 0x3F8CCCCD, + 0x40466666, 0x40466666 } + }, + { .name = "New Age", + .vals = { 0x00000000, 0x00000000, 0x40000000, + 0x40000000, 0x00000000, 0x00000000, + 0x00000000, 0x3F8CCCCD, 0x40000000, + 0x40000000, 0x40000000 } + }, + { .name = "Pop", + .vals = { 0x00000000, 0xBFCCCCCD, 0x00000000, + 0x40000000, 0x40000000, 0x00000000, + 0xBF99999A, 0xBF99999A, 0x00000000, + 0x40466666, 0x40C00000 } + }, + { .name = "Rock", + .vals = { 0x00000000, 0xBF99999A, 0xBF99999A, + 0x3F8CCCCD, 0x40000000, 0xBF99999A, + 0xBF99999A, 0x00000000, 0x00000000, + 0x40800000, 0x40800000 } + }, + { .name = "Vocal", + .vals = { 0x00000000, 0xC0000000, 0xBF99999A, + 0xBF99999A, 0x00000000, 0x40466666, + 0x40800000, 0x40466666, 0x00000000, + 0x00000000, 0x3F8CCCCD } + } +}; + +/* DSP commands for the alternative ca0132 codec's output modes */ + +struct ca0132_alt_out_set { + char *name; /*preset name*/ + unsigned char commands; + unsigned int mids[ALT_OUT_SET_MAX_COMMANDS]; + unsigned int reqs[ALT_OUT_SET_MAX_COMMANDS]; + unsigned int vals[ALT_OUT_SET_MAX_COMMANDS]; +}; + +static struct ca0132_alt_out_set alt_out_presets[] = { + { .name = "Line Out", + .commands = 7, + .mids = { 0x96, 0x96, 0x96, 0x8F, + 0x96, 0x96, 0x96 }, + .reqs = { 0x19, 0x17, 0x18, 0x01, + 0x1F, 0x15, 0x3A }, + .vals = { 0x3F000000, 0x42A00000, 0x00000000, + 0x00000000, 0x00000000, 0x00000000, + 0x00000000 } + }, + { .name = "Headphone", + .commands = 7, + .mids = { 0x96, 0x96, 0x96, 0x8F, + 0x96, 0x96, 0x96 }, + .reqs = { 0x19, 0x17, 0x18, 0x01, + 0x1F, 0x15, 0x3A }, + .vals = { 0x3F000000, 0x42A00000, 0x00000000, + 0x00000000, 0x00000000, 0x00000000, + 0x00000000 } + }, + { .name = "Surround", + .commands = 8, + .mids = { 0x96, 0x8F, 0x96, 0x96, + 0x96, 0x96, 0x96, 0x96 }, + .reqs = { 0x18, 0x01, 0x1F, 0x15, + 0x3A, 0x1A, 0x1B, 0x1C }, + .vals = { 0x00000000, 0x00000000, 0x00000000, + 0x00000000, 0x00000000, 0x00000000, + 0x00000000, 0x00000000 } + } +}; + +#define DSP_VOL_OUT 0 +#define DSP_VOL_IN 1 + +struct ct_dsp_volume_ctl { + hda_nid_t vnid; + int mid; /*effect module ID*/ + unsigned int reqs[3]; /*effect module request*/ +}; + +static struct ct_dsp_volume_ctl ca0132_alt_vol_ctls[] = { + { .vnid = VNID_SPK, + .mid = 0x32, + .reqs = {3, 4, 2} + }, + { .vnid = VNID_MIC, + .mid = 0x37, + .reqs = {1, 2, 3} + } +}; + enum hda_cmd_vendor_io { /* for DspIO node */ VENDOR_DSPIO_SCP_WRITE_DATA_LOW = 0x000, VENDOR_DSPIO_SCP_WRITE_DATA_HIGH = 0x100, - VENDOR_DSPIO_STATUS = 0xF01, + VENDOR_DSPIO_STATUS = 0xF01, VENDOR_DSPIO_SCP_POST_READ_DATA = 0x702, - VENDOR_DSPIO_SCP_READ_DATA = 0xF02, - VENDOR_DSPIO_DSP_INIT = 0x703, + VENDOR_DSPIO_SCP_READ_DATA = 0xF02, + VENDOR_DSPIO_DSP_INIT = 0x703, VENDOR_DSPIO_SCP_POST_COUNT_QUERY = 0x704, - VENDOR_DSPIO_SCP_READ_COUNT = 0xF04, + VENDOR_DSPIO_SCP_READ_COUNT = 0xF04, /* for ChipIO node */ - VENDOR_CHIPIO_ADDRESS_LOW = 0x000, - VENDOR_CHIPIO_ADDRESS_HIGH = 0x100, - VENDOR_CHIPIO_STREAM_FORMAT = 0x200, - VENDOR_CHIPIO_DATA_LOW = 0x300, - VENDOR_CHIPIO_DATA_HIGH = 0x400, + VENDOR_CHIPIO_ADDRESS_LOW = 0x000, + VENDOR_CHIPIO_ADDRESS_HIGH = 0x100, + VENDOR_CHIPIO_STREAM_FORMAT = 0x200, + VENDOR_CHIPIO_DATA_LOW = 0x300, + VENDOR_CHIPIO_DATA_HIGH = 0x400, - VENDOR_CHIPIO_GET_PARAMETER = 0xF00, - VENDOR_CHIPIO_STATUS = 0xF01, - VENDOR_CHIPIO_HIC_POST_READ = 0x702, - VENDOR_CHIPIO_HIC_READ_DATA = 0xF03, + VENDOR_CHIPIO_GET_PARAMETER = 0xF00, + VENDOR_CHIPIO_STATUS = 0xF01, + VENDOR_CHIPIO_HIC_POST_READ = 0x702, + VENDOR_CHIPIO_HIC_READ_DATA = 0xF03, - VENDOR_CHIPIO_8051_DATA_WRITE = 0x707, - VENDOR_CHIPIO_8051_DATA_READ = 0xF07, + VENDOR_CHIPIO_8051_DATA_WRITE = 0x707, + VENDOR_CHIPIO_8051_DATA_READ = 0xF07, VENDOR_CHIPIO_CT_EXTENSIONS_ENABLE = 0x70A, VENDOR_CHIPIO_CT_EXTENSIONS_GET = 0xF0A, - VENDOR_CHIPIO_PLL_PMU_WRITE = 0x70C, - VENDOR_CHIPIO_PLL_PMU_READ = 0xF0C, - VENDOR_CHIPIO_8051_ADDRESS_LOW = 0x70D, + VENDOR_CHIPIO_PLL_PMU_WRITE = 0x70C, + VENDOR_CHIPIO_PLL_PMU_READ = 0xF0C, + VENDOR_CHIPIO_8051_ADDRESS_LOW = 0x70D, VENDOR_CHIPIO_8051_ADDRESS_HIGH = 0x70E, - VENDOR_CHIPIO_FLAG_SET = 0x70F, - VENDOR_CHIPIO_FLAGS_GET = 0xF0F, - VENDOR_CHIPIO_PARAM_SET = 0x710, - VENDOR_CHIPIO_PARAM_GET = 0xF10, + VENDOR_CHIPIO_FLAG_SET = 0x70F, + VENDOR_CHIPIO_FLAGS_GET = 0xF0F, + VENDOR_CHIPIO_PARAM_SET = 0x710, + VENDOR_CHIPIO_PARAM_GET = 0xF10, VENDOR_CHIPIO_PORT_ALLOC_CONFIG_SET = 0x711, - VENDOR_CHIPIO_PORT_ALLOC_SET = 0x712, - VENDOR_CHIPIO_PORT_ALLOC_GET = 0xF12, - VENDOR_CHIPIO_PORT_FREE_SET = 0x713, + VENDOR_CHIPIO_PORT_ALLOC_SET = 0x712, + VENDOR_CHIPIO_PORT_ALLOC_GET = 0xF12, + VENDOR_CHIPIO_PORT_FREE_SET = 0x713, - VENDOR_CHIPIO_PARAM_EX_ID_GET = 0xF17, - VENDOR_CHIPIO_PARAM_EX_ID_SET = 0x717, + VENDOR_CHIPIO_PARAM_EX_ID_GET = 0xF17, + VENDOR_CHIPIO_PARAM_EX_ID_SET = 0x717, VENDOR_CHIPIO_PARAM_EX_VALUE_GET = 0xF18, VENDOR_CHIPIO_PARAM_EX_VALUE_SET = 0x718, - VENDOR_CHIPIO_DMIC_CTL_SET = 0x788, - VENDOR_CHIPIO_DMIC_CTL_GET = 0xF88, - VENDOR_CHIPIO_DMIC_PIN_SET = 0x789, - VENDOR_CHIPIO_DMIC_PIN_GET = 0xF89, - VENDOR_CHIPIO_DMIC_MCLK_SET = 0x78A, - VENDOR_CHIPIO_DMIC_MCLK_GET = 0xF8A, + VENDOR_CHIPIO_DMIC_CTL_SET = 0x788, + VENDOR_CHIPIO_DMIC_CTL_GET = 0xF88, + VENDOR_CHIPIO_DMIC_PIN_SET = 0x789, + VENDOR_CHIPIO_DMIC_PIN_GET = 0xF89, + VENDOR_CHIPIO_DMIC_MCLK_SET = 0x78A, + VENDOR_CHIPIO_DMIC_MCLK_GET = 0xF8A, - VENDOR_CHIPIO_EAPD_SEL_SET = 0x78D + VENDOR_CHIPIO_EAPD_SEL_SET = 0x78D }; /* @@ -536,41 +729,41 @@ enum hda_cmd_vendor_io { */ enum control_flag_id { /* Connection manager stream setup is bypassed/enabled */ - CONTROL_FLAG_C_MGR = 0, + CONTROL_FLAG_C_MGR = 0, /* DSP DMA is bypassed/enabled */ - CONTROL_FLAG_DMA = 1, + CONTROL_FLAG_DMA = 1, /* 8051 'idle' mode is disabled/enabled */ - CONTROL_FLAG_IDLE_ENABLE = 2, + CONTROL_FLAG_IDLE_ENABLE = 2, /* Tracker for the SPDIF-in path is bypassed/enabled */ - CONTROL_FLAG_TRACKER = 3, + CONTROL_FLAG_TRACKER = 3, /* DigitalOut to Spdif2Out connection is disabled/enabled */ - CONTROL_FLAG_SPDIF2OUT = 4, + CONTROL_FLAG_SPDIF2OUT = 4, /* Digital Microphone is disabled/enabled */ - CONTROL_FLAG_DMIC = 5, + CONTROL_FLAG_DMIC = 5, /* ADC_B rate is 48 kHz/96 kHz */ - CONTROL_FLAG_ADC_B_96KHZ = 6, + CONTROL_FLAG_ADC_B_96KHZ = 6, /* ADC_C rate is 48 kHz/96 kHz */ - CONTROL_FLAG_ADC_C_96KHZ = 7, + CONTROL_FLAG_ADC_C_96KHZ = 7, /* DAC rate is 48 kHz/96 kHz (affects all DACs) */ - CONTROL_FLAG_DAC_96KHZ = 8, + CONTROL_FLAG_DAC_96KHZ = 8, /* DSP rate is 48 kHz/96 kHz */ - CONTROL_FLAG_DSP_96KHZ = 9, + CONTROL_FLAG_DSP_96KHZ = 9, /* SRC clock is 98 MHz/196 MHz (196 MHz forces rate to 96 KHz) */ - CONTROL_FLAG_SRC_CLOCK_196MHZ = 10, + CONTROL_FLAG_SRC_CLOCK_196MHZ = 10, /* SRC rate is 48 kHz/96 kHz (48 kHz disabled when clock is 196 MHz) */ - CONTROL_FLAG_SRC_RATE_96KHZ = 11, + CONTROL_FLAG_SRC_RATE_96KHZ = 11, /* Decode Loop (DSP->SRC->DSP) is disabled/enabled */ - CONTROL_FLAG_DECODE_LOOP = 12, + CONTROL_FLAG_DECODE_LOOP = 12, /* De-emphasis filter on DAC-1 disabled/enabled */ - CONTROL_FLAG_DAC1_DEEMPHASIS = 13, + CONTROL_FLAG_DAC1_DEEMPHASIS = 13, /* De-emphasis filter on DAC-2 disabled/enabled */ - CONTROL_FLAG_DAC2_DEEMPHASIS = 14, + CONTROL_FLAG_DAC2_DEEMPHASIS = 14, /* De-emphasis filter on DAC-3 disabled/enabled */ - CONTROL_FLAG_DAC3_DEEMPHASIS = 15, + CONTROL_FLAG_DAC3_DEEMPHASIS = 15, /* High-pass filter on ADC_B disabled/enabled */ - CONTROL_FLAG_ADC_B_HIGH_PASS = 16, + CONTROL_FLAG_ADC_B_HIGH_PASS = 16, /* High-pass filter on ADC_C disabled/enabled */ - CONTROL_FLAG_ADC_C_HIGH_PASS = 17, + CONTROL_FLAG_ADC_C_HIGH_PASS = 17, /* Common mode on Port_A disabled/enabled */ CONTROL_FLAG_PORT_A_COMMON_MODE = 18, /* Common mode on Port_D disabled/enabled */ @@ -580,7 +773,7 @@ enum control_flag_id { /* Impedance for ramp generator on Port_D, 16 Ohm/10K Ohm */ CONTROL_FLAG_PORT_D_10KOHM_LOAD = 21, /* ASI rate is 48kHz/96kHz */ - CONTROL_FLAG_ASI_96KHZ = 22, + CONTROL_FLAG_ASI_96KHZ = 22, /* DAC power settings able to control attached ports no/yes */ CONTROL_FLAG_DACS_CONTROL_PORTS = 23, /* Clock Stop OK reporting is disabled/enabled */ @@ -594,9 +787,9 @@ enum control_flag_id { */ enum control_param_id { /* 0: None, 1: Mic1In*/ - CONTROL_PARAM_VIP_SOURCE = 1, + CONTROL_PARAM_VIP_SOURCE = 1, /* 0: force HDA, 1: allow DSP if HDA Spdif1Out stream is idle */ - CONTROL_PARAM_SPDIF1_SOURCE = 2, + CONTROL_PARAM_SPDIF1_SOURCE = 2, /* Port A output stage gain setting to use when 16 Ohm output * impedance is selected*/ CONTROL_PARAM_PORTA_160OHM_GAIN = 8, @@ -607,27 +800,27 @@ enum control_param_id { /* Stream Control */ /* Select stream with the given ID */ - CONTROL_PARAM_STREAM_ID = 24, + CONTROL_PARAM_STREAM_ID = 24, /* Source connection point for the selected stream */ CONTROL_PARAM_STREAM_SOURCE_CONN_POINT = 25, /* Destination connection point for the selected stream */ CONTROL_PARAM_STREAM_DEST_CONN_POINT = 26, /* Number of audio channels in the selected stream */ - CONTROL_PARAM_STREAMS_CHANNELS = 27, + CONTROL_PARAM_STREAMS_CHANNELS = 27, /*Enable control for the selected stream */ - CONTROL_PARAM_STREAM_CONTROL = 28, + CONTROL_PARAM_STREAM_CONTROL = 28, /* Connection Point Control */ /* Select connection point with the given ID */ - CONTROL_PARAM_CONN_POINT_ID = 29, + CONTROL_PARAM_CONN_POINT_ID = 29, /* Connection point sample rate */ CONTROL_PARAM_CONN_POINT_SAMPLE_RATE = 30, /* Node Control */ /* Select HDA node with the given ID */ - CONTROL_PARAM_NODE_ID = 31 + CONTROL_PARAM_NODE_ID = 31 }; /* @@ -635,9 +828,9 @@ enum control_param_id { */ enum hda_vendor_status_dspio { /* Success */ - VENDOR_STATUS_DSPIO_OK = 0x00, + VENDOR_STATUS_DSPIO_OK = 0x00, /* Busy, unable to accept new command, the host must retry */ - VENDOR_STATUS_DSPIO_BUSY = 0x01, + VENDOR_STATUS_DSPIO_BUSY = 0x01, /* SCP command queue is full */ VENDOR_STATUS_DSPIO_SCP_COMMAND_QUEUE_FULL = 0x02, /* SCP response queue is empty */ @@ -658,24 +851,24 @@ enum hda_vendor_status_chipio { * CA0132 sample rate */ enum ca0132_sample_rate { - SR_6_000 = 0x00, - SR_8_000 = 0x01, - SR_9_600 = 0x02, - SR_11_025 = 0x03, - SR_16_000 = 0x04, - SR_22_050 = 0x05, - SR_24_000 = 0x06, - SR_32_000 = 0x07, - SR_44_100 = 0x08, - SR_48_000 = 0x09, - SR_88_200 = 0x0A, - SR_96_000 = 0x0B, - SR_144_000 = 0x0C, - SR_176_400 = 0x0D, - SR_192_000 = 0x0E, - SR_384_000 = 0x0F, - - SR_COUNT = 0x10, + SR_6_000 = 0x00, + SR_8_000 = 0x01, + SR_9_600 = 0x02, + SR_11_025 = 0x03, + SR_16_000 = 0x04, + SR_22_050 = 0x05, + SR_24_000 = 0x06, + SR_32_000 = 0x07, + SR_44_100 = 0x08, + SR_48_000 = 0x09, + SR_88_200 = 0x0A, + SR_96_000 = 0x0B, + SR_144_000 = 0x0C, + SR_176_400 = 0x0D, + SR_192_000 = 0x0E, + SR_384_000 = 0x0F, + + SR_COUNT = 0x10, SR_RATE_UNKNOWN = 0x1F }; @@ -684,7 +877,7 @@ enum dsp_download_state { DSP_DOWNLOAD_FAILED = -1, DSP_DOWNLOAD_INIT = 0, DSP_DOWNLOADING = 1, - DSP_DOWNLOADED = 2 + DSP_DOWNLOADED = 2 }; /* retrieve parameters from hda format */ @@ -703,6 +896,7 @@ struct ca0132_spec { const struct hda_verb *base_init_verbs; const struct hda_verb *base_exit_verbs; const struct hda_verb *chip_init_verbs; + const struct hda_verb *sbz_init_verbs; struct hda_verb *spec_init_verbs; struct auto_pin_cfg autocfg; @@ -719,6 +913,7 @@ struct ca0132_spec { hda_nid_t shared_mic_nid; hda_nid_t shared_out_nid; hda_nid_t unsol_tag_hp; + hda_nid_t unsol_tag_front_hp; /* alt_ca0132 specific */ hda_nid_t unsol_tag_amic1; /* chip access */ @@ -734,6 +929,19 @@ struct ca0132_spec { unsigned int scp_resp_header; unsigned int scp_resp_data[4]; unsigned int scp_resp_count; + bool dsp_download_wakeup; + /* + * true if alt-firmware is detected, if no alt-firmware is detected + * the default ctefx.bin file will be used. + */ + bool alt_firmware_present; + bool sbz_failure_entered; + + /* + * Sound Blaster Z PCI region 2 iomem, used for input and output + * switching, and other stuff I don't fully understand. + */ + void __iomem *mem_base; /* mixer and effects related */ unsigned char dmic_ctl; @@ -746,6 +954,16 @@ struct ca0132_spec { long effects_switch[EFFECTS_COUNT]; long voicefx_val; long cur_mic_boost; + /* Values related to the extra ca0132_alt controls */ + unsigned char in_enum_val; + unsigned char out_enum_val; + unsigned char mic_boost_enum_val; + unsigned char smart_volume_setting; + long fx_ctl_val[EFFECT_LEVEL_SLIDERS]; + long xbass_xover_freq; + long eq_preset_val; + unsigned int tlv[4]; + struct hda_vmaster_mute_hook vmaster_mute; struct hda_codec *codec; struct delayed_work unsol_hp_work; @@ -762,6 +980,8 @@ struct ca0132_spec { enum { QUIRK_NONE, QUIRK_ALIENWARE, + QUIRK_SBZ, + QUIRK_R3DI, }; static const struct hda_pintbl alienware_pincfgs[] = { @@ -778,10 +998,42 @@ static const struct hda_pintbl alienware_pincfgs[] = { {} }; +/* Sound Blaster Z pin configs taken from Windows Driver */ +static const struct hda_pintbl sbz_pincfgs[] = { + { 0x0b, 0x01017010 }, /* Port G -- Lineout FRONT L/R */ + { 0x0c, 0x014510f0 }, /* SPDIF Out 1 */ + { 0x0d, 0x014510f0 }, /* Digital Out */ + { 0x0e, 0x01c510f0 }, /* SPDIF In */ + { 0x0f, 0x0221701f }, /* Port A -- BackPanel HP */ + { 0x10, 0x01017012 }, /* Port D -- Center/LFE or FP Hp */ + { 0x11, 0x01017014 }, /* Port B -- LineMicIn2 / Rear L/R */ + { 0x12, 0x01a170f0 }, /* Port C -- LineIn1 */ + { 0x13, 0x908700f0 }, /* What U Hear In*/ + { 0x18, 0x50d000f0 }, /* N/A */ + {} +}; + +/* Recon3D integrated pin configs taken from Windows Driver */ +static const struct hda_pintbl r3di_pincfgs[] = { + { 0x0b, 0x01014110 }, /* Port G -- Lineout FRONT L/R */ + { 0x0c, 0x014510f0 }, /* SPDIF Out 1 */ + { 0x0d, 0x014510f0 }, /* Digital Out */ + { 0x0e, 0x41c520f0 }, /* SPDIF In */ + { 0x0f, 0x0221401f }, /* Port A -- BackPanel HP */ + { 0x10, 0x01016011 }, /* Port D -- Center/LFE or FP Hp */ + { 0x11, 0x01011014 }, /* Port B -- LineMicIn2 / Rear L/R */ + { 0x12, 0x02a090f0 }, /* Port C -- LineIn1 */ + { 0x13, 0x908700f0 }, /* What U Hear In*/ + { 0x18, 0x500000f0 }, /* N/A */ + {} +}; + static const struct snd_pci_quirk ca0132_quirks[] = { SND_PCI_QUIRK(0x1028, 0x0685, "Alienware 15 2015", QUIRK_ALIENWARE), SND_PCI_QUIRK(0x1028, 0x0688, "Alienware 17 2015", QUIRK_ALIENWARE), SND_PCI_QUIRK(0x1028, 0x0708, "Alienware 15 R2 2016", QUIRK_ALIENWARE), + SND_PCI_QUIRK(0x1102, 0x0010, "Sound Blaster Z", QUIRK_SBZ), + SND_PCI_QUIRK(0x1458, 0xA036, "Recon3Di", QUIRK_R3DI), {} }; @@ -965,6 +1217,29 @@ static int chipio_write(struct hda_codec *codec, } /* + * Write given value to the given address through the chip I/O widget. + * not protected by the Mutex + */ +static int chipio_write_no_mutex(struct hda_codec *codec, + unsigned int chip_addx, const unsigned int data) +{ + int err; + + + /* write the address, and if successful proceed to write data */ + err = chipio_write_address(codec, chip_addx); + if (err < 0) + goto exit; + + err = chipio_write_data(codec, data); + if (err < 0) + goto exit; + +exit: + return err; +} + +/* * Write multiple values to the given address through the chip I/O widget. * protected by the Mutex */ @@ -1039,6 +1314,9 @@ static void chipio_set_control_param(struct hda_codec *codec, struct ca0132_spec *spec = codec->spec; int val; + codec_dbg(codec, "chipio_param ID: %d val: 0x%02x", + param_id, param_val); + if ((param_id < 32) && (param_val < 8)) { val = (param_val << 5) | (param_id); snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, @@ -1058,6 +1336,86 @@ static void chipio_set_control_param(struct hda_codec *codec, } /* + * Set chip parameters through the chip I/O widget. NO MUTEX. + */ +static void chipio_set_control_param_no_mutex(struct hda_codec *codec, + enum control_param_id param_id, int param_val) +{ + int val; + + codec_dbg(codec, "chipio_param ID: %d val: 0x%02x", + param_id, param_val); + + if ((param_id < 32) && (param_val < 8)) { + val = (param_val << 5) | (param_id); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_PARAM_SET, val); + } else { + if (chipio_send(codec, VENDOR_CHIPIO_STATUS, 0) == 0) { + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_PARAM_EX_ID_SET, + param_id); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_PARAM_EX_VALUE_SET, + param_val); + } + } +} +/* + * Connect stream to a source point, and then connect + * that source point to a destination point. + */ +static void chipio_set_stream_source_dest(struct hda_codec *codec, + int streamid, int source_point, int dest_point) +{ + + chipio_set_control_param_no_mutex(codec, + CONTROL_PARAM_STREAM_ID, streamid); + chipio_set_control_param_no_mutex(codec, + CONTROL_PARAM_STREAM_SOURCE_CONN_POINT, source_point); + chipio_set_control_param_no_mutex(codec, + CONTROL_PARAM_STREAM_DEST_CONN_POINT, dest_point); +} + +/* + * Set number of channels in the selected stream. + */ +static void chipio_set_stream_channels(struct hda_codec *codec, + int streamid, unsigned int channels) +{ + chipio_set_control_param_no_mutex(codec, CONTROL_PARAM_STREAM_ID, + streamid); + chipio_set_control_param_no_mutex(codec, CONTROL_PARAM_STREAMS_CHANNELS, + channels); +} + +/* + * Enable/Disable audio stream. + */ +static void chipio_set_stream_control(struct hda_codec *codec, + int streamid, int enable) +{ + codec_dbg(codec, "Stream: 0x%02x Val: %d", streamid, enable); + chipio_set_control_param_no_mutex(codec, + CONTROL_PARAM_STREAM_ID, streamid); + chipio_set_control_param_no_mutex(codec, + CONTROL_PARAM_STREAM_CONTROL, enable); +} + + +/* + * Set sampling rate of the connection point. NO MUTEX. + */ +static void chipio_set_conn_rate_no_mutex(struct hda_codec *codec, + int connid, enum ca0132_sample_rate rate) +{ + chipio_set_control_param_no_mutex(codec, + CONTROL_PARAM_CONN_POINT_ID, connid); + chipio_set_control_param_no_mutex(codec, + CONTROL_PARAM_CONN_POINT_SAMPLE_RATE, rate); +} + +/* * Set sampling rate of the connection point. */ static void chipio_set_conn_rate(struct hda_codec *codec, @@ -1420,8 +1778,8 @@ static int dspio_send_scp_message(struct hda_codec *codec, * Returns zero or a negative error code. */ static int dspio_scp(struct hda_codec *codec, - int mod_id, int req, int dir, void *data, unsigned int len, - void *reply, unsigned int *reply_len) + int mod_id, int src_id, int req, int dir, void *data, + unsigned int len, void *reply, unsigned int *reply_len) { int status = 0; struct scp_msg scp_send, scp_reply; @@ -1445,13 +1803,16 @@ static int dspio_scp(struct hda_codec *codec, return -EINVAL; } - scp_send.hdr = make_scp_header(mod_id, 0x20, (dir == SCP_GET), req, + scp_send.hdr = make_scp_header(mod_id, src_id, (dir == SCP_GET), req, 0, 0, 0, len/sizeof(unsigned int)); if (data != NULL && len > 0) { len = min((unsigned int)(sizeof(scp_send.data)), len); memcpy(scp_send.data, data, len); } + codec_dbg(codec, "scp_cmd mod: %02x req: %02x src: %02x\n", + mod_id, req, src_id); + ret_bytes = 0; send_size = sizeof(unsigned int) + len; status = dspio_send_scp_message(codec, (unsigned char *)&scp_send, @@ -1502,15 +1863,24 @@ static int dspio_scp(struct hda_codec *codec, * Set DSP parameters */ static int dspio_set_param(struct hda_codec *codec, int mod_id, - int req, void *data, unsigned int len) + int src_id, int req, void *data, unsigned int len) { - return dspio_scp(codec, mod_id, req, SCP_SET, data, len, NULL, NULL); + return dspio_scp(codec, mod_id, src_id, req, SCP_SET, data, len, + NULL, NULL); } static int dspio_set_uint_param(struct hda_codec *codec, int mod_id, int req, unsigned int data) { - return dspio_set_param(codec, mod_id, req, &data, sizeof(unsigned int)); + return dspio_set_param(codec, mod_id, 0x20, req, &data, + sizeof(unsigned int)); +} + +static int dspio_set_uint_param_no_source(struct hda_codec *codec, int mod_id, + int req, unsigned int data) +{ + return dspio_set_param(codec, mod_id, 0x00, req, &data, + sizeof(unsigned int)); } /* @@ -1522,8 +1892,9 @@ static int dspio_alloc_dma_chan(struct hda_codec *codec, unsigned int *dma_chan) unsigned int size = sizeof(dma_chan); codec_dbg(codec, " dspio_alloc_dma_chan() -- begin\n"); - status = dspio_scp(codec, MASTERCONTROL, MASTERCONTROL_ALLOC_DMA_CHAN, - SCP_GET, NULL, 0, dma_chan, &size); + status = dspio_scp(codec, MASTERCONTROL, 0x20, + MASTERCONTROL_ALLOC_DMA_CHAN, SCP_GET, NULL, 0, + dma_chan, &size); if (status < 0) { codec_dbg(codec, "dspio_alloc_dma_chan: SCP Failed\n"); @@ -1552,8 +1923,9 @@ static int dspio_free_dma_chan(struct hda_codec *codec, unsigned int dma_chan) codec_dbg(codec, " dspio_free_dma_chan() -- begin\n"); codec_dbg(codec, "dspio_free_dma_chan: chan=%d\n", dma_chan); - status = dspio_scp(codec, MASTERCONTROL, MASTERCONTROL_ALLOC_DMA_CHAN, - SCP_SET, &dma_chan, sizeof(dma_chan), NULL, &dummy); + status = dspio_scp(codec, MASTERCONTROL, 0x20, + MASTERCONTROL_ALLOC_DMA_CHAN, SCP_SET, &dma_chan, + sizeof(dma_chan), NULL, &dummy); if (status < 0) { codec_dbg(codec, "dspio_free_dma_chan: SCP Failed\n"); @@ -1800,7 +2172,7 @@ static int dsp_dma_setup(struct hda_codec *codec, return -ENXIO; } - codec_dbg(codec, " dsp_dma_setup() start reg pgm\n"); + codec_dbg(codec, " dsp_dma_setup() start reg pgm\n"); addr_field = dsp_addx << DSPDMAC_DMACFG_DBADR_LOBIT; incr_field = 0; @@ -1820,7 +2192,7 @@ static int dsp_dma_setup(struct hda_codec *codec, codec_dbg(codec, "write DMACFG Reg fail\n"); return status; } - codec_dbg(codec, " dsp_dma_setup() Write DMACFG\n"); + codec_dbg(codec, " dsp_dma_setup() Write DMACFG\n"); adr_ofs = (count - 1) << (DSPDMAC_DSPADROFS_BOFS_LOBIT + (code ? 0 : 1)); @@ -1831,7 +2203,7 @@ static int dsp_dma_setup(struct hda_codec *codec, codec_dbg(codec, "write DSPADROFS Reg fail\n"); return status; } - codec_dbg(codec, " dsp_dma_setup() Write DSPADROFS\n"); + codec_dbg(codec, " dsp_dma_setup() Write DSPADROFS\n"); base_cnt = (count - 1) << DSPDMAC_XFRCNT_BCNT_LOBIT; @@ -1845,7 +2217,7 @@ static int dsp_dma_setup(struct hda_codec *codec, codec_dbg(codec, "write XFRCNT Reg fail\n"); return status; } - codec_dbg(codec, " dsp_dma_setup() Write XFRCNT\n"); + codec_dbg(codec, " dsp_dma_setup() Write XFRCNT\n"); codec_dbg(codec, "ChipA=0x%x, cnt=0x%x, DMACFG=0x%x, " @@ -1876,7 +2248,7 @@ static int dsp_dma_start(struct hda_codec *codec, codec_dbg(codec, "read CHNLSTART reg fail\n"); return status; } - codec_dbg(codec, "-- dsp_dma_start() Read CHNLSTART\n"); + codec_dbg(codec, "-- dsp_dma_start() Read CHNLSTART\n"); reg &= ~(DSPDMAC_CHNLSTART_EN_MASK | DSPDMAC_CHNLSTART_DIS_MASK); @@ -2075,8 +2447,8 @@ struct dma_engine { enum dma_state { - DMA_STATE_STOP = 0, - DMA_STATE_RUN = 1 + DMA_STATE_STOP = 0, + DMA_STATE_RUN = 1 }; static int dma_convert_to_hda_format(struct hda_codec *codec, @@ -2575,14 +2947,16 @@ static int dspxfr_image(struct hda_codec *codec, */ static void dspload_post_setup(struct hda_codec *codec) { + struct ca0132_spec *spec = codec->spec; codec_dbg(codec, "---- dspload_post_setup ------\n"); + if (spec->quirk == QUIRK_NONE) { + /*set DSP speaker to 2.0 configuration*/ + chipio_write(codec, XRAM_XRAM_INST_OFFSET(0x18), 0x08080080); + chipio_write(codec, XRAM_XRAM_INST_OFFSET(0x19), 0x3f800000); - /*set DSP speaker to 2.0 configuration*/ - chipio_write(codec, XRAM_XRAM_INST_OFFSET(0x18), 0x08080080); - chipio_write(codec, XRAM_XRAM_INST_OFFSET(0x19), 0x3f800000); - - /*update write pointer*/ - chipio_write(codec, XRAM_XRAM_INST_OFFSET(0x29), 0x00000002); + /*update write pointer*/ + chipio_write(codec, XRAM_XRAM_INST_OFFSET(0x29), 0x00000002); + } } /** @@ -2675,6 +3049,7 @@ static bool dspload_is_loaded(struct hda_codec *codec) static bool dspload_wait_loaded(struct hda_codec *codec) { + struct ca0132_spec *spec = codec->spec; unsigned long timeout = jiffies + msecs_to_jiffies(2000); do { @@ -2684,12 +3059,199 @@ static bool dspload_wait_loaded(struct hda_codec *codec) } msleep(20); } while (time_before(jiffies, timeout)); - + if (spec->dsp_download_wakeup == 1) { + codec_dbg(codec, "Waking up from sleep, reload the DSP."); + return false; + } codec_err(codec, "ca0132 failed to download DSP\n"); return false; } /* + * Setup GPIO for the other variants of Core3D. + */ + +/* + * Sets up the GPIO pins so that they are discoverable. If this isn't done, + * the card shows as having no GPIO pins. + */ +static void ca0132_gpio_init(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + + switch (spec->quirk) { + case QUIRK_SBZ: + snd_hda_codec_write(codec, 0x01, 0, 0x793, 0x00); + snd_hda_codec_write(codec, 0x01, 0, 0x794, 0x53); + snd_hda_codec_write(codec, 0x01, 0, 0x790, 0x23); + break; + case QUIRK_R3DI: + snd_hda_codec_write(codec, 0x01, 0, 0x793, 0x00); + snd_hda_codec_write(codec, 0x01, 0, 0x794, 0x5B); + break; + } + +} + +/* Sets the GPIO for audio output. */ +static void ca0132_gpio_setup(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + + switch (spec->quirk) { + case QUIRK_SBZ: + snd_hda_codec_write(codec, 0x01, 0, + AC_VERB_SET_GPIO_DIRECTION, 0x07); + snd_hda_codec_write(codec, 0x01, 0, + AC_VERB_SET_GPIO_MASK, 0x07); + snd_hda_codec_write(codec, 0x01, 0, + AC_VERB_SET_GPIO_DATA, 0x04); + snd_hda_codec_write(codec, 0x01, 0, + AC_VERB_SET_GPIO_DATA, 0x06); + break; + case QUIRK_R3DI: + snd_hda_codec_write(codec, 0x01, 0, + AC_VERB_SET_GPIO_DIRECTION, 0x1E); + snd_hda_codec_write(codec, 0x01, 0, + AC_VERB_SET_GPIO_MASK, 0x1F); + snd_hda_codec_write(codec, 0x01, 0, + AC_VERB_SET_GPIO_DATA, 0x0C); + break; + } +} + +/* On shutdown, sends commands in sets of three */ +static void sbz_gpio_shutdown_commands(struct hda_codec *codec, int dir, + int mask, int data) +{ + if (dir >= 0) + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DIRECTION, + dir); + if (mask >= 0) + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_MASK, + mask); + + if (data >= 0) + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, + data); +} + +/* + * GPIO control functions for the Recon3D integrated. + */ + +enum r3di_gpio_bit { + /* Bit 1 - Switch between front/rear mic. 0 = rear, 1 = front */ + R3DI_MIC_SELECT_BIT = 1, + /* Bit 2 - Switch between headphone/line out. 0 = Headphone, 1 = Line */ + R3DI_OUT_SELECT_BIT = 2, + /* I dunno what this actually does, but it stays on until the dsp + * is downloaded. + */ + R3DI_GPIO_DSP_DOWNLOADING = 3, + /* Same as above, no clue what it does, but it comes on after the dsp + * is downloaded. + */ + R3DI_GPIO_DSP_DOWNLOADED = 4 +}; + +enum r3di_mic_select { + /* Set GPIO bit 1 to 0 for rear mic */ + R3DI_REAR_MIC = 0, + /* Set GPIO bit 1 to 1 for front microphone*/ + R3DI_FRONT_MIC = 1 +}; + +enum r3di_out_select { + /* Set GPIO bit 2 to 0 for headphone */ + R3DI_HEADPHONE_OUT = 0, + /* Set GPIO bit 2 to 1 for speaker */ + R3DI_LINE_OUT = 1 +}; + +enum r3di_dsp_status { + /* Set GPIO bit 3 to 1 until DSP is downloaded */ + R3DI_DSP_DOWNLOADING = 0, + /* Set GPIO bit 4 to 1 once DSP is downloaded */ + R3DI_DSP_DOWNLOADED = 1 +}; + +static void r3di_gpio_mic_set(struct hda_codec *codec, + enum r3di_mic_select cur_mic) +{ + unsigned int cur_gpio; + + /* Get the current GPIO Data setup */ + cur_gpio = snd_hda_codec_read(codec, 0x01, 0, + AC_VERB_GET_GPIO_DATA, 0); + + switch (cur_mic) { + case R3DI_REAR_MIC: + cur_gpio &= ~(1 << R3DI_MIC_SELECT_BIT); + break; + case R3DI_FRONT_MIC: + cur_gpio |= (1 << R3DI_MIC_SELECT_BIT); + break; + } + snd_hda_codec_write(codec, codec->core.afg, 0, + AC_VERB_SET_GPIO_DATA, cur_gpio); + +} + +static void r3di_gpio_out_set(struct hda_codec *codec, + enum r3di_out_select cur_out) +{ + unsigned int cur_gpio; + + /* Get the current GPIO Data setup */ + cur_gpio = snd_hda_codec_read(codec, 0x01, 0, + AC_VERB_GET_GPIO_DATA, 0); + + switch (cur_out) { + case R3DI_HEADPHONE_OUT: + cur_gpio &= ~(1 << R3DI_OUT_SELECT_BIT); + break; + case R3DI_LINE_OUT: + cur_gpio |= (1 << R3DI_OUT_SELECT_BIT); + break; + } + snd_hda_codec_write(codec, codec->core.afg, 0, + AC_VERB_SET_GPIO_DATA, cur_gpio); + +} + +static void r3di_gpio_dsp_status_set(struct hda_codec *codec, + enum r3di_dsp_status dsp_status) +{ + unsigned int cur_gpio; + + /* Get the current GPIO Data setup */ + cur_gpio = snd_hda_codec_read(codec, 0x01, 0, + AC_VERB_GET_GPIO_DATA, 0); + + switch (dsp_status) { + case R3DI_DSP_DOWNLOADING: + cur_gpio |= (1 << R3DI_GPIO_DSP_DOWNLOADING); + snd_hda_codec_write(codec, codec->core.afg, 0, + AC_VERB_SET_GPIO_DATA, cur_gpio); + break; + case R3DI_DSP_DOWNLOADED: + /* Set DOWNLOADING bit to 0. */ + cur_gpio &= ~(1 << R3DI_GPIO_DSP_DOWNLOADING); + + snd_hda_codec_write(codec, codec->core.afg, 0, + AC_VERB_SET_GPIO_DATA, cur_gpio); + + cur_gpio |= (1 << R3DI_GPIO_DSP_DOWNLOADED); + break; + } + + snd_hda_codec_write(codec, codec->core.afg, 0, + AC_VERB_SET_GPIO_DATA, cur_gpio); + +} + +/* * PCM callbacks */ static int ca0132_playback_pcm_prepare(struct hda_pcm_stream *hinfo, @@ -2701,7 +3263,6 @@ static int ca0132_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct ca0132_spec *spec = codec->spec; snd_hda_codec_setup_stream(codec, spec->dacs[0], stream_tag, 0, format); - return 0; } @@ -2713,7 +3274,6 @@ static int ca0132_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, if (spec->dsp_state == DSP_DOWNLOADING) return 0; - /*If Playback effects are on, allow stream some time to flush *effects tail*/ if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID]) @@ -2866,8 +3426,90 @@ static unsigned int ca0132_capture_pcm_delay(struct hda_pcm_stream *info, CA0132_CODEC_VOL_MONO(xname, nid, 3, dir) #define CA0132_CODEC_MUTE(xname, nid, dir) \ CA0132_CODEC_MUTE_MONO(xname, nid, 3, dir) +/* + * Lookup table with decibel values for the DSP. When volume is changed in + * Windows, the DSP is also sent the dB value in floating point. In Windows, + * these values have decimal points, probably because the Windows driver + * actually uses floating point. We can't here, so I made a lookup table of + * values -99 to 9. -99 is the start for the vmaster out control, and for the + * microphone, the value begins at -90. So on microphone lookups, start at + * 9. + */ + +static unsigned int float_vol_db_lookup[] = { +0xC2B40000, 0xC2B20000, 0xC2B00000, 0xC2AE0000, 0xC2AC0000, 0xC2AA0000, +0xC2A80000, 0xC2A60000, 0xC2A40000, 0xC2A20000, 0xC2A00000, 0xC29E0000, +0xC29C0000, 0xC29A0000, 0xC2980000, 0xC2960000, 0xC2940000, 0xC2920000, +0xC2900000, 0xC28E0000, 0xC28C0000, 0xC28A0000, 0xC2880000, 0xC2860000, +0xC2840000, 0xC2820000, 0xC2800000, 0xC27C0000, 0xC2780000, 0xC2740000, +0xC2700000, 0xC26C0000, 0xC2680000, 0xC2640000, 0xC2600000, 0xC25C0000, +0xC2580000, 0xC2540000, 0xC2500000, 0xC24C0000, 0xC2480000, 0xC2440000, +0xC2400000, 0xC23C0000, 0xC2380000, 0xC2340000, 0xC2300000, 0xC22C0000, +0xC2280000, 0xC2240000, 0xC2200000, 0xC21C0000, 0xC2180000, 0xC2140000, +0xC2100000, 0xC20C0000, 0xC2080000, 0xC2040000, 0xC2000000, 0xC1F80000, +0xC1F00000, 0xC1E80000, 0xC1E00000, 0xC1D80000, 0xC1D00000, 0xC1C80000, +0xC1C00000, 0xC1B80000, 0xC1B00000, 0xC1A80000, 0xC1A00000, 0xC1980000, +0xC1900000, 0xC1880000, 0xC1800000, 0xC1700000, 0xC1600000, 0xC1500000, +0xC1400000, 0xC1300000, 0xC1200000, 0xC1100000, 0xC1000000, 0xC0E00000, +0xC0C00000, 0xC0A00000, 0xC0800000, 0xC0400000, 0xC0000000, 0xBF800000, +0x00000000, 0x3F800000, 0x40000000, 0x40400000, 0x40800000, 0x40A00000, +0x40C00000, 0x40E00000, 0x41000000, 0x41100000 +}; + + +/* + * This table counts from float 0 to 1 in increments of .01, which is + * useful for a few different sliders. + */ + +static unsigned int float_zero_to_one_lookup[] = { +0x00000000, 0x3C23D70A, 0x3CA3D70A, 0x3CF5C28F, 0x3D23D70A, 0x3D4CCCCD, +0x3D75C28F, 0x3D8F5C29, 0x3DA3D70A, 0x3DB851EC, 0x3DCCCCCD, 0x3DE147AE, +0x3DF5C28F, 0x3E051EB8, 0x3E0F5C29, 0x3E19999A, 0x3E23D70A, 0x3E2E147B, +0x3E3851EC, 0x3E428F5C, 0x3E4CCCCD, 0x3E570A3D, 0x3E6147AE, 0x3E6B851F, +0x3E75C28F, 0x3E800000, 0x3E851EB8, 0x3E8A3D71, 0x3E8F5C29, 0x3E947AE1, +0x3E99999A, 0x3E9EB852, 0x3EA3D70A, 0x3EA8F5C3, 0x3EAE147B, 0x3EB33333, +0x3EB851EC, 0x3EBD70A4, 0x3EC28F5C, 0x3EC7AE14, 0x3ECCCCCD, 0x3ED1EB85, +0x3ED70A3D, 0x3EDC28F6, 0x3EE147AE, 0x3EE66666, 0x3EEB851F, 0x3EF0A3D7, +0x3EF5C28F, 0x3EFAE148, 0x3F000000, 0x3F028F5C, 0x3F051EB8, 0x3F07AE14, +0x3F0A3D71, 0x3F0CCCCD, 0x3F0F5C29, 0x3F11EB85, 0x3F147AE1, 0x3F170A3D, +0x3F19999A, 0x3F1C28F6, 0x3F1EB852, 0x3F2147AE, 0x3F23D70A, 0x3F266666, +0x3F28F5C3, 0x3F2B851F, 0x3F2E147B, 0x3F30A3D7, 0x3F333333, 0x3F35C28F, +0x3F3851EC, 0x3F3AE148, 0x3F3D70A4, 0x3F400000, 0x3F428F5C, 0x3F451EB8, +0x3F47AE14, 0x3F4A3D71, 0x3F4CCCCD, 0x3F4F5C29, 0x3F51EB85, 0x3F547AE1, +0x3F570A3D, 0x3F59999A, 0x3F5C28F6, 0x3F5EB852, 0x3F6147AE, 0x3F63D70A, +0x3F666666, 0x3F68F5C3, 0x3F6B851F, 0x3F6E147B, 0x3F70A3D7, 0x3F733333, +0x3F75C28F, 0x3F7851EC, 0x3F7AE148, 0x3F7D70A4, 0x3F800000 +}; + +/* + * This table counts from float 10 to 1000, which is the range of the x-bass + * crossover slider in Windows. + */ + +static unsigned int float_xbass_xover_lookup[] = { +0x41200000, 0x41A00000, 0x41F00000, 0x42200000, 0x42480000, 0x42700000, +0x428C0000, 0x42A00000, 0x42B40000, 0x42C80000, 0x42DC0000, 0x42F00000, +0x43020000, 0x430C0000, 0x43160000, 0x43200000, 0x432A0000, 0x43340000, +0x433E0000, 0x43480000, 0x43520000, 0x435C0000, 0x43660000, 0x43700000, +0x437A0000, 0x43820000, 0x43870000, 0x438C0000, 0x43910000, 0x43960000, +0x439B0000, 0x43A00000, 0x43A50000, 0x43AA0000, 0x43AF0000, 0x43B40000, +0x43B90000, 0x43BE0000, 0x43C30000, 0x43C80000, 0x43CD0000, 0x43D20000, +0x43D70000, 0x43DC0000, 0x43E10000, 0x43E60000, 0x43EB0000, 0x43F00000, +0x43F50000, 0x43FA0000, 0x43FF0000, 0x44020000, 0x44048000, 0x44070000, +0x44098000, 0x440C0000, 0x440E8000, 0x44110000, 0x44138000, 0x44160000, +0x44188000, 0x441B0000, 0x441D8000, 0x44200000, 0x44228000, 0x44250000, +0x44278000, 0x442A0000, 0x442C8000, 0x442F0000, 0x44318000, 0x44340000, +0x44368000, 0x44390000, 0x443B8000, 0x443E0000, 0x44408000, 0x44430000, +0x44458000, 0x44480000, 0x444A8000, 0x444D0000, 0x444F8000, 0x44520000, +0x44548000, 0x44570000, 0x44598000, 0x445C0000, 0x445E8000, 0x44610000, +0x44638000, 0x44660000, 0x44688000, 0x446B0000, 0x446D8000, 0x44700000, +0x44728000, 0x44750000, 0x44778000, 0x447A0000 +}; + /* The following are for tuning of products */ + #ifdef ENABLE_TUNING_CONTROLS static unsigned int voice_focus_vals_lookup[] = { @@ -2900,26 +3542,6 @@ static unsigned int voice_focus_vals_lookup[] = { 0x43300000, 0x43310000, 0x43320000, 0x43330000, 0x43340000 }; -static unsigned int mic_svm_vals_lookup[] = { -0x00000000, 0x3C23D70A, 0x3CA3D70A, 0x3CF5C28F, 0x3D23D70A, 0x3D4CCCCD, -0x3D75C28F, 0x3D8F5C29, 0x3DA3D70A, 0x3DB851EC, 0x3DCCCCCD, 0x3DE147AE, -0x3DF5C28F, 0x3E051EB8, 0x3E0F5C29, 0x3E19999A, 0x3E23D70A, 0x3E2E147B, -0x3E3851EC, 0x3E428F5C, 0x3E4CCCCD, 0x3E570A3D, 0x3E6147AE, 0x3E6B851F, -0x3E75C28F, 0x3E800000, 0x3E851EB8, 0x3E8A3D71, 0x3E8F5C29, 0x3E947AE1, -0x3E99999A, 0x3E9EB852, 0x3EA3D70A, 0x3EA8F5C3, 0x3EAE147B, 0x3EB33333, -0x3EB851EC, 0x3EBD70A4, 0x3EC28F5C, 0x3EC7AE14, 0x3ECCCCCD, 0x3ED1EB85, -0x3ED70A3D, 0x3EDC28F6, 0x3EE147AE, 0x3EE66666, 0x3EEB851F, 0x3EF0A3D7, -0x3EF5C28F, 0x3EFAE148, 0x3F000000, 0x3F028F5C, 0x3F051EB8, 0x3F07AE14, -0x3F0A3D71, 0x3F0CCCCD, 0x3F0F5C29, 0x3F11EB85, 0x3F147AE1, 0x3F170A3D, -0x3F19999A, 0x3F1C28F6, 0x3F1EB852, 0x3F2147AE, 0x3F23D70A, 0x3F266666, -0x3F28F5C3, 0x3F2B851F, 0x3F2E147B, 0x3F30A3D7, 0x3F333333, 0x3F35C28F, -0x3F3851EC, 0x3F3AE148, 0x3F3D70A4, 0x3F400000, 0x3F428F5C, 0x3F451EB8, -0x3F47AE14, 0x3F4A3D71, 0x3F4CCCCD, 0x3F4F5C29, 0x3F51EB85, 0x3F547AE1, -0x3F570A3D, 0x3F59999A, 0x3F5C28F6, 0x3F5EB852, 0x3F6147AE, 0x3F63D70A, -0x3F666666, 0x3F68F5C3, 0x3F6B851F, 0x3F6E147B, 0x3F70A3D7, 0x3F733333, -0x3F75C28F, 0x3F7851EC, 0x3F7AE148, 0x3F7D70A4, 0x3F800000 -}; - static unsigned int equalizer_vals_lookup[] = { 0xC1C00000, 0xC1B80000, 0xC1B00000, 0xC1A80000, 0xC1A00000, 0xC1980000, 0xC1900000, 0xC1880000, 0xC1800000, 0xC1700000, 0xC1600000, 0xC1500000, @@ -2942,7 +3564,7 @@ static int tuning_ctl_set(struct hda_codec *codec, hda_nid_t nid, break; snd_hda_power_up(codec); - dspio_set_param(codec, ca0132_tuning_ctls[i].mid, + dspio_set_param(codec, ca0132_tuning_ctls[i].mid, 0x20, ca0132_tuning_ctls[i].req, &(lookup[idx]), sizeof(unsigned int)); snd_hda_power_down(codec); @@ -3028,7 +3650,7 @@ static int mic_svm_ctl_put(struct snd_kcontrol *kcontrol, spec->cur_ctl_vals[idx] = *valp; idx = *valp; - tuning_ctl_set(codec, nid, mic_svm_vals_lookup, idx); + tuning_ctl_set(codec, nid, float_zero_to_one_lookup, idx); return 0; } @@ -3068,9 +3690,11 @@ static int equalizer_ctl_put(struct snd_kcontrol *kcontrol, return 1; } +/*FIXME Figure out why the program will not compile with these. */ +/* static const DECLARE_TLV_DB_SCALE(voice_focus_db_scale, 2000, 100, 0); static const DECLARE_TLV_DB_SCALE(eq_db_scale, -2400, 100, 0); - +*/ static int add_tuning_control(struct hda_codec *codec, hda_nid_t pnid, hda_nid_t nid, const char *name, int dir) @@ -3089,7 +3713,7 @@ static int add_tuning_control(struct hda_codec *codec, knew.info = voice_focus_ctl_info; knew.get = tuning_ctl_get; knew.put = voice_focus_ctl_put; - knew.tlv.p = voice_focus_db_scale; +// knew.tlv.p = voice_focus_db_scale; break; case MIC_SVM: knew.info = mic_svm_ctl_info; @@ -3100,7 +3724,7 @@ static int add_tuning_control(struct hda_codec *codec, knew.info = equalizer_ctl_info; knew.get = tuning_ctl_get; knew.put = equalizer_ctl_put; - knew.tlv.p = eq_db_scale; +// knew.tlv.p = eq_db_scale; break; default: return 0; @@ -3111,6 +3735,7 @@ static int add_tuning_control(struct hda_codec *codec, return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec)); } + static int add_tuning_ctls(struct hda_codec *codec) { int i; @@ -3134,7 +3759,7 @@ static void ca0132_init_tuning_defaults(struct hda_codec *codec) struct ca0132_spec *spec = codec->spec; int i; - /* Wedge Angle defaults to 30. 10 below is 30 - 20. 20 is min. */ + /* Wedge Angle defaults to 30. 10 below is 30 - 20. 20 is min. */ spec->cur_ctl_vals[WEDGE_ANGLE - TUNING_CTL_START_NID] = 10; /* SVM level defaults to 0.74. */ spec->cur_ctl_vals[SVM_LEVEL - TUNING_CTL_START_NID] = 74; @@ -3146,11 +3771,446 @@ static void ca0132_init_tuning_defaults(struct hda_codec *codec) #endif /*ENABLE_TUNING_CONTROLS*/ /* + * Below I've added controls to mess with the effect levels, I've only enabled + * them on the Sound Blaster Z, but they would probably also work on the + * Chromebook. I figured they were probably tuned specifically for it, and left + * out for a reason. + */ + +/* Sets DSP effect level from the sliders above the controls */ +static int ca0132_alt_slider_ctl_set(struct hda_codec *codec, hda_nid_t nid, + unsigned int *lookup, int idx) +{ + int i = 0; + unsigned int y; + /* For X_BASS, req 2 is actually crossover freq instead of + * effect level. + */ + if (nid == X_BASS) + y = 2; + else + y = 1; + + snd_hda_power_up(codec); + if (nid == XBASS_XOVER) { + for (i = 0; i < OUT_EFFECTS_COUNT; i++) + if (ca0132_effects[i].nid == X_BASS) + break; + + dspio_set_param(codec, ca0132_effects[i].mid, 0x20, + ca0132_effects[i].reqs[1], + &(lookup[idx - 1]), sizeof(unsigned int)); + } else { + /* Find the actual effect structure */ + for (i = 0; i < OUT_EFFECTS_COUNT; i++) + if (nid == ca0132_effects[i].nid) + break; + + dspio_set_param(codec, ca0132_effects[i].mid, 0x20, + ca0132_effects[i].reqs[y], + &(lookup[idx]), sizeof(unsigned int)); + } + + snd_hda_power_down(codec); + + return 0; +} + +static int ca0132_alt_xbass_xover_slider_ctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + long *valp = ucontrol->value.integer.value; + + *valp = spec->xbass_xover_freq; + return 0; +} + +static int ca0132_alt_slider_ctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + hda_nid_t nid = get_amp_nid(kcontrol); + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + long *valp = ucontrol->value.integer.value; + int idx = nid - OUT_EFFECT_START_NID; + + *valp = spec->fx_ctl_val[idx]; + return 0; +} + +/* + * The X-bass crossover starts at 10hz, so the min is 1. The + * frequency is set in multiples of 10. + */ +static int ca0132_alt_xbass_xover_slider_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 1; + uinfo->value.integer.max = 100; + uinfo->value.integer.step = 1; + + return 0; +} + +static int ca0132_alt_effect_slider_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int chs = get_amp_channels(kcontrol); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = chs == 3 ? 2 : 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 100; + uinfo->value.integer.step = 1; + + return 0; +} + +static int ca0132_alt_xbass_xover_slider_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + hda_nid_t nid = get_amp_nid(kcontrol); + long *valp = ucontrol->value.integer.value; + int idx; + + /* any change? */ + if (spec->xbass_xover_freq == *valp) + return 0; + + spec->xbass_xover_freq = *valp; + + idx = *valp; + ca0132_alt_slider_ctl_set(codec, nid, float_xbass_xover_lookup, idx); + + return 0; +} + +static int ca0132_alt_effect_slider_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + hda_nid_t nid = get_amp_nid(kcontrol); + long *valp = ucontrol->value.integer.value; + int idx; + + idx = nid - EFFECT_START_NID; + /* any change? */ + if (spec->fx_ctl_val[idx] == *valp) + return 0; + + spec->fx_ctl_val[idx] = *valp; + + idx = *valp; + ca0132_alt_slider_ctl_set(codec, nid, float_zero_to_one_lookup, idx); + + return 0; +} + +static int ca0132_alt_add_effect_slider(struct hda_codec *codec, hda_nid_t nid, + const char *pfx, int dir) +{ + char *fx = "FX:"; + char namestr[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + int type = dir ? HDA_INPUT : HDA_OUTPUT; + struct snd_kcontrol_new knew = + HDA_CODEC_VOLUME_MONO(namestr, nid, 1, 0, type); + + sprintf(namestr, "%s %s %s Volume", fx, pfx, dirstr[dir]); + + knew.tlv.c = 0; + knew.tlv.p = 0; + + switch (nid) { + case XBASS_XOVER: + knew.info = ca0132_alt_xbass_xover_slider_info; + knew.get = ca0132_alt_xbass_xover_slider_ctl_get; + knew.put = ca0132_alt_xbass_xover_slider_put; + break; + default: + knew.info = ca0132_alt_effect_slider_info; + knew.get = ca0132_alt_slider_ctl_get; + knew.put = ca0132_alt_effect_slider_put; + knew.private_value = + HDA_COMPOSE_AMP_VAL(nid, 1, 0, type); + break; + } + + return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec)); +} + +/* + * Sets the internal DSP decibel level. Currently only the SBZ sets internal + * capture volume level, and all other alternative codecs set the playback + * volume level. + */ +static void ca0132_alt_dsp_volume_put(struct hda_codec *codec, hda_nid_t nid) +{ + struct ca0132_spec *spec = codec->spec; + unsigned int dsp_dir; + unsigned int lookup_val; + + if (spec->quirk == QUIRK_R3DI && nid == VNID_MIC) + return; + + if (nid == VNID_SPK) + dsp_dir = DSP_VOL_OUT; + else + dsp_dir = DSP_VOL_IN; + + lookup_val = spec->vnode_lvol[nid - VNODE_START_NID]; + + dspio_set_uint_param(codec, + ca0132_alt_vol_ctls[dsp_dir].mid, + ca0132_alt_vol_ctls[dsp_dir].reqs[0], + float_vol_db_lookup[lookup_val]); + + lookup_val = spec->vnode_rvol[nid - VNODE_START_NID]; + + dspio_set_uint_param(codec, + ca0132_alt_vol_ctls[dsp_dir].mid, + ca0132_alt_vol_ctls[dsp_dir].reqs[1], + float_vol_db_lookup[lookup_val]); + + dspio_set_uint_param(codec, + ca0132_alt_vol_ctls[dsp_dir].mid, + ca0132_alt_vol_ctls[dsp_dir].reqs[2], FLOAT_0); +} + +/* + * Select the active output. + * If autodetect is enabled, output will be selected based on jack detection. + * If a jack is detected in the headphone connector, then headphone will be + * selected. Otherwise, it defaults to line-out. + * If autodetect is disabled, output will be set based on the output select + * enum control. It can either be surround, line out, or headphone. + */ + +static int ca0132_alt_select_out(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + unsigned int pin_ctl; + int jack_present; + int auto_jack; + unsigned int i; + unsigned int tmp; + int err; + /* Default Headphone is rear headphone */ + hda_nid_t headphone_nid = spec->out_pins[1]; + + codec_dbg(codec, "ca0132_alt_select_out\n"); + + snd_hda_power_up_pm(codec); + + auto_jack = spec->vnode_lswitch[VNID_HP_ASEL - VNODE_START_NID]; + + /* + * If headphone rear or front is plugged in, set to headphone. + * If neither is plugged in, set to rear line out. Only if + * hp/speaker auto detect is enabled. + */ + if (auto_jack) { + jack_present = snd_hda_jack_detect(codec, spec->unsol_tag_hp) || + snd_hda_jack_detect(codec, spec->unsol_tag_front_hp); + + if (jack_present) + spec->cur_out_type = HEADPHONE_OUT; + else + spec->cur_out_type = SPEAKER_OUT; + } else + spec->cur_out_type = spec->out_enum_val; + + /* Begin DSP output switch */ + tmp = FLOAT_1; + err = dspio_set_uint_param(codec, 0x96, 0x3A, tmp); + if (err < 0) + goto exit; + + switch (spec->cur_out_type) { + case SPEAKER_OUT: + codec_dbg(codec, "ca0132_alt_select_out speaker\n"); + /*speaker out config*/ + switch (spec->quirk) { + case QUIRK_SBZ: + /* + * I'm pretty positive these are some sort of GPIO, + * where the 4th hex digit is the number of the GPIO, + * and the second is on or off. These are used for + * switching outputs, and most importantly, front + * microphone + */ + writew(0x0007, spec->mem_base + 0x320); + writew(0x0104, spec->mem_base + 0x320); + writew(0x0101, spec->mem_base + 0x320); + chipio_set_control_param(codec, 0x0D, 0x18); + break; + case QUIRK_R3DI: + chipio_set_control_param(codec, 0x0D, 0x24); + r3di_gpio_out_set(codec, R3DI_LINE_OUT); + break; + } + + /* disable headphone node */ + pin_ctl = snd_hda_codec_read(codec, spec->out_pins[1], 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + snd_hda_set_pin_ctl(codec, spec->out_pins[1], + pin_ctl & ~PIN_HP); + /* enable line-out node */ + pin_ctl = snd_hda_codec_read(codec, spec->out_pins[0], 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + snd_hda_set_pin_ctl(codec, spec->out_pins[0], + pin_ctl | PIN_OUT); + /* Enable EAPD */ + snd_hda_codec_write(codec, spec->out_pins[0], 0, + AC_VERB_SET_EAPD_BTLENABLE, 0x01); + + /* If PlayEnhancement is enabled, set different source */ + if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID]) + dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_1); + else + dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_8); + + /* run through the output dsp commands for line-out */ + for (i = 0; i < alt_out_presets[spec->cur_out_type].commands; + i++) { + err = dspio_set_uint_param(codec, + alt_out_presets[spec->cur_out_type].mids[i], + alt_out_presets[spec->cur_out_type].reqs[i], + alt_out_presets[spec->cur_out_type].vals[i]); + + if (err < 0) + goto exit; + } + break; + case HEADPHONE_OUT: + codec_dbg(codec, "ca0132_alt_select_out hp\n"); + /* Headphone out config*/ + switch (spec->quirk) { + case QUIRK_SBZ: + writew(0x0107, spec->mem_base + 0x320); + writew(0x0104, spec->mem_base + 0x320); + writew(0x0001, spec->mem_base + 0x320); + chipio_set_control_param(codec, 0x0D, 0x12); + break; + case QUIRK_R3DI: + chipio_set_control_param(codec, 0x0D, 0x21); + r3di_gpio_out_set(codec, R3DI_HEADPHONE_OUT); + break; + } + + snd_hda_codec_write(codec, spec->out_pins[0], 0, + AC_VERB_SET_EAPD_BTLENABLE, 0x00); + + /* disable speaker*/ + pin_ctl = snd_hda_codec_read(codec, spec->out_pins[0], 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + snd_hda_set_pin_ctl(codec, spec->out_pins[0], + pin_ctl & ~PIN_HP); + + /* enable headphone, either front or rear */ + + if (snd_hda_jack_detect(codec, spec->unsol_tag_front_hp)) + headphone_nid = spec->out_pins[2]; + else if (snd_hda_jack_detect(codec, spec->unsol_tag_hp)) + headphone_nid = spec->out_pins[1]; + + pin_ctl = snd_hda_codec_read(codec, headphone_nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + snd_hda_set_pin_ctl(codec, headphone_nid, + pin_ctl | PIN_HP); + + if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID]) + dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_1); + else + dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_0); + + for (i = 0; i < alt_out_presets[spec->cur_out_type].commands; + i++) { + err = dspio_set_uint_param(codec, + alt_out_presets[spec->cur_out_type].mids[i], + alt_out_presets[spec->cur_out_type].reqs[i], + alt_out_presets[spec->cur_out_type].vals[i]); + + if (err < 0) + goto exit; + } + break; + case SURROUND_OUT: + codec_dbg(codec, "ca0132_alt_select_out Surround\n"); + /* Surround out config*/ + switch (spec->quirk) { + case QUIRK_SBZ: + writew(0x0007, spec->mem_base + 0x320); + writew(0x0104, spec->mem_base + 0x320); + writew(0x0101, spec->mem_base + 0x320); + chipio_set_control_param(codec, 0x0D, 0x18); + break; + case QUIRK_R3DI: + chipio_set_control_param(codec, 0x0D, 0x24); + r3di_gpio_out_set(codec, R3DI_LINE_OUT); + break; + } + /* enable line out node */ + pin_ctl = snd_hda_codec_read(codec, spec->out_pins[0], 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + snd_hda_set_pin_ctl(codec, spec->out_pins[0], + pin_ctl | PIN_OUT); + /* Disable headphone out */ + pin_ctl = snd_hda_codec_read(codec, spec->out_pins[1], 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + snd_hda_set_pin_ctl(codec, spec->out_pins[1], + pin_ctl & ~PIN_HP); + /* Enable EAPD on line out */ + snd_hda_codec_write(codec, spec->out_pins[0], 0, + AC_VERB_SET_EAPD_BTLENABLE, 0x01); + /* enable center/lfe out node */ + pin_ctl = snd_hda_codec_read(codec, spec->out_pins[2], 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + snd_hda_set_pin_ctl(codec, spec->out_pins[2], + pin_ctl | PIN_OUT); + /* Now set rear surround node as out. */ + pin_ctl = snd_hda_codec_read(codec, spec->out_pins[3], 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + snd_hda_set_pin_ctl(codec, spec->out_pins[3], + pin_ctl | PIN_OUT); + + if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID]) + dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_1); + else + dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_8); + + for (i = 0; i < alt_out_presets[spec->cur_out_type].commands; + i++) { + err = dspio_set_uint_param(codec, + alt_out_presets[spec->cur_out_type].mids[i], + alt_out_presets[spec->cur_out_type].reqs[i], + alt_out_presets[spec->cur_out_type].vals[i]); + + if (err < 0) + goto exit; + } + break; + } + +exit: + snd_hda_power_down_pm(codec); + + return err < 0 ? err : 0; +} + +/* * Select the active output. * If autodetect is enabled, output will be selected based on jack detection. * If jack inserted, headphone will be selected, else built-in speakers * If autodetect is disabled, output will be selected based on selection. */ + static int ca0132_select_out(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; @@ -3180,12 +4240,12 @@ static int ca0132_select_out(struct hda_codec *codec) if (spec->cur_out_type == SPEAKER_OUT) { codec_dbg(codec, "ca0132_select_out speaker\n"); /*speaker out config*/ - tmp = FLOAT_ONE; + tmp = FLOAT_1; err = dspio_set_uint_param(codec, 0x80, 0x04, tmp); if (err < 0) goto exit; /*enable speaker EQ*/ - tmp = FLOAT_ONE; + tmp = FLOAT_1; err = dspio_set_uint_param(codec, 0x8f, 0x00, tmp); if (err < 0) goto exit; @@ -3213,12 +4273,12 @@ static int ca0132_select_out(struct hda_codec *codec) } else { codec_dbg(codec, "ca0132_select_out hp\n"); /*headphone out config*/ - tmp = FLOAT_ZERO; + tmp = FLOAT_0; err = dspio_set_uint_param(codec, 0x80, 0x04, tmp); if (err < 0) goto exit; /*disable speaker EQ*/ - tmp = FLOAT_ZERO; + tmp = FLOAT_0; err = dspio_set_uint_param(codec, 0x8f, 0x00, tmp); if (err < 0) goto exit; @@ -3256,8 +4316,18 @@ static void ca0132_unsol_hp_delayed(struct work_struct *work) struct ca0132_spec *spec = container_of( to_delayed_work(work), struct ca0132_spec, unsol_hp_work); struct hda_jack_tbl *jack; + switch (spec->quirk) { + case QUIRK_SBZ: + ca0132_alt_select_out(spec->codec); + break; + case QUIRK_R3DI: + ca0132_alt_select_out(spec->codec); + break; + case QUIRK_NONE: + ca0132_select_out(spec->codec); + break; + } - ca0132_select_out(spec->codec); jack = snd_hda_jack_tbl_get(spec->codec, spec->unsol_tag_hp); if (jack) { jack->block_report = 0; @@ -3266,8 +4336,77 @@ static void ca0132_unsol_hp_delayed(struct work_struct *work) } static void ca0132_set_dmic(struct hda_codec *codec, int enable); +static void resume_mic1(struct hda_codec *codec, unsigned int oldval); +static int stop_mic1(struct hda_codec *codec); static int ca0132_mic_boost_set(struct hda_codec *codec, long val); static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val); +static int ca0132_cvoice_switch_set(struct hda_codec *codec); +static int ca0132_alt_mic_boost_set(struct hda_codec *codec, long val); + +static int ca0132_alt_set_vipsource(struct hda_codec *codec, int val) +{ + struct ca0132_spec *spec = codec->spec; + unsigned int tmp; + + if (spec->dsp_state != DSP_DOWNLOADED) + return 0; + + codec_dbg(codec, "ca0132_alt_set_vipsource"); + + chipio_set_stream_control(codec, 0x03, 0); + chipio_set_stream_control(codec, 0x04, 0); + + /* if CrystalVoice is off, vipsource should be 0 */ + if (!spec->effects_switch[CRYSTAL_VOICE - EFFECT_START_NID] || + (val == 0) || spec->in_enum_val == REAR_LINE_IN) { + codec_dbg(codec, "ca0132_alt_set_vipsource off."); + chipio_set_control_param(codec, CONTROL_PARAM_VIP_SOURCE, 0); + + tmp = FLOAT_0; + dspio_set_uint_param(codec, 0x80, 0x05, tmp); + + chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000); + chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000); + if (spec->quirk == QUIRK_R3DI) + chipio_set_conn_rate(codec, 0x0F, SR_96_000); + + + if (spec->in_enum_val == REAR_LINE_IN) + tmp = FLOAT_0; + else { + if (spec->quirk == QUIRK_SBZ) + tmp = FLOAT_3; + else + tmp = FLOAT_1; + } + + dspio_set_uint_param(codec, 0x80, 0x00, tmp); + + } else { + codec_dbg(codec, "ca0132_alt_set_vipsource on."); + chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_16_000); + chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_16_000); + if (spec->quirk == QUIRK_R3DI) + chipio_set_conn_rate(codec, 0x0F, SR_16_000); + + if (spec->effects_switch[VOICE_FOCUS - EFFECT_START_NID]) + tmp = FLOAT_2; + else + tmp = FLOAT_1; + dspio_set_uint_param(codec, 0x80, 0x00, tmp); + + tmp = FLOAT_1; + dspio_set_uint_param(codec, 0x80, 0x05, tmp); + + msleep(20); + chipio_set_control_param(codec, CONTROL_PARAM_VIP_SOURCE, val); + } + + chipio_set_stream_control(codec, 0x03, 1); + chipio_set_stream_control(codec, 0x04, 1); + + return 1; +} /* * Select the active VIP source @@ -3283,25 +4422,27 @@ static int ca0132_set_vipsource(struct hda_codec *codec, int val) /* if CrystalVoice if off, vipsource should be 0 */ if (!spec->effects_switch[CRYSTAL_VOICE - EFFECT_START_NID] || (val == 0)) { + codec_dbg(codec, "ca0132_set_vipsource off."); chipio_set_control_param(codec, CONTROL_PARAM_VIP_SOURCE, 0); chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000); chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000); if (spec->cur_mic_type == DIGITAL_MIC) - tmp = FLOAT_TWO; + tmp = FLOAT_2; else - tmp = FLOAT_ONE; + tmp = FLOAT_1; dspio_set_uint_param(codec, 0x80, 0x00, tmp); - tmp = FLOAT_ZERO; + tmp = FLOAT_0; dspio_set_uint_param(codec, 0x80, 0x05, tmp); } else { + codec_dbg(codec, "ca0132_set_vipsource on."); chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_16_000); chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_16_000); if (spec->cur_mic_type == DIGITAL_MIC) - tmp = FLOAT_TWO; + tmp = FLOAT_2; else - tmp = FLOAT_ONE; + tmp = FLOAT_1; dspio_set_uint_param(codec, 0x80, 0x00, tmp); - tmp = FLOAT_ONE; + tmp = FLOAT_1; dspio_set_uint_param(codec, 0x80, 0x05, tmp); msleep(20); chipio_set_control_param(codec, CONTROL_PARAM_VIP_SOURCE, val); @@ -3363,6 +4504,129 @@ static int ca0132_select_mic(struct hda_codec *codec) } /* + * Select the active input. + * Mic detection isn't used, because it's kind of pointless on the SBZ. + * The front mic has no jack-detection, so the only way to switch to it + * is to do it manually in alsamixer. + */ + +static int ca0132_alt_select_in(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + unsigned int tmp; + + codec_dbg(codec, "ca0132_alt_select_in\n"); + + snd_hda_power_up_pm(codec); + + chipio_set_stream_control(codec, 0x03, 0); + chipio_set_stream_control(codec, 0x04, 0); + + spec->cur_mic_type = spec->in_enum_val; + + switch (spec->cur_mic_type) { + case REAR_MIC: + switch (spec->quirk) { + case QUIRK_SBZ: + writew(0x0000, spec->mem_base + 0x320); + tmp = FLOAT_3; + break; + case QUIRK_R3DI: + r3di_gpio_mic_set(codec, R3DI_REAR_MIC); + tmp = FLOAT_1; + break; + default: + tmp = FLOAT_1; + break; + } + + chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000); + chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000); + + if (spec->quirk == QUIRK_R3DI) + chipio_set_conn_rate(codec, 0x0F, SR_96_000); + + dspio_set_uint_param(codec, 0x80, 0x00, tmp); + + chipio_set_stream_control(codec, 0x03, 1); + chipio_set_stream_control(codec, 0x04, 1); + + if (spec->quirk == QUIRK_SBZ) { + chipio_write(codec, 0x18B098, 0x0000000C); + chipio_write(codec, 0x18B09C, 0x0000000C); + } + + ca0132_alt_mic_boost_set(codec, spec->mic_boost_enum_val); + break; + case REAR_LINE_IN: + ca0132_alt_mic_boost_set(codec, 0); + switch (spec->quirk) { + case QUIRK_SBZ: + writew(0x0000, spec->mem_base + 0x320); + break; + case QUIRK_R3DI: + r3di_gpio_mic_set(codec, R3DI_REAR_MIC); + break; + } + + chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000); + chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000); + + if (spec->quirk == QUIRK_R3DI) + chipio_set_conn_rate(codec, 0x0F, SR_96_000); + + tmp = FLOAT_0; + dspio_set_uint_param(codec, 0x80, 0x00, tmp); + + if (spec->quirk == QUIRK_SBZ) { + chipio_write(codec, 0x18B098, 0x00000000); + chipio_write(codec, 0x18B09C, 0x00000000); + } + + chipio_set_stream_control(codec, 0x03, 1); + chipio_set_stream_control(codec, 0x04, 1); + break; + case FRONT_MIC: + switch (spec->quirk) { + case QUIRK_SBZ: + writew(0x0100, spec->mem_base + 0x320); + writew(0x0005, spec->mem_base + 0x320); + tmp = FLOAT_3; + break; + case QUIRK_R3DI: + r3di_gpio_mic_set(codec, R3DI_FRONT_MIC); + tmp = FLOAT_1; + break; + default: + tmp = FLOAT_1; + break; + } + + chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000); + chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000); + if (spec->quirk == QUIRK_R3DI) + chipio_set_conn_rate(codec, 0x0F, SR_96_000); + + dspio_set_uint_param(codec, 0x80, 0x00, tmp); + + chipio_set_stream_control(codec, 0x03, 1); + chipio_set_stream_control(codec, 0x04, 1); + + if (spec->quirk == QUIRK_SBZ) { + chipio_write(codec, 0x18B098, 0x0000000C); + chipio_write(codec, 0x18B09C, 0x000000CC); + } + ca0132_alt_mic_boost_set(codec, spec->mic_boost_enum_val); + break; + } + ca0132_cvoice_switch_set(codec); + + snd_hda_power_down_pm(codec); + return 0; + +} + +/* * Check if VNODE settings take effect immediately. */ static bool ca0132_is_vnode_effective(struct hda_codec *codec, @@ -3391,7 +4655,7 @@ static bool ca0132_is_vnode_effective(struct hda_codec *codec, /* * The following functions are control change helpers. -* They return 0 if no changed. Return 1 if changed. +* They return 0 if no changed. Return 1 if changed. */ static int ca0132_voicefx_set(struct hda_codec *codec, int enable) { @@ -3401,9 +4665,9 @@ static int ca0132_voicefx_set(struct hda_codec *codec, int enable) /* based on CrystalVoice state to enable VoiceFX. */ if (enable) { tmp = spec->effects_switch[CRYSTAL_VOICE - EFFECT_START_NID] ? - FLOAT_ONE : FLOAT_ZERO; + FLOAT_1 : FLOAT_0; } else { - tmp = FLOAT_ZERO; + tmp = FLOAT_0; } dspio_set_uint_param(codec, ca0132_voicefx.mid, @@ -3418,7 +4682,7 @@ static int ca0132_voicefx_set(struct hda_codec *codec, int enable) static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val) { struct ca0132_spec *spec = codec->spec; - unsigned int on; + unsigned int on, tmp; int num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT; int err = 0; int idx = nid - EFFECT_START_NID; @@ -3435,19 +4699,63 @@ static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val) /* for in effect, qualify with CrystalVoice */ if ((nid >= IN_EFFECT_START_NID) && (nid < IN_EFFECT_END_NID)) { - /* if CrystalVoice if off, turn off in effects. */ + /* if CrystalVoice is off, turn off in effects. */ if (!spec->effects_switch[CRYSTAL_VOICE - EFFECT_START_NID]) val = 0; + /* Voice Focus applies to 2-ch Mic, Digital Mic */ - if ((nid == VOICE_FOCUS) && (spec->cur_mic_type != DIGITAL_MIC)) + if ((nid == VOICE_FOCUS) && (spec->cur_mic_type != DIGITAL_MIC) + && (spec->quirk != QUIRK_SBZ)) val = 0; + + /* If Voice Focus on SBZ, set to two channel. */ + if ((nid == VOICE_FOCUS) && (spec->quirk == QUIRK_SBZ) + && (spec->cur_mic_type != REAR_LINE_IN)) { + if (spec->effects_switch[CRYSTAL_VOICE - + EFFECT_START_NID]) { + + if (spec->effects_switch[VOICE_FOCUS - + EFFECT_START_NID]) + tmp = FLOAT_2; + else + tmp = FLOAT_1; + + dspio_set_uint_param(codec, 0x80, 0x00, tmp); + } + } + + /* + * For SBZ noise reduction, there's an extra command + * to module ID 0x47. No clue why. + */ + if ((nid == NOISE_REDUCTION) && (spec->quirk == QUIRK_SBZ) + && (spec->cur_mic_type != REAR_LINE_IN)) { + if (spec->effects_switch[CRYSTAL_VOICE - + EFFECT_START_NID]) { + if (spec->effects_switch[NOISE_REDUCTION - + EFFECT_START_NID]) + tmp = FLOAT_1; + else + tmp = FLOAT_0; + } else + tmp = FLOAT_0; + + dspio_set_uint_param(codec, 0x47, 0x00, tmp); + } + + /* If rear line in disable effects. */ + if (spec->quirk != QUIRK_NONE && + spec->in_enum_val == REAR_LINE_IN) { + val = 0; + } + } codec_dbg(codec, "ca0132_effect_set: nid=0x%x, val=%ld\n", nid, val); - on = (val == 0) ? FLOAT_ZERO : FLOAT_ONE; + on = (val == 0) ? FLOAT_0 : FLOAT_1; err = dspio_set_uint_param(codec, ca0132_effects[idx].mid, ca0132_effects[idx].reqs[0], on); @@ -3469,6 +4777,9 @@ static int ca0132_pe_switch_set(struct hda_codec *codec) codec_dbg(codec, "ca0132_pe_switch_set: val=%ld\n", spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID]); + if (spec->quirk != QUIRK_NONE) + ca0132_alt_select_out(codec); + i = OUT_EFFECT_START_NID - EFFECT_START_NID; nid = OUT_EFFECT_START_NID; /* PE affects all out effects */ @@ -3515,6 +4826,11 @@ static int ca0132_cvoice_switch_set(struct hda_codec *codec) codec_dbg(codec, "ca0132_cvoice_switch_set: val=%ld\n", spec->effects_switch[CRYSTAL_VOICE - EFFECT_START_NID]); + if (spec->quirk != QUIRK_NONE) { + if (spec->in_enum_val == REAR_LINE_IN) + codec_dbg(codec, "Line-in, no CrystalVoice."); + } + i = IN_EFFECT_START_NID - EFFECT_START_NID; nid = IN_EFFECT_START_NID; /* CrystalVoice affects all in effects */ @@ -3526,7 +4842,10 @@ static int ca0132_cvoice_switch_set(struct hda_codec *codec) /* set correct vipsource */ oldval = stop_mic1(codec); - ret |= ca0132_set_vipsource(codec, 1); + if (spec->quirk != QUIRK_NONE) + ret |= ca0132_alt_set_vipsource(codec, 1); + else + ret |= ca0132_set_vipsource(codec, 1); resume_mic1(codec, oldval); return ret; } @@ -3546,6 +4865,16 @@ static int ca0132_mic_boost_set(struct hda_codec *codec, long val) return ret; } +static int ca0132_alt_mic_boost_set(struct hda_codec *codec, long val) +{ + struct ca0132_spec *spec = codec->spec; + int ret = 0; + + ret = snd_hda_codec_amp_update(codec, spec->input_pins[0], 0, + HDA_INPUT, 0, HDA_AMP_VOLMASK, val); + return ret; +} + static int ca0132_vnode_switch_set(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -3560,9 +4889,21 @@ static int ca0132_vnode_switch_set(struct snd_kcontrol *kcontrol, if (nid == VNID_HP_SEL) { auto_jack = spec->vnode_lswitch[VNID_HP_ASEL - VNODE_START_NID]; - if (!auto_jack) - ca0132_select_out(codec); - return 1; + if (!auto_jack) { + switch (spec->quirk) { + case QUIRK_SBZ: + ca0132_alt_select_out(codec); + break; + case QUIRK_R3DI: + ca0132_alt_select_out(codec); + break; + case QUIRK_NONE: + ca0132_select_out(codec); + break; + } + + return 1; + } } if (nid == VNID_AMIC1_SEL) { @@ -3574,7 +4915,18 @@ static int ca0132_vnode_switch_set(struct snd_kcontrol *kcontrol, } if (nid == VNID_HP_ASEL) { - ca0132_select_out(codec); + + switch (spec->quirk) { + case QUIRK_SBZ: + ca0132_alt_select_out(codec); + break; + case QUIRK_R3DI: + ca0132_alt_select_out(codec); + break; + case QUIRK_NONE: + ca0132_select_out(codec); + break; + } return 1; } @@ -3603,10 +4955,236 @@ static int ca0132_vnode_switch_set(struct snd_kcontrol *kcontrol, } /* End of control change helpers. */ +/* + * Mic Boost Enum for alternative ca0132 codecs. I didn't like that the original + * only has off or full 30 dB, and didn't like making a volume slider that has + * traditional 0-100 in alsamixer that goes in big steps. I like enum better. + */ + +#define MIC_BOOST_NUM_OF_STEPS 4 +#define MIC_BOOST_ENUM_MAX_STRLEN 10 + +static int ca0132_alt_mic_boost_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + char *sfx = "dB"; + char namestr[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = MIC_BOOST_NUM_OF_STEPS; + if (uinfo->value.enumerated.item >= MIC_BOOST_NUM_OF_STEPS) + uinfo->value.enumerated.item = MIC_BOOST_NUM_OF_STEPS - 1; + sprintf(namestr, "%d %s", (uinfo->value.enumerated.item * 10), sfx); + strcpy(uinfo->value.enumerated.name, namestr); + return 0; +} + +static int ca0132_alt_mic_boost_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + + ucontrol->value.enumerated.item[0] = spec->mic_boost_enum_val; + return 0; +} + +static int ca0132_alt_mic_boost_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + int sel = ucontrol->value.enumerated.item[0]; + unsigned int items = MIC_BOOST_NUM_OF_STEPS; + + if (sel >= items) + return 0; + + codec_dbg(codec, "ca0132_alt_mic_boost: boost=%d\n", + sel); + + spec->mic_boost_enum_val = sel; + + ca0132_alt_mic_boost_set(codec, spec->mic_boost_enum_val); + + return 1; +} + + +/* + * Input Select Control for alternative ca0132 codecs. This exists because + * front microphone has no auto-detect, and we need a way to set the rear + * as line-in + */ + +static int ca0132_alt_input_source_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = IN_SRC_NUM_OF_INPUTS; + if (uinfo->value.enumerated.item >= IN_SRC_NUM_OF_INPUTS) + uinfo->value.enumerated.item = IN_SRC_NUM_OF_INPUTS - 1; + strcpy(uinfo->value.enumerated.name, + in_src_str[uinfo->value.enumerated.item]); + return 0; +} + +static int ca0132_alt_input_source_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + + ucontrol->value.enumerated.item[0] = spec->in_enum_val; + return 0; +} + +static int ca0132_alt_input_source_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + int sel = ucontrol->value.enumerated.item[0]; + unsigned int items = IN_SRC_NUM_OF_INPUTS; + + if (sel >= items) + return 0; + + codec_dbg(codec, "ca0132_alt_input_select: sel=%d, preset=%s\n", + sel, in_src_str[sel]); + + spec->in_enum_val = sel; + + ca0132_alt_select_in(codec); + + return 1; +} + +/* Sound Blaster Z Output Select Control */ + +static int ca0132_alt_output_select_get_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = SBZ_NUM_OF_OUTPUTS; + if (uinfo->value.enumerated.item >= SBZ_NUM_OF_OUTPUTS) + uinfo->value.enumerated.item = SBZ_NUM_OF_OUTPUTS - 1; + strcpy(uinfo->value.enumerated.name, + alt_out_presets[uinfo->value.enumerated.item].name); + return 0; +} + +static int ca0132_alt_output_select_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + + ucontrol->value.enumerated.item[0] = spec->out_enum_val; + return 0; +} + +static int ca0132_alt_output_select_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + int sel = ucontrol->value.enumerated.item[0]; + unsigned int items = SBZ_NUM_OF_OUTPUTS; + unsigned int auto_jack; + + if (sel >= items) + return 0; + + codec_dbg(codec, "ca0132_alt_output_select: sel=%d, preset=%s\n", + sel, alt_out_presets[sel].name); + + spec->out_enum_val = sel; + + auto_jack = spec->vnode_lswitch[VNID_HP_ASEL - VNODE_START_NID]; + + if (!auto_jack) + ca0132_alt_select_out(codec); + + return 1; +} + +/* + * Smart Volume output setting control. Three different settings, Normal, + * which takes the value from the smart volume slider. The two others, loud + * and night, disregard the slider value and have uneditable values. + */ + +static int ca0132_alt_svm_setting_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = NUM_OF_SVM_SETTINGS; + if (uinfo->value.enumerated.item >= NUM_OF_SVM_SETTINGS) + uinfo->value.enumerated.item = NUM_OF_SVM_SETTINGS - 1; + strcpy(uinfo->value.enumerated.name, + out_svm_set_enum_str[uinfo->value.enumerated.item]); + return 0; +} + +static int ca0132_alt_svm_setting_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + + ucontrol->value.enumerated.item[0] = spec->smart_volume_setting; + return 0; +} + +static int ca0132_alt_svm_setting_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + int sel = ucontrol->value.enumerated.item[0]; + unsigned int items = NUM_OF_SVM_SETTINGS; + unsigned int idx = SMART_VOLUME - EFFECT_START_NID; + unsigned int tmp; + + if (sel >= items) + return 0; + + codec_dbg(codec, "ca0132_alt_svm_setting: sel=%d, preset=%s\n", + sel, out_svm_set_enum_str[sel]); + + spec->smart_volume_setting = sel; + + switch (sel) { + case 0: + tmp = FLOAT_0; + break; + case 1: + tmp = FLOAT_1; + break; + case 2: + tmp = FLOAT_2; + break; + default: + tmp = FLOAT_0; + break; + } + /* Req 2 is the Smart Volume Setting req. */ + dspio_set_uint_param(codec, ca0132_effects[idx].mid, + ca0132_effects[idx].reqs[2], tmp); + + return 1; +} + static int ca0132_voicefx_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - unsigned int items = ARRAY_SIZE(ca0132_voicefx_presets); + unsigned int items = sizeof(ca0132_voicefx_presets) + / sizeof(struct ct_voicefx_preset); uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; @@ -3635,8 +5213,10 @@ static int ca0132_voicefx_put(struct snd_kcontrol *kcontrol, struct ca0132_spec *spec = codec->spec; int i, err = 0; int sel = ucontrol->value.enumerated.item[0]; + unsigned int items = sizeof(ca0132_voicefx_presets) + / sizeof(struct ct_voicefx_preset); - if (sel >= ARRAY_SIZE(ca0132_voicefx_presets)) + if (sel >= items) return 0; codec_dbg(codec, "ca0132_voicefx_put: sel=%d, preset=%s\n", @@ -3663,6 +5243,68 @@ static int ca0132_voicefx_put(struct snd_kcontrol *kcontrol, return 1; } +/* Sound Blaster Z EQ preset controls */ + +static int ca0132_alt_eq_preset_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + unsigned int items = sizeof(ca0132_alt_eq_presets) + / sizeof(struct ct_eq_preset); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = items; + if (uinfo->value.enumerated.item >= items) + uinfo->value.enumerated.item = items - 1; + strcpy(uinfo->value.enumerated.name, + ca0132_alt_eq_presets[uinfo->value.enumerated.item].name); + return 0; +} + +static int ca0132_alt_eq_preset_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + + ucontrol->value.enumerated.item[0] = spec->eq_preset_val; + return 0; +} + +static int ca0132_alt_eq_preset_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + int i, err = 0; + int sel = ucontrol->value.enumerated.item[0]; + unsigned int items = sizeof(ca0132_alt_eq_presets) + / sizeof(struct ct_eq_preset); + + if (sel >= items) + return 0; + + codec_dbg(codec, "ca0132_alt_eq_preset_put: sel=%d, preset=%s\n", + sel, ca0132_alt_eq_presets[sel].name); + + /* + * Idx 0 is default. + * Default needs to qualify with CrystalVoice state. + */ + for (i = 0; i < EQ_PRESET_MAX_PARAM_COUNT; i++) { + err = dspio_set_uint_param(codec, ca0132_alt_eq_enum.mid, + ca0132_alt_eq_enum.reqs[i], + ca0132_alt_eq_presets[sel].vals[i]); + if (err < 0) + break; + } + + if (err >= 0) + spec->eq_preset_val = sel; + + return 1; +} + static int ca0132_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -3714,6 +5356,7 @@ static int ca0132_switch_put(struct snd_kcontrol *kcontrol, nid, *valp); snd_hda_power_up(codec); + /* vnode */ if ((nid >= VNODE_START_NID) && (nid < VNODE_END_NID)) { if (ch & 1) { @@ -3755,8 +5398,13 @@ static int ca0132_switch_put(struct snd_kcontrol *kcontrol, spec->cur_mic_boost = *valp; /* Mic boost does not apply to Digital Mic */ - if (spec->cur_mic_type != DIGITAL_MIC) - changed = ca0132_mic_boost_set(codec, *valp); + if (spec->quirk == QUIRK_SBZ || spec->quirk == QUIRK_R3DI) { + if (spec->in_enum_val != REAR_LINE_IN) + changed = ca0132_mic_boost_set(codec, *valp); + } else { + if (spec->cur_mic_type != DIGITAL_MIC) + changed = ca0132_mic_boost_set(codec, *valp); + } goto exit; } @@ -3856,6 +5504,7 @@ static int ca0132_volume_put(struct snd_kcontrol *kcontrol, unsigned long pval; snd_hda_power_up(codec); + ca0132_alt_dsp_volume_put(codec, nid); mutex_lock(&codec->control_mutex); pval = kcontrol->private_value; kcontrol->private_value = HDA_COMPOSE_AMP_VAL(shared_nid, ch, @@ -3907,29 +5556,148 @@ static int ca0132_volume_tlv(struct snd_kcontrol *kcontrol, int op_flag, return err; } +/* + * Added the FX: prefix for the SBZ, because otherwise the surround effect + * takes over the actual surround volume switch. Plus, makes it more clear + * what the switches mean. Kept it only on the SBZ just in case it would + * disturb the Chromebook. + */ static int add_fx_switch(struct hda_codec *codec, hda_nid_t nid, const char *pfx, int dir) { + struct ca0132_spec *spec = codec->spec; + char *fx = "FX:"; char namestr[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; int type = dir ? HDA_INPUT : HDA_OUTPUT; struct snd_kcontrol_new knew = CA0132_CODEC_MUTE_MONO(namestr, nid, 1, type); - sprintf(namestr, "%s %s Switch", pfx, dirstr[dir]); + if ((spec->quirk != QUIRK_NONE) && (nid <= IN_EFFECT_END_NID)) + sprintf(namestr, "%s %s %s Switch", fx, pfx, dirstr[dir]); + else + sprintf(namestr, "%s %s Switch", pfx, dirstr[dir]); + return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec)); } -static int add_voicefx(struct hda_codec *codec) +static int add_voicefx(struct hda_codec *codec) +{ + struct snd_kcontrol_new knew = + HDA_CODEC_MUTE_MONO(ca0132_voicefx.name, + VOICEFX, 1, 0, HDA_INPUT); + knew.info = ca0132_voicefx_info; + knew.get = ca0132_voicefx_get; + knew.put = ca0132_voicefx_put; + return snd_hda_ctl_add(codec, VOICEFX, snd_ctl_new1(&knew, codec)); +} + +/* Create the Sound Blaster Z EQ Preset control */ + +static int add_ca0132_alt_eq_presets(struct hda_codec *codec) +{ + struct snd_kcontrol_new knew = + HDA_CODEC_MUTE_MONO(ca0132_alt_eq_enum.name, + EQ_PRESET_ENUM, 1, 0, HDA_OUTPUT); + knew.info = ca0132_alt_eq_preset_info; + knew.get = ca0132_alt_eq_preset_get; + knew.put = ca0132_alt_eq_preset_put; + return snd_hda_ctl_add(codec, EQ_PRESET_ENUM, + snd_ctl_new1(&knew, codec)); +} + +/* + * Create an Input Source enumerated control for the alternate ca0132 codecs + * because the front microphone has no auto-detect, and Line-in has to be set + * somehow. + */ + +static int ca0132_alt_add_input_enum(struct hda_codec *codec) +{ + struct snd_kcontrol_new knew = + HDA_CODEC_MUTE_MONO("Input Source", + INPUT_SOURCE_ENUM, 1, 0, HDA_INPUT); + knew.info = ca0132_alt_input_source_info; + knew.get = ca0132_alt_input_source_get; + knew.put = ca0132_alt_input_source_put; + return snd_hda_ctl_add(codec, INPUT_SOURCE_ENUM, + snd_ctl_new1(&knew, codec)); +} + +/* + * Create an Output Select enumerated control for codecs with surround + * out capabilities. + */ + +static int ca0132_alt_add_output_enum(struct hda_codec *codec) +{ + struct snd_kcontrol_new knew = + HDA_CODEC_MUTE_MONO("Output Select", + OUTPUT_SOURCE_ENUM, 1, 0, HDA_OUTPUT); + knew.info = ca0132_alt_output_select_get_info; + knew.get = ca0132_alt_output_select_get; + knew.put = ca0132_alt_output_select_put; + return snd_hda_ctl_add(codec, OUTPUT_SOURCE_ENUM, + snd_ctl_new1(&knew, codec)); +} + +/* + * Add enumerated control for the three different settings of the smart volume + * output effect. Normal just uses the slider value, and loud and night are + * their own things that ignore that value. + */ + +static int ca0132_alt_add_svm_enum(struct hda_codec *codec) +{ + struct snd_kcontrol_new knew = + HDA_CODEC_MUTE_MONO("FX: Smart Volume Setting", + SMART_VOLUME_ENUM, 1, 0, HDA_OUTPUT); + knew.info = ca0132_alt_svm_setting_info; + knew.get = ca0132_alt_svm_setting_get; + knew.put = ca0132_alt_svm_setting_put; + return snd_hda_ctl_add(codec, SMART_VOLUME_ENUM, + snd_ctl_new1(&knew, codec)); + +} + +static int ca0132_alt_add_mic_boost_enum(struct hda_codec *codec) { struct snd_kcontrol_new knew = - HDA_CODEC_MUTE_MONO(ca0132_voicefx.name, - VOICEFX, 1, 0, HDA_INPUT); - knew.info = ca0132_voicefx_info; - knew.get = ca0132_voicefx_get; - knew.put = ca0132_voicefx_put; - return snd_hda_ctl_add(codec, VOICEFX, snd_ctl_new1(&knew, codec)); + HDA_CODEC_MUTE_MONO("Mic Boost Capture Switch", + MIC_BOOST_ENUM, 1, 0, HDA_INPUT); + knew.info = ca0132_alt_mic_boost_info; + knew.get = ca0132_alt_mic_boost_get; + knew.put = ca0132_alt_mic_boost_put; + return snd_hda_ctl_add(codec, MIC_BOOST_ENUM, + snd_ctl_new1(&knew, codec)); + } /* + * Need to create slave controls for the alternate codecs that have surround + * capabilities. + */ +static const char * const ca0132_alt_slave_pfxs[] = { + "Front", "Surround", "Center", "LFE", NULL, +}; + +/* + * Also need special channel map, because the default one is incorrect. + * I think this has to do with the pin for rear surround being 0x11, + * and the center/lfe being 0x10. Usually the pin order is the opposite. + */ +const struct snd_pcm_chmap_elem ca0132_alt_chmaps[] = { + { .channels = 2, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR } }, + { .channels = 4, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_RL, SNDRV_CHMAP_RR } }, + { .channels = 6, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_FC, SNDRV_CHMAP_LFE, + SNDRV_CHMAP_RL, SNDRV_CHMAP_RR } }, + { } +}; + +/* * When changing Node IDs for Mixer Controls below, make sure to update * Node IDs in ca0132_config() as well. */ @@ -3955,10 +5723,38 @@ static struct snd_kcontrol_new ca0132_mixer[] = { { } /* end */ }; +/* + * Separate mixer for alternate codecs,, includes controls for the surround + * nodes, and removes unused things, like Analog-Mic2 and Amic auto-detect. + * Leaving the HDA_CODEC_VOLUME and MUTE incase you want to remove the DSP + * volume settings. They work pretty well for now, only issue is a TLV + * mismatch that I can't figure out. + */ +static struct snd_kcontrol_new ca0132_alt_mixer[] = { +// HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0, HDA_OUTPUT), +// HDA_CODEC_MUTE("Front Playback Switch", 0x02, 0, HDA_OUTPUT), + CA0132_CODEC_VOL("Front Playback Volume", VNID_SPK, HDA_OUTPUT), + CA0132_CODEC_MUTE("Front Playback Switch", VNID_SPK, HDA_OUTPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x04, 0, HDA_OUTPUT), + HDA_CODEC_MUTE("Surround Playback Switch", 0x04, 0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x03, 1, 0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x03, 1, 0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x03, 2, 0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x03, 2, 0, HDA_OUTPUT), + CA0132_CODEC_VOL("Capture Volume", VNID_MIC, HDA_INPUT), + CA0132_CODEC_MUTE("Capture Switch", VNID_MIC, HDA_INPUT), + HDA_CODEC_VOLUME("What U Hear Capture Volume", 0x0a, 0, HDA_INPUT), + HDA_CODEC_MUTE("What U Hear Capture Switch", 0x0a, 0, HDA_INPUT), + CA0132_CODEC_MUTE_MONO("HP/Speaker Auto Detect Playback Switch", + VNID_HP_ASEL, 1, HDA_OUTPUT), + { } /* end */ +}; + static int ca0132_build_controls(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; - int i, num_fx; + struct hda_pcm *pcm; + int i, num_fx, num_sliders; int err = 0; /* Add Mixer controls */ @@ -3967,8 +5763,22 @@ static int ca0132_build_controls(struct hda_codec *codec) if (err < 0) return err; } + /* Setup vmaster with surround slaves for desktop ca0132 devices */ + if (spec->quirk != QUIRK_NONE) { + snd_hda_set_vmaster_tlv(codec, spec->dacs[0], HDA_OUTPUT, + spec->tlv); + snd_hda_add_vmaster(codec, "Master Playback Volume", + spec->tlv, ca0132_alt_slave_pfxs, + "Playback Volume"); + err = __snd_hda_add_vmaster(codec, "Master Playback Switch", + NULL, ca0132_alt_slave_pfxs, + "Playback Switch", + true, &spec->vmaster_mute.sw_kctl); + + } - /* Add in and out effects controls. + /* + * Add in and out effects controls. * VoiceFX, PE and CrystalVoice are added separately. */ num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT; @@ -3979,21 +5789,63 @@ static int ca0132_build_controls(struct hda_codec *codec) if (err < 0) return err; } + /* + * Change the names for the non-chromebook versions to make it clearer + * what these buttons actually do. Left them alone on the chromebook + * just in case. Also, add output select enum. + */ + if (spec->quirk != QUIRK_NONE) { + ca0132_alt_add_svm_enum(codec); + ca0132_alt_add_output_enum(codec); + add_ca0132_alt_eq_presets(codec); + err = add_fx_switch(codec, PLAY_ENHANCEMENT, + "Enable OutFX", 0); + if (err < 0) + return err; - err = add_fx_switch(codec, PLAY_ENHANCEMENT, "PlayEnhancement", 0); - if (err < 0) - return err; + err = add_fx_switch(codec, CRYSTAL_VOICE, + "Enable InFX", 1); + if (err < 0) + return err; + } else { + err = add_fx_switch(codec, PLAY_ENHANCEMENT, + "PlayEnhancement", 0); + if (err < 0) + return err; - err = add_fx_switch(codec, CRYSTAL_VOICE, "CrystalVoice", 1); - if (err < 0) - return err; + err = add_fx_switch(codec, CRYSTAL_VOICE, + "CrystalVoice", 1); + if (err < 0) + return err; + } add_voicefx(codec); + /* Add the input select enum, and all of the effect sliders. */ + if (spec->quirk != QUIRK_NONE) { + ca0132_alt_add_mic_boost_enum(codec); + ca0132_alt_add_input_enum(codec); + + num_sliders = OUT_EFFECTS_COUNT - 1; + for (i = 0; i < num_sliders; i++) { + err = ca0132_alt_add_effect_slider(codec, + ca0132_effects[i].nid, + ca0132_effects[i].name, + ca0132_effects[i].direct); + if (err < 0) + return err; + } + + err = ca0132_alt_add_effect_slider(codec, XBASS_XOVER, + "X-Bass Crossover", EFX_DIR_OUT); + if (err < 0) + return err; + } #ifdef ENABLE_TUNING_CONTROLS add_tuning_ctls(codec); #endif + err = snd_hda_jack_add_kctls(codec, &spec->autocfg); if (err < 0) return err; @@ -4014,6 +5866,27 @@ static int ca0132_build_controls(struct hda_codec *codec) if (err < 0) return err; } + + /* + * if there is a possibility of having 6 channels, set up the proper + * chmap. The SBZ has a funky setup that needs FR,FL,FC,LFE,RR,RL + * instead of the more common FR,FL,RR,RL,FC,LFE. Not sure why. + */ + list_for_each_entry(pcm, &codec->pcm_list_head, list) { + struct hda_pcm_stream *hinfo = + &pcm->stream[SNDRV_PCM_STREAM_PLAYBACK]; + struct snd_pcm_chmap *chmap; + const struct snd_pcm_chmap_elem *elem; + + elem = ca0132_alt_chmaps; + if (hinfo->channels_max == 6) { + err = snd_pcm_add_chmap_ctls(pcm->pcm, + SNDRV_PCM_STREAM_PLAYBACK, + elem, hinfo->channels_max, 0, &chmap); + if (err < 0) + codec_dbg(codec, "snd_pcm_add_chmap_ctls failed!"); + } + } return 0; } @@ -4065,23 +5938,30 @@ static int ca0132_build_pcms(struct hda_codec *codec) struct ca0132_spec *spec = codec->spec; struct hda_pcm *info; + info = snd_hda_codec_pcm_new(codec, "CA0132 Analog"); if (!info) return -ENOMEM; + info->own_chmap = true; + info->stream[SNDRV_PCM_STREAM_PLAYBACK].chmap = ca0132_alt_chmaps; info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ca0132_pcm_analog_playback; info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->dacs[0]; info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = - spec->multiout.max_channels; + spec->multiout.max_channels; info->stream[SNDRV_PCM_STREAM_CAPTURE] = ca0132_pcm_analog_capture; info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = 1; info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[0]; - info = snd_hda_codec_pcm_new(codec, "CA0132 Analog Mic-In2"); - if (!info) - return -ENOMEM; - info->stream[SNDRV_PCM_STREAM_CAPTURE] = ca0132_pcm_analog_capture; - info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = 1; - info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[1]; + /* SBZ doesn't use this ADC. */ + if (spec->quirk == QUIRK_NONE) { + info = snd_hda_codec_pcm_new(codec, "CA0132 Analog Mic-In2"); + if (!info) + return -ENOMEM; + info->stream[SNDRV_PCM_STREAM_CAPTURE] = + ca0132_pcm_analog_capture; + info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = 1; + info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[1]; + } info = snd_hda_codec_pcm_new(codec, "CA0132 What U Hear"); if (!info) @@ -4113,6 +5993,7 @@ static int ca0132_build_pcms(struct hda_codec *codec) static void init_output(struct hda_codec *codec, hda_nid_t pin, hda_nid_t dac) { + if (pin) { snd_hda_set_pin_ctl(codec, pin, PIN_HP); if (get_wcaps(codec, pin) & AC_WCAP_OUT_AMP) @@ -4171,7 +6052,7 @@ static void ca0132_set_dmic(struct hda_codec *codec, int enable) ca0132_set_vipsource(codec, 0); if (enable) { /* set DMic input as 2-ch */ - tmp = FLOAT_TWO; + tmp = FLOAT_2; dspio_set_uint_param(codec, 0x80, 0x00, tmp); val = spec->dmic_ctl; @@ -4183,7 +6064,7 @@ static void ca0132_set_dmic(struct hda_codec *codec, int enable) chipio_set_control_flag(codec, CONTROL_FLAG_DMIC, 1); } else { /* set AMic input as mono */ - tmp = FLOAT_ONE; + tmp = FLOAT_1; dspio_set_uint_param(codec, 0x80, 0x00, tmp); val = spec->dmic_ctl; @@ -4213,7 +6094,7 @@ static void ca0132_init_dmic(struct hda_codec *codec) /* MCLK uses MPIO1, set to enable. * Bit 2-0: MPIO select - * Bit 3: set to disable + * Bit 3: set to disable * Bit 7-4: reserved */ val = 0x01; @@ -4222,9 +6103,9 @@ static void ca0132_init_dmic(struct hda_codec *codec) /* Data1 uses MPIO3. Data2 not use * Bit 2-0: Data1 MPIO select - * Bit 3: set disable Data1 + * Bit 3: set disable Data1 * Bit 6-4: Data2 MPIO select - * Bit 7: set disable Data2 + * Bit 7: set disable Data2 */ val = 0x83; snd_hda_codec_write(codec, spec->input_pins[0], 0, @@ -4232,10 +6113,10 @@ static void ca0132_init_dmic(struct hda_codec *codec) /* Use Ch-0 and Ch-1. Rate is 48K, mode 1. Disable DMic first. * Bit 3-0: Channel mask - * Bit 4: set for 48KHz, clear for 32KHz - * Bit 5: mode - * Bit 6: set to select Data2, clear for Data1 - * Bit 7: set to enable DMic, clear for AMic + * Bit 4: set for 48KHz, clear for 32KHz + * Bit 5: mode + * Bit 6: set to select Data2, clear for Data1 + * Bit 7: set to enable DMic, clear for AMic */ val = 0x23; /* keep a copy of dmic ctl val for enable/disable dmic purpuse */ @@ -4288,6 +6169,219 @@ static void ca0132_refresh_widget_caps(struct hda_codec *codec) } /* + * Recon3Di r3di_setup_defaults sub functions. + */ + +static void r3di_dsp_scp_startup(struct hda_codec *codec) +{ + unsigned int tmp; + + tmp = 0x00000000; + dspio_set_uint_param_no_source(codec, 0x80, 0x0A, tmp); + + tmp = 0x00000001; + dspio_set_uint_param_no_source(codec, 0x80, 0x0B, tmp); + + tmp = 0x00000004; + dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); + + tmp = 0x00000005; + dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); + + tmp = 0x00000000; + dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); + +} + +static void r3di_dsp_initial_mic_setup(struct hda_codec *codec) +{ + unsigned int tmp; + + /* Mic 1 Setup */ + chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000); + chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000); + /* This ConnPointID is unique to Recon3Di. Haven't seen it elsewhere */ + chipio_set_conn_rate(codec, 0x0F, SR_96_000); + tmp = FLOAT_1; + dspio_set_uint_param(codec, 0x80, 0x00, tmp); + + /* Mic 2 Setup, even though it isn't connected on SBZ */ + chipio_set_conn_rate(codec, MEM_CONNID_MICIN2, SR_96_000); + chipio_set_conn_rate(codec, MEM_CONNID_MICOUT2, SR_96_000); + chipio_set_conn_rate(codec, 0x0F, SR_96_000); + tmp = FLOAT_0; + dspio_set_uint_param(codec, 0x80, 0x01, tmp); +} + +/* + * Initialize Sound Blaster Z analog microphones. + */ +static void sbz_init_analog_mics(struct hda_codec *codec) +{ + unsigned int tmp; + + /* Mic 1 Setup */ + chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000); + chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000); + tmp = FLOAT_3; + dspio_set_uint_param(codec, 0x80, 0x00, tmp); + + /* Mic 2 Setup, even though it isn't connected on SBZ */ + chipio_set_conn_rate(codec, MEM_CONNID_MICIN2, SR_96_000); + chipio_set_conn_rate(codec, MEM_CONNID_MICOUT2, SR_96_000); + tmp = FLOAT_0; + dspio_set_uint_param(codec, 0x80, 0x01, tmp); + +} + +/* + * Sets the source of stream 0x14 to connpointID 0x48, and the destination + * connpointID to 0x91. If this isn't done, the destination is 0x71, and + * you get no sound. I'm guessing this has to do with the Sound Blaster Z + * having an updated DAC, which changes the destination to that DAC. + */ +static void sbz_connect_streams(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + + mutex_lock(&spec->chipio_mutex); + + codec_dbg(codec, "Connect Streams entered, mutex locked and loaded.\n"); + + chipio_set_stream_channels(codec, 0x0C, 6); + chipio_set_stream_control(codec, 0x0C, 1); + + /* This value is 0x43 for 96khz, and 0x83 for 192khz. */ + chipio_write_no_mutex(codec, 0x18a020, 0x00000043); + + /* Setup stream 0x14 with it's source and destination points */ + chipio_set_stream_source_dest(codec, 0x14, 0x48, 0x91); + chipio_set_conn_rate_no_mutex(codec, 0x48, SR_96_000); + chipio_set_conn_rate_no_mutex(codec, 0x91, SR_96_000); + chipio_set_stream_channels(codec, 0x14, 2); + chipio_set_stream_control(codec, 0x14, 1); + + codec_dbg(codec, "Connect Streams exited, mutex released.\n"); + + mutex_unlock(&spec->chipio_mutex); + +} + +/* + * Write data through ChipIO to setup proper stream destinations. + * Not sure how it exactly works, but it seems to direct data + * to different destinations. Example is f8 to c0, e0 to c0. + * All I know is, if you don't set these, you get no sound. + */ +static void sbz_chipio_startup_data(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + + mutex_lock(&spec->chipio_mutex); + + codec_dbg(codec, "Startup Data entered, mutex locked and loaded.\n"); + + /* These control audio output */ + chipio_write_no_mutex(codec, 0x190060, 0x0001f8c0); + chipio_write_no_mutex(codec, 0x190064, 0x0001f9c1); + chipio_write_no_mutex(codec, 0x190068, 0x0001fac6); + chipio_write_no_mutex(codec, 0x19006c, 0x0001fbc7); + /* Signal to update I think */ + chipio_write_no_mutex(codec, 0x19042c, 0x00000001); + + chipio_set_stream_channels(codec, 0x0C, 6); + chipio_set_stream_control(codec, 0x0C, 1); + /* No clue what these control */ + chipio_write_no_mutex(codec, 0x190030, 0x0001e0c0); + chipio_write_no_mutex(codec, 0x190034, 0x0001e1c1); + chipio_write_no_mutex(codec, 0x190038, 0x0001e4c2); + chipio_write_no_mutex(codec, 0x19003c, 0x0001e5c3); + chipio_write_no_mutex(codec, 0x190040, 0x0001e2c4); + chipio_write_no_mutex(codec, 0x190044, 0x0001e3c5); + chipio_write_no_mutex(codec, 0x190048, 0x0001e8c6); + chipio_write_no_mutex(codec, 0x19004c, 0x0001e9c7); + chipio_write_no_mutex(codec, 0x190050, 0x0001ecc8); + chipio_write_no_mutex(codec, 0x190054, 0x0001edc9); + chipio_write_no_mutex(codec, 0x190058, 0x0001eaca); + chipio_write_no_mutex(codec, 0x19005c, 0x0001ebcb); + + chipio_write_no_mutex(codec, 0x19042c, 0x00000001); + + codec_dbg(codec, "Startup Data exited, mutex released.\n"); + + mutex_unlock(&spec->chipio_mutex); +} +/* + * Sound Blaster Z uses these after DSP is loaded. Weird SCP commands + * without a 0x20 source like normal. + */ +static void sbz_dsp_scp_startup(struct hda_codec *codec) +{ + unsigned int tmp; + + tmp = 0x00000003; + dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); + + tmp = 0x00000000; + dspio_set_uint_param_no_source(codec, 0x80, 0x0A, tmp); + + tmp = 0x00000001; + dspio_set_uint_param_no_source(codec, 0x80, 0x0B, tmp); + + tmp = 0x00000004; + dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); + + tmp = 0x00000005; + dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); + + tmp = 0x00000000; + dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); + +} + +static void sbz_dsp_initial_mic_setup(struct hda_codec *codec) +{ + unsigned int tmp; + + chipio_set_stream_control(codec, 0x03, 0); + chipio_set_stream_control(codec, 0x04, 0); + + chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000); + chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000); + + tmp = FLOAT_3; + dspio_set_uint_param(codec, 0x80, 0x00, tmp); + + chipio_set_stream_control(codec, 0x03, 1); + chipio_set_stream_control(codec, 0x04, 1); + + chipio_write(codec, 0x18b098, 0x0000000c); + chipio_write(codec, 0x18b09C, 0x0000000c); +} + +/* + * This is for the extra volume verbs 0x797 (left) and 0x798 (right). These add + * extra precision for decibel values. If you had the dB value in floating point + * you would take the value after the decimal point, multiply by 64, and divide + * by 2. So for 8.59, it's (59 * 64) / 100. Useful if someone wanted to + * implement fixed point or floating point dB volumes. For now, I'll set them + * to 0 just incase a value has lingered from a boot into Windows. + */ + +static void ca0132_alt_vol_setup(struct hda_codec *codec) +{ + + snd_hda_codec_write(codec, 0x02, 0, 0x797, 0x00); + snd_hda_codec_write(codec, 0x02, 0, 0x798, 0x00); + snd_hda_codec_write(codec, 0x03, 0, 0x797, 0x00); + snd_hda_codec_write(codec, 0x03, 0, 0x798, 0x00); + snd_hda_codec_write(codec, 0x04, 0, 0x797, 0x00); + snd_hda_codec_write(codec, 0x04, 0, 0x798, 0x00); + snd_hda_codec_write(codec, 0x07, 0, 0x797, 0x00); + snd_hda_codec_write(codec, 0x07, 0, 0x798, 0x00); +} + +/* * Setup default parameters for DSP */ static void ca0132_setup_defaults(struct hda_codec *codec) @@ -4311,24 +6405,138 @@ static void ca0132_setup_defaults(struct hda_codec *codec) } /*remove DSP headroom*/ - tmp = FLOAT_ZERO; + tmp = FLOAT_0; dspio_set_uint_param(codec, 0x96, 0x3C, tmp); /*set speaker EQ bypass attenuation*/ dspio_set_uint_param(codec, 0x8f, 0x01, tmp); /* set AMic1 and AMic2 as mono mic */ - tmp = FLOAT_ONE; + tmp = FLOAT_1; dspio_set_uint_param(codec, 0x80, 0x00, tmp); dspio_set_uint_param(codec, 0x80, 0x01, tmp); /* set AMic1 as CrystalVoice input */ - tmp = FLOAT_ONE; + tmp = FLOAT_1; dspio_set_uint_param(codec, 0x80, 0x05, tmp); /* set WUH source */ - tmp = FLOAT_TWO; + tmp = FLOAT_2; + dspio_set_uint_param(codec, 0x31, 0x00, tmp); + +} + +/* + * Setup default parameters for Recon3Di DSP. + */ + +static void r3di_setup_defaults(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + unsigned int tmp; + int num_fx; + int idx, i; + + if (spec->dsp_state != DSP_DOWNLOADED) + return; + + r3di_dsp_scp_startup(codec); + + r3di_dsp_initial_mic_setup(codec); + + /*remove DSP headroom*/ + tmp = FLOAT_0; + dspio_set_uint_param(codec, 0x96, 0x3C, tmp); + + /* set WUH source */ + tmp = FLOAT_2; + dspio_set_uint_param(codec, 0x31, 0x00, tmp); + + /* Set speaker source? */ + dspio_set_uint_param(codec, 0x32, 0x00, tmp); + + r3di_gpio_dsp_status_set(codec, R3DI_DSP_DOWNLOADED); + + /* Setup effect defaults */ + num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT + 1; + for (idx = 0; idx < num_fx; idx++) { + for (i = 0; i <= ca0132_effects[idx].params; i++) { + dspio_set_uint_param(codec, ca0132_effects[idx].mid, + ca0132_effects[idx].reqs[i], + ca0132_effects[idx].def_vals[i]); + } + } + +} + +/* + * Setup default parameters for the Sound Blaster Z DSP. A lot more going on + * than the Chromebook setup. + */ +static void sbz_setup_defaults(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + unsigned int tmp, stream_format; + int num_fx; + int idx, i; + + if (spec->dsp_state != DSP_DOWNLOADED) + return; + + sbz_dsp_scp_startup(codec); + sbz_init_analog_mics(codec); + sbz_connect_streams(codec); + sbz_chipio_startup_data(codec); + + chipio_set_stream_control(codec, 0x03, 1); + chipio_set_stream_control(codec, 0x04, 1); + + /* + * Sets internal input loopback to off, used to have a switch to + * enable input loopback, but turned out to be way too buggy. + */ + tmp = FLOAT_1; + dspio_set_uint_param(codec, 0x37, 0x08, tmp); + dspio_set_uint_param(codec, 0x37, 0x10, tmp); + + /*remove DSP headroom*/ + tmp = FLOAT_0; + dspio_set_uint_param(codec, 0x96, 0x3C, tmp); + + /* set WUH source */ + tmp = FLOAT_2; dspio_set_uint_param(codec, 0x31, 0x00, tmp); + + /* Set speaker source? */ + dspio_set_uint_param(codec, 0x32, 0x00, tmp); + + sbz_dsp_initial_mic_setup(codec); + + + /* out, in effects + voicefx */ + num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT + 1; + for (idx = 0; idx < num_fx; idx++) { + for (i = 0; i <= ca0132_effects[idx].params; i++) { + dspio_set_uint_param(codec, ca0132_effects[idx].mid, + ca0132_effects[idx].reqs[i], + ca0132_effects[idx].def_vals[i]); + } + } + + /* + * Have to make a stream to bind the sound output to, otherwise + * you'll get dead audio. Before I did this, it would bind to an + * audio input, and would never work + */ + stream_format = snd_hdac_calc_stream_format(48000, 2, + SNDRV_PCM_FORMAT_S32_LE, 32, 0); + + snd_hda_codec_setup_stream(codec, spec->dacs[0], spec->dsp_stream_id, + 0, stream_format); + + snd_hda_codec_cleanup_stream(codec, spec->dacs[0]); + + return; } /* @@ -4336,12 +6544,33 @@ static void ca0132_setup_defaults(struct hda_codec *codec) */ static void ca0132_init_flags(struct hda_codec *codec) { - chipio_set_control_flag(codec, CONTROL_FLAG_IDLE_ENABLE, 0); - chipio_set_control_flag(codec, CONTROL_FLAG_PORT_A_COMMON_MODE, 0); - chipio_set_control_flag(codec, CONTROL_FLAG_PORT_D_COMMON_MODE, 0); - chipio_set_control_flag(codec, CONTROL_FLAG_PORT_A_10KOHM_LOAD, 0); - chipio_set_control_flag(codec, CONTROL_FLAG_PORT_D_10KOHM_LOAD, 0); - chipio_set_control_flag(codec, CONTROL_FLAG_ADC_C_HIGH_PASS, 1); + struct ca0132_spec *spec = codec->spec; + + if (spec->quirk == QUIRK_SBZ || spec->quirk == QUIRK_R3DI) { + chipio_set_control_flag(codec, CONTROL_FLAG_DSP_96KHZ, 1); + chipio_set_control_flag(codec, CONTROL_FLAG_DAC_96KHZ, 1); + chipio_set_control_flag(codec, CONTROL_FLAG_ADC_B_96KHZ, 1); + chipio_set_control_flag(codec, CONTROL_FLAG_ADC_C_96KHZ, 1); + chipio_set_control_flag(codec, CONTROL_FLAG_SRC_RATE_96KHZ, 1); + chipio_set_control_flag(codec, CONTROL_FLAG_IDLE_ENABLE, 0); + chipio_set_control_flag(codec, CONTROL_FLAG_SPDIF2OUT, 0); + chipio_set_control_flag(codec, + CONTROL_FLAG_PORT_D_10KOHM_LOAD, 0); + chipio_set_control_flag(codec, + CONTROL_FLAG_PORT_A_10KOHM_LOAD, 1); + } else { + chipio_set_control_flag(codec, CONTROL_FLAG_IDLE_ENABLE, 0); + chipio_set_control_flag(codec, + CONTROL_FLAG_PORT_A_COMMON_MODE, 0); + chipio_set_control_flag(codec, + CONTROL_FLAG_PORT_D_COMMON_MODE, 0); + chipio_set_control_flag(codec, + CONTROL_FLAG_PORT_A_10KOHM_LOAD, 0); + chipio_set_control_flag(codec, + CONTROL_FLAG_PORT_D_10KOHM_LOAD, 0); + chipio_set_control_flag(codec, + CONTROL_FLAG_ADC_C_HIGH_PASS, 1); + } } /* @@ -4349,6 +6578,17 @@ static void ca0132_init_flags(struct hda_codec *codec) */ static void ca0132_init_params(struct hda_codec *codec) { + struct ca0132_spec *spec = codec->spec; + + if (spec->quirk == QUIRK_SBZ || spec->quirk == QUIRK_R3DI) { + chipio_set_conn_rate(codec, MEM_CONNID_WUH, SR_48_000); + chipio_set_conn_rate(codec, 0x0B, SR_48_000); + chipio_set_control_param(codec, CONTROL_PARAM_SPDIF1_SOURCE, 0); + /* unsure of what param 0 is */ + chipio_set_control_param(codec, 0, 0); + chipio_set_control_param(codec, CONTROL_PARAM_VIP_SOURCE, 0); + } + chipio_set_control_param(codec, CONTROL_PARAM_PORTA_160OHM_GAIN, 6); chipio_set_control_param(codec, CONTROL_PARAM_PORTD_160OHM_GAIN, 6); } @@ -4367,14 +6607,54 @@ static void ca0132_set_dsp_msr(struct hda_codec *codec, bool is96k) chipio_set_conn_rate(codec, MEM_CONNID_WUH, SR_48_000); } +/* + * This function selects the firmware and begins the download. I have added + * the ability to load alternate firmware for different variants. Both the + * Recon3Di and the Sound Blaster Z have been tested and confirmed working + * with the default firmware as well, but I want to use the firmware from the + * Windows driver for peace of mind. This may not be the appropriate thing, + * and I would understand it's removal. + */ static bool ca0132_download_dsp_images(struct hda_codec *codec) { bool dsp_loaded = false; + struct ca0132_spec *spec = codec->spec; const struct dsp_image_seg *dsp_os_image; const struct firmware *fw_entry; + switch (spec->quirk) { + case QUIRK_SBZ: + if (request_firmware(&fw_entry, SBZ_EFX_FILE, + codec->card->dev) != 0) { + codec_dbg(codec, "SBZ alt firmware not detected. "); + spec->alt_firmware_present = false; + } else { + codec_dbg(codec, "Sound Blaster Z firmware selected."); + spec->alt_firmware_present = true; + } + break; + case QUIRK_R3DI: + if (request_firmware(&fw_entry, R3DI_EFX_FILE, + codec->card->dev) != 0) { + codec_dbg(codec, "Recon3Di alt firmware not detected."); + spec->alt_firmware_present = false; + } else { + codec_dbg(codec, "Recon3Di firmware selected."); + spec->alt_firmware_present = true; + } + break; + default: + spec->alt_firmware_present = false; + break; + } + /* Use default ctefx.bin if no alt firmware is detected */ + if (spec->alt_firmware_present == false) { + codec_dbg(codec, "Default firmware selected."); + if (request_firmware(&fw_entry, EFX_FILE, + codec->card->dev) != 0) + return false; + } - if (request_firmware(&fw_entry, EFX_FILE, codec->card->dev) != 0) - return false; + codec_dbg(codec, "Inside ca0132_download_dsp_images"); dsp_os_image = (struct dsp_image_seg *)(fw_entry->data); if (dspload_image(codec, dsp_os_image, 0, 0, true, 0)) { @@ -4402,13 +6682,17 @@ static void ca0132_download_dsp(struct hda_codec *codec) return; /* don't retry failures */ chipio_enable_clocks(codec); - spec->dsp_state = DSP_DOWNLOADING; - if (!ca0132_download_dsp_images(codec)) - spec->dsp_state = DSP_DOWNLOAD_FAILED; - else - spec->dsp_state = DSP_DOWNLOADED; + if (spec->dsp_state != DSP_DOWNLOADED) { + spec->dsp_state = DSP_DOWNLOADING; - if (spec->dsp_state == DSP_DOWNLOADED) + if (!ca0132_download_dsp_images(codec)) + spec->dsp_state = DSP_DOWNLOAD_FAILED; + else + spec->dsp_state = DSP_DOWNLOADED; + } + + /* Don't do this on SBZ, it's already done earlier. */ + if (spec->dsp_state == DSP_DOWNLOADED && spec->quirk == QUIRK_NONE) ca0132_set_dsp_msr(codec, true); } @@ -4431,7 +6715,8 @@ static void hp_callback(struct hda_codec *codec, struct hda_jack_callback *cb) struct ca0132_spec *spec = codec->spec; struct hda_jack_tbl *tbl; - /* Delay enabling the HP amp, to let the mic-detection + /* + * Delay enabling the HP amp, to let the mic-detection * state machine run. */ cancel_delayed_work_sync(&spec->unsol_hp_work); @@ -4454,6 +6739,59 @@ static void ca0132_init_unsol(struct hda_codec *codec) amic_callback); snd_hda_jack_detect_enable_callback(codec, UNSOL_TAG_DSP, ca0132_process_dsp_response); + /* Front headphone jack detection */ + if (spec->quirk == QUIRK_SBZ || spec->quirk == QUIRK_R3DI) + snd_hda_jack_detect_enable_callback(codec, + spec->unsol_tag_front_hp, hp_callback); +} + +/* + * Sound Blaster Z exit specific commands. + */ +static void sbz_region2_exit(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + unsigned int i; + + for (i = 0; i < 4; i++) + writeb(0x0, spec->mem_base + 0x100); + + for (i = 0; i < 8; i++) + writeb(0xb3, spec->mem_base + 0x304); + + writew(0x0000, spec->mem_base + 0x320); + writew(0x0001, spec->mem_base + 0x320); + writew(0x0104, spec->mem_base + 0x320); + writew(0x0005, spec->mem_base + 0x320); + writew(0x0007, spec->mem_base + 0x320); +} + +static void sbz_set_pin_ctl_default(struct hda_codec *codec) +{ + hda_nid_t pins[5] = {0x0B, 0x0C, 0x0E, 0x12, 0x13}; + unsigned int i; + + snd_hda_codec_write(codec, 0x11, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40); + + for (i = 0; i < 5; i++) { + snd_hda_codec_write(codec, pins[i], 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00); + } + +} + +static void sbz_clear_unsolicited(struct hda_codec *codec) +{ + hda_nid_t pins[7] = {0x0B, 0x0E, 0x0F, 0x10, 0x11, 0x12, 0x13}; + unsigned int i; + + for (i = 0; i < 7; i++) { + snd_hda_codec_write(codec, pins[i], 0, + AC_VERB_SET_UNSOLICITED_ENABLE, 0x00); + } + + return; } /* @@ -4476,8 +6814,9 @@ static struct hda_verb ca0132_base_exit_verbs[] = { {} }; -/* Other verbs tables. Sends after DSP download. */ -static struct hda_verb ca0132_init_verbs0[] = { +/* Other verbs tables. Sends after DSP download. */ + +static struct hda_verb ca0132_init_verbs[] = { /* chip init verbs */ {0x15, 0x70D, 0xF0}, {0x15, 0x70E, 0xFE}, @@ -4506,8 +6845,27 @@ static struct hda_verb ca0132_init_verbs0[] = { {0x15, 0x546, 0xC9}, {0x15, 0x53B, 0xCE}, {0x15, 0x5E8, 0xC9}, - {0x15, 0x717, 0x0D}, - {0x15, 0x718, 0x20}, + {} +}; + +/* Extra init verbs for SBZ */ +static struct hda_verb sbz_init_verbs[] = { + {0x15, 0x70D, 0x20}, + {0x15, 0x70E, 0x19}, + {0x15, 0x707, 0x00}, + {0x15, 0x539, 0xCE}, + {0x15, 0x546, 0xC9}, + {0x15, 0x70D, 0xB7}, + {0x15, 0x70E, 0x09}, + {0x15, 0x707, 0x10}, + {0x15, 0x70D, 0xAF}, + {0x15, 0x70E, 0x09}, + {0x15, 0x707, 0x01}, + {0x15, 0x707, 0x05}, + {0x15, 0x70D, 0x73}, + {0x15, 0x70E, 0x09}, + {0x15, 0x707, 0x14}, + {0x15, 0x6FF, 0xC4}, {} }; @@ -4521,7 +6879,12 @@ static void ca0132_init_chip(struct hda_codec *codec) mutex_init(&spec->chipio_mutex); spec->cur_out_type = SPEAKER_OUT; - spec->cur_mic_type = DIGITAL_MIC; + + if (spec->quirk == QUIRK_NONE) + spec->cur_mic_type = DIGITAL_MIC; + else + spec->cur_mic_type = REAR_MIC; + spec->cur_mic_boost = 0; for (i = 0; i < VNODES_COUNT; i++) { @@ -4539,6 +6902,15 @@ static void ca0132_init_chip(struct hda_codec *codec) on = (unsigned int)ca0132_effects[i].reqs[0]; spec->effects_switch[i] = on ? 1 : 0; } + /* + * Sets defaults for the effect slider controls, only for alternative + * ca0132 codecs. Also sets x-bass crossover frequency to 80hz. + */ + if (spec->quirk != QUIRK_NONE) { + spec->xbass_xover_freq = 8; + for (i = 0; i < EFFECT_LEVEL_SLIDERS; i++) + spec->fx_ctl_val[i] = effect_slider_defaults[i]; + } spec->voicefx_val = 0; spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID] = 1; @@ -4549,6 +6921,56 @@ static void ca0132_init_chip(struct hda_codec *codec) #endif } +/* Set all the GPIO to off before shutdown to prevent loud pop noise. */ + +static void r3di_gpio_shutdown(struct hda_codec *codec) +{ + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 0x00); +} + +static void sbz_exit_chip(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + + chipio_set_stream_control(codec, 0x03, 0); + chipio_set_stream_control(codec, 0x04, 0); + + /* Mess with GPIO */ + sbz_gpio_shutdown_commands(codec, 0x07, 0x07, -1); + sbz_gpio_shutdown_commands(codec, 0x07, 0x07, 0x05); + sbz_gpio_shutdown_commands(codec, 0x07, 0x07, 0x01); + + chipio_set_stream_control(codec, 0x14, 0); + chipio_set_stream_control(codec, 0x0C, 0); + + chipio_set_conn_rate(codec, 0x41, SR_192_000); + chipio_set_conn_rate(codec, 0x91, SR_192_000); + + chipio_write(codec, 0x18a020, 0x00000083); + + sbz_gpio_shutdown_commands(codec, 0x07, 0x07, 0x03); + sbz_gpio_shutdown_commands(codec, 0x07, 0x07, 0x07); + sbz_gpio_shutdown_commands(codec, 0x07, 0x07, 0x06); + + chipio_set_stream_control(codec, 0x0C, 0); + + chipio_set_control_param(codec, 0x0D, 0x24); + + sbz_clear_unsolicited(codec); + sbz_set_pin_ctl_default(codec); + + snd_hda_codec_write(codec, spec->out_pins[0], 0, + AC_VERB_SET_EAPD_BTLENABLE, 0x00); + + if (dspload_is_loaded(codec)) + dsp_reset(codec); + + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_CT_EXTENSIONS_ENABLE, 0x00); + + sbz_region2_exit(codec); +} + static void ca0132_exit_chip(struct hda_codec *codec) { /* put any chip cleanup stuffs here. */ @@ -4557,28 +6979,156 @@ static void ca0132_exit_chip(struct hda_codec *codec) dsp_reset(codec); } +/* Extra card variant specific commands that don't really fit anywhere else. */ + +static void sbz_pre_dsp_setup(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + + writel(0x00820680, spec->mem_base + 0x01C); + writel(0x00820680, spec->mem_base + 0x01C); + + snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xfc); + snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xfd); + snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xfe); + snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xff); + + chipio_write(codec, 0x18b0a4, 0x000000c2); +} + +static void r3di_pre_dsp_setup(struct hda_codec *codec) +{ + chipio_write(codec, 0x18b0a4, 0x000000c2); + + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x1E); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_ADDRESS_HIGH, 0x1C); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_DATA_WRITE, 0x5B); + + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x20); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_ADDRESS_HIGH, 0x19); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_DATA_WRITE, 0x00); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_DATA_WRITE, 0x40); +} + +/* + * These are sent before the DSP is downloaded. Not sure + * what they do, or if they're necessary. Could possibly + * be removed. Figure they're better to leave in. + */ +static void sbz_region2_startup(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + + writel(0x00000000, spec->mem_base + 0x400); + writel(0x00000000, spec->mem_base + 0x408); + writel(0x00000000, spec->mem_base + 0x40C); + writel(0x00880680, spec->mem_base + 0x01C); + writel(0x00000083, spec->mem_base + 0xC0C); + writel(0x00000030, spec->mem_base + 0xC00); + writel(0x00000000, spec->mem_base + 0xC04); + writel(0x00000003, spec->mem_base + 0xC0C); + writel(0x00000003, spec->mem_base + 0xC0C); + writel(0x00000003, spec->mem_base + 0xC0C); + writel(0x00000003, spec->mem_base + 0xC0C); + writel(0x000000C1, spec->mem_base + 0xC08); + writel(0x000000F1, spec->mem_base + 0xC08); + writel(0x00000001, spec->mem_base + 0xC08); + writel(0x000000C7, spec->mem_base + 0xC08); + writel(0x000000C1, spec->mem_base + 0xC08); + writel(0x00000080, spec->mem_base + 0xC04); +} + +/* + * Extra init functions for alternative ca0132 codecs. Done + * here so they don't clutter up the main ca0132_init function + * anymore than they have to. + */ +static void ca0132_alt_init(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + + ca0132_alt_vol_setup(codec); + + switch (spec->quirk) { + case QUIRK_SBZ: + codec_dbg(codec, "SBZ alt_init"); + ca0132_gpio_init(codec); + sbz_pre_dsp_setup(codec); + snd_hda_sequence_write(codec, spec->chip_init_verbs); + snd_hda_sequence_write(codec, spec->sbz_init_verbs); + break; + case QUIRK_R3DI: + codec_dbg(codec, "R3DI alt_init"); + ca0132_gpio_init(codec); + ca0132_gpio_setup(codec); + r3di_gpio_dsp_status_set(codec, R3DI_DSP_DOWNLOADING); + r3di_pre_dsp_setup(codec); + snd_hda_sequence_write(codec, spec->chip_init_verbs); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, 0x6FF, 0xC4); + break; + } +} + +static void sbz_check_startup_success(struct hda_codec *codec); + static int ca0132_init(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; int i; + if (spec->dsp_state == DSP_DOWNLOADED) + return 0; + if (spec->dsp_state != DSP_DOWNLOAD_FAILED) spec->dsp_state = DSP_DOWNLOAD_INIT; spec->curr_chip_addx = INVALID_CHIP_ADDRESS; + if (spec->quirk == QUIRK_SBZ) + sbz_region2_startup(codec); + snd_hda_power_up_pm(codec); ca0132_init_unsol(codec); - ca0132_init_params(codec); ca0132_init_flags(codec); snd_hda_sequence_write(codec, spec->base_init_verbs); + if (spec->quirk != QUIRK_NONE) + ca0132_alt_init(codec); + ca0132_download_dsp(codec); ca0132_refresh_widget_caps(codec); - ca0132_setup_defaults(codec); - ca0132_init_analog_mic2(codec); - ca0132_init_dmic(codec); + + if (spec->quirk == QUIRK_SBZ) + writew(0x0107, spec->mem_base + 0x320); + + switch (spec->quirk) { + case QUIRK_SBZ: + sbz_setup_defaults(codec); + /* + * The bug this fixes is hard to reproduce, and I have + * only tested it with alt firmware. It may work for regular + * firmware, but I am not sure. It does not break with regular + * firmware though, so I'll leave it in. + */ + sbz_check_startup_success(codec); + break; + case QUIRK_R3DI: + r3di_setup_defaults(codec); + break; + default: + ca0132_setup_defaults(codec); + ca0132_init_analog_mic2(codec); + ca0132_init_dmic(codec); + break; + } for (i = 0; i < spec->num_outputs; i++) init_output(codec, spec->out_pins[i], spec->dacs[0]); @@ -4590,11 +7140,32 @@ static int ca0132_init(struct hda_codec *codec) init_input(codec, cfg->dig_in_pin, spec->dig_in); - snd_hda_sequence_write(codec, spec->chip_init_verbs); - snd_hda_sequence_write(codec, spec->spec_init_verbs); + if (spec->quirk == QUIRK_NONE) { + snd_hda_sequence_write(codec, spec->chip_init_verbs); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_PARAM_EX_ID_SET, 0x0D); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_PARAM_EX_VALUE_SET, 0x20); + } - ca0132_select_out(codec); - ca0132_select_mic(codec); + if (spec->quirk == QUIRK_SBZ) + ca0132_gpio_setup(codec); + + snd_hda_sequence_write(codec, spec->spec_init_verbs); + switch (spec->quirk) { + case QUIRK_SBZ: + ca0132_alt_select_out(codec); + ca0132_alt_select_in(codec); + break; + case QUIRK_R3DI: + ca0132_alt_select_out(codec); + ca0132_alt_select_in(codec); + break; + case QUIRK_NONE: + ca0132_select_out(codec); + ca0132_select_mic(codec); + break; + } snd_hda_jack_report_sync(codec); @@ -4603,25 +7174,170 @@ static int ca0132_init(struct hda_codec *codec) return 0; } +/* + * The Sound Blaster Z was randomly not working on startup despite everything + * else seeming to go fine. I do not know what causes this, but I found that + * a reboot was enough to fix it, the DSP doesn't need a full shutdown to clear. + * This function will attempt to reload the DSP three times after detecting a + * failure, and give up after that. This has worked for me once, and took two + * reloads to fix the problem. Will have to see how it works for others. + */ + +static void sbz_check_startup_success(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + unsigned int dsp_data_check[4]; + unsigned int cur_address = 0x390; + unsigned int i; + unsigned int failure = 0; + unsigned int reload = 3; + + if (spec->sbz_failure_entered) + return; + + spec->sbz_failure_entered = true; + + for (i = 0; i < 4; i++) { + chipio_read(codec, cur_address, &dsp_data_check[i]); + cur_address += 0x4; + } + for (i = 0; i < 4; i++) { + if (dsp_data_check[i] == 0xa1a2a3a4) + failure = 1; + } + + codec_dbg(codec, "SBZ failure = %d ", failure); + + /* + * While the failure condition is true, and we haven't reached our + * three reload limit, continue trying to reload the driver and + * fix the issue. + */ + while (failure && (reload != 0)) { + codec_dbg(codec, "Reloading... Tries left: %d", reload); + sbz_exit_chip(codec); + spec->dsp_state = DSP_DOWNLOAD_INIT; + codec->patch_ops.init(codec); + failure = 0; + for (i = 0; i < 4; i++) { + chipio_read(codec, cur_address, &dsp_data_check[i]); + cur_address += 0x4; + } + for (i = 0; i < 4; i++) { + if (dsp_data_check[i] == 0xa1a2a3a4) + failure = 1; + } + reload--; + } + + if (!failure) + return; + + codec_info(codec, "Sound Blaster Z DSP Failure. Try a full shutdown."); +} + static void ca0132_free(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; cancel_delayed_work_sync(&spec->unsol_hp_work); snd_hda_power_up(codec); - snd_hda_sequence_write(codec, spec->base_exit_verbs); - ca0132_exit_chip(codec); + switch (spec->quirk) { + case QUIRK_SBZ: + sbz_exit_chip(codec); + /* unmap BAR region 2. */ + iounmap(spec->mem_base); + break; + case QUIRK_R3DI: + r3di_gpio_shutdown(codec); + break; + default: + snd_hda_sequence_write(codec, spec->base_exit_verbs); + ca0132_exit_chip(codec); + break; + } snd_hda_power_down(codec); kfree(spec->spec_init_verbs); kfree(codec->spec); } +static int ca0132_resume(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + hda_nid_t nid; + int i; + bool dsp_loaded; + + codec_dbg(codec, "ca0132 resume entered. dsp_download_wakeup: %d", + spec->dsp_download_wakeup); + /* + * Check and make sure the DSP isn't already loaded. For some + * reason, the SBZ triggers suspend after the build_controls + * patch op has been triggered. This should prevent it from + * downloading the dsp twice, there may be a more sensible + * solution that finds what triggers suspend, but for now, + * this works. + */ + dsp_loaded = dspload_is_loaded(codec); + + if (dsp_loaded) { + spec->dsp_download_wakeup = 0; + return 0; + } + + if (spec->dsp_download_wakeup) { + spec->dsp_state = DSP_DOWNLOAD_INIT; + spec->dsp_download_wakeup = 0; + } + + switch (spec->quirk) { + case QUIRK_ALIENWARE: + snd_hda_apply_pincfgs(codec, alienware_pincfgs); + break; + case QUIRK_SBZ: + snd_hda_apply_pincfgs(codec, sbz_pincfgs); + break; + case QUIRK_R3DI: + snd_hda_apply_pincfgs(codec, r3di_pincfgs); + } + + codec->patch_ops.init(codec); + + /* Update out effects */ + i = OUT_EFFECT_START_NID - EFFECT_START_NID; + nid = OUT_EFFECT_START_NID; + for (; nid < OUT_EFFECT_END_NID; nid++, i++) + ca0132_effects_set(codec, nid, spec->effects_switch[i]); + + return 0; + +} + +static int ca0132_suspend(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + + codec_dbg(codec, "ca0132 suspend entered."); + + spec->dsp_download_wakeup = 1; + + return 0; +} + +static void ca0132_reboot_notify(struct hda_codec *codec) +{ + codec->patch_ops.free(codec); +} + static const struct hda_codec_ops ca0132_patch_ops = { .build_controls = ca0132_build_controls, .build_pcms = ca0132_build_pcms, .init = ca0132_init, .free = ca0132_free, .unsol_event = snd_hda_jack_unsol_event, + .suspend = ca0132_suspend, + .resume = ca0132_resume, + .reboot_notify = ca0132_reboot_notify, }; static void ca0132_config(struct hda_codec *codec) @@ -4635,9 +7351,14 @@ static void ca0132_config(struct hda_codec *codec) spec->multiout.dac_nids = spec->dacs; spec->multiout.num_dacs = 3; - spec->multiout.max_channels = 2; - if (spec->quirk == QUIRK_ALIENWARE) { + if (spec->quirk == QUIRK_NONE) + spec->multiout.max_channels = 2; + else + spec->multiout.max_channels = 6; + + switch (spec->quirk) { + case QUIRK_ALIENWARE: codec_dbg(codec, "ca0132_config: QUIRK_ALIENWARE applied.\n"); snd_hda_apply_pincfgs(codec, alienware_pincfgs); @@ -4657,7 +7378,71 @@ static void ca0132_config(struct hda_codec *codec) spec->input_pins[2] = 0x13; spec->shared_mic_nid = 0x7; spec->unsol_tag_amic1 = 0x11; - } else { + break; + case QUIRK_SBZ: + codec_dbg(codec, "ca0132_config: QUIRK_SBZ applied.\n"); + snd_hda_apply_pincfgs(codec, sbz_pincfgs); + + spec->num_outputs = 2; + spec->out_pins[0] = 0x0B; /* Line out */ + spec->out_pins[1] = 0x0F; /* Rear headphone out */ + spec->out_pins[2] = 0x10; /* Front Headphone / Center/LFE*/ + spec->out_pins[3] = 0x11; /* Rear surround */ + spec->shared_out_nid = 0x2; + spec->unsol_tag_hp = spec->out_pins[1]; + spec->unsol_tag_front_hp = spec->out_pins[2]; + + spec->adcs[0] = 0x7; /* Rear Mic / Line-in */ + spec->adcs[1] = 0x8; /* Front Mic, but only if no DSP */ + spec->adcs[2] = 0xa; /* what u hear */ + + spec->num_inputs = 2; + spec->input_pins[0] = 0x12; /* Rear Mic / Line-in */ + spec->input_pins[1] = 0x13; /* What U Hear */ + spec->shared_mic_nid = 0x7; + spec->unsol_tag_amic1 = spec->input_pins[0]; + + /* SPDIF I/O */ + spec->dig_out = 0x05; + spec->multiout.dig_out_nid = spec->dig_out; + cfg->dig_out_pins[0] = 0x0c; + cfg->dig_outs = 1; + cfg->dig_out_type[0] = HDA_PCM_TYPE_SPDIF; + spec->dig_in = 0x09; + cfg->dig_in_pin = 0x0e; + cfg->dig_in_type = HDA_PCM_TYPE_SPDIF; + break; + case QUIRK_R3DI: + codec_dbg(codec, "ca0132_config: QUIRK_R3DI applied.\n"); + snd_hda_apply_pincfgs(codec, r3di_pincfgs); + + spec->num_outputs = 2; + spec->out_pins[0] = 0x0B; /* Line out */ + spec->out_pins[1] = 0x0F; /* Rear headphone out */ + spec->out_pins[2] = 0x10; /* Front Headphone / Center/LFE*/ + spec->out_pins[3] = 0x11; /* Rear surround */ + spec->shared_out_nid = 0x2; + spec->unsol_tag_hp = spec->out_pins[1]; + spec->unsol_tag_front_hp = spec->out_pins[2]; + + spec->adcs[0] = 0x07; /* Rear Mic / Line-in */ + spec->adcs[1] = 0x08; /* Front Mic, but only if no DSP */ + spec->adcs[2] = 0x0a; /* what u hear */ + + spec->num_inputs = 2; + spec->input_pins[0] = 0x12; /* Rear Mic / Line-in */ + spec->input_pins[1] = 0x13; /* What U Hear */ + spec->shared_mic_nid = 0x7; + spec->unsol_tag_amic1 = spec->input_pins[0]; + + /* SPDIF I/O */ + spec->dig_out = 0x05; + spec->multiout.dig_out_nid = spec->dig_out; + cfg->dig_out_pins[0] = 0x0c; + cfg->dig_outs = 1; + cfg->dig_out_type[0] = HDA_PCM_TYPE_SPDIF; + break; + default: spec->num_outputs = 2; spec->out_pins[0] = 0x0b; /* speaker out */ spec->out_pins[1] = 0x10; /* headphone out */ @@ -4684,7 +7469,9 @@ static void ca0132_config(struct hda_codec *codec) spec->dig_in = 0x09; cfg->dig_in_pin = 0x0e; cfg->dig_in_type = HDA_PCM_TYPE_SPDIF; + break; } + } static int ca0132_prepare_verbs(struct hda_codec *codec) @@ -4693,7 +7480,10 @@ static int ca0132_prepare_verbs(struct hda_codec *codec) #define NUM_SPEC_VERBS 4 struct ca0132_spec *spec = codec->spec; - spec->chip_init_verbs = ca0132_init_verbs0; + spec->chip_init_verbs = ca0132_init_verbs; + if (spec->quirk == QUIRK_SBZ) + spec->sbz_init_verbs = sbz_init_verbs; + spec->spec_init_verbs = kzalloc(sizeof(struct hda_verb) * NUM_SPEC_VERBS, GFP_KERNEL); if (!spec->spec_init_verbs) return -ENOMEM; @@ -4732,6 +7522,7 @@ static int ca0132_prepare_verbs(struct hda_codec *codec) return 0; } + static int patch_ca0132(struct hda_codec *codec) { struct ca0132_spec *spec; @@ -4749,6 +7540,7 @@ static int patch_ca0132(struct hda_codec *codec) codec->patch_ops = ca0132_patch_ops; codec->pcm_format_first = 1; codec->no_sticky_stream = 1; + spec->dsp_download_wakeup = 0; /* Detect codec quirk */ quirk = snd_pci_quirk_lookup(codec->bus->pci, ca0132_quirks); @@ -4757,9 +7549,32 @@ static int patch_ca0132(struct hda_codec *codec) else spec->quirk = QUIRK_NONE; + /* Setup BAR Region 2 for Sound Blaster Z */ + if (spec->quirk == QUIRK_SBZ) { + spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); + if (spec->mem_base == NULL) { + codec_dbg(codec, "pci_iomap failed!"); + codec_dbg(codec, "perhaps this is not an SBZ?"); + spec->quirk = QUIRK_NONE; + } + } + spec->dsp_state = DSP_DOWNLOAD_INIT; spec->num_mixers = 1; - spec->mixers[0] = ca0132_mixer; + + switch (spec->quirk) { + case QUIRK_SBZ: + spec->mixers[0] = ca0132_alt_mixer; + snd_hda_codec_set_name(codec, "Sound Blaster Z"); + break; + case QUIRK_R3DI: + spec->mixers[0] = ca0132_alt_mixer; + snd_hda_codec_set_name(codec, "Recon3Di"); + break; + case QUIRK_NONE: + spec->mixers[0] = ca0132_mixer; + break; + } spec->base_init_verbs = ca0132_base_init_verbs; spec->base_exit_verbs = ca0132_base_exit_verbs; -- 2.7.4

3 4

Re: [alsa-devel] [PATCH 09/15] net: irda: pxaficp_ir: remove the dmaengine compat need
by Greg Kroah-Hartman 23 Apr '18

23 Apr '18

On Mon, Apr 02, 2018 at 04:26:50PM +0200, Robert Jarzmik wrote: > As the pxa architecture switched towards the dmaengine slave map, the > old compatibility mechanism to acquire the dma requestor line number and > priority are not needed anymore. > > This patch simplifies the dma resource acquisition, using the more > generic function dma_request_slave_channel(). > > Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> > --- > drivers/staging/irda/drivers/pxaficp_ir.c | 14 ++------------ > 1 file changed, 2 insertions(+), 12 deletions(-) This file is no longer in Linus's tree :) thanks, greg k-h

1 0

Re: [alsa-devel] [PATCH 3/8] [media] v4l: rcar_fdp1: Change platform dependency to ARCH_RENESAS
by Laurent Pinchart 23 Apr '18

23 Apr '18

Hi Geert, On Saturday, 21 April 2018 11:07:11 EEST Laurent Pinchart wrote: > On Friday, 20 April 2018 16:28:29 EEST Geert Uytterhoeven wrote: > > The Renesas Fine Display Processor driver is used on Renesas R-Car SoCs > > only. Since commit 9b5ba0df4ea4f940 ("ARM: shmobile: Introduce > > ARCH_RENESAS") is ARCH_RENESAS a more appropriate platform dependency > > than the legacy ARCH_SHMOBILE, hence use the former. > > > > This will allow to drop ARCH_SHMOBILE on ARM and ARM64 in the near > > future. > > > > Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> > > Acked-by: Laurent Pinchart <laurent.pinchart(a)ideasonboard.com> > > How would you like to get this merged ? Unless you would like to merge the whole series in one go, I'll take this in my tree as I have a conflicting patch I would like to submit for v4.18. > > --- > > > > drivers/media/platform/Kconfig | 2 +- > > 1 file changed, 1 insertion(+), 1 deletion(-) > > > > diff --git a/drivers/media/platform/Kconfig > > b/drivers/media/platform/Kconfig index f9235e8f8e962d2e..7ad4725f9d1f9627 > > 100644 > > --- a/drivers/media/platform/Kconfig > > +++ b/drivers/media/platform/Kconfig > > @@ -396,7 +396,7 @@ config VIDEO_SH_VEU > > config VIDEO_RENESAS_FDP1 > > tristate "Renesas Fine Display Processor" > > depends on VIDEO_DEV && VIDEO_V4L2 && HAS_DMA > > - depends on ARCH_SHMOBILE || COMPILE_TEST > > + depends on ARCH_RENESAS || COMPILE_TEST > > depends on (!ARCH_RENESAS && !VIDEO_RENESAS_FCP) || VIDEO_RENESAS_FCP > > select VIDEOBUF2_DMA_CONTIG > > select V4L2_MEM2MEM_DEV -- Regards, Laurent Pinchart

2 1

[alsa-devel] [PATCH 0/4] More COMPILE_TEST patches to build all media drivers on x86_64
by Mauro Carvalho Chehab 23 Apr '18

23 Apr '18

While now all media drivers build with COMPILE_TEST on i386, there are still a few of them that don't build on x86_64. Making them compiling there is just a matter of touching Kconfig files. While here, fix smatch warnings when building the siano driver on big endian architectures. Mauro Carvalho Chehab (4): media: radio: allow building ISA drivers with COMPILE_TEST media: sta2x11_vip: allow build with COMPILE_TEST sound, media: allow building ISA drivers it with COMPILE_TEST media: siano: get rid of __le32/__le16 cast warnings drivers/media/common/siano/smsendian.c | 14 ++++++------ drivers/media/pci/sta2x11/Kconfig | 2 +- drivers/media/radio/Kconfig | 41 +++++++++++++++++++++------------- sound/isa/Kconfig | 9 ++++---- 4 files changed, 39 insertions(+), 27 deletions(-) -- 2.14.3

4 11

[alsa-devel] [PATCH] ALSA: pcm: Return negative delays from SNDRV_PCM_IOCTL_DELAY.
by Jeffery Miller 23 Apr '18

23 Apr '18

The commit c2c86a97175f ("ALSA: pcm: Remove set_fs() in PCM core code") changed SNDRV_PCM_IOCTL_DELAY to return an inconsistent error instead of a negative delay. Originally the call would succeed and return the negative delay. The Chromium OS Audio Server (CRAS) gets confused and hangs when the error is returned instead of the negative delay. Help CRAS avoid the issue by rolling back the behavior to return a negative delay instead of an error. Fixes: c2c86a97175f ("ALSA: pcm: Remove set_fs() in PCM core code") Signed-off-by: Jeffery Miller <jmiller(a)neverware.com> --- This patch changes the SNDRV_PCM_IOCTL_DELAY ioctl return behavior to be similar to what it was in an earlier kernel. I encountered this when upgrading from the 4.4 kernel to 4.14. Chrome Audio Server (cras) does not handle the error return as gracefully as it was handling the successful but negative delays. The intent of this patch is to simply revert the behavior of the call to match what it was previously. It has been applied and tested with the 4.14 kernel. I did confirm this is still an issue and is resolved with the patch on the tiwai/sound.git master branch. sound/core/pcm_compat.c | 7 ++++--- sound/core/pcm_native.c | 23 +++++++++++------------ 2 files changed, 15 insertions(+), 15 deletions(-) diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c index 0be248543f1e..077008a8e1d5 100644 --- a/sound/core/pcm_compat.c +++ b/sound/core/pcm_compat.c @@ -27,10 +27,11 @@ static int snd_pcm_ioctl_delay_compat(struct snd_pcm_substream *substream, s32 __user *src) { snd_pcm_sframes_t delay; + int err; - delay = snd_pcm_delay(substream); - if (delay < 0) - return delay; + err = snd_pcm_delay(substream, &delay); + if (err) + return err; if (put_user(delay, src)) return -EFAULT; return 0; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 7585444352df..d2ba181c666e 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2654,7 +2654,8 @@ static int snd_pcm_hwsync(struct snd_pcm_substream *substream) return err; } -static snd_pcm_sframes_t snd_pcm_delay(struct snd_pcm_substream *substream) +static int snd_pcm_delay(struct snd_pcm_substream *substream, + snd_pcm_sframes_t *delay) { int err; snd_pcm_sframes_t n = 0; @@ -2664,7 +2665,9 @@ static snd_pcm_sframes_t snd_pcm_delay(struct snd_pcm_substream *substream) if (!err) n = snd_pcm_calc_delay(substream); snd_pcm_stream_unlock_irq(substream); - return err < 0 ? err : n; + if (!err) + *delay = n; + return err; } static int snd_pcm_sync_ptr(struct snd_pcm_substream *substream, @@ -2866,11 +2869,13 @@ static int snd_pcm_common_ioctl(struct file *file, return snd_pcm_hwsync(substream); case SNDRV_PCM_IOCTL_DELAY: { - snd_pcm_sframes_t delay = snd_pcm_delay(substream); + snd_pcm_sframes_t delay; snd_pcm_sframes_t __user *res = arg; + int err; - if (delay < 0) - return delay; + err = snd_pcm_delay(substream, &delay); + if (err) + return err; if (put_user(delay, res)) return -EFAULT; return 0; @@ -2958,13 +2963,7 @@ int snd_pcm_kernel_ioctl(struct snd_pcm_substream *substream, case SNDRV_PCM_IOCTL_DROP: return snd_pcm_drop(substream); case SNDRV_PCM_IOCTL_DELAY: - { - result = snd_pcm_delay(substream); - if (result < 0) - return result; - *frames = result; - return 0; - } + return snd_pcm_delay(substream, frames); default: return -EINVAL; } -- 2.14.3

2 1

[alsa-devel] [PATCH] ALSA: core: Report audio_tstamp in snd_pcm_sync_ptr
by Takashi Iwai 22 Apr '18

22 Apr '18

From: David Henningsson <diwic(a)ubuntu.com> It looks like a simple mistake that this struct member was forgotten. Audio_tstamp isn't used much, and on some archs (such as x86) this ioctl is not used by default, so that might be the reason why this has slipped for so long. Fixes: 4eeaaeaea1ce ("ALSA: core: add hooks for audio timestamps") Signed-off-by: David Henningsson <diwic(a)ubuntu.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_native.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 35ffcce..632228b 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2751,6 +2751,7 @@ static int snd_pcm_sync_ptr(struct snd_pcm_substream *substream, sync_ptr.s.status.hw_ptr = status->hw_ptr; sync_ptr.s.status.tstamp = status->tstamp; sync_ptr.s.status.suspended_state = status->suspended_state; + sync_ptr.s.status.audio_tstamp = status->audio_tstamp; snd_pcm_stream_unlock_irq(substream); if (copy_to_user(_sync_ptr, &sync_ptr, sizeof(sync_ptr))) return -EFAULT; -- 2.7.4

2 1

Re: [alsa-devel] [PATCH 04/15] media: pxa_camera: remove the dmaengine compat need
by Hans Verkuil 22 Apr '18

22 Apr '18

On 04/02/2018 04:26 PM, Robert Jarzmik wrote: > From: Robert Jarzmik <robert.jarzmik(a)renault.com> > > As the pxa architecture switched towards the dmaengine slave map, the > old compatibility mechanism to acquire the dma requestor line number and > priority are not needed anymore. > > This patch simplifies the dma resource acquisition, using the more > generic function dma_request_slave_channel(). > > Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> Acked-by: Hans Verkuil <hans.verkuil(a)cisco.com> Regards, Hans > --- > drivers/media/platform/pxa_camera.c | 22 +++------------------- > 1 file changed, 3 insertions(+), 19 deletions(-) > > diff --git a/drivers/media/platform/pxa_camera.c b/drivers/media/platform/pxa_camera.c > index c71a00736541..4c82d1880753 100644 > --- a/drivers/media/platform/pxa_camera.c > +++ b/drivers/media/platform/pxa_camera.c > @@ -2357,8 +2357,6 @@ static int pxa_camera_probe(struct platform_device *pdev) > .src_maxburst = 8, > .direction = DMA_DEV_TO_MEM, > }; > - dma_cap_mask_t mask; > - struct pxad_param params; > char clk_name[V4L2_CLK_NAME_SIZE]; > int irq; > int err = 0, i; > @@ -2432,34 +2430,20 @@ static int pxa_camera_probe(struct platform_device *pdev) > pcdev->base = base; > > /* request dma */ > - dma_cap_zero(mask); > - dma_cap_set(DMA_SLAVE, mask); > - dma_cap_set(DMA_PRIVATE, mask); > - > - params.prio = 0; > - params.drcmr = 68; > - pcdev->dma_chans[0] = > - dma_request_slave_channel_compat(mask, pxad_filter_fn, > - &params, &pdev->dev, "CI_Y"); > + pcdev->dma_chans[0] = dma_request_slave_channel(&pdev->dev, "CI_Y"); > if (!pcdev->dma_chans[0]) { > dev_err(&pdev->dev, "Can't request DMA for Y\n"); > return -ENODEV; > } > > - params.drcmr = 69; > - pcdev->dma_chans[1] = > - dma_request_slave_channel_compat(mask, pxad_filter_fn, > - &params, &pdev->dev, "CI_U"); > + pcdev->dma_chans[1] = dma_request_slave_channel(&pdev->dev, "CI_U"); > if (!pcdev->dma_chans[1]) { > dev_err(&pdev->dev, "Can't request DMA for Y\n"); > err = -ENODEV; > goto exit_free_dma_y; > } > > - params.drcmr = 70; > - pcdev->dma_chans[2] = > - dma_request_slave_channel_compat(mask, pxad_filter_fn, > - &params, &pdev->dev, "CI_V"); > + pcdev->dma_chans[2] = dma_request_slave_channel(&pdev->dev, "CI_V"); > if (!pcdev->dma_chans[2]) { > dev_err(&pdev->dev, "Can't request DMA for V\n"); > err = -ENODEV; >

1 0

[alsa-devel] [PATCH] ALSA: add more quirks for DSD interfaces
by Daniel Mack 22 Apr '18

22 Apr '18

Based on a downstream patch from Harry ten Berge. Signed-off-by: Daniel Mack <daniel(a)zonque.org> Reported-and-tested-by: wenyi(a)tianyu-wool.com Original-by: Harry ten Berge <htenberge(a)gmail.com> --- sound/usb/quirks.c | 27 +++++++++++++++++++++++++++ 1 file changed, 27 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index acbeb52f6fd6..5681767cc0d5 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1327,20 +1327,47 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, /* XMOS based USB DACs */ switch (chip->usb_id) { + case USB_ID(0x1511, 0x0037): /* AURALiC VEGA */ + case USB_ID(0x20b1, 0x0002): /* Wyred 4 Sound DAC-2 DSD */ + case USB_ID(0x20b1, 0x2004): /* Matrix Audio X-SPDIF 2 */ case USB_ID(0x20b1, 0x3008): /* iFi Audio micro/nano iDSD */ case USB_ID(0x20b1, 0x2008): /* Matrix Audio X-Sabre */ case USB_ID(0x20b1, 0x300a): /* Matrix Audio Mini-i Pro */ case USB_ID(0x22d9, 0x0416): /* OPPO HA-1 */ + case USB_ID(0x22d9, 0x0436): /* OPPO Sonica */ + case USB_ID(0x22d9, 0x0461): /* OPPO UDP-205 */ + case USB_ID(0x2522, 0x0012): /* LH Labs VI DAC Infinity */ + case USB_ID(0x25ce, 0x001f): /* Mytek Brooklyn DAC */ + case USB_ID(0x25ce, 0x0021): /* Mytek Manhattan DAC */ + case USB_ID(0x25ce, 0x8025): /* Mytek Brooklyn DAC+ */ case USB_ID(0x2772, 0x0230): /* Pro-Ject Pre Box S2 Digital */ if (fp->altsetting == 2) return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; + case USB_ID(0x0d8c, 0x0316): /* Hegel HD12 DSD */ + case USB_ID(0x16b0, 0x06b2): /* NuPrime DAC-10 */ + case USB_ID(0x16d0, 0x0733): /* Furutech ADL Stratos */ + case USB_ID(0x16d0, 0x09db): /* NuPrime Audio DAC-9 */ + case USB_ID(0x1db5, 0x0003): /* Bryston BDA3 */ case USB_ID(0x20b1, 0x000a): /* Gustard DAC-X20U */ + case USB_ID(0x20b1, 0x2005): /* Denafrips Ares DAC */ case USB_ID(0x20b1, 0x2009): /* DIYINHK DSD DXD 384kHz USB to I2S/DSD */ case USB_ID(0x20b1, 0x2023): /* JLsounds I2SoverUSB */ + case USB_ID(0x20b1, 0x3021): /* Eastern El. MiniMax Tube DAC Supreme */ case USB_ID(0x20b1, 0x3023): /* Aune X1S 32BIT/384 DSD DAC */ + case USB_ID(0x20b1, 0x302d): /* Unison Research Unico CD Due */ + case USB_ID(0x20b1, 0x3036): /* Holo Springs Level 3 R2R DAC */ + case USB_ID(0x20b1, 0x307b): /* CH Precision C1 DAC */ + case USB_ID(0x20b1, 0x3086): /* Singxer F-1 converter board */ + case USB_ID(0x22d9, 0x0426): /* OPPO HA-2 */ + case USB_ID(0x22e1, 0xca01): /* HDTA Serenade DSD */ + case USB_ID(0x249c, 0x9326): /* M2Tech Young MkIII */ case USB_ID(0x2616, 0x0106): /* PS Audio NuWave DAC */ + case USB_ID(0x2622, 0x0041): /* Audiolab M-DAC+ */ + case USB_ID(0x27f7, 0x3002): /* W4S DAC-2v2SE */ + case USB_ID(0x29a2, 0x0086): /* Mutec MC3+ USB */ + case USB_ID(0x6b42, 0x0042): /* MSB Technology */ if (fp->altsetting == 3) return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; -- 2.14.3

3 2

[alsa-devel] Old firmware version for Dice II ASIC
by Takashi Sakamoto 21 Apr '18

21 Apr '18

Hi Clemens, Recently I get Alesis Multimix 12 FireWire and work for ALSA dice driver to support it. This unit uses ASIC of 'WaveFront DICE II STD' and TCAT extended application protocol is not supported. Unfortunately, the driver in v4.17-rc1 can't handle this unit, due to validation of address sections for global space. $ dmesg snd_dice fw1.0: Sound card registration failed: -19 (sound/firewire/dice/dice-transaction.c) 265 static int get_subaddrs(struct snd_dice *dice) 266 { 267 static const int min_values[10] = { 268 10, 0x64 / 4, 269 10, 0x18 / 4, 270 10, 0x18 / 4, 271 0, 0, 272 0, 0, 273 }; ... 290 err = snd_fw_transaction(dice->unit, TCODE_READ_BLOCK_REQUEST, 291 DICE_PRIVATE_SPACE, pointers, 292 sizeof(__be32) * ARRAY_SIZE(min_values), 0); 293 if (err < 0) 294 goto end; 295 296 for (i = 0; i < ARRAY_SIZE(min_values); ++i) { 297 data = be32_to_cpu(pointers[i]); 298 if (data < min_values[i] || data >= 0x40000) { 299 err = -ENODEV; 300 goto end; 301 } 302 } The condition 'data < min_values[i]' is evaluated as true at a second iteration (i = 1). A quadlet of 0xffffe0000004-7 has 0x000018 and this is less than 0x19 (= 0x64 / 4) in fact. $ ./firewire-request /dev/fw1 read 0xffffe0000000 28 result: 000: 00 00 00 0a 00 00 00 18 00 00 00 22 00 00 00 8a result: 010: 00 00 00 ac 00 00 01 12 00 00 00 00 00 00 00 00 result: 020: 00 00 00 00 00 00 00 00 In line 183 of 'sound/firewire/dice/dice-interface.h', I can see below comments: (sound/firewire/dice/dice-interface.h) 175 #define GLOBAL_SAMPLE_RATE 0x05c ... 181 #define GLOBAL_VERSION 0x060 182 183 /* Some old firmware versions do not have the following global registers: */ ... 188 #define GLOBAL_CLOCK_CAPABILITIES 0x064 But this is not proper in my case because the maximum offset on global sub-address space is 0x05c in quadlet unit. Well, this unit supports 44.1/48.0 kHz and use internal/Rx1 as its sampling clock source according to user manual. This is what in 'check_clock_caps()'. (sound/firewire/dice/dice.c) 96 static int check_clock_caps(struct snd_dice *dice) 97 { 101 /* some very old firmwares don't tell about their clock support */ 102 if (dice->clock_caps > 0) { ... 109 } else { 110 /* this should be supported by any device */ 111 dice->clock_caps = CLOCK_CAP_RATE_44100 | 112 CLOCK_CAP_RATE_48000 | 113 CLOCK_CAP_SOURCE_ARX1 | 114 CLOCK_CAP_SOURCE_INTERNAL; 115 } In my opinion, GLOBAL_VERSION offset belongs to global address space for newer firmware version. Would I request any comment to you for this idea? Regards Takashi Sakamoto

2 4

[alsa-devel] [PATCH 1/5] compress: remove dead code _is_codec_supported()
by Vinod Koul 21 Apr '18

21 Apr '18

The _is_codec_supported() was dead and none using it, so remove this and eliminate unused function warning compress.c:145:13: warning: ‘_is_codec_supported’ defined but not used [-Wunused-function] We can take from git if user appears Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- src/lib/compress.c | 53 ----------------------------------------------------- 1 file changed, 53 deletions(-) diff --git a/src/lib/compress.c b/src/lib/compress.c index 8ae6bbda7a89..934690e39ed0 100644 --- a/src/lib/compress.c +++ b/src/lib/compress.c @@ -142,49 +142,6 @@ static int get_compress_version(struct compress *compress) return version; } -static bool _is_codec_supported(struct compress *compress, struct compr_config *config, - const struct snd_compr_caps *caps) -{ - bool codec = false; - unsigned int i; - - for (i = 0; i < caps->num_codecs; i++) { - if (caps->codecs[i] == config->codec->id) { - /* found the codec */ - codec = true; - break; - } - } - if (codec == false) { - oops(compress, ENXIO, "this codec is not supported"); - return false; - } - - if (config->fragment_size < caps->min_fragment_size) { - oops(compress, EINVAL, "requested fragment size %d is below min supported %d", - config->fragment_size, caps->min_fragment_size); - return false; - } - if (config->fragment_size > caps->max_fragment_size) { - oops(compress, EINVAL, "requested fragment size %d is above max supported %d", - config->fragment_size, caps->max_fragment_size); - return false; - } - if (config->fragments < caps->min_fragments) { - oops(compress, EINVAL, "requested fragments %d are below min supported %d", - config->fragments, caps->min_fragments); - return false; - } - if (config->fragments > caps->max_fragments) { - oops(compress, EINVAL, "requested fragments %d are above max supported %d", - config->fragments, caps->max_fragments); - return false; - } - - /* TODO: match the codec properties */ - return true; -} - static bool _is_codec_type_supported(int fd, struct snd_codec *codec) { struct snd_compr_caps caps; @@ -271,16 +228,6 @@ struct compress *compress_open(unsigned int card, unsigned int device, config->fragments = caps.max_fragments; } -#if 0 - /* FIXME need to turn this On when DSP supports - * and treat in no support case - */ - if (_is_codec_supported(compress, config, &caps) == false) { - oops(compress, errno, "codec not supported\n"); - goto codec_fail; - } -#endif - memcpy(compress->config, config, sizeof(*compress->config)); fill_compress_params(config, &params); -- 2.7.4

2 7

[alsa-devel] [PATCH alsa-lib v4 0/4] alsa-lib: ASoC: topology: Improve hw_configs
by Kirill Marinushkin 20 Apr '18

20 Apr '18

Hello Takashi, This patch series waits since 20th of February, because it modifies UAPI headers and should be done in sync with Linux and ALSA. I see that previously there was no clear understanding, how to do this in sync. We discussed with you on 02 Apr 2018, and you suggested the following steps: Step 1. A person from Intel, who knows the topic (Pierre), adds his "Reviewed-by" or "Acked-by" tags Step 2. I clarify with Mark to apply the Linux part (different patch series) Step 3. After that, I clarify with you to apply the ALSA part (this patch series) The step 1 is done. Old patches are reviewed by Pierre, so they have his "Reviewed-by" tag. New patches are suggested by him, and we agreed on the implementation, so they have his "Acked-by" tag. The step 2 is done. Mark applied the corresponding part to his Linux tree [1]. Now we are on the step 3. If you don't have anything against this patch series, please apply it to ALSA. Best Regards, Kirill [1] https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git Kirill Marinushkin (4): ASoC: topology: Fix bclk and fsync inversion in set_link_hw_format() ASoC: topology: Add missing clock gating parameter when parsing hw_configs ASoC: topology: Add definitions for mclk_direction values ASoC: topology: Add alias conf parameter names for hw_configs include/sound/asoc.h | 29 +++++++++-- include/topology.h | 8 +-- src/conf/topology/broadwell/broadwell.conf | 4 +- src/topology/pcm.c | 81 +++++++++++++++++++++++------- 4 files changed, 95 insertions(+), 27 deletions(-) -- 2.13.6

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[alsa-devel] [PATCH][next] ASoC: rt5668: fix incorrect 'and' operator
by Colin King 20 Apr '18

20 Apr '18

From: Colin Ian King <colin.king(a)canonical.com> Currently logical and is being used instead of bitwise and. Fix this. Detected by CoverityScan, CID#1468008 ("Logical vs bitwise operator") Fixes: d59fb2856223 ("ASoC: rt5668: add rt5668B codec driver") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/rt5668.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5668.c b/sound/soc/codecs/rt5668.c index 52a343f96eb2..3c19d03f2446 100644 --- a/sound/soc/codecs/rt5668.c +++ b/sound/soc/codecs/rt5668.c @@ -1194,7 +1194,7 @@ static int set_filter_clk(struct snd_soc_dapm_widget *w, int ref, val, reg, idx = -EINVAL; static const int div[] = {1, 2, 3, 4, 6, 8, 12, 16, 24, 32, 48}; - val = snd_soc_component_read32(component, RT5668_GPIO_CTRL_1) && + val = snd_soc_component_read32(component, RT5668_GPIO_CTRL_1) & RT5668_GP4_PIN_MASK; if (w->shift == RT5668_PWR_ADC_S1F_BIT && val == RT5668_GP4_PIN_ADCDAT2) -- 2.17.0

3 2

[alsa-devel] [PATCH 00/61] tree-wide: simplify getting .drvdata
by Wolfram Sang 20 Apr '18

20 Apr '18

I got tired of fixing this in Renesas drivers manually, so I took the big hammer. Remove this cumbersome code pattern which got copy-pasted too much already: - struct platform_device *pdev = to_platform_device(dev); - struct ep93xx_keypad *keypad = platform_get_drvdata(pdev); + struct ep93xx_keypad *keypad = dev_get_drvdata(dev); I send this out as one patch per directory per subsystem. I think they should be applied individually. If you prefer a broken out series per subsystem, I can provide this as well. Just mail me. A branch (tested by buildbot; only with all commits squashed into one commit before) can be found here: git://git.kernel.org/pub/scm/linux/kernel/git/wsa/linux.git coccinelle/get_drvdata Open for other comments, suggestions, too, of course. Here is the cocci-script I created (after <n> iterations by manually checking samples): @@ struct device* d; identifier pdev; expression *ptr; @@ ( - struct platform_device *pdev = to_platform_device(d); | - struct platform_device *pdev; ... - pdev = to_platform_device(d); ) <... when != pdev - &pdev->dev + d ...> ptr = - platform_get_drvdata(pdev) + dev_get_drvdata(d) <... when != pdev - &pdev->dev + d ...> Kind regards, Wolfram Wolfram Sang (61): ARM: plat-samsung: simplify getting .drvdata ata: simplify getting .drvdata auxdisplay: simplify getting .drvdata bus: simplify getting .drvdata clk: samsung: simplify getting .drvdata crypto: simplify getting .drvdata dma: simplify getting .drvdata dmaengine: dw: simplify getting .drvdata dmaengine: qcom: simplify getting .drvdata gpio: simplify getting .drvdata gpu: drm: msm: simplify getting .drvdata gpu: drm: msm: adreno: simplify getting .drvdata gpu: drm: msm: disp: mdp5: simplify getting .drvdata gpu: drm: msm: dsi: simplify getting .drvdata gpu: drm: omapdrm: displays: simplify getting .drvdata gpu: drm: vc4: simplify getting .drvdata hid: simplify getting .drvdata iio: common: cros_ec_sensors: simplify getting .drvdata iio: common: hid-sensors: simplify getting .drvdata input: keyboard: simplify getting .drvdata input: misc: simplify getting .drvdata input: mouse: simplify getting .drvdata input: touchscreen: simplify getting .drvdata iommu: simplify getting .drvdata media: platform: am437x: simplify getting .drvdata media: platform: exynos4-is: simplify getting .drvdata media: platform: s5p-mfc: simplify getting .drvdata mmc: host: simplify getting .drvdata mtd: devices: simplify getting .drvdata mtd: nand: onenand: simplify getting .drvdata net: dsa: simplify getting .drvdata net: ethernet: cadence: simplify getting .drvdata net: ethernet: davicom: simplify getting .drvdata net: ethernet: smsc: simplify getting .drvdata net: ethernet: ti: simplify getting .drvdata net: ethernet: wiznet: simplify getting .drvdata perf: simplify getting .drvdata pinctrl: simplify getting .drvdata pinctrl: intel: simplify getting .drvdata platform: x86: simplify getting .drvdata power: supply: simplify getting .drvdata ptp: simplify getting .drvdata pwm: simplify getting .drvdata rtc: simplify getting .drvdata slimbus: simplify getting .drvdata spi: simplify getting .drvdata staging: greybus: simplify getting .drvdata staging: iio: adc: simplify getting .drvdata staging: nvec: simplify getting .drvdata thermal: simplify getting .drvdata thermal: int340x_thermal: simplify getting .drvdata thermal: st: simplify getting .drvdata tty: serial: simplify getting .drvdata uio: simplify getting .drvdata usb: mtu3: simplify getting .drvdata usb: phy: simplify getting .drvdata video: fbdev: simplify getting .drvdata video: fbdev: omap2: omapfb: displays: simplify getting .drvdata watchdog: simplify getting .drvdata net: dsa: simplify getting .drvdata ASoC: atmel: simplify getting .drvdata arch/arm/plat-samsung/adc.c | 3 +- drivers/ata/pata_samsung_cf.c | 8 ++--- drivers/auxdisplay/arm-charlcd.c | 6 ++-- drivers/bus/brcmstb_gisb.c | 12 +++---- drivers/clk/samsung/clk-s3c2410-dclk.c | 6 ++-- drivers/crypto/exynos-rng.c | 6 ++-- drivers/crypto/picoxcell_crypto.c | 6 ++-- drivers/dma/at_hdmac.c | 9 ++--- drivers/dma/at_xdmac.c | 9 ++--- drivers/dma/dw/platform.c | 6 ++-- drivers/dma/fsldma.c | 6 ++-- drivers/dma/idma64.c | 6 ++-- drivers/dma/qcom/hidma.c | 3 +- drivers/dma/qcom/hidma_mgmt_sys.c | 6 ++-- drivers/dma/ste_dma40.c | 12 +++---- drivers/dma/txx9dmac.c | 8 ++--- drivers/gpio/gpio-dwapb.c | 6 ++-- drivers/gpio/gpio-lynxpoint.c | 3 +- drivers/gpio/gpio-omap.c | 12 +++---- drivers/gpio/gpio-tegra.c | 6 ++-- drivers/gpio/gpio-zynq.c | 6 ++-- drivers/gpu/drm/msm/adreno/adreno_device.c | 6 ++-- drivers/gpu/drm/msm/disp/mdp5/mdp5_kms.c | 6 ++-- drivers/gpu/drm/msm/dsi/dsi_host.c | 6 ++-- drivers/gpu/drm/msm/msm_drv.c | 3 +- drivers/gpu/drm/omapdrm/displays/panel-dsi-cm.c | 18 ++++------ drivers/gpu/drm/vc4/vc4_drv.c | 3 +- drivers/hid/hid-sensor-custom.c | 12 +++---- .../common/cros_ec_sensors/cros_ec_sensors_core.c | 6 ++-- .../iio/common/hid-sensors/hid-sensor-trigger.c | 9 ++--- drivers/input/keyboard/ep93xx_keypad.c | 10 +++--- drivers/input/keyboard/imx_keypad.c | 10 +++--- drivers/input/keyboard/lpc32xx-keys.c | 6 ++-- drivers/input/keyboard/matrix_keypad.c | 10 +++--- drivers/input/keyboard/omap4-keypad.c | 10 +++--- drivers/input/keyboard/pmic8xxx-keypad.c | 6 ++-- drivers/input/keyboard/pxa27x_keypad.c | 10 +++--- drivers/input/keyboard/samsung-keypad.c | 12 +++---- drivers/input/keyboard/snvs_pwrkey.c | 10 +++--- drivers/input/keyboard/spear-keyboard.c | 10 +++--- drivers/input/keyboard/st-keyscan.c | 6 ++-- drivers/input/keyboard/tegra-kbc.c | 10 +++--- drivers/input/misc/max77693-haptic.c | 6 ++-- drivers/input/misc/max8997_haptic.c | 3 +- drivers/input/misc/palmas-pwrbutton.c | 6 ++-- drivers/input/misc/regulator-haptic.c | 6 ++-- drivers/input/misc/twl4030-vibra.c | 3 +- drivers/input/misc/twl6040-vibra.c | 3 +- drivers/input/mouse/navpoint.c | 6 ++-- drivers/input/touchscreen/imx6ul_tsc.c | 6 ++-- drivers/iommu/qcom_iommu.c | 6 ++-- drivers/media/platform/am437x/am437x-vpfe.c | 6 ++-- drivers/media/platform/exynos4-is/media-dev.c | 6 ++-- drivers/media/platform/exynos4-is/mipi-csis.c | 6 ++-- drivers/media/platform/s5p-mfc/s5p_mfc.c | 6 ++-- drivers/mmc/host/davinci_mmc.c | 6 ++-- drivers/mmc/host/sdhci-of-arasan.c | 6 ++-- drivers/mmc/host/wmt-sdmmc.c | 6 ++-- drivers/mtd/devices/docg3.c | 3 +- drivers/mtd/nand/onenand/samsung.c | 6 ++-- drivers/net/dsa/bcm_sf2.c | 6 ++-- drivers/net/dsa/qca8k.c | 6 ++-- drivers/net/ethernet/cadence/macb_main.c | 6 ++-- drivers/net/ethernet/davicom/dm9000.c | 6 ++-- drivers/net/ethernet/smsc/smc91x.c | 3 +- drivers/net/ethernet/ti/cpsw.c | 6 ++-- drivers/net/ethernet/ti/davinci_emac.c | 6 ++-- drivers/net/ethernet/wiznet/w5300.c | 6 ++-- drivers/perf/arm_spe_pmu.c | 6 ++-- drivers/pinctrl/intel/pinctrl-baytrail.c | 6 ++-- drivers/pinctrl/intel/pinctrl-cherryview.c | 6 ++-- drivers/pinctrl/intel/pinctrl-intel.c | 6 ++-- drivers/pinctrl/pinctrl-amd.c | 6 ++-- drivers/pinctrl/pinctrl-at91-pio4.c | 6 ++-- drivers/platform/x86/asus-laptop.c | 3 +- drivers/platform/x86/asus-wmi.c | 3 +- drivers/platform/x86/samsung-laptop.c | 3 +- drivers/power/supply/gpio-charger.c | 3 +- drivers/ptp/ptp_dte.c | 6 ++-- drivers/pwm/pwm-atmel-tcb.c | 6 ++-- drivers/pwm/pwm-rcar.c | 3 +- drivers/rtc/rtc-bq4802.c | 6 ++-- drivers/rtc/rtc-ds1216.c | 6 ++-- drivers/rtc/rtc-ds1511.c | 9 ++--- drivers/rtc/rtc-ds1553.c | 15 +++----- drivers/rtc/rtc-ds1685.c | 21 ++++------- drivers/rtc/rtc-ds1742.c | 6 ++-- drivers/rtc/rtc-lpc32xx.c | 16 ++++----- drivers/rtc/rtc-m48t59.c | 41 +++++++++------------- drivers/rtc/rtc-mv.c | 3 +- drivers/rtc/rtc-mxc.c | 21 ++++------- drivers/rtc/rtc-pcap.c | 15 +++----- drivers/rtc/rtc-sh.c | 15 +++----- drivers/rtc/rtc-stk17ta8.c | 15 +++----- drivers/rtc/rtc-test.c | 3 +- drivers/rtc/rtc-zynqmp.c | 10 +++--- drivers/slimbus/qcom-ctrl.c | 6 ++-- drivers/spi/spi-cadence.c | 6 ++-- drivers/spi/spi-zynqmp-gqspi.c | 6 ++-- drivers/staging/greybus/arche-platform.c | 3 +- drivers/staging/iio/adc/ad7606_par.c | 6 ++-- drivers/staging/nvec/nvec.c | 6 ++-- drivers/thermal/int340x_thermal/int3400_thermal.c | 9 ++--- drivers/thermal/rockchip_thermal.c | 8 ++--- drivers/thermal/spear_thermal.c | 8 ++--- drivers/thermal/st/st_thermal.c | 6 ++-- drivers/thermal/zx2967_thermal.c | 6 ++-- drivers/tty/serial/imx.c | 18 ++++------ drivers/tty/serial/qcom_geni_serial.c | 6 ++-- drivers/tty/serial/st-asc.c | 6 ++-- drivers/tty/serial/xilinx_uartps.c | 6 ++-- drivers/uio/uio_fsl_elbc_gpcm.c | 6 ++-- drivers/usb/mtu3/mtu3_plat.c | 6 ++-- drivers/usb/phy/phy-am335x.c | 6 ++-- drivers/video/fbdev/auo_k190x.c | 12 +++---- .../fbdev/omap2/omapfb/displays/panel-dsi-cm.c | 18 ++++------ drivers/video/fbdev/sh_mobile_lcdcfb.c | 6 ++-- drivers/video/fbdev/sh_mobile_meram.c | 6 ++-- drivers/watchdog/cadence_wdt.c | 6 ++-- drivers/watchdog/of_xilinx_wdt.c | 6 ++-- drivers/watchdog/wdat_wdt.c | 6 ++-- net/dsa/legacy.c | 6 ++-- sound/soc/atmel/atmel_ssc_dai.c | 6 ++-- 123 files changed, 319 insertions(+), 607 deletions(-) -- 2.11.0

3 4

[alsa-devel] [PATCH 0/7] ASoC: Intel: Replace GFP_ATOMIC with GFP_KERNEL
by Vinod Koul 20 Apr '18

20 Apr '18

Jia-Ju Bai recently fixed few Intel drivers which were using GFP_ATOMIC wrongly, this series fixes the remaining ones. Vinod Koul (7): ASoC: Intel: byt-max98090: Replace GFP_ATOMIC with GFP_KERNEL ASoC: Intel: bytcht_es8316: Replace GFP_ATOMIC with GFP_KERNEL ASoC: Intel: bytcr_rt5651: Replace GFP_ATOMIC with GFP_KERNEL ASoC: Intel: cht_bsw_nau8824: Replace GFP_ATOMIC with GFP_KERNEL ASoC: Intel: kbl_da7219_max98357a: Replace GFP_ATOMIC with GFP_KERNEL ASoC: Intel: kbl_rt5663_max98927: Replace GFP_ATOMIC with GFP_KERNEL ASoC: Intel: kbl_rt5663_rt5514_max98927: Replace GFP_ATOMIC with GFP_KERNEL sound/soc/intel/boards/byt-max98090.c | 2 +- sound/soc/intel/boards/bytcht_es8316.c | 2 +- sound/soc/intel/boards/bytcr_rt5651.c | 2 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98357a.c | 2 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 2 +- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 2 +- 7 files changed, 7 insertions(+), 7 deletions(-) -- 2.7.4

2 14

[alsa-devel] [PATCH] ASoC: sh: Drop SUPERH platform dependency
by Geert Uytterhoeven 20 Apr '18

20 Apr '18

The SIU sound peripheral is used only on SuperH SH-Mobile platforms. As both SUPERH and ARCH_SHMOBILE are set for these platforms, the SUPERH dependency can be dropped. Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- sound/soc/sh/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index c1b7fb91e3063f2b..0ae0800bf3a8b471 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -28,7 +28,7 @@ config SND_SOC_SH4_FSI config SND_SOC_SH4_SIU tristate - depends on (SUPERH || ARCH_SHMOBILE) && HAVE_CLK + depends on ARCH_SHMOBILE && HAVE_CLK select DMA_ENGINE select DMADEVICES select SH_DMAE -- 2.7.4

2 1

[alsa-devel] Sample format changes by plugin while playing
by Robert Tiemann 20 Apr '18

20 Apr '18

Hi, I have a general question about ALSA plugins: Is it possible for an ALSA plugin to reconfigure the sample rate and sample format of its slave in such a way that the software playing through that plugin wouldn't notice? That is, given some audio stream, the plugin shall be allowed to change the sample rate and format based on control data extracted from the audio data pushed through the plugin. In my imagination, any output format changes should be completely transparent to the sound player; the player would always deliver a stream of audio data at a constant rate, and once in a while the plugin would change the hardware settings. For sure there would be an audible gap during reconfiguration, but that would be OK for me. Now, is this possible at all? Best regards, Robert

3 7

[alsa-devel] general protection fault in snd_rawmidi_ioctl_compat
by syzbot 19 Apr '18

19 Apr '18

Hello, syzbot hit the following crash on upstream commit b284d4d5a6785f8cd07eda2646a95782373cd01e (Tue Apr 10 19:25:30 2018 +0000) Merge tag 'ceph-for-4.17-rc1' of git://github.com/ceph/ceph-client syzbot dashboard link: https://syzkaller.appspot.com/bug?extid=f7a0348affc3b67bc617 Unfortunately, I don't have any reproducer for this crash yet. Raw console output: https://syzkaller.appspot.com/x/log.txt?id=5208906934517760 Kernel config: https://syzkaller.appspot.com/x/.config?id=-1223000601505858474 compiler: gcc (GCC) 8.0.1 20180301 (experimental) IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+f7a0348affc3b67bc617(a)syzkaller.appspotmail.com It will help syzbot understand when the bug is fixed. See footer for details. If you forward the report, please keep this part and the footer. kasan: CONFIG_KASAN_INLINE enabled kasan: GPF could be caused by NULL-ptr deref or user memory access general protection fault: 0000 [#1] SMP KASAN Dumping ftrace buffer: (ftrace buffer empty) Modules linked in: CPU: 1 PID: 21671 Comm: syz-executor7 Not tainted 4.16.0+ #17 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 RIP: 0010:snd_rawmidi_ioctl_status_x32 sound/core/rawmidi_compat.c:125 [inline] RIP: 0010:snd_rawmidi_ioctl_compat+0x51d/0xba0 sound/core/rawmidi_compat.c:168 RSP: 0018:ffff88018ccdfc08 EFLAGS: 00010202 RAX: dffffc0000000000 RBX: 0000000000000000 RCX: ffffc900054cb000 RDX: 000000000000000b RSI: ffffffff85a61885 RDI: 0000000000000058 RBP: ffff88018ccdfd80 R08: ffff8801af6f0740 R09: 0000000000000000 R10: 0000000000000003 R11: 0000000000000000 R12: 1ffff1003199bf83 R13: 0000000000930000 R14: ffff88018ccdfcf8 R15: 0000000000000001 FS: 00007fa64d466700(0000) GS:ffff8801db100000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00007fa30835b518 CR3: 00000001879ef000 CR4: 00000000001406e0 DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000400 Call Trace: C_SYSC_ioctl fs/compat_ioctl.c:1461 [inline] compat_SyS_ioctl+0xa37/0x33b0 fs/compat_ioctl.c:1407 do_syscall_64+0x29e/0x9d0 arch/x86/entry/common.c:287 entry_SYSCALL_64_after_hwframe+0x42/0xb7 RIP: 0033:0x455259 RSP: 002b:00007fa64d465c68 EFLAGS: 00000246 ORIG_RAX: 0000000000000202 RAX: ffffffffffffffda RBX: 00007fa64d4666d4 RCX: 0000000000455259 RDX: 0000000000930000 RSI: 00000000c0305720 RDI: 0000000000000013 RBP: 000000000072bf58 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000020000180 R11: 0000000000000246 R12: 00000000ffffffff R13: 0000000000000641 R14: 00000000006fc6b8 R15: 0000000000000001 Code: df 48 89 fa 48 c1 ea 03 80 3c 02 00 0f 85 1c 06 00 00 48 b8 00 00 00 00 00 fc ff df 48 8b 5b 08 48 8d 7b 58 48 89 fa 48 c1 ea 03 <80> 3c 02 00 0f 85 e5 05 00 00 48 8b 7b 58 4c 89 f6 e8 5d cc ff RIP: snd_rawmidi_ioctl_status_x32 sound/core/rawmidi_compat.c:125 [inline] RSP: ffff88018ccdfc08 RIP: snd_rawmidi_ioctl_compat+0x51d/0xba0 sound/core/rawmidi_compat.c:168 RSP: ffff88018ccdfc08 ---[ end trace 7dcd43c411f49842 ]--- --- This bug is generated by a dumb bot. It may contain errors. See https://goo.gl/tpsmEJ for details. Direct all questions to syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. If you forgot to add the Reported-by tag, once the fix for this bug is merged into any tree, please reply to this email with: #syz fix: exact-commit-title To mark this as a duplicate of another syzbot report, please reply with: #syz dup: exact-subject-of-another-report If it's a one-off invalid bug report, please reply with: #syz invalid Note: if the crash happens again, it will cause creation of a new bug report. Note: all commands must start from beginning of the line in the email body.

2 1

[alsa-devel] [PATCH] ASoC: Intel: Disable SND_SOC_INTEL_BAYTRAIL when SND_SST_ATOM_HIFI2_PLATFORM is enabled
by Hans de Goede 19 Apr '18

19 Apr '18

The sound/soc/intel/common/sst-acpi.c code only tries to load the "baytrail-pcm-audio" driver (and supporting board drivers) when SND_SST_ATOM_HIFI2_PLATFORM is not enabled, since otherwise these are handled by snd-soc-sst-atom-hifi2-platform.ko. Since these thus will never be used when SND_SST_ATOM_HIFI2_PLATFORM is enabled, building these drivers when it is enabled is useless. Add a Kconfig dependency to reflect this, so that SND_SOC_INTEL_BAYTRAIL cannot be enabled when SND_SST_ATOM_HIFI2_PLATFORM is also enabled. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index ceb105cbd461..640f9bc9bd46 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -61,7 +61,7 @@ config SND_SOC_INTEL_HASWELL config SND_SOC_INTEL_BAYTRAIL tristate "Baytrail (legacy) Platforms" - depends on DMADEVICES && ACPI + depends on DMADEVICES && ACPI && SND_SST_ATOM_HIFI2_PLATFORM=n select SND_SOC_INTEL_SST select SND_SOC_INTEL_SST_ACPI select SND_SOC_INTEL_SST_FIRMWARE -- 2.17.0

3 4

[alsa-devel] Applied "ASoC: topology: Revert recent changes while boot errors are investigated" to the asoc tree
by Mark Brown 19 Apr '18

19 Apr '18

The patch ASoC: topology: Revert recent changes while boot errors are investigated has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 291bfb928863d496e25c785e132a8fbfb32341a8 Mon Sep 17 00:00:00 2001 From: Mark Brown <broonie(a)kernel.org> Date: Thu, 19 Apr 2018 12:14:10 +0100 Subject: [PATCH] ASoC: topology: Revert recent changes while boot errors are investigated Krzysztof Kozlowski reported a NULL dereference in _instantiate_card() on Odroid XU3 and XU boards which he bisected to 45f8cb57da0d7 (ASoC: core: Allow topology to override machine driver FE DAI link config). Revert that commit for now, along with f11a5c27f928 (ASoC: core: Add name prefix for machines with topology rewrites) due to dependency issues, in order to keep things booting cleanly in -next. Reported-by: Krzysztof Kozlowski <krzk(a)kernel.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/soc.h | 12 ------ sound/soc/soc-core.c | 87 ++------------------------------------------ sound/soc/soc-pcm.c | 12 ------ 3 files changed, 3 insertions(+), 108 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 3676d0a8f532..ad266d7e9553 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1012,14 +1012,6 @@ struct snd_soc_platform_driver { /* platform stream compress ops */ const struct snd_compr_ops *compr_ops; - - /* this platform uses topology and ignore machine driver FEs */ - const char *ignore_machine; - const char *topology_name_prefix; - int (*be_hw_params_fixup)(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params); - bool use_dai_pcm_id; /* use the DAI link PCM ID as PCM device number */ - int be_pcm_base; /* base device ID for all BE PCMs */ }; struct snd_soc_dai_link_component { @@ -1126,9 +1118,6 @@ struct snd_soc_dai_link { /* pmdown_time is ignored at stop */ unsigned int ignore_pmdown_time:1; - /* Do not create a PCM for this DAI link (Backend link) */ - unsigned int ignore:1; - struct list_head list; /* DAI link list of the soc card */ struct snd_soc_dobj dobj; /* For topology */ }; @@ -1168,7 +1157,6 @@ struct snd_soc_card { const char *long_name; const char *driver_name; char dmi_longname[80]; - char topology_shortname[32]; struct device *dev; struct snd_card *snd_card; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index aa1c33b3cce0..bf7ca32ab31f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1050,9 +1050,6 @@ static int soc_bind_dai_link(struct snd_soc_card *card, const char *platform_name; int i; - if (dai_link->ignore) - return 0; - dev_dbg(card->dev, "ASoC: binding %s\n", dai_link->name); if (soc_is_dai_link_bound(card, dai_link)) { @@ -1675,7 +1672,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card, { struct snd_soc_dai_link *dai_link = rtd->dai_link; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int i, ret, num; + int i, ret; dev_dbg(card->dev, "ASoC: probe %s dai link %d late %d\n", card->name, rtd->num, order); @@ -1721,23 +1718,9 @@ static int soc_probe_link_dais(struct snd_soc_card *card, soc_dpcm_debugfs_add(rtd); #endif - /* - * most drivers will register their PCMs using DAI link ordering but - * topology based drivers can use the DAI link id field to set PCM - * device number and then use rtd + a base offset of the BEs. - */ - if (rtd->platform->driver->use_dai_pcm_id) { - if (rtd->dai_link->no_pcm) - num = rtd->platform->driver->be_pcm_base + rtd->num; - else - num = rtd->dai_link->id; - } else { - num = rtd->num; - } - if (cpu_dai->driver->compress_new) { /*create compress_device"*/ - ret = cpu_dai->driver->compress_new(rtd, num); + ret = cpu_dai->driver->compress_new(rtd, rtd->num); if (ret < 0) { dev_err(card->dev, "ASoC: can't create compress %s\n", dai_link->stream_name); @@ -1747,7 +1730,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card, if (!dai_link->params) { /* create the pcm */ - ret = soc_new_pcm(rtd, num); + ret = soc_new_pcm(rtd, rtd->num); if (ret < 0) { dev_err(card->dev, "ASoC: can't create pcm %s :%d\n", dai_link->stream_name, ret); @@ -2093,67 +2076,6 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour) EXPORT_SYMBOL_GPL(snd_soc_set_dmi_name); #endif /* CONFIG_DMI */ -static void soc_check_tplg_fes(struct snd_soc_card *card) -{ - struct snd_soc_platform *platform; - struct snd_soc_dai_link *dai_link; - int i; - - list_for_each_entry(platform, &platform_list, list) { - - /* does this platform override FEs ? */ - if (!platform->driver->ignore_machine) - continue; - - /* for this machine ? */ - if (strcmp(platform->driver->ignore_machine, - card->dev->driver->name)) - continue; - - /* machine matches, so override the rtd data */ - for (i = 0; i < card->num_links; i++) { - - dai_link = &card->dai_link[i]; - - /* ignore this FE */ - if (dai_link->dynamic) { - dai_link->ignore = true; - continue; - } - - dev_info(card->dev, "info: override FE DAI link %s\n", - card->dai_link[i].name); - - /* override platform */ - dai_link->platform_name = platform->component.name; - dai_link->cpu_dai_name = platform->component.name; - - /* convert non BE into BE */ - dai_link->no_pcm = 1; - dai_link->dpcm_playback = 1; - dai_link->dpcm_capture = 1; - - /* override any BE fixups */ - dai_link->be_hw_params_fixup = - platform->driver->be_hw_params_fixup; - - /* most BE links dont set stream name, so set it to - * dai link name if it's NULL to help bind widgets. - */ - if (!dai_link->stream_name) - dai_link->stream_name = dai_link->name; - } - - /* Inform userspace we are using alternate topology */ - if (platform->driver->topology_name_prefix) { - snprintf(card->topology_shortname, 32, "%s-%s", - platform->driver->topology_name_prefix, - card->name); - card->name = card->topology_shortname; - } - } -} - static int snd_soc_instantiate_card(struct snd_soc_card *card) { struct snd_soc_codec *codec; @@ -2164,9 +2086,6 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) mutex_lock(&client_mutex); mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_INIT); - /* check whether any platform is ignore machine FE and using topology */ - soc_check_tplg_fes(card); - /* bind DAIs */ for (i = 0; i < card->num_links; i++) { ret = soc_bind_dai_link(card, &card->dai_link[i]); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 4ce489165a6d..68d9dc930096 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -909,20 +909,8 @@ int soc_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; int ret; - /* perform any topology hw_params fixups before DAI */ - if (rtd->dai_link->be_hw_params_fixup) { - ret = rtd->dai_link->be_hw_params_fixup(rtd, params); - if (ret < 0) { - dev_err(rtd->dev, - "ASoC: hw_params topology fixup failed %d\n", - ret); - return ret; - } - } - if (dai->driver->ops->hw_params) { ret = dai->driver->ops->hw_params(substream, params, dai); if (ret < 0) { -- 2.17.0

1 0

[alsa-devel] [BUG] NULL pointer on next-20180419 on Odroid XU3 (snd_soc_register_card)
by Krzysztof Kozlowski 19 Apr '18

19 Apr '18

Hi Mark and Liam, On today's next-20180419, NULL pointer exception appeared in snd_soc_instantiate_card() on Odroid XU3 and XU boards (Samsung, Exynos5422, Exynos5410, max98090 codec). Built with exynos_defconfig (ARM). I bisected it to: 45f8cb57da0d7a9ead4b39d7f5def333a5b0c08b is the first bad commit commit 45f8cb57da0d7a9ead4b39d7f5def333a5b0c08b Author: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Date: Tue Mar 27 14:30:40 2018 +0100 ASoC: core: Allow topology to override machine driver FE DAI link config. Machine drivers statically define a number of DAI links that currently cannot be changed or removed by topology. This means PCMs and platform components cannot be changed by topology at runtime AND machine drivers are tightly coupled to topology. This patch allows topology to override the machine driver DAI link config in order to reuse machine drivers with different topologies and platform components. The patch supports :- 1) create new FE PCMs with a topology defined PCM ID. 2) destroy existing static FE PCMs 3) change the platform component driver. 4) assign any new HW params fixups. The patch requires no changes to the machine drivers, but does add some platform component flags that the platform component driver can assign before loading topologies. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> :040000 040000 25fc03e81975655bd0f90500b2578551742a2fba 28715f6094887d961cb84b26a4d56f478c1ace89 M include :040000 040000 e387f83c522f7cfc71c0f7669752b4475e3ab55e baad16725177cc3a4bb6c70afb2416d4f1b7f620 M sound Full OOPS: [ 6.172383] max98090 5-0010: MAX98090 REVID=0x43 [ 6.182548] max98090 5-0010: use default 2.8v micbias [ 6.196790] Unable to handle kernel NULL pointer dereference at virtual address 00000004 [ 6.203639] pgd = 7cab35d7 [ 6.206125] [00000004] *pgd=00000000 [ 6.209748] Internal error: Oops: 5 [#1] PREEMPT SMP ARM [ 6.214958] Modules linked in: [ 6.217982] CPU: 0 PID: 62 Comm: kworker/0:2 Not tainted 4.17.0-rc1-next-20180419 #41 [ 6.225797] Hardware name: SAMSUNG EXYNOS (Flattened Device Tree) [ 6.231877] Workqueue: events deferred_probe_work_func [ 6.236973] PC is at snd_soc_register_card+0xacc/0x1048 [ 6.242165] LR is at 0x3 [ 6.244663] pc : [<c0783b44>] lr : [<00000003>] psr: 60000013 [ 6.250917] sp : eeb25d98 ip : 00000000 fp : ecc18800 [ 6.256113] r10: ee0ec810 r9 : 00000002 r8 : ee0eca60 [ 6.261312] r7 : ee0ec968 r6 : ecc18e14 r5 : ee0ec900 r4 : c0d73e24 [ 6.267817] r3 : 00000000 r2 : 5d521c4d r1 : ee0ecc24 r0 : 00000000 [ 6.274315] Flags: nZCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment none [ 6.281421] Control: 10c5387d Table: 4000406a DAC: 00000051 [ 6.287138] Process kworker/0:2 (pid: 62, stack limit = 0xf8128fcf) [ 6.293377] Stack: (0xeeb25d98 to 0xeeb26000) [ 6.297704] 5d80: 00000000 ee0ec890 [ 6.305868] 5da0: c1008448 ee0eca08 ee0ec898 c0284318 ee0ec984 c10804fc c0dc0a38 014080c0 [ 6.314013] 5dc0: ee0c6200 5d521c4d 00000000 ee0ec810 ee0c88d0 ee0ec810 ee05e810 ef01f7a4 [ 6.322159] 5de0: c10898e8 00000000 00000022 c0793cc0 ee0ec810 00000000 ee05e810 ee0eca60 [ 6.330304] 5e00: ef01f7a4 c07a6b80 ee05e810 ee05e810 c10898e8 fffffdfb c18136f8 c0578400 [ 6.338450] 5e20: ee05e810 c18136f4 00000000 c10b3320 c18136f8 c0575f4c 00000001 c0578a40 [ 6.346595] 5e40: c10898e8 00000000 eeb25e94 c0576240 c1008448 00000001 c0d78588 c10b3320 [ 6.354741] 5e60: c10c6d60 c0573f24 fffffff3 ee8a7ed4 ed55f3d4 5d521c4d ee05e810 ee05e810 [ 6.362886] 5e80: c1008448 ee05e844 c1064414 c0575b24 c1064414 ee05e810 00000001 5d521c4d [ 6.371031] 5ea0: ee05e810 ee05e810 c1064738 c1064414 eeb24000 c0574de4 ee05e810 c10643c8 [ 6.379177] 5ec0: c10643c8 c05753a0 ffffe000 00000000 60000013 c1064438 eeaaf100 eef65fc0 [ 6.387323] 5ee0: eeb25f1c c100846c eef69200 c1008448 c10ab850 c01471d8 00000001 00000000 [ 6.395468] 5f00: c0147114 60000093 00000000 00000000 c10aa247 eef65fc0 c0147978 c1064438 [ 6.403614] 5f20: c120f104 00000000 c0d7887c 5d521c4d eef65fc0 eeaaf100 eef65fc0 eeaaf118 [ 6.411759] 5f40: 00000008 c1005900 eef65ff4 eeb24000 eef65fc0 c01478d8 eea9f340 00000000 [ 6.419905] 5f60: ffffe000 eeb23000 eea9f340 00000000 eeb24000 eeaaf100 c01478a0 eeb23038 [ 6.428050] 5f80: eeb03ea0 c014e1f4 eeb24000 eea9f340 c014e0bc 00000000 00000000 00000000 [ 6.436195] 5fa0: 00000000 00000000 00000000 c01010b4 00000000 00000000 00000000 00000000 [ 6.444340] 5fc0: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 [ 6.452486] 5fe0: 00000000 00000000 00000000 00000000 00000013 00000000 00000000 00000000 [ 6.460643] [<c0783b44>] (snd_soc_register_card) from [<c0793cc0>] (devm_snd_soc_register_card+0x38/0x70) [ 6.470179] [<c0793cc0>] (devm_snd_soc_register_card) from [<c07a6b80>] (odroid_audio_probe+0x1a4/0x210) [ 6.479621] [<c07a6b80>] (odroid_audio_probe) from [<c0578400>] (platform_drv_probe+0x4c/0xb0) [ 6.488193] [<c0578400>] (platform_drv_probe) from [<c0575f4c>] (driver_probe_device+0x2e0/0x4b4) [ 6.497032] [<c0575f4c>] (driver_probe_device) from [<c0573f24>] (bus_for_each_drv+0x74/0xb8) [ 6.505521] [<c0573f24>] (bus_for_each_drv) from [<c0575b24>] (__device_attach+0xd4/0x168) [ 6.513753] [<c0575b24>] (__device_attach) from [<c0574de4>] (bus_probe_device+0x88/0x90) [ 6.521898] [<c0574de4>] (bus_probe_device) from [<c05753a0>] (deferred_probe_work_func+0x64/0x18c) [ 6.530917] [<c05753a0>] (deferred_probe_work_func) from [<c01471d8>] (process_one_work+0x220/0x8e8) [ 6.540012] [<c01471d8>] (process_one_work) from [<c01478d8>] (worker_thread+0x38/0x584) [ 6.548073] [<c01478d8>] (worker_thread) from [<c014e1f4>] (kthread+0x138/0x168) [ 6.555433] [<c014e1f4>] (kthread) from [<c01010b4>] (ret_from_fork+0x14/0x20) [ 6.562616] Exception stack(0xeeb25fb0 to 0xeeb25ff8) [ 6.567633] 5fa0: 00000000 00000000 00000000 00000000 [ 6.575794] 5fc0: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 [ 6.583939] 5fe0: 00000000 00000000 00000000 00000000 00000013 00000000 [ 6.590521] Code: e5d83059 e3130080 1a0000ec e59b35a4 (e5932004) [ 6.596677] ---[ end trace 97faeced889b58b2 ]--- [ 6.963181] smsc95xx 1-1.1:1.0 eth0: link up, 100Mbps, full-duplex, lpa 0xC1E1 Full log is here: http://www.krzk.eu/#/builders/1/builds/1769/steps/10/logs/serial0 Bisect log: git bisect start # bad: [2754896635e43cdfaf49175a2df23c8bc0eab37a] Add linux-next specific files for 20180419 git bisect bad 2754896635e43cdfaf49175a2df23c8bc0eab37a # good: [a27fc14219f2e3c4a46ba9177b04d9b52c875532] Merge branch 'parisc-4.17-3' of git://git.kernel.org/pub/scm/linux/kernel/git/deller/parisc-linux git bisect good a27fc14219f2e3c4a46ba9177b04d9b52c875532 # good: [d5d9427e3e758fdb55c71a224d587d752661c405] Merge remote-tracking branch 'v4l-dvb/master' git bisect good d5d9427e3e758fdb55c71a224d587d752661c405 # bad: [4fcbe47f55bb71b44724ec00626e28efc24b7bad] Merge remote-tracking branch 'security/next-testing' git bisect bad 4fcbe47f55bb71b44724ec00626e28efc24b7bad # good: [c0c9267abfaf150c68138271f7fe014e5002d7cf] Merge remote-tracking branch 'sound/for-next' git bisect good c0c9267abfaf150c68138271f7fe014e5002d7cf # bad: [726ada29f866aa17d76929b89dc7601300c5a1d6] Merge remote-tracking branch 'block/for-next' git bisect bad 726ada29f866aa17d76929b89dc7601300c5a1d6 # good: [651e4890930d10d009a9a8b829a7177670975ec7] ASoC: Intel: Skylake: Unify the fw ops for SKL and KBL git bisect good 651e4890930d10d009a9a8b829a7177670975ec7 # good: [28f66e6a6a2528369f22d319870fc1b619d05ca2] Merge branch 'for-4.17/block' into for-next git bisect good 28f66e6a6a2528369f22d319870fc1b619d05ca2 # bad: [24ada03555505205b0c8b8b796d52926600bf947] ASoC: topology: Fix build errors git bisect bad 24ada03555505205b0c8b8b796d52926600bf947 # bad: [45f8cb57da0d7a9ead4b39d7f5def333a5b0c08b] ASoC: core: Allow topology to override machine driver FE DAI link config. git bisect bad 45f8cb57da0d7a9ead4b39d7f5def333a5b0c08b # good: [112c60b333ec9ddc7e940116f5e9fdc0d1a9706a] ASoC: Intel: Skylake: Reset stream to link mapping git bisect good 112c60b333ec9ddc7e940116f5e9fdc0d1a9706a # good: [7b0037fa2d6048284e90de4131fc53c0ac1d4430] ASoC: nau8824: user configuration of key detection git bisect good 7b0037fa2d6048284e90de4131fc53c0ac1d4430 # first bad commit: [45f8cb57da0d7a9ead4b39d7f5def333a5b0c08b] ASoC: core: Allow topology to override machine driver FE DAI link config. krzk@PC-kkoz:~/dev/linux$ Best regards, Krzysztof

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda/realtek - set PINCFG_HEADSET_MIC to parse_flags
by Hui Wang 19 Apr '18

19 Apr '18

Otherwise, the pin will be regarded as microphone, and the jack name is "Mic Phantom", it is always on in the pulseaudio even nothing is plugged into the jack. So the UI is confusing to users since the microphone always shows up in the UI even there is no microphone plugged. After adding this flag, the jack name is "Headset Mic Phantom", then the pulseaudio can handle its detection correctly. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index aef1f52..c3b63b7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6370,6 +6370,8 @@ static const struct hda_fixup alc269_fixups[] = { { 0x19, 0x01a1913c }, /* use as headset mic, without its own jack detect */ { } }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MIC }, }; -- 2.7.4

2 4

[alsa-devel] [PATCH 0/2] Add support for the Axe-Fx III implicit fb setup
by Alberto Aguirre 19 Apr '18

19 Apr '18

Some consolidation of repeated code in set_sync_ep_implicit_fb_quirk and a quirk to support implicit feedback for the Axe-FX III Alberto Aguirre (2): ALSA: usb-audio: simplify set_sync_ep_implicit_fb_quirk ALSA: usb-audio: add implicit fb quirk for Axe-Fx III sound/usb/pcm.c | 56 ++++++++++++++++++++++++-------------------------------- 1 file changed, 24 insertions(+), 32 deletions(-) -- 2.14.1

2 3

[alsa-devel] [PATCH][tinycompress] crecord: Use whole buffer requested by user
by Charles Keepax 19 Apr '18

19 Apr '18

Currently, crecord will only actually use 1 fragment worth of its internal buffer no matter what the user requests. Correct this so that the whole buffer requested by the user is used for copying data. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- src/utils/crecord.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/src/utils/crecord.c b/src/utils/crecord.c index e051c2d..8728c5d 100644 --- a/src/utils/crecord.c +++ b/src/utils/crecord.c @@ -370,15 +370,15 @@ static void capture_samples(char *name, unsigned int card, unsigned int device, if (verbose) fprintf(finfo, "%s: Opened compress device\n", __func__); - size = config.fragment_size; - buffer = malloc(size * config.fragments); + size = config.fragments * config.fragment_size; + buffer = malloc(size); if (!buffer) { fprintf(stderr, "Unable to allocate %d bytes\n", size); goto comp_exit; } fprintf(finfo, "Recording file %s On Card %u device %u, with buffer of %lu bytes\n", - name, card, device, buffer_size); + name, card, device, size); fprintf(finfo, "Codec %u Format %u Channels %u, %u Hz\n", codec.id, codec.format, codec.ch_out, rate); -- 2.11.0

2 1

[alsa-devel] [PATCH v2 2/3] ASoC: amd: dma driver changes for BT I2S instance
by Vijendar Mukunda 19 Apr '18

19 Apr '18

With in ACP, There are three I2S controllers can be configured/enabled ( I2S SP, I2S MICSP, I2S BT). Default enabled I2S controller instance is I2S SP. This patch provides required changes to support I2S BT controller Instance. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- v1->v2: defined i2s instance macros in acp header file sound/soc/amd/Kconfig | 1 + sound/soc/amd/acp-pcm-dma.c | 388 +++++++++++++++++++++++++++----------------- sound/soc/amd/acp.h | 67 +++++++- 3 files changed, 298 insertions(+), 158 deletions(-) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 6cbf9cf..6b7c620 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -1,5 +1,6 @@ config SND_SOC_AMD_ACP tristate "AMD Audio Coprocessor support" + select SND_DESIGNWARE_PCM help This option enables ACP DMA support on AMD platform. diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 5ffe2ef..7c392fe 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -23,6 +23,8 @@ #include <drm/amd_asic_type.h> #include "acp.h" +#include "../dwc/local.h" + #define DRV_NAME "acp_audio_dma" #define PLAYBACK_MIN_NUM_PERIODS 2 @@ -37,13 +39,14 @@ #define MAX_BUFFER (PLAYBACK_MAX_PERIOD_SIZE * PLAYBACK_MAX_NUM_PERIODS) #define MIN_BUFFER MAX_BUFFER -#define ST_PLAYBACK_MAX_PERIOD_SIZE 8192 +#define ST_PLAYBACK_MAX_PERIOD_SIZE 4096 #define ST_CAPTURE_MAX_PERIOD_SIZE ST_PLAYBACK_MAX_PERIOD_SIZE #define ST_MAX_BUFFER (ST_PLAYBACK_MAX_PERIOD_SIZE * PLAYBACK_MAX_NUM_PERIODS) #define ST_MIN_BUFFER ST_MAX_BUFFER #define DRV_NAME "acp_audio_dma" + static const struct snd_pcm_hardware acp_pcm_hardware_playback = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | @@ -317,54 +320,21 @@ static void acp_pte_config(void __iomem *acp_mmio, struct page *pg, } static void config_acp_dma(void __iomem *acp_mmio, - struct audio_substream_data *audio_config, + struct audio_substream_data *rtd, u32 asic_type) { - u32 pte_offset, sram_bank; - u16 ch1, ch2, destination, dma_dscr_idx; - - if (audio_config->direction == SNDRV_PCM_STREAM_PLAYBACK) { - pte_offset = ACP_PLAYBACK_PTE_OFFSET; - ch1 = SYSRAM_TO_ACP_CH_NUM; - ch2 = ACP_TO_I2S_DMA_CH_NUM; - sram_bank = ACP_SHARED_RAM_BANK_1_ADDRESS; - destination = TO_ACP_I2S_1; - - } else { - pte_offset = ACP_CAPTURE_PTE_OFFSET; - ch1 = SYSRAM_TO_ACP_CH_NUM; - ch2 = ACP_TO_I2S_DMA_CH_NUM; - switch (asic_type) { - case CHIP_STONEY: - sram_bank = ACP_SHARED_RAM_BANK_3_ADDRESS; - break; - default: - sram_bank = ACP_SHARED_RAM_BANK_5_ADDRESS; - } - destination = FROM_ACP_I2S_1; - } - - acp_pte_config(acp_mmio, audio_config->pg, audio_config->num_of_pages, - pte_offset); - if (audio_config->direction == SNDRV_PCM_STREAM_PLAYBACK) - dma_dscr_idx = PLAYBACK_START_DMA_DESCR_CH12; - else - dma_dscr_idx = CAPTURE_START_DMA_DESCR_CH14; - + acp_pte_config(acp_mmio, rtd->pg, rtd->num_of_pages, + rtd->pte_offset); /* Configure System memory <-> ACP SRAM DMA descriptors */ - set_acp_sysmem_dma_descriptors(acp_mmio, audio_config->size, - audio_config->direction, pte_offset, ch1, - sram_bank, dma_dscr_idx, asic_type); - - if (audio_config->direction == SNDRV_PCM_STREAM_PLAYBACK) - dma_dscr_idx = PLAYBACK_START_DMA_DESCR_CH13; - else - dma_dscr_idx = CAPTURE_START_DMA_DESCR_CH15; + set_acp_sysmem_dma_descriptors(acp_mmio, rtd->size, + rtd->direction, rtd->pte_offset, + rtd->ch1, rtd->sram_bank, + rtd->dma_dscr_idx_1, asic_type); /* Configure ACP SRAM <-> I2S DMA descriptors */ - set_acp_to_i2s_dma_descriptors(acp_mmio, audio_config->size, - audio_config->direction, sram_bank, - destination, ch2, dma_dscr_idx, - asic_type); + set_acp_to_i2s_dma_descriptors(acp_mmio, rtd->size, + rtd->direction, rtd->sram_bank, + rtd->destination, rtd->ch2, + rtd->dma_dscr_idx_2, asic_type); } /* Start a given DMA channel transfer */ @@ -390,6 +360,9 @@ static void acp_dma_start(void __iomem *acp_mmio, case ACP_TO_I2S_DMA_CH_NUM: case ACP_TO_SYSRAM_CH_NUM: case I2S_TO_ACP_DMA_CH_NUM: + case ACP_TO_I2S_DMA_BT_INSTANCE_CH_NUM: + case ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM: + case I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM: dma_ctrl |= ACP_DMA_CNTL_0__DMAChIOCEn_MASK; break; default: @@ -670,6 +643,24 @@ static irqreturn_t dma_irq_handler(int irq, void *arg) acp_mmio, mmACP_EXTERNAL_INTR_STAT); } + if ((intr_flag & BIT(ACP_TO_I2S_DMA_BT_INSTANCE_CH_NUM)) != 0) { + valid_irq = true; + if (acp_reg_read(acp_mmio, mmACP_DMA_CUR_DSCR_9) == + PLAYBACK_START_DMA_DESCR_CH9) + dscr_idx = PLAYBACK_END_DMA_DESCR_CH8; + else + dscr_idx = PLAYBACK_START_DMA_DESCR_CH8; + config_acp_dma_channel(acp_mmio, + SYSRAM_TO_ACP_BT_INSTANCE_CH_NUM, + dscr_idx, 1, 0); + acp_dma_start(acp_mmio, SYSRAM_TO_ACP_BT_INSTANCE_CH_NUM, + false); + snd_pcm_period_elapsed(irq_data->play_i2sbt_stream); + acp_reg_write((intr_flag & + BIT(ACP_TO_I2S_DMA_BT_INSTANCE_CH_NUM)) << 16, + acp_mmio, mmACP_EXTERNAL_INTR_STAT); + } + if ((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) != 0) { valid_irq = true; if (acp_reg_read(acp_mmio, mmACP_DMA_CUR_DSCR_15) == @@ -692,6 +683,31 @@ static irqreturn_t dma_irq_handler(int irq, void *arg) acp_mmio, mmACP_EXTERNAL_INTR_STAT); } + if ((intr_flag & BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) != 0) { + valid_irq = true; + if (acp_reg_read(acp_mmio, mmACP_DMA_CUR_DSCR_11) == + CAPTURE_START_DMA_DESCR_CH11) + dscr_idx = CAPTURE_END_DMA_DESCR_CH10; + else + dscr_idx = CAPTURE_START_DMA_DESCR_CH10; + config_acp_dma_channel(acp_mmio, + ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM, + dscr_idx, 1, 0); + acp_dma_start(acp_mmio, ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM, + false); + acp_reg_write((intr_flag & + BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) << 16, + acp_mmio, mmACP_EXTERNAL_INTR_STAT); + } + + if ((intr_flag & BIT(ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM)) != 0) { + valid_irq = true; + snd_pcm_period_elapsed(irq_data->capture_i2sbt_stream); + acp_reg_write((intr_flag & + BIT(ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM)) << 16, + acp_mmio, mmACP_EXTERNAL_INTR_STAT); + } + if (valid_irq) return IRQ_HANDLED; else @@ -707,6 +723,7 @@ static int acp_dma_open(struct snd_pcm_substream *substream) struct snd_soc_component *component = snd_soc_rtdcom_lookup(prtd, DRV_NAME); struct audio_drv_data *intr_data = dev_get_drvdata(component->dev); + struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(prtd->cpu_dai); struct audio_substream_data *adata = kzalloc(sizeof(struct audio_substream_data), GFP_KERNEL); if (!adata) @@ -720,6 +737,8 @@ static int acp_dma_open(struct snd_pcm_substream *substream) default: runtime->hw = acp_pcm_hardware_playback; } + adata->i2s_play_instance = dev->i2s_instance; + adata->bytescount = 0; } else { switch (intr_data->asic_type) { case CHIP_STONEY: @@ -728,6 +747,8 @@ static int acp_dma_open(struct snd_pcm_substream *substream) default: runtime->hw = acp_pcm_hardware_capture; } + adata->i2s_capture_instance = dev->i2s_instance; + adata->bytescount = 0; } ret = snd_pcm_hw_constraint_integer(runtime, @@ -747,11 +768,46 @@ static int acp_dma_open(struct snd_pcm_substream *substream) * This enablement is not required for another stream, if current * stream is not closed */ - if (!intr_data->play_i2ssp_stream && !intr_data->capture_i2ssp_stream) + if (!intr_data->play_i2ssp_stream && !intr_data->capture_i2ssp_stream && + !intr_data->play_i2sbt_stream && !intr_data->capture_i2sbt_stream) acp_reg_write(1, adata->acp_mmio, mmACP_EXTERNAL_INTR_ENB); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - intr_data->play_i2ssp_stream = substream; + switch (adata->i2s_play_instance) { + case I2S_BT_INSTANCE: + adata->pte_offset = ACP_ST_BT_PLAYBACK_PTE_OFFSET; + adata->ch1 = SYSRAM_TO_ACP_BT_INSTANCE_CH_NUM; + adata->ch2 = ACP_TO_I2S_DMA_BT_INSTANCE_CH_NUM; + adata->sram_bank = ACP_SRAM_BANK_3_ADDRESS; + adata->destination = TO_BLUETOOTH; + adata->dma_dscr_idx_1 = PLAYBACK_START_DMA_DESCR_CH8; + adata->dma_dscr_idx_2 = PLAYBACK_START_DMA_DESCR_CH9; + adata->byte_cnt_high = + mmACP_I2S_BT_TRANSMIT_BYTE_CNT_HIGH; + adata->byte_cnt_low = + mmACP_I2S_BT_TRANSMIT_BYTE_CNT_LOW; + intr_data->play_i2sbt_stream = substream; + break; + case I2S_SP_INSTANCE: + default: + switch (intr_data->asic_type) { + case CHIP_STONEY: + adata->pte_offset = ACP_ST_PLAYBACK_PTE_OFFSET; + break; + default: + adata->pte_offset = ACP_PLAYBACK_PTE_OFFSET; + } + adata->ch1 = SYSRAM_TO_ACP_CH_NUM; + adata->ch2 = ACP_TO_I2S_DMA_CH_NUM; + adata->sram_bank = ACP_SRAM_BANK_1_ADDRESS; + adata->destination = TO_ACP_I2S_1; + adata->dma_dscr_idx_1 = PLAYBACK_START_DMA_DESCR_CH12; + adata->dma_dscr_idx_2 = PLAYBACK_START_DMA_DESCR_CH13; + adata->byte_cnt_high = mmACP_I2S_TRANSMIT_BYTE_CNT_HIGH; + adata->byte_cnt_low = mmACP_I2S_TRANSMIT_BYTE_CNT_LOW; + intr_data->play_i2ssp_stream = substream; + break; + } /* * For Stoney, Memory gating is disabled,i.e SRAM Banks * won't be turned off. The default state for SRAM banks is ON. @@ -763,7 +819,40 @@ static int acp_dma_open(struct snd_pcm_substream *substream) bank, true); } } else { - intr_data->capture_i2ssp_stream = substream; + switch (adata->i2s_capture_instance) { + case I2S_BT_INSTANCE: + adata->pte_offset = ACP_ST_BT_CAPTURE_PTE_OFFSET; + adata->ch1 = ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM; + adata->ch2 = I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM; + adata->sram_bank = ACP_SRAM_BANK_4_ADDRESS; + adata->destination = FROM_BLUETOOTH; + adata->dma_dscr_idx_1 = CAPTURE_START_DMA_DESCR_CH10; + adata->dma_dscr_idx_2 = CAPTURE_START_DMA_DESCR_CH11; + adata->byte_cnt_high = + mmACP_I2S_BT_RECEIVE_BYTE_CNT_HIGH; + adata->byte_cnt_low = + mmACP_I2S_BT_RECEIVE_BYTE_CNT_LOW; + intr_data->capture_i2sbt_stream = substream; + break; + case I2S_SP_INSTANCE: + default: + adata->pte_offset = ACP_CAPTURE_PTE_OFFSET; + adata->ch1 = SYSRAM_TO_ACP_CH_NUM; + adata->ch2 = ACP_TO_I2S_DMA_CH_NUM; + switch (intr_data->asic_type) { + case CHIP_STONEY: + adata->sram_bank = ACP_SRAM_BANK_2_ADDRESS; + break; + default: + adata->sram_bank = ACP_SRAM_BANK_5_ADDRESS; + } + adata->destination = FROM_ACP_I2S_1; + adata->dma_dscr_idx_1 = CAPTURE_START_DMA_DESCR_CH14; + adata->dma_dscr_idx_2 = CAPTURE_START_DMA_DESCR_CH15; + adata->byte_cnt_high = mmACP_I2S_RECEIVED_BYTE_CNT_HIGH; + adata->byte_cnt_low = mmACP_I2S_RECEIVED_BYTE_CNT_LOW; + intr_data->capture_i2ssp_stream = substream; + } if (intr_data->asic_type != CHIP_STONEY) { for (bank = 5; bank <= 8; bank++) acp_set_sram_bank_state(intr_data->acp_mmio, @@ -797,10 +886,25 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, if (adata->asic_type == CHIP_STONEY) { val = acp_reg_read(adata->acp_mmio, mmACP_I2S_16BIT_RESOLUTION_EN); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - val |= ACP_I2S_SP_16BIT_RESOLUTION_EN; - else - val |= ACP_I2S_MIC_16BIT_RESOLUTION_EN; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + switch (rtd->i2s_play_instance) { + case I2S_BT_INSTANCE: + val |= ACP_I2S_BT_16BIT_RESOLUTION_EN; + break; + case I2S_SP_INSTANCE: + default: + val |= ACP_I2S_SP_16BIT_RESOLUTION_EN; + } + } else { + switch (rtd->i2s_capture_instance) { + case I2S_BT_INSTANCE: + val |= ACP_I2S_BT_16BIT_RESOLUTION_EN; + break; + case I2S_SP_INSTANCE: + default: + val |= ACP_I2S_MIC_16BIT_RESOLUTION_EN; + } + } acp_reg_write(val, adata->acp_mmio, mmACP_I2S_16BIT_RESOLUTION_EN); } @@ -837,26 +941,15 @@ static int acp_dma_hw_free(struct snd_pcm_substream *substream) return snd_pcm_lib_free_pages(substream); } -static u64 acp_get_byte_count(void __iomem *acp_mmio, int stream) +static u64 acp_get_byte_count(struct audio_substream_data *rtd) { - union acp_dma_count playback_dma_count; - union acp_dma_count capture_dma_count; - u64 bytescount = 0; + union acp_dma_count byte_count; - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - playback_dma_count.bcount.high = acp_reg_read(acp_mmio, - mmACP_I2S_TRANSMIT_BYTE_CNT_HIGH); - playback_dma_count.bcount.low = acp_reg_read(acp_mmio, - mmACP_I2S_TRANSMIT_BYTE_CNT_LOW); - bytescount = playback_dma_count.bytescount; - } else { - capture_dma_count.bcount.high = acp_reg_read(acp_mmio, - mmACP_I2S_RECEIVED_BYTE_CNT_HIGH); - capture_dma_count.bcount.low = acp_reg_read(acp_mmio, - mmACP_I2S_RECEIVED_BYTE_CNT_LOW); - bytescount = capture_dma_count.bytescount; - } - return bytescount; + byte_count.bcount.high = acp_reg_read(rtd->acp_mmio, + rtd->byte_cnt_high); + byte_count.bcount.low = acp_reg_read(rtd->acp_mmio, + rtd->byte_cnt_low); + return byte_count.bytescount; } static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) @@ -872,15 +965,10 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) return -EINVAL; buffersize = frames_to_bytes(runtime, runtime->buffer_size); - bytescount = acp_get_byte_count(rtd->acp_mmio, substream->stream); + bytescount = acp_get_byte_count(rtd); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (bytescount > rtd->i2ssp_renderbytescount) - bytescount = bytescount - rtd->i2ssp_renderbytescount; - } else { - if (bytescount > rtd->i2ssp_capturebytescount) - bytescount = bytescount - rtd->i2ssp_capturebytescount; - } + if (bytescount > rtd->bytescount) + bytescount = bytescount - rtd->bytescount; pos = do_div(bytescount, buffersize); return bytes_to_frames(runtime, pos); } @@ -898,21 +986,14 @@ static int acp_dma_prepare(struct snd_pcm_substream *substream) if (!rtd) return -EINVAL; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - config_acp_dma_channel(rtd->acp_mmio, SYSRAM_TO_ACP_CH_NUM, - PLAYBACK_START_DMA_DESCR_CH12, - NUM_DSCRS_PER_CHANNEL, 0); - config_acp_dma_channel(rtd->acp_mmio, ACP_TO_I2S_DMA_CH_NUM, - PLAYBACK_START_DMA_DESCR_CH13, - NUM_DSCRS_PER_CHANNEL, 0); - } else { - config_acp_dma_channel(rtd->acp_mmio, ACP_TO_SYSRAM_CH_NUM, - CAPTURE_START_DMA_DESCR_CH14, - NUM_DSCRS_PER_CHANNEL, 0); - config_acp_dma_channel(rtd->acp_mmio, I2S_TO_ACP_DMA_CH_NUM, - CAPTURE_START_DMA_DESCR_CH15, - NUM_DSCRS_PER_CHANNEL, 0); - } + config_acp_dma_channel(rtd->acp_mmio, + rtd->ch1, + rtd->dma_dscr_idx_1, + NUM_DSCRS_PER_CHANNEL, 0); + config_acp_dma_channel(rtd->acp_mmio, + rtd->ch2, + rtd->dma_dscr_idx_2, + NUM_DSCRS_PER_CHANNEL, 0); return 0; } @@ -934,15 +1015,13 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: - bytescount = acp_get_byte_count(rtd->acp_mmio, - substream->stream); + bytescount = acp_get_byte_count(rtd); + if (rtd->bytescount == 0) + rtd->bytescount = bytescount; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (rtd->i2ssp_renderbytescount == 0) - rtd->i2ssp_renderbytescount = bytescount; - acp_dma_start(rtd->acp_mmio, - SYSRAM_TO_ACP_CH_NUM, false); + acp_dma_start(rtd->acp_mmio, rtd->ch1, false); while (acp_reg_read(rtd->acp_mmio, mmACP_DMA_CH_STS) & - BIT(SYSRAM_TO_ACP_CH_NUM)) { + BIT(rtd->ch1)) { if (!loops--) { dev_err(component->dev, "acp dma start timeout\n"); @@ -950,40 +1029,21 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) } cpu_relax(); } - - acp_dma_start(rtd->acp_mmio, - ACP_TO_I2S_DMA_CH_NUM, true); - - } else { - if (rtd->i2ssp_capturebytescount == 0) - rtd->i2ssp_capturebytescount = bytescount; - acp_dma_start(rtd->acp_mmio, - I2S_TO_ACP_DMA_CH_NUM, true); } + acp_dma_start(rtd->acp_mmio, rtd->ch2, true); ret = 0; break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: - /* - * Need to stop only circular DMA channels : - * ACP_TO_I2S_DMA_CH_NUM / I2S_TO_ACP_DMA_CH_NUM. Non-circular - * channels will stopped automatically after its transfer - * completes : SYSRAM_TO_ACP_CH_NUM / ACP_TO_SYSRAM_CH_NUM - */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - ret = acp_dma_stop(rtd->acp_mmio, - SYSRAM_TO_ACP_CH_NUM); - ret = acp_dma_stop(rtd->acp_mmio, - ACP_TO_I2S_DMA_CH_NUM); - rtd->i2ssp_renderbytescount = 0; + acp_dma_stop(rtd->acp_mmio, rtd->ch1); + ret = acp_dma_stop(rtd->acp_mmio, rtd->ch2); } else { - ret = acp_dma_stop(rtd->acp_mmio, - I2S_TO_ACP_DMA_CH_NUM); - ret = acp_dma_stop(rtd->acp_mmio, - ACP_TO_SYSRAM_CH_NUM); - rtd->i2ssp_capturebytescount = 0; + acp_dma_stop(rtd->acp_mmio, rtd->ch2); + ret = acp_dma_stop(rtd->acp_mmio, rtd->ch1); } + rtd->bytescount = 0; break; default: ret = -EINVAL; @@ -1028,27 +1088,40 @@ static int acp_dma_close(struct snd_pcm_substream *substream) DRV_NAME); struct audio_drv_data *adata = dev_get_drvdata(component->dev); - kfree(rtd); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - adata->play_i2ssp_stream = NULL; - /* - * For Stoney, Memory gating is disabled,i.e SRAM Banks - * won't be turned off. The default state for SRAM banks is ON. - * Setting SRAM bank state code skipped for STONEY platform. - * added condition checks for Carrizo platform only - */ - if (adata->asic_type != CHIP_STONEY) { - for (bank = 1; bank <= 4; bank++) - acp_set_sram_bank_state(adata->acp_mmio, bank, - false); + switch (rtd->i2s_play_instance) { + case I2S_BT_INSTANCE: + adata->play_i2sbt_stream = NULL; + break; + case I2S_SP_INSTANCE: + default: + adata->play_i2ssp_stream = NULL; + /* + * For Stoney, Memory gating is disabled,i.e SRAM Banks + * won't be turned off. The default state for SRAM banks + * is ON.Setting SRAM bank state code skipped for STONEY + * platform. Added condition checks for Carrizo platform + * only. + */ + if (adata->asic_type != CHIP_STONEY) { + for (bank = 1; bank <= 4; bank++) + acp_set_sram_bank_state(adata->acp_mmio, + bank, false); + } } } else { - adata->capture_i2ssp_stream = NULL; - if (adata->asic_type != CHIP_STONEY) { - for (bank = 5; bank <= 8; bank++) - acp_set_sram_bank_state(adata->acp_mmio, bank, - false); + switch (rtd->i2s_capture_instance) { + case I2S_BT_INSTANCE: + adata->capture_i2sbt_stream = NULL; + break; + case I2S_SP_INSTANCE: + default: + adata->capture_i2ssp_stream = NULL; + if (adata->asic_type != CHIP_STONEY) { + for (bank = 5; bank <= 8; bank++) + acp_set_sram_bank_state(adata->acp_mmio, + bank, false); + } } } @@ -1056,8 +1129,10 @@ static int acp_dma_close(struct snd_pcm_substream *substream) * Disable ACP irq, when the current stream is being closed and * another stream is also not active. */ - if (!adata->play_i2ssp_stream && !adata->capture_i2ssp_stream) + if (!adata->play_i2ssp_stream && !adata->capture_i2ssp_stream && + !adata->play_i2sbt_stream && !adata->capture_i2sbt_stream) acp_reg_write(0, adata->acp_mmio, mmACP_EXTERNAL_INTR_ENB); + kfree(rtd); return 0; } @@ -1110,6 +1185,8 @@ static int acp_audio_probe(struct platform_device *pdev) audio_drv_data->play_i2ssp_stream = NULL; audio_drv_data->capture_i2ssp_stream = NULL; + audio_drv_data->play_i2sbt_stream = NULL; + audio_drv_data->capture_i2sbt_stream = NULL; audio_drv_data->asic_type = *pdata; @@ -1166,6 +1243,7 @@ static int acp_pcm_resume(struct device *dev) { u16 bank; int status; + struct audio_substream_data *rtd; struct audio_drv_data *adata = dev_get_drvdata(dev); status = acp_init(adata->acp_mmio, adata->asic_type); @@ -1185,9 +1263,8 @@ static int acp_pcm_resume(struct device *dev) acp_set_sram_bank_state(adata->acp_mmio, bank, true); } - config_acp_dma(adata->acp_mmio, - adata->play_i2ssp_stream->runtime->private_data, - adata->asic_type); + rtd = adata->play_i2ssp_stream->runtime->private_data; + config_acp_dma(adata->acp_mmio, rtd, adata->asic_type); } if (adata->capture_i2ssp_stream && adata->capture_i2ssp_stream->runtime) { @@ -1196,9 +1273,20 @@ static int acp_pcm_resume(struct device *dev) acp_set_sram_bank_state(adata->acp_mmio, bank, true); } - config_acp_dma(adata->acp_mmio, - adata->capture_i2ssp_stream->runtime->private_data, - adata->asic_type); + rtd = adata->capture_i2ssp_stream->runtime->private_data; + config_acp_dma(adata->acp_mmio, rtd, adata->asic_type); + } + if (adata->asic_type != CHIP_CARRIZO) { + if (adata->play_i2sbt_stream && + adata->play_i2sbt_stream->runtime) { + rtd = adata->play_i2sbt_stream->runtime->private_data; + config_acp_dma(adata->acp_mmio, rtd, adata->asic_type); + } + if (adata->capture_i2sbt_stream && + adata->capture_i2sbt_stream->runtime) { + rtd = adata->capture_i2sbt_stream->runtime->private_data; + config_acp_dma(adata->acp_mmio, rtd, adata->asic_type); + } } acp_reg_write(1, adata->acp_mmio, mmACP_EXTERNAL_INTR_ENB); return 0; diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index 0e6089b..95c39a3 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -10,17 +10,30 @@ #define ACP_PLAYBACK_PTE_OFFSET 10 #define ACP_CAPTURE_PTE_OFFSET 0 +/* Playback and Capture Offset for Stoney */ +#define ACP_ST_PLAYBACK_PTE_OFFSET 0x04 +#define ACP_ST_CAPTURE_PTE_OFFSET 0x00 +#define ACP_ST_BT_PLAYBACK_PTE_OFFSET 0x08 +#define ACP_ST_BT_CAPTURE_PTE_OFFSET 0x0c + #define ACP_GARLIC_CNTL_DEFAULT 0x00000FB4 #define ACP_ONION_CNTL_DEFAULT 0x00000FB4 #define ACP_PHYSICAL_BASE 0x14000 -/* Playback SRAM address (as a destination in dma descriptor) */ -#define ACP_SHARED_RAM_BANK_1_ADDRESS 0x4002000 - -/* Capture SRAM address (as a source in dma descriptor) */ -#define ACP_SHARED_RAM_BANK_5_ADDRESS 0x400A000 -#define ACP_SHARED_RAM_BANK_3_ADDRESS 0x4006000 +/* + * In case of I2S SP controller instance, Stoney uses SRAM bank 1 for + * playback and SRAM Bank 2 for capture where as in case of BT I2S + * Instance, Stoney uses SRAM Bank 3 for playback & SRAM Bank 4 will + * be used for capture. Carrizo uses I2S SP controller instance. SRAM Banks + * 1, 2, 3, 4 will be used for playback & SRAM Banks 5, 6, 7, 8 will be used + * for capture scenario. + */ +#define ACP_SRAM_BANK_1_ADDRESS 0x4002000 +#define ACP_SRAM_BANK_2_ADDRESS 0x4004000 +#define ACP_SRAM_BANK_3_ADDRESS 0x4006000 +#define ACP_SRAM_BANK_4_ADDRESS 0x4008000 +#define ACP_SRAM_BANK_5_ADDRESS 0x400A000 #define ACP_DMA_RESET_TIME 10000 #define ACP_CLOCK_EN_TIME_OUT_VALUE 0x000000FF @@ -35,8 +48,13 @@ #define TO_ACP_I2S_1 0x2 #define TO_ACP_I2S_2 0x4 +#define TO_BLUETOOTH 0x3 #define FROM_ACP_I2S_1 0xa #define FROM_ACP_I2S_2 0xb +#define FROM_BLUETOOTH 0xb + +#define I2S_SP_INSTANCE 0x01 +#define I2S_BT_INSTANCE 0x02 #define ACP_TILE_ON_MASK 0x03 #define ACP_TILE_OFF_MASK 0x02 @@ -57,6 +75,14 @@ #define ACP_TO_SYSRAM_CH_NUM 14 #define I2S_TO_ACP_DMA_CH_NUM 15 +/* Playback DMA Channels for I2S BT instance */ +#define SYSRAM_TO_ACP_BT_INSTANCE_CH_NUM 8 +#define ACP_TO_I2S_DMA_BT_INSTANCE_CH_NUM 9 + +/* Capture DMA Channels for I2S BT Instance */ +#define ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM 10 +#define I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM 11 + #define NUM_DSCRS_PER_CHANNEL 2 #define PLAYBACK_START_DMA_DESCR_CH12 0 @@ -69,9 +95,22 @@ #define CAPTURE_START_DMA_DESCR_CH15 6 #define CAPTURE_END_DMA_DESCR_CH15 7 +/* I2S BT Instance DMA Descriptors */ +#define PLAYBACK_START_DMA_DESCR_CH8 8 +#define PLAYBACK_END_DMA_DESCR_CH8 9 +#define PLAYBACK_START_DMA_DESCR_CH9 10 +#define PLAYBACK_END_DMA_DESCR_CH9 11 + +#define CAPTURE_START_DMA_DESCR_CH10 12 +#define CAPTURE_END_DMA_DESCR_CH10 13 +#define CAPTURE_START_DMA_DESCR_CH11 14 +#define CAPTURE_END_DMA_DESCR_CH11 15 + #define mmACP_I2S_16BIT_RESOLUTION_EN 0x5209 #define ACP_I2S_MIC_16BIT_RESOLUTION_EN 0x01 #define ACP_I2S_SP_16BIT_RESOLUTION_EN 0x02 +#define ACP_I2S_BT_16BIT_RESOLUTION_EN 0x04 + enum acp_dma_priority_level { /* 0x0 Specifies the DMA channel is given normal priority */ ACP_DMA_PRIORITY_LEVEL_NORMAL = 0x0, @@ -84,16 +123,28 @@ struct audio_substream_data { struct page *pg; unsigned int order; u16 num_of_pages; + u16 i2s_play_instance; + u16 i2s_capture_instance; u16 direction; + u16 ch1; + u16 ch2; + u16 destination; + u16 dma_dscr_idx_1; + u16 dma_dscr_idx_2; + u32 pte_offset; + u32 sram_bank; + u32 byte_cnt_high; + u32 byte_cnt_low; uint64_t size; - u64 i2ssp_renderbytescount; - u64 i2ssp_capturebytescount; + u64 bytescount; void __iomem *acp_mmio; }; struct audio_drv_data { struct snd_pcm_substream *play_i2ssp_stream; struct snd_pcm_substream *capture_i2ssp_stream; + struct snd_pcm_substream *play_i2sbt_stream; + struct snd_pcm_substream *capture_i2sbt_stream; void __iomem *acp_mmio; u32 asic_type; }; -- 2.7.4

3 2

[alsa-devel] [PATCH v2 1/3] ASoC: dwc: I2S Controller instance param added
by Vijendar Mukunda 18 Apr '18

18 Apr '18

When multiple I2S controller instances created, i2s_instance parameter refers to i2s controller instance value. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- v1->v2: removed acp dma driver specific macros from dwc header file include/sound/designware_i2s.h | 4 ++++ sound/soc/dwc/dwc-i2s.c | 1 + sound/soc/dwc/local.h | 1 + 3 files changed, 6 insertions(+) diff --git a/include/sound/designware_i2s.h b/include/sound/designware_i2s.h index 830f5ca..2a94e7c 100644 --- a/include/sound/designware_i2s.h +++ b/include/sound/designware_i2s.h @@ -44,6 +44,10 @@ struct i2s_platform_data { int channel; u32 snd_fmts; u32 snd_rates; + /* i2s_instance parameter returns I2S controller instance value + * when multiple I2S controllers instantiated + */ + u32 i2s_instance; #define DW_I2S_QUIRK_COMP_REG_OFFSET (1 << 0) #define DW_I2S_QUIRK_COMP_PARAM1 (1 << 1) diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index 65112b9..58f81a4 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -553,6 +553,7 @@ static int dw_configure_dai_by_pd(struct dw_i2s_dev *dev, if (dev->quirks & DW_I2S_QUIRK_16BIT_IDX_OVERRIDE) idx = 1; + dev->i2s_instance = pdata->i2s_instance; /* Set DMA slaves info */ dev->play_dma_data.pd.data = pdata->play_dma_data; dev->capture_dma_data.pd.data = pdata->capture_dma_data; diff --git a/sound/soc/dwc/local.h b/sound/soc/dwc/local.h index 91dc70a..e89e464 100644 --- a/sound/soc/dwc/local.h +++ b/sound/soc/dwc/local.h @@ -98,6 +98,7 @@ struct dw_i2s_dev { u32 ccr; u32 xfer_resolution; u32 fifo_th; + u32 i2s_instance; /* data related to DMA transfers b/w i2s and DMAC */ union dw_i2s_snd_dma_data play_dma_data; -- 2.7.4

3 3

[alsa-devel] Applied "ASoC: topology: Fix build errors" to the asoc tree
by Mark Brown 18 Apr '18

18 Apr '18

The patch ASoC: topology: Fix build errors has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 24ada03555505205b0c8b8b796d52926600bf947 Mon Sep 17 00:00:00 2001 From: Mark Brown <broonie(a)kernel.org> Date: Wed, 18 Apr 2018 15:40:41 +0100 Subject: [PATCH] ASoC: topology: Fix build errors The two commits: 81e9b0a07889 ASoC: topology: Give more data to clients via callbacks 28aa6f7779f7 ASoC: topology: Add callback for DAPM route load/unload break the build so revert them. Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/soc-topology.h | 30 ++++++------------ sound/soc/intel/skylake/skl-pcm.c | 7 ++-- sound/soc/intel/skylake/skl-topology.c | 5 ++- sound/soc/intel/skylake/skl-topology.h | 20 +++++++++--- sound/soc/soc-topology.c | 44 ++++++++------------------ 5 files changed, 42 insertions(+), 64 deletions(-) diff --git a/include/sound/soc-topology.h b/include/sound/soc-topology.h index 401ef2c45d6c..f552c3f56368 100644 --- a/include/sound/soc-topology.h +++ b/include/sound/soc-topology.h @@ -30,9 +30,6 @@ struct snd_soc_dapm_context; struct snd_soc_card; struct snd_kcontrol_new; struct snd_soc_dai_link; -struct snd_soc_dai_driver; -struct snd_soc_dai; -struct snd_soc_dapm_route; /* object scan be loaded and unloaded in groups with identfying indexes */ #define SND_SOC_TPLG_INDEX_ALL 0 /* ID that matches all FW objects */ @@ -112,44 +109,35 @@ struct snd_soc_tplg_widget_events { struct snd_soc_tplg_ops { /* external kcontrol init - used for any driver specific init */ - int (*control_load)(struct snd_soc_component *, int index, + int (*control_load)(struct snd_soc_component *, struct snd_kcontrol_new *, struct snd_soc_tplg_ctl_hdr *); int (*control_unload)(struct snd_soc_component *, struct snd_soc_dobj *); - /* DAPM graph route element loading and unloading */ - int (*dapm_route_load)(struct snd_soc_component *, int index, - struct snd_soc_dapm_route *route); - int (*dapm_route_unload)(struct snd_soc_component *, - struct snd_soc_dobj *); - /* external widget init - used for any driver specific init */ - int (*widget_load)(struct snd_soc_component *, int index, + int (*widget_load)(struct snd_soc_component *, struct snd_soc_dapm_widget *, struct snd_soc_tplg_dapm_widget *); - int (*widget_ready)(struct snd_soc_component *, int index, + int (*widget_ready)(struct snd_soc_component *, struct snd_soc_dapm_widget *, struct snd_soc_tplg_dapm_widget *); int (*widget_unload)(struct snd_soc_component *, struct snd_soc_dobj *); /* FE DAI - used for any driver specific init */ - int (*dai_load)(struct snd_soc_component *, int index, - struct snd_soc_dai_driver *dai_drv, - struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai); - + int (*dai_load)(struct snd_soc_component *, + struct snd_soc_dai_driver *dai_drv); int (*dai_unload)(struct snd_soc_component *, struct snd_soc_dobj *); /* DAI link - used for any driver specific init */ - int (*link_load)(struct snd_soc_component *, int index, - struct snd_soc_dai_link *link, - struct snd_soc_tplg_link_config *cfg); + int (*link_load)(struct snd_soc_component *, + struct snd_soc_dai_link *link); int (*link_unload)(struct snd_soc_component *, struct snd_soc_dobj *); /* callback to handle vendor bespoke data */ - int (*vendor_load)(struct snd_soc_component *, int index, + int (*vendor_load)(struct snd_soc_component *, struct snd_soc_tplg_hdr *); int (*vendor_unload)(struct snd_soc_component *, struct snd_soc_tplg_hdr *); @@ -158,7 +146,7 @@ struct snd_soc_tplg_ops { void (*complete)(struct snd_soc_component *); /* manifest - optional to inform component of manifest */ - int (*manifest)(struct snd_soc_component *, int index, + int (*manifest)(struct snd_soc_component *, struct snd_soc_tplg_manifest *); /* vendor specific kcontrol handlers available for binding */ diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 1f4dd08d36c5..afa86b9e4dcf 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -1017,11 +1017,10 @@ static struct snd_soc_dai_driver skl_platform_dai[] = { }, }; -int skl_dai_load(struct snd_soc_component *cmp, int index, - struct snd_soc_dai_driver *dai_drv, - struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai) +int skl_dai_load(struct snd_soc_component *cmp, + struct snd_soc_dai_driver *pcm_dai) { - dai_drv->ops = &skl_pcm_dai_ops; + pcm_dai->ops = &skl_pcm_dai_ops; return 0; } diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 6ac081f1f215..3b1dca419883 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -2851,7 +2851,7 @@ void skl_cleanup_resources(struct skl *skl) * information to the driver about module and pipeline parameters which DSP * FW expects like ids, resource values, formats etc */ -static int skl_tplg_widget_load(struct snd_soc_component *cmpnt, int index, +static int skl_tplg_widget_load(struct snd_soc_component *cmpnt, struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w) { @@ -2958,7 +2958,6 @@ static int skl_init_enum_data(struct device *dev, struct soc_enum *se, } static int skl_tplg_control_load(struct snd_soc_component *cmpnt, - int index, struct snd_kcontrol_new *kctl, struct snd_soc_tplg_ctl_hdr *hdr) { @@ -3447,7 +3446,7 @@ static int skl_tplg_get_manifest_data(struct snd_soc_tplg_manifest *manifest, return 0; } -static int skl_manifest_load(struct snd_soc_component *cmpnt, int index, +static int skl_manifest_load(struct snd_soc_component *cmpnt, struct snd_soc_tplg_manifest *manifest) { struct hdac_ext_bus *ebus = snd_soc_component_get_drvdata(cmpnt); diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h index 77857c598eed..b1e0667c0ae0 100644 --- a/sound/soc/intel/skylake/skl-topology.h +++ b/sound/soc/intel/skylake/skl-topology.h @@ -221,9 +221,18 @@ struct skl_mod_inst_map { u16 inst_id; }; +struct skl_uuid_inst_map { + u16 inst_id; + u16 reserved; + uuid_le mod_uuid; +} __packed; + struct skl_kpb_params { u32 num_modules; - struct skl_mod_inst_map map[0]; + union { + struct skl_mod_inst_map map[0]; + struct skl_uuid_inst_map map_uuid[0]; + } u; }; struct skl_module_inst_id { @@ -460,7 +469,7 @@ int skl_dsp_set_dma_control(struct skl_sst *ctx, u32 *caps, u32 caps_size, u32 node_id); void skl_tplg_set_be_dmic_config(struct snd_soc_dai *dai, struct skl_pipe_params *params, int stream); -int skl_tplg_init(struct snd_soc_platform *platform, +int skl_tplg_init(struct snd_soc_component *component, struct hdac_ext_bus *ebus); struct skl_module_cfg *skl_tplg_fe_get_cpr_module( struct snd_soc_dai *dai, int stream); @@ -503,7 +512,8 @@ int skl_pcm_host_dma_prepare(struct device *dev, int skl_pcm_link_dma_prepare(struct device *dev, struct skl_pipe_params *params); -int skl_dai_load(struct snd_soc_component *, int index, - struct snd_soc_dai_driver *dai_drv, - struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai); +int skl_dai_load(struct snd_soc_component *cmp, + struct snd_soc_dai_driver *pcm_dai); +void skl_tplg_add_moduleid_in_bind_params(struct skl *skl, + struct snd_soc_dapm_widget *w); #endif diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index b95a9ab0b526..de08693be9e1 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -315,7 +315,7 @@ static int soc_tplg_vendor_load_(struct soc_tplg *tplg, int ret = 0; if (tplg->comp && tplg->ops && tplg->ops->vendor_load) - ret = tplg->ops->vendor_load(tplg->comp, tplg->index, hdr); + ret = tplg->ops->vendor_load(tplg->comp, hdr); else { dev_err(tplg->dev, "ASoC: no vendor load callback for ID %d\n", hdr->vendor_type); @@ -347,8 +347,7 @@ static int soc_tplg_widget_load(struct soc_tplg *tplg, struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w) { if (tplg->comp && tplg->ops && tplg->ops->widget_load) - return tplg->ops->widget_load(tplg->comp, tplg->index, w, - tplg_w); + return tplg->ops->widget_load(tplg->comp, w, tplg_w); return 0; } @@ -359,30 +358,27 @@ static int soc_tplg_widget_ready(struct soc_tplg *tplg, struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w) { if (tplg->comp && tplg->ops && tplg->ops->widget_ready) - return tplg->ops->widget_ready(tplg->comp, tplg->index, w, - tplg_w); + return tplg->ops->widget_ready(tplg->comp, w, tplg_w); return 0; } /* pass DAI configurations to component driver for extra initialization */ static int soc_tplg_dai_load(struct soc_tplg *tplg, - struct snd_soc_dai_driver *dai_drv, - struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai) + struct snd_soc_dai_driver *dai_drv) { if (tplg->comp && tplg->ops && tplg->ops->dai_load) - return tplg->ops->dai_load(tplg->comp, tplg->index, dai_drv, - pcm, dai); + return tplg->ops->dai_load(tplg->comp, dai_drv); return 0; } /* pass link configurations to component driver for extra initialization */ static int soc_tplg_dai_link_load(struct soc_tplg *tplg, - struct snd_soc_dai_link *link, struct snd_soc_tplg_link_config *cfg) + struct snd_soc_dai_link *link) { if (tplg->comp && tplg->ops && tplg->ops->link_load) - return tplg->ops->link_load(tplg->comp, tplg->index, link, cfg); + return tplg->ops->link_load(tplg->comp, link); return 0; } @@ -703,8 +699,7 @@ static int soc_tplg_init_kcontrol(struct soc_tplg *tplg, struct snd_kcontrol_new *k, struct snd_soc_tplg_ctl_hdr *hdr) { if (tplg->comp && tplg->ops && tplg->ops->control_load) - return tplg->ops->control_load(tplg->comp, tplg->index, k, - hdr); + return tplg->ops->control_load(tplg->comp, k, hdr); return 0; } @@ -1161,17 +1156,6 @@ static int soc_tplg_kcontrol_elems_load(struct soc_tplg *tplg, return 0; } -/* optionally pass new dynamic kcontrol to component driver. */ -static int soc_tplg_add_route(struct soc_tplg *tplg, - struct snd_soc_dapm_route *route) -{ - if (tplg->comp && tplg->ops && tplg->ops->dapm_route_load) - return tplg->ops->dapm_route_load(tplg->comp, tplg->index, - route); - - return 0; -} - static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg, struct snd_soc_tplg_hdr *hdr) { @@ -1220,8 +1204,6 @@ static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg, else route.control = elem->control; - soc_tplg_add_route(tplg, &route); - /* add route, but keep going if some fail */ snd_soc_dapm_add_routes(dapm, &route, 1); } @@ -1776,7 +1758,7 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg, dai_drv->compress_new = snd_soc_new_compress; /* pass control to component driver for optional further init */ - ret = soc_tplg_dai_load(tplg, dai_drv, pcm, NULL); + ret = soc_tplg_dai_load(tplg, dai_drv); if (ret < 0) { dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n"); kfree(dai_drv); @@ -1846,7 +1828,7 @@ static int soc_tplg_fe_link_create(struct soc_tplg *tplg, set_link_flags(link, pcm->flag_mask, pcm->flags); /* pass control to component driver for optional further init */ - ret = soc_tplg_dai_link_load(tplg, link, NULL); + ret = soc_tplg_dai_link_load(tplg, link); if (ret < 0) { dev_err(tplg->comp->dev, "ASoC: FE link loading failed\n"); kfree(link); @@ -2154,7 +2136,7 @@ static int soc_tplg_link_config(struct soc_tplg *tplg, set_link_flags(link, cfg->flag_mask, cfg->flags); /* pass control to component driver for optional further init */ - ret = soc_tplg_dai_link_load(tplg, link, cfg); + ret = soc_tplg_dai_link_load(tplg, link); if (ret < 0) { dev_err(tplg->dev, "ASoC: physical link loading failed\n"); return ret; @@ -2276,7 +2258,7 @@ static int soc_tplg_dai_config(struct soc_tplg *tplg, set_dai_flags(dai_drv, d->flag_mask, d->flags); /* pass control to component driver for optional further init */ - ret = soc_tplg_dai_load(tplg, dai_drv, NULL, dai); + ret = soc_tplg_dai_load(tplg, dai_drv); if (ret < 0) { dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n"); return ret; @@ -2382,7 +2364,7 @@ static int soc_tplg_manifest_load(struct soc_tplg *tplg, /* pass control to component driver for optional further init */ if (tplg->comp && tplg->ops && tplg->ops->manifest) - return tplg->ops->manifest(tplg->comp, tplg->index, _manifest); + return tplg->ops->manifest(tplg->comp, _manifest); if (!abi_match) /* free the duplicated one */ kfree(_manifest); -- 2.17.0

1 0

[alsa-devel] [asoc:for-4.18 48/51] sound/soc/intel/skylake/skl-pcm.c:1351:4: error: implicit declaration of function 'skl_tplg_add_moduleid_in_bind_params'; did you mean 'skl_tplg_update_pipe_params'?
by kbuild test robot 18 Apr '18

18 Apr '18

Hi Liam, FYI, the error/warning still remains. tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.18 head: 221dd96c30a7c65b24ead7fdd7645abb99506ce2 commit: 81e9b0a078894841a50a8dd666fd64ca452a2a50 [48/51] ASoC: topology: Give more data to clients via callbacks config: x86_64-federa-25 (attached as .config) compiler: gcc-7 (Debian 7.3.0-16) 7.3.0 reproduce: git checkout 81e9b0a078894841a50a8dd666fd64ca452a2a50 # save the attached .config to linux build tree make ARCH=x86_64 All errors (new ones prefixed by >>): sound/soc/intel/skylake/skl-pcm.c: In function 'skl_populate_modules': >> sound/soc/intel/skylake/skl-pcm.c:1351:4: error: implicit declaration of function 'skl_tplg_add_moduleid_in_bind_params'; did you mean 'skl_tplg_update_pipe_params'? [-Werror=implicit-function-declaration] skl_tplg_add_moduleid_in_bind_params(skl, w); ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ skl_tplg_update_pipe_params sound/soc/intel/skylake/skl-pcm.c: In function 'skl_platform_soc_probe': >> sound/soc/intel/skylake/skl-pcm.c:1372:23: error: passing argument 1 of 'skl_tplg_init' from incompatible pointer type [-Werror=incompatible-pointer-types] ret = skl_tplg_init(component, ebus); ^~~~~~~~~ In file included from sound/soc/intel/skylake/skl-pcm.c:28:0: sound/soc/intel/skylake/skl-topology.h:463:5: note: expected 'struct snd_soc_platform *' but argument is of type 'struct snd_soc_component *' int skl_tplg_init(struct snd_soc_platform *platform, ^~~~~~~~~~~~~ cc1: some warnings being treated as errors -- sound/soc/intel/skylake/skl-topology.c: In function 'skl_fill_sink_instance_id': >> sound/soc/intel/skylake/skl-topology.c:833:45: error: 'struct skl_kpb_params' has no member named 'u' struct skl_mod_inst_map *inst = kpb_params->u.map; ^~ sound/soc/intel/skylake/skl-topology.c: In function 'skl_tplg_find_moduleid_from_uuid': sound/soc/intel/skylake/skl-topology.c:953:17: error: 'struct skl_kpb_params' has no member named 'u' &uuid_params->u.map_uuid[i].mod_uuid); ^~ sound/soc/intel/skylake/skl-topology.c:959:10: error: 'struct skl_kpb_params' has no member named 'u' params->u.map[i].mod_id = module_id; ^~ sound/soc/intel/skylake/skl-topology.c:960:10: error: 'struct skl_kpb_params' has no member named 'u' params->u.map[i].inst_id = ^~ sound/soc/intel/skylake/skl-topology.c:961:16: error: 'struct skl_kpb_params' has no member named 'u' uuid_params->u.map_uuid[i].inst_id; ^~ sound/soc/intel/skylake/skl-topology.c: At top level: >> sound/soc/intel/skylake/skl-topology.c:3544:5: error: conflicting types for 'skl_tplg_init' int skl_tplg_init(struct snd_soc_component *component, struct hdac_ext_bus *ebus) ^~~~~~~~~~~~~ In file included from sound/soc/intel/skylake/skl-topology.c:27:0: sound/soc/intel/skylake/skl-topology.h:463:5: note: previous declaration of 'skl_tplg_init' was here int skl_tplg_init(struct snd_soc_platform *platform, ^~~~~~~~~~~~~ vim +1351 sound/soc/intel/skylake/skl-pcm.c b26199ea Jeeja KP 2017-03-24 1330 64cb1d0a Vinod Koul 2016-08-10 1331 static int skl_populate_modules(struct skl *skl) 64cb1d0a Vinod Koul 2016-08-10 1332 { 64cb1d0a Vinod Koul 2016-08-10 1333 struct skl_pipeline *p; 64cb1d0a Vinod Koul 2016-08-10 1334 struct skl_pipe_module *m; 64cb1d0a Vinod Koul 2016-08-10 1335 struct snd_soc_dapm_widget *w; 64cb1d0a Vinod Koul 2016-08-10 1336 struct skl_module_cfg *mconfig; b26199ea Jeeja KP 2017-03-24 1337 int ret = 0; 64cb1d0a Vinod Koul 2016-08-10 1338 64cb1d0a Vinod Koul 2016-08-10 1339 list_for_each_entry(p, &skl->ppl_list, node) { 64cb1d0a Vinod Koul 2016-08-10 1340 list_for_each_entry(m, &p->pipe->w_list, node) { 64cb1d0a Vinod Koul 2016-08-10 1341 w = m->w; 64cb1d0a Vinod Koul 2016-08-10 1342 mconfig = w->priv; 64cb1d0a Vinod Koul 2016-08-10 1343 b26199ea Jeeja KP 2017-03-24 1344 ret = skl_get_module_info(skl, mconfig); 64cb1d0a Vinod Koul 2016-08-10 1345 if (ret < 0) { 64cb1d0a Vinod Koul 2016-08-10 1346 dev_err(skl->skl_sst->dev, b26199ea Jeeja KP 2017-03-24 1347 "query module info failed\n"); b26199ea Jeeja KP 2017-03-24 1348 return ret; 64cb1d0a Vinod Koul 2016-08-10 1349 } f7a9f772 Sriram Periyasamy 2018-01-27 1350 f7a9f772 Sriram Periyasamy 2018-01-27 @1351 skl_tplg_add_moduleid_in_bind_params(skl, w); 64cb1d0a Vinod Koul 2016-08-10 1352 } 64cb1d0a Vinod Koul 2016-08-10 1353 } b26199ea Jeeja KP 2017-03-24 1354 64cb1d0a Vinod Koul 2016-08-10 1355 return ret; 64cb1d0a Vinod Koul 2016-08-10 1356 } 64cb1d0a Vinod Koul 2016-08-10 1357 56b03b4c Kuninori Morimoto 2018-01-29 1358 static int skl_platform_soc_probe(struct snd_soc_component *component) b663a8c5 Jeeja KP 2015-10-07 1359 { 56b03b4c Kuninori Morimoto 2018-01-29 1360 struct hdac_ext_bus *ebus = dev_get_drvdata(component->dev); fe3f4442 Dharageswari R 2016-06-03 1361 struct skl *skl = ebus_to_skl(ebus); 78cdbbda Vinod Koul 2016-07-26 1362 const struct skl_dsp_ops *ops; fe3f4442 Dharageswari R 2016-06-03 1363 int ret; b663a8c5 Jeeja KP 2015-10-07 1364 56b03b4c Kuninori Morimoto 2018-01-29 1365 pm_runtime_get_sync(component->dev); ec8ae570 Vinod Koul 2016-08-04 1366 if ((ebus_to_hbus(ebus))->ppcap) { 56b03b4c Kuninori Morimoto 2018-01-29 1367 skl->component = component; 5cdf6c09 Vinod Koul 2017-06-30 1368 5cdf6c09 Vinod Koul 2017-06-30 1369 /* init debugfs */ 5cdf6c09 Vinod Koul 2017-06-30 1370 skl->debugfs = skl_debugfs_init(skl); 5cdf6c09 Vinod Koul 2017-06-30 1371 56b03b4c Kuninori Morimoto 2018-01-29 @1372 ret = skl_tplg_init(component, ebus); fe3f4442 Dharageswari R 2016-06-03 1373 if (ret < 0) { 56b03b4c Kuninori Morimoto 2018-01-29 1374 dev_err(component->dev, "Failed to init topology!\n"); fe3f4442 Dharageswari R 2016-06-03 1375 return ret; fe3f4442 Dharageswari R 2016-06-03 1376 } 78cdbbda Vinod Koul 2016-07-26 1377 78cdbbda Vinod Koul 2016-07-26 1378 /* load the firmwares, since all is set */ 78cdbbda Vinod Koul 2016-07-26 1379 ops = skl_get_dsp_ops(skl->pci->device); 78cdbbda Vinod Koul 2016-07-26 1380 if (!ops) 78cdbbda Vinod Koul 2016-07-26 1381 return -EIO; 78cdbbda Vinod Koul 2016-07-26 1382 78cdbbda Vinod Koul 2016-07-26 1383 if (skl->skl_sst->is_first_boot == false) { 56b03b4c Kuninori Morimoto 2018-01-29 1384 dev_err(component->dev, "DSP reports first boot done!!!\n"); 78cdbbda Vinod Koul 2016-07-26 1385 return -EIO; 78cdbbda Vinod Koul 2016-07-26 1386 } 78cdbbda Vinod Koul 2016-07-26 1387 fc9fdd61 Sanyog Kale 2018-03-13 1388 /* fc9fdd61 Sanyog Kale 2018-03-13 1389 * Disable dynamic clock and power gating during firmware fc9fdd61 Sanyog Kale 2018-03-13 1390 * and library download fc9fdd61 Sanyog Kale 2018-03-13 1391 */ 56b03b4c Kuninori Morimoto 2018-01-29 1392 skl->skl_sst->enable_miscbdcge(component->dev, false); fc9fdd61 Sanyog Kale 2018-03-13 1393 skl->skl_sst->clock_power_gating(component->dev, false); d5cc0a1f Pardha Saradhi K 2018-01-02 1394 56b03b4c Kuninori Morimoto 2018-01-29 1395 ret = ops->init_fw(component->dev, skl->skl_sst); 56b03b4c Kuninori Morimoto 2018-01-29 1396 skl->skl_sst->enable_miscbdcge(component->dev, true); fc9fdd61 Sanyog Kale 2018-03-13 1397 skl->skl_sst->clock_power_gating(component->dev, true); 78cdbbda Vinod Koul 2016-07-26 1398 if (ret < 0) { 56b03b4c Kuninori Morimoto 2018-01-29 1399 dev_err(component->dev, "Failed to boot first fw: %d\n", ret); 78cdbbda Vinod Koul 2016-07-26 1400 return ret; 78cdbbda Vinod Koul 2016-07-26 1401 } 64cb1d0a Vinod Koul 2016-08-10 1402 skl_populate_modules(skl); a26a3f53 Pardha Saradhi K 2016-11-03 1403 skl->skl_sst->update_d0i3c = skl_update_d0i3c; cb729d80 G Kranthi 2017-03-13 1404 skl_dsp_enable_notification(skl->skl_sst, false); 9452314d Pradeep Tewani 2017-12-06 1405 9452314d Pradeep Tewani 2017-12-06 1406 if (skl->cfg.astate_cfg != NULL) { 9452314d Pradeep Tewani 2017-12-06 1407 skl_dsp_set_astate_cfg(skl->skl_sst, 9452314d Pradeep Tewani 2017-12-06 1408 skl->cfg.astate_cfg->count, 9452314d Pradeep Tewani 2017-12-06 1409 skl->cfg.astate_cfg); 9452314d Pradeep Tewani 2017-12-06 1410 } fe3f4442 Dharageswari R 2016-06-03 1411 } 56b03b4c Kuninori Morimoto 2018-01-29 1412 pm_runtime_mark_last_busy(component->dev); 56b03b4c Kuninori Morimoto 2018-01-29 1413 pm_runtime_put_autosuspend(component->dev); b663a8c5 Jeeja KP 2015-10-07 1414 b663a8c5 Jeeja KP 2015-10-07 1415 return 0; b663a8c5 Jeeja KP 2015-10-07 1416 } 56b03b4c Kuninori Morimoto 2018-01-29 1417 :::::: The code at line 1351 was first introduced by commit :::::: f7a9f77256e7fbd6150651bfab44f60c39f0b7a2 ASoC: Intel: Skylake: Find module id from UUID for bind params :::::: TO: Sriram Periyasamy <sriramx.periyasamy(a)intel.com> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH v2 3/3] ASoC: amd: enabling bt i2s config after acp reset
by Vijendar Mukunda 18 Apr '18

18 Apr '18

On ST/CZ based platforms, for specific platform bt uart mux to be defined for bt i2s. By default, these pins will be used for uart. After acp reset , it requires to reprogram bt i2s config mux pins to enable bt i2s instance. added bt i2s enablement sequence during acp init. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- v1->v2: fixed kbuild errors sound/soc/amd/acp-da7219-max98357a.c | 2 ++ sound/soc/amd/acp-pcm-dma.c | 9 +++++++++ sound/soc/amd/acp.h | 1 + 3 files changed, 12 insertions(+) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index b205c78..9ff2138 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -44,6 +44,7 @@ static struct snd_soc_jack cz_jack; struct clk *da7219_dai_clk; +extern int bt_pad_enable; static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) { @@ -251,6 +252,7 @@ static int cz_probe(struct platform_device *pdev) cz_card.name, ret); return ret; } + bt_pad_enable = device_property_read_bool(&pdev->dev, "bt-pad-enable"); return 0; } diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 7c392fe..b52c660 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -46,6 +46,8 @@ #define DRV_NAME "acp_audio_dma" +bool bt_pad_enable = false; +EXPORT_SYMBOL(bt_pad_enable); static const struct snd_pcm_hardware acp_pcm_hardware_playback = { .info = SNDRV_PCM_INFO_INTERLEAVED | @@ -525,6 +527,13 @@ static int acp_init(void __iomem *acp_mmio, u32 asic_type) val &= ~ACP_SOFT_RESET__SoftResetAud_MASK; acp_reg_write(val, acp_mmio, mmACP_SOFT_RESET); + /* For BT instance change pins from UART to BT */ + if (bt_pad_enable) { + val = acp_reg_read(acp_mmio, mmACP_BT_UART_PAD_SEL); + val |= ACP_BT_UART_PAD_SELECT_MASK; + acp_reg_write(val, acp_mmio, mmACP_BT_UART_PAD_SEL); + } + /* initiailize Onion control DAGB register */ acp_reg_write(ACP_ONION_CNTL_DEFAULT, acp_mmio, mmACP_AXI2DAGB_ONION_CNTL); diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index 95c39a3..520a08f 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -110,6 +110,7 @@ #define ACP_I2S_MIC_16BIT_RESOLUTION_EN 0x01 #define ACP_I2S_SP_16BIT_RESOLUTION_EN 0x02 #define ACP_I2S_BT_16BIT_RESOLUTION_EN 0x04 +#define ACP_BT_UART_PAD_SELECT_MASK 0x1 enum acp_dma_priority_level { /* 0x0 Specifies the DMA channel is given normal priority */ -- 2.7.4

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[alsa-devel] [PATCH 1/4] ASoC: dwc: I2S Controller instance param added
by Vijendar Mukunda 18 Apr '18

18 Apr '18

When multiple I2S controller instances created, i2s_instance parameter refers to i2s controller instance value. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Reviewed-by: Alex Deucher <alexander.deucher(a)amd.com> --- include/sound/designware_i2s.h | 6 ++++++ sound/soc/dwc/dwc-i2s.c | 1 + sound/soc/dwc/local.h | 1 + 3 files changed, 8 insertions(+) diff --git a/include/sound/designware_i2s.h b/include/sound/designware_i2s.h index 830f5ca..8113759 100644 --- a/include/sound/designware_i2s.h +++ b/include/sound/designware_i2s.h @@ -44,6 +44,10 @@ struct i2s_platform_data { int channel; u32 snd_fmts; u32 snd_rates; + /* i2s_instance parameter returns I2S controller instance value + * when multiple I2S controllers instantiated + */ + u32 i2s_instance; #define DW_I2S_QUIRK_COMP_REG_OFFSET (1 << 0) #define DW_I2S_QUIRK_COMP_PARAM1 (1 << 1) @@ -74,5 +78,7 @@ struct i2s_dma_data { #define FOUR_CHANNEL_SUPPORT 4 /* up to 3.1 */ #define SIX_CHANNEL_SUPPORT 6 /* up to 5.1 */ #define EIGHT_CHANNEL_SUPPORT 8 /* up to 7.1 */ +#define I2S_SP_INSTANCE 1 +#define I2S_BT_INSTANCE 2 #endif /* __SOUND_DESIGNWARE_I2S_H */ diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index 65112b9..58f81a4 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -553,6 +553,7 @@ static int dw_configure_dai_by_pd(struct dw_i2s_dev *dev, if (dev->quirks & DW_I2S_QUIRK_16BIT_IDX_OVERRIDE) idx = 1; + dev->i2s_instance = pdata->i2s_instance; /* Set DMA slaves info */ dev->play_dma_data.pd.data = pdata->play_dma_data; dev->capture_dma_data.pd.data = pdata->capture_dma_data; diff --git a/sound/soc/dwc/local.h b/sound/soc/dwc/local.h index 91dc70a..e89e464 100644 --- a/sound/soc/dwc/local.h +++ b/sound/soc/dwc/local.h @@ -98,6 +98,7 @@ struct dw_i2s_dev { u32 ccr; u32 xfer_resolution; u32 fifo_th; + u32 i2s_instance; /* data related to DMA transfers b/w i2s and DMAC */ union dw_i2s_snd_dma_data play_dma_data; -- 2.7.4

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[alsa-devel] [asoc:for-4.18 48/50] sound/soc/intel/skylake/skl-pcm.c:1372:37: sparse: incorrect type in argument 1 (different base types)
by kbuild test robot 17 Apr '18

17 Apr '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.18 head: 5db6aab6f36f7560dc95f7ca340d5632b7a3be6a commit: 81e9b0a078894841a50a8dd666fd64ca452a2a50 [48/50] ASoC: topology: Give more data to clients via callbacks reproduce: # apt-get install sparse git checkout 81e9b0a078894841a50a8dd666fd64ca452a2a50 make ARCH=x86_64 allmodconfig make C=1 CF=-D__CHECK_ENDIAN__ sparse warnings: (new ones prefixed by >>) sound/soc/intel/skylake/skl-pcm.c:159:43: sparse: incorrect type in argument 3 (different base types) @@ expected unsigned int [unsigned] format @@ got restricted snd_unsigned int [unsigned] format @@ sound/soc/intel/skylake/skl-pcm.c:159:43: expected unsigned int [unsigned] format sound/soc/intel/skylake/skl-pcm.c:159:43: got restricted snd_pcm_format_t [usertype] format sound/soc/intel/skylake/skl-pcm.c:195:47: sparse: incorrect type in argument 3 (different base types) @@ expected unsigned int [unsigned] format @@ got restricted snd_unsigned int [unsigned] format @@ sound/soc/intel/skylake/skl-pcm.c:195:47: expected unsigned int [unsigned] format sound/soc/intel/skylake/skl-pcm.c:195:47: got restricted snd_pcm_format_t [usertype] format sound/soc/intel/skylake/skl-pcm.c:1351:25: sparse: undefined identifier 'skl_tplg_add_moduleid_in_bind_params' >> sound/soc/intel/skylake/skl-pcm.c:1372:37: sparse: incorrect type in argument 1 (different base types) @@ expected struct snd_soc_platform *platform @@ got sstruct snd_soc_platform *platform @@ sound/soc/intel/skylake/skl-pcm.c:1372:37: expected struct snd_soc_platform *platform sound/soc/intel/skylake/skl-pcm.c:1372:37: got struct snd_soc_component *component >> sound/soc/intel/skylake/skl-pcm.c:1351:61: sparse: call with no type! sound/soc/intel/skylake/skl-pcm.c: In function 'skl_populate_modules': sound/soc/intel/skylake/skl-pcm.c:1351:4: error: implicit declaration of function 'skl_tplg_add_moduleid_in_bind_params'; did you mean 'skl_tplg_update_pipe_params'? [-Werror=implicit-function-declaration] skl_tplg_add_moduleid_in_bind_params(skl, w); ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ skl_tplg_update_pipe_params sound/soc/intel/skylake/skl-pcm.c: In function 'skl_platform_soc_probe': sound/soc/intel/skylake/skl-pcm.c:1372:23: error: passing argument 1 of 'skl_tplg_init' from incompatible pointer type [-Werror=incompatible-pointer-types] ret = skl_tplg_init(component, ebus); ^~~~~~~~~ In file included from sound/soc/intel/skylake/skl-pcm.c:28:0: sound/soc/intel/skylake/skl-topology.h:463:5: note: expected 'struct snd_soc_platform *' but argument is of type 'struct snd_soc_component *' int skl_tplg_init(struct snd_soc_platform *platform, ^~~~~~~~~~~~~ cc1: some warnings being treated as errors vim +1372 sound/soc/intel/skylake/skl-pcm.c b26199ea Jeeja KP 2017-03-24 1330 64cb1d0a Vinod Koul 2016-08-10 1331 static int skl_populate_modules(struct skl *skl) 64cb1d0a Vinod Koul 2016-08-10 1332 { 64cb1d0a Vinod Koul 2016-08-10 1333 struct skl_pipeline *p; 64cb1d0a Vinod Koul 2016-08-10 1334 struct skl_pipe_module *m; 64cb1d0a Vinod Koul 2016-08-10 1335 struct snd_soc_dapm_widget *w; 64cb1d0a Vinod Koul 2016-08-10 1336 struct skl_module_cfg *mconfig; b26199ea Jeeja KP 2017-03-24 1337 int ret = 0; 64cb1d0a Vinod Koul 2016-08-10 1338 64cb1d0a Vinod Koul 2016-08-10 1339 list_for_each_entry(p, &skl->ppl_list, node) { 64cb1d0a Vinod Koul 2016-08-10 1340 list_for_each_entry(m, &p->pipe->w_list, node) { 64cb1d0a Vinod Koul 2016-08-10 1341 w = m->w; 64cb1d0a Vinod Koul 2016-08-10 1342 mconfig = w->priv; 64cb1d0a Vinod Koul 2016-08-10 1343 b26199ea Jeeja KP 2017-03-24 1344 ret = skl_get_module_info(skl, mconfig); 64cb1d0a Vinod Koul 2016-08-10 1345 if (ret < 0) { 64cb1d0a Vinod Koul 2016-08-10 1346 dev_err(skl->skl_sst->dev, b26199ea Jeeja KP 2017-03-24 1347 "query module info failed\n"); b26199ea Jeeja KP 2017-03-24 1348 return ret; 64cb1d0a Vinod Koul 2016-08-10 1349 } f7a9f772 Sriram Periyasamy 2018-01-27 1350 f7a9f772 Sriram Periyasamy 2018-01-27 @1351 skl_tplg_add_moduleid_in_bind_params(skl, w); 64cb1d0a Vinod Koul 2016-08-10 1352 } 64cb1d0a Vinod Koul 2016-08-10 1353 } b26199ea Jeeja KP 2017-03-24 1354 64cb1d0a Vinod Koul 2016-08-10 1355 return ret; 64cb1d0a Vinod Koul 2016-08-10 1356 } 64cb1d0a Vinod Koul 2016-08-10 1357 56b03b4c Kuninori Morimoto 2018-01-29 1358 static int skl_platform_soc_probe(struct snd_soc_component *component) b663a8c5 Jeeja KP 2015-10-07 1359 { 56b03b4c Kuninori Morimoto 2018-01-29 1360 struct hdac_ext_bus *ebus = dev_get_drvdata(component->dev); fe3f4442 Dharageswari R 2016-06-03 1361 struct skl *skl = ebus_to_skl(ebus); 78cdbbda Vinod Koul 2016-07-26 1362 const struct skl_dsp_ops *ops; fe3f4442 Dharageswari R 2016-06-03 1363 int ret; b663a8c5 Jeeja KP 2015-10-07 1364 56b03b4c Kuninori Morimoto 2018-01-29 1365 pm_runtime_get_sync(component->dev); ec8ae570 Vinod Koul 2016-08-04 1366 if ((ebus_to_hbus(ebus))->ppcap) { 56b03b4c Kuninori Morimoto 2018-01-29 1367 skl->component = component; 5cdf6c09 Vinod Koul 2017-06-30 1368 5cdf6c09 Vinod Koul 2017-06-30 1369 /* init debugfs */ 5cdf6c09 Vinod Koul 2017-06-30 1370 skl->debugfs = skl_debugfs_init(skl); 5cdf6c09 Vinod Koul 2017-06-30 1371 56b03b4c Kuninori Morimoto 2018-01-29 @1372 ret = skl_tplg_init(component, ebus); fe3f4442 Dharageswari R 2016-06-03 1373 if (ret < 0) { 56b03b4c Kuninori Morimoto 2018-01-29 1374 dev_err(component->dev, "Failed to init topology!\n"); fe3f4442 Dharageswari R 2016-06-03 1375 return ret; fe3f4442 Dharageswari R 2016-06-03 1376 } 78cdbbda Vinod Koul 2016-07-26 1377 78cdbbda Vinod Koul 2016-07-26 1378 /* load the firmwares, since all is set */ 78cdbbda Vinod Koul 2016-07-26 1379 ops = skl_get_dsp_ops(skl->pci->device); 78cdbbda Vinod Koul 2016-07-26 1380 if (!ops) 78cdbbda Vinod Koul 2016-07-26 1381 return -EIO; 78cdbbda Vinod Koul 2016-07-26 1382 78cdbbda Vinod Koul 2016-07-26 1383 if (skl->skl_sst->is_first_boot == false) { 56b03b4c Kuninori Morimoto 2018-01-29 1384 dev_err(component->dev, "DSP reports first boot done!!!\n"); 78cdbbda Vinod Koul 2016-07-26 1385 return -EIO; 78cdbbda Vinod Koul 2016-07-26 1386 } 78cdbbda Vinod Koul 2016-07-26 1387 fc9fdd61 Sanyog Kale 2018-03-13 1388 /* fc9fdd61 Sanyog Kale 2018-03-13 1389 * Disable dynamic clock and power gating during firmware fc9fdd61 Sanyog Kale 2018-03-13 1390 * and library download fc9fdd61 Sanyog Kale 2018-03-13 1391 */ 56b03b4c Kuninori Morimoto 2018-01-29 1392 skl->skl_sst->enable_miscbdcge(component->dev, false); fc9fdd61 Sanyog Kale 2018-03-13 1393 skl->skl_sst->clock_power_gating(component->dev, false); d5cc0a1f Pardha Saradhi K 2018-01-02 1394 56b03b4c Kuninori Morimoto 2018-01-29 1395 ret = ops->init_fw(component->dev, skl->skl_sst); 56b03b4c Kuninori Morimoto 2018-01-29 1396 skl->skl_sst->enable_miscbdcge(component->dev, true); fc9fdd61 Sanyog Kale 2018-03-13 1397 skl->skl_sst->clock_power_gating(component->dev, true); 78cdbbda Vinod Koul 2016-07-26 1398 if (ret < 0) { 56b03b4c Kuninori Morimoto 2018-01-29 1399 dev_err(component->dev, "Failed to boot first fw: %d\n", ret); 78cdbbda Vinod Koul 2016-07-26 1400 return ret; 78cdbbda Vinod Koul 2016-07-26 1401 } 64cb1d0a Vinod Koul 2016-08-10 1402 skl_populate_modules(skl); a26a3f53 Pardha Saradhi K 2016-11-03 1403 skl->skl_sst->update_d0i3c = skl_update_d0i3c; cb729d80 G Kranthi 2017-03-13 1404 skl_dsp_enable_notification(skl->skl_sst, false); 9452314d Pradeep Tewani 2017-12-06 1405 9452314d Pradeep Tewani 2017-12-06 1406 if (skl->cfg.astate_cfg != NULL) { 9452314d Pradeep Tewani 2017-12-06 1407 skl_dsp_set_astate_cfg(skl->skl_sst, 9452314d Pradeep Tewani 2017-12-06 1408 skl->cfg.astate_cfg->count, 9452314d Pradeep Tewani 2017-12-06 1409 skl->cfg.astate_cfg); 9452314d Pradeep Tewani 2017-12-06 1410 } fe3f4442 Dharageswari R 2016-06-03 1411 } 56b03b4c Kuninori Morimoto 2018-01-29 1412 pm_runtime_mark_last_busy(component->dev); 56b03b4c Kuninori Morimoto 2018-01-29 1413 pm_runtime_put_autosuspend(component->dev); b663a8c5 Jeeja KP 2015-10-07 1414 b663a8c5 Jeeja KP 2015-10-07 1415 return 0; b663a8c5 Jeeja KP 2015-10-07 1416 } 56b03b4c Kuninori Morimoto 2018-01-29 1417 :::::: The code at line 1372 was first introduced by commit :::::: 56b03b4c4f5e8e1a44328b2df75bfb31fc4c3609 ASoC: intel: skylake: replace platform to component :::::: TO: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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Re: [alsa-devel] [PATCH v2 00/21] Allow compile-testing NO_DMA (drivers)
by Geert Uytterhoeven 17 Apr '18

17 Apr '18

Hi Rob, On Thu, Apr 5, 2018 at 2:32 AM, Rob Herring <robherring2(a)gmail.com> wrote: > On Fri, Mar 16, 2018 at 8:51 AM, Geert Uytterhoeven > <geert(a)linux-m68k.org> wrote: >> If NO_DMA=y, get_dma_ops() returns a reference to the non-existing >> symbol bad_dma_ops, thus causing a link failure if it is ever used. >> >> The intention of this is twofold: >> 1. To catch users of the DMA API on systems that do no support the DMA >> mapping API, >> 2. To avoid building drivers that cannot work on such systems anyway. >> >> However, the disadvantage is that we have to keep on adding dependencies >> on HAS_DMA all over the place. >> >> Thanks to the COMPILE_TEST symbol, lots of drivers now depend on one or >> more platform dependencies (that imply HAS_DMA) || COMPILE_TEST, thus >> already covering intention #2. Having to add an explicit dependency on >> HAS_DMA here is cumbersome, and hinders compile-testing. > > The same can be said for CONFIG_IOMEM and CONFIG_OF. Any plans to > remove those too? CONFIG_IOMEM is mostly just a !CONFIG_UM option. Perhaps, if time permits... Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

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[alsa-devel] [asoc:for-4.18 48/50] sound/soc/intel/skylake/skl-pcm.c:1351:4: error: implicit declaration of function 'skl_tplg_add_moduleid_in_bind_params'; did you mean 'skl_tplg_update_pipe_params'?
by kbuild test robot 17 Apr '18

17 Apr '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.18 head: 5db6aab6f36f7560dc95f7ca340d5632b7a3be6a commit: 81e9b0a078894841a50a8dd666fd64ca452a2a50 [48/50] ASoC: topology: Give more data to clients via callbacks config: i386-randconfig-x014-201815 (attached as .config) compiler: gcc-7 (Debian 7.3.0-1) 7.3.0 reproduce: git checkout 81e9b0a078894841a50a8dd666fd64ca452a2a50 # save the attached .config to linux build tree make ARCH=i386 All errors (new ones prefixed by >>): sound/soc/intel/skylake/skl-pcm.c: In function 'skl_populate_modules': >> sound/soc/intel/skylake/skl-pcm.c:1351:4: error: implicit declaration of function 'skl_tplg_add_moduleid_in_bind_params'; did you mean 'skl_tplg_update_pipe_params'? [-Werror=implicit-function-declaration] skl_tplg_add_moduleid_in_bind_params(skl, w); ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ skl_tplg_update_pipe_params sound/soc/intel/skylake/skl-pcm.c: In function 'skl_platform_soc_probe': >> sound/soc/intel/skylake/skl-pcm.c:1372:23: error: passing argument 1 of 'skl_tplg_init' from incompatible pointer type [-Werror=incompatible-pointer-types] ret = skl_tplg_init(component, ebus); ^~~~~~~~~ In file included from sound/soc/intel/skylake/skl-pcm.c:28:0: sound/soc/intel/skylake/skl-topology.h:463:5: note: expected 'struct snd_soc_platform *' but argument is of type 'struct snd_soc_component *' int skl_tplg_init(struct snd_soc_platform *platform, ^~~~~~~~~~~~~ Cyclomatic Complexity 5 include/linux/compiler.h:__read_once_size Cyclomatic Complexity 5 include/linux/compiler.h:__write_once_size Cyclomatic Complexity 4 include/linux/string.h:memcpy Cyclomatic Complexity 4 include/linux/string.h:memcmp Cyclomatic Complexity 3 include/linux/string.h:kmemdup Cyclomatic Complexity 1 include/linux/uuid.h:uuid_le_cmp Cyclomatic Complexity 1 arch/x86/include/asm/bitops.h:__ffs Cyclomatic Complexity 1 arch/x86/include/asm/bitops.h:fls Cyclomatic Complexity 1 include/linux/log2.h:__ilog2_u32 Cyclomatic Complexity 2 arch/x86/include/asm/div64.h:div_u64_rem Cyclomatic Complexity 1 include/linux/list.h:INIT_LIST_HEAD Cyclomatic Complexity 1 include/linux/list.h:__list_del Cyclomatic Complexity 1 include/linux/list.h:list_empty Cyclomatic Complexity 1 include/asm-generic/getorder.h:__get_order Cyclomatic Complexity 1 include/linux/math64.h:div_u64 Cyclomatic Complexity 1 include/linux/spinlock.h:spinlock_check Cyclomatic Complexity 1 include/linux/spinlock.h:spin_unlock_irqrestore Cyclomatic Complexity 1 include/linux/time32.h:timespec64_to_timespec Cyclomatic Complexity 1 include/linux/device.h:dev_get_drvdata Cyclomatic Complexity 1 arch/x86/include/asm/io.h:readl Cyclomatic Complexity 67 include/linux/slab.h:kmalloc_large Cyclomatic Complexity 3 include/linux/slab.h:kmalloc Cyclomatic Complexity 1 include/linux/slab.h:kzalloc Cyclomatic Complexity 1 include/linux/pm_runtime.h:pm_runtime_mark_last_busy Cyclomatic Complexity 1 include/sound/pcm.h:bytes_to_frames Cyclomatic Complexity 1 include/sound/pcm.h:snd_pcm_trigger_done Cyclomatic Complexity 1 include/sound/pcm.h:hw_param_mask_c Cyclomatic Complexity 1 include/sound/pcm.h:hw_param_interval_c Cyclomatic Complexity 1 include/sound/pcm.h:params_channels Cyclomatic Complexity 1 include/sound/pcm.h:params_rate Cyclomatic Complexity 1 include/sound/pcm.h:params_buffer_bytes Cyclomatic Complexity 3 include/sound/pcm_params.h:snd_mask_min Cyclomatic Complexity 1 include/sound/pcm_params.h:params_format Cyclomatic Complexity 2 include/sound/soc-dai.h:snd_soc_dai_get_dma_data Cyclomatic Complexity 2 include/sound/soc-dai.h:snd_soc_dai_set_dma_data Cyclomatic Complexity 1 include/sound/hda_register.h:snd_hdac_stream_get_pos_lpib Cyclomatic Complexity 1 include/sound/hda_register.h:snd_hdac_stream_get_pos_posbuf Cyclomatic Complexity 1 sound/soc/intel/skylake/skl-pcm.c:get_hdac_ext_stream Cyclomatic Complexity 2 sound/soc/intel/skylake/skl-pcm.c:skl_get_host_stream_type Cyclomatic Complexity 4 sound/soc/intel/skylake/skl-pcm.c:skl_adjust_codec_delay Cyclomatic Complexity 2 sound/soc/intel/skylake/skl-topology.h:get_skl_ctx Cyclomatic Complexity 2 sound/soc/intel/skylake/skl-pcm.c:get_bus_ctx Cyclomatic Complexity 5 sound/soc/intel/skylake/skl-pcm.c:skl_decoupled_trigger Cyclomatic Complexity 11 sound/soc/intel/skylake/skl-pcm.c:skl_pcm_trigger Cyclomatic Complexity 1 sound/soc/intel/skylake/skl-pcm.c:skl_substream_free_pages Cyclomatic Complexity 4 sound/soc/intel/skylake/skl-pcm.c:skl_pcm_hw_free Cyclomatic Complexity 1 sound/soc/intel/skylake/skl-pcm.c:skl_substream_alloc_pages Cyclomatic Complexity 7 sound/soc/intel/skylake/skl-pcm.c:skl_pcm_hw_params Cyclomatic Complexity 7 sound/soc/intel/skylake/skl-pcm.c:skl_set_suspend_active Cyclomatic Complexity 6 sound/soc/intel/skylake/skl-pcm.c:skl_pcm_close Cyclomatic Complexity 1 sound/soc/intel/skylake/skl-pcm.c:skl_set_pcm_constrains Cyclomatic Complexity 7 sound/soc/intel/skylake/skl-pcm.c:skl_pcm_open Cyclomatic Complexity 1 include/linux/timekeeping32.h:ktime_get_ts Cyclomatic Complexity 1 include/linux/timekeeping32.h:getrawmonotonic Cyclomatic Complexity 1 include/linux/timekeeping32.h:getnstimeofday Cyclomatic Complexity 3 include/sound/pcm.h:snd_pcm_gettime Cyclomatic Complexity 4 sound/soc/intel/skylake/skl-pcm.c:skl_get_time_info Cyclomatic Complexity 3 sound/soc/intel/skylake/skl-pcm.c:skl_platform_pcm_pointer Cyclomatic Complexity 15 sound/soc/intel/skylake/skl-pcm.c:skl_coupled_trigger Cyclomatic Complexity 2 sound/soc/intel/skylake/skl-pcm.c:skl_platform_pcm_trigger Cyclomatic Complexity 2 sound/soc/intel/skylake/skl-pcm.c:skl_platform_open Cyclomatic Complexity 1 sound/soc/intel/skylake/skl-pcm.c:skl_pcm_free Cyclomatic Complexity 6 sound/soc/intel/skylake/skl-pcm.c:skl_pcm_new Cyclomatic Complexity 1 include/linux/pm_runtime.h:pm_runtime_get_sync Cyclomatic Complexity 19 sound/soc/intel/skylake/skl-pcm.c:skl_get_module_info Cyclomatic Complexity 8 sound/soc/intel/skylake/skl-pcm.c:skl_populate_modules Cyclomatic Complexity 1 include/linux/pm_runtime.h:pm_runtime_put_autosuspend Cyclomatic Complexity 8 sound/soc/intel/skylake/skl-pcm.c:skl_platform_soc_probe Cyclomatic Complexity 1 sound/soc/intel/skylake/skl-pcm.c:skl_be_hw_params Cyclomatic Complexity 5 sound/soc/intel/skylake/skl-pcm.c:skl_link_pcm_trigger Cyclomatic Complexity 4 sound/soc/intel/skylake/skl-pcm.c:skl_link_pcm_prepare Cyclomatic Complexity 3 sound/soc/intel/skylake/skl-pcm.c:skl_link_hw_free Cyclomatic Complexity 4 sound/soc/intel/skylake/skl-pcm.c:skl_link_hw_params Cyclomatic Complexity 2 include/linux/list.h:__list_del_entry Cyclomatic Complexity 1 include/linux/list.h:list_del Cyclomatic Complexity 6 sound/soc/intel/skylake/skl-pcm.c:skl_pcm_host_dma_prepare Cyclomatic Complexity 8 sound/soc/intel/skylake/skl-pcm.c:skl_pcm_prepare Cyclomatic Complexity 8 sound/soc/intel/skylake/skl-pcm.c:skl_pcm_link_dma_prepare Cyclomatic Complexity 1 sound/soc/intel/skylake/skl-pcm.c:skl_dai_load Cyclomatic Complexity 6 sound/soc/intel/skylake/skl-pcm.c:skl_platform_register Cyclomatic Complexity 7 sound/soc/intel/skylake/skl-pcm.c:skl_platform_unregister cc1: some warnings being treated as errors -- sound/soc/intel/skylake/skl-topology.c: In function 'skl_fill_sink_instance_id': >> sound/soc/intel/skylake/skl-topology.c:833:45: error: 'struct skl_kpb_params' has no member named 'u' struct skl_mod_inst_map *inst = kpb_params->u.map; ^~ sound/soc/intel/skylake/skl-topology.c: In function 'skl_tplg_find_moduleid_from_uuid': sound/soc/intel/skylake/skl-topology.c:953:17: error: 'struct skl_kpb_params' has no member named 'u' &uuid_params->u.map_uuid[i].mod_uuid); ^~ sound/soc/intel/skylake/skl-topology.c:959:10: error: 'struct skl_kpb_params' has no member named 'u' params->u.map[i].mod_id = module_id; ^~ sound/soc/intel/skylake/skl-topology.c:960:10: error: 'struct skl_kpb_params' has no member named 'u' params->u.map[i].inst_id = ^~ sound/soc/intel/skylake/skl-topology.c:961:16: error: 'struct skl_kpb_params' has no member named 'u' uuid_params->u.map_uuid[i].inst_id; ^~ sound/soc/intel/skylake/skl-topology.c: At top level: >> sound/soc/intel/skylake/skl-topology.c:3544:5: error: conflicting types for 'skl_tplg_init' int skl_tplg_init(struct snd_soc_component *component, struct hdac_ext_bus *ebus) ^~~~~~~~~~~~~ In file included from sound/soc/intel/skylake/skl-topology.c:27:0: sound/soc/intel/skylake/skl-topology.h:463:5: note: previous declaration of 'skl_tplg_init' was here int skl_tplg_init(struct snd_soc_platform *platform, ^~~~~~~~~~~~~ vim +1351 sound/soc/intel/skylake/skl-pcm.c b26199ea Jeeja KP 2017-03-24 1330 64cb1d0a Vinod Koul 2016-08-10 1331 static int skl_populate_modules(struct skl *skl) 64cb1d0a Vinod Koul 2016-08-10 1332 { 64cb1d0a Vinod Koul 2016-08-10 1333 struct skl_pipeline *p; 64cb1d0a Vinod Koul 2016-08-10 1334 struct skl_pipe_module *m; 64cb1d0a Vinod Koul 2016-08-10 1335 struct snd_soc_dapm_widget *w; 64cb1d0a Vinod Koul 2016-08-10 1336 struct skl_module_cfg *mconfig; b26199ea Jeeja KP 2017-03-24 1337 int ret = 0; 64cb1d0a Vinod Koul 2016-08-10 1338 64cb1d0a Vinod Koul 2016-08-10 1339 list_for_each_entry(p, &skl->ppl_list, node) { 64cb1d0a Vinod Koul 2016-08-10 1340 list_for_each_entry(m, &p->pipe->w_list, node) { 64cb1d0a Vinod Koul 2016-08-10 1341 w = m->w; 64cb1d0a Vinod Koul 2016-08-10 1342 mconfig = w->priv; 64cb1d0a Vinod Koul 2016-08-10 1343 b26199ea Jeeja KP 2017-03-24 1344 ret = skl_get_module_info(skl, mconfig); 64cb1d0a Vinod Koul 2016-08-10 1345 if (ret < 0) { 64cb1d0a Vinod Koul 2016-08-10 1346 dev_err(skl->skl_sst->dev, b26199ea Jeeja KP 2017-03-24 1347 "query module info failed\n"); b26199ea Jeeja KP 2017-03-24 1348 return ret; 64cb1d0a Vinod Koul 2016-08-10 1349 } f7a9f772 Sriram Periyasamy 2018-01-27 1350 f7a9f772 Sriram Periyasamy 2018-01-27 @1351 skl_tplg_add_moduleid_in_bind_params(skl, w); 64cb1d0a Vinod Koul 2016-08-10 1352 } 64cb1d0a Vinod Koul 2016-08-10 1353 } b26199ea Jeeja KP 2017-03-24 1354 64cb1d0a Vinod Koul 2016-08-10 1355 return ret; 64cb1d0a Vinod Koul 2016-08-10 1356 } 64cb1d0a Vinod Koul 2016-08-10 1357 56b03b4c Kuninori Morimoto 2018-01-29 1358 static int skl_platform_soc_probe(struct snd_soc_component *component) b663a8c5 Jeeja KP 2015-10-07 1359 { 56b03b4c Kuninori Morimoto 2018-01-29 1360 struct hdac_ext_bus *ebus = dev_get_drvdata(component->dev); fe3f4442 Dharageswari R 2016-06-03 1361 struct skl *skl = ebus_to_skl(ebus); 78cdbbda Vinod Koul 2016-07-26 1362 const struct skl_dsp_ops *ops; fe3f4442 Dharageswari R 2016-06-03 1363 int ret; b663a8c5 Jeeja KP 2015-10-07 1364 56b03b4c Kuninori Morimoto 2018-01-29 1365 pm_runtime_get_sync(component->dev); ec8ae570 Vinod Koul 2016-08-04 1366 if ((ebus_to_hbus(ebus))->ppcap) { 56b03b4c Kuninori Morimoto 2018-01-29 1367 skl->component = component; 5cdf6c09 Vinod Koul 2017-06-30 1368 5cdf6c09 Vinod Koul 2017-06-30 1369 /* init debugfs */ 5cdf6c09 Vinod Koul 2017-06-30 1370 skl->debugfs = skl_debugfs_init(skl); 5cdf6c09 Vinod Koul 2017-06-30 1371 56b03b4c Kuninori Morimoto 2018-01-29 @1372 ret = skl_tplg_init(component, ebus); fe3f4442 Dharageswari R 2016-06-03 1373 if (ret < 0) { 56b03b4c Kuninori Morimoto 2018-01-29 1374 dev_err(component->dev, "Failed to init topology!\n"); fe3f4442 Dharageswari R 2016-06-03 1375 return ret; fe3f4442 Dharageswari R 2016-06-03 1376 } 78cdbbda Vinod Koul 2016-07-26 1377 78cdbbda Vinod Koul 2016-07-26 1378 /* load the firmwares, since all is set */ 78cdbbda Vinod Koul 2016-07-26 1379 ops = skl_get_dsp_ops(skl->pci->device); 78cdbbda Vinod Koul 2016-07-26 1380 if (!ops) 78cdbbda Vinod Koul 2016-07-26 1381 return -EIO; 78cdbbda Vinod Koul 2016-07-26 1382 78cdbbda Vinod Koul 2016-07-26 1383 if (skl->skl_sst->is_first_boot == false) { 56b03b4c Kuninori Morimoto 2018-01-29 1384 dev_err(component->dev, "DSP reports first boot done!!!\n"); 78cdbbda Vinod Koul 2016-07-26 1385 return -EIO; 78cdbbda Vinod Koul 2016-07-26 1386 } 78cdbbda Vinod Koul 2016-07-26 1387 fc9fdd61 Sanyog Kale 2018-03-13 1388 /* fc9fdd61 Sanyog Kale 2018-03-13 1389 * Disable dynamic clock and power gating during firmware fc9fdd61 Sanyog Kale 2018-03-13 1390 * and library download fc9fdd61 Sanyog Kale 2018-03-13 1391 */ 56b03b4c Kuninori Morimoto 2018-01-29 1392 skl->skl_sst->enable_miscbdcge(component->dev, false); fc9fdd61 Sanyog Kale 2018-03-13 1393 skl->skl_sst->clock_power_gating(component->dev, false); d5cc0a1f Pardha Saradhi K 2018-01-02 1394 56b03b4c Kuninori Morimoto 2018-01-29 1395 ret = ops->init_fw(component->dev, skl->skl_sst); 56b03b4c Kuninori Morimoto 2018-01-29 1396 skl->skl_sst->enable_miscbdcge(component->dev, true); fc9fdd61 Sanyog Kale 2018-03-13 1397 skl->skl_sst->clock_power_gating(component->dev, true); 78cdbbda Vinod Koul 2016-07-26 1398 if (ret < 0) { 56b03b4c Kuninori Morimoto 2018-01-29 1399 dev_err(component->dev, "Failed to boot first fw: %d\n", ret); 78cdbbda Vinod Koul 2016-07-26 1400 return ret; 78cdbbda Vinod Koul 2016-07-26 1401 } 64cb1d0a Vinod Koul 2016-08-10 1402 skl_populate_modules(skl); a26a3f53 Pardha Saradhi K 2016-11-03 1403 skl->skl_sst->update_d0i3c = skl_update_d0i3c; cb729d80 G Kranthi 2017-03-13 1404 skl_dsp_enable_notification(skl->skl_sst, false); 9452314d Pradeep Tewani 2017-12-06 1405 9452314d Pradeep Tewani 2017-12-06 1406 if (skl->cfg.astate_cfg != NULL) { 9452314d Pradeep Tewani 2017-12-06 1407 skl_dsp_set_astate_cfg(skl->skl_sst, 9452314d Pradeep Tewani 2017-12-06 1408 skl->cfg.astate_cfg->count, 9452314d Pradeep Tewani 2017-12-06 1409 skl->cfg.astate_cfg); 9452314d Pradeep Tewani 2017-12-06 1410 } fe3f4442 Dharageswari R 2016-06-03 1411 } 56b03b4c Kuninori Morimoto 2018-01-29 1412 pm_runtime_mark_last_busy(component->dev); 56b03b4c Kuninori Morimoto 2018-01-29 1413 pm_runtime_put_autosuspend(component->dev); b663a8c5 Jeeja KP 2015-10-07 1414 b663a8c5 Jeeja KP 2015-10-07 1415 return 0; b663a8c5 Jeeja KP 2015-10-07 1416 } 56b03b4c Kuninori Morimoto 2018-01-29 1417 :::::: The code at line 1351 was first introduced by commit :::::: f7a9f77256e7fbd6150651bfab44f60c39f0b7a2 ASoC: Intel: Skylake: Find module id from UUID for bind params :::::: TO: Sriram Periyasamy <sriramx.periyasamy(a)intel.com> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH 2/4] ASoC: amd: fixed checkpatch pl warnings
by Vijendar Mukunda 17 Apr '18

17 Apr '18

fixed checkpatch pl warnings. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/acp-pcm-dma.c | 259 ++++++++++++++++++++++++-------------------- sound/soc/amd/acp.h | 22 ++-- 2 files changed, 153 insertions(+), 128 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 540088d..5ffe2ef 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -130,7 +130,8 @@ static void acp_reg_write(u32 val, void __iomem *acp_mmio, u32 reg) writel(val, acp_mmio + (reg * 4)); } -/* Configure a given dma channel parameters - enable/disable, +/* + * Configure a given dma channel parameters - enable/disable, * number of descriptors, priority */ static void config_acp_dma_channel(void __iomem *acp_mmio, u8 ch_num, @@ -149,11 +150,12 @@ static void config_acp_dma_channel(void __iomem *acp_mmio, u8 ch_num, & dscr_strt_idx), acp_mmio, mmACP_DMA_DSCR_STRT_IDX_0 + ch_num); - /* program a DMA channel with the number of descriptors to be + /* + * program a DMA channel with the number of descriptors to be * processed in the transfer - */ + */ acp_reg_write(ACP_DMA_DSCR_CNT_0__DMAChDscrCnt_MASK & num_dscrs, - acp_mmio, mmACP_DMA_DSCR_CNT_0 + ch_num); + acp_mmio, mmACP_DMA_DSCR_CNT_0 + ch_num); /* set DMA channel priority */ acp_reg_write(priority_level, acp_mmio, mmACP_DMA_PRIO_0 + ch_num); @@ -180,13 +182,15 @@ static void config_dma_descriptor_in_sram(void __iomem *acp_mmio, acp_reg_write(descr_info->xfer_val, acp_mmio, mmACP_SRBM_Targ_Idx_Data); } -/* Initialize the DMA descriptor information for transfer between +/* + * Initialize the DMA descriptor information for transfer between * system memory <-> ACP SRAM */ static void set_acp_sysmem_dma_descriptors(void __iomem *acp_mmio, - u32 size, int direction, u32 pte_offset, - u16 ch, u32 sram_bank, - u16 dma_dscr_idx, u32 asic_type) + u32 size, int direction, + u32 pte_offset, u16 ch, + u32 sram_bank, u16 dma_dscr_idx, + u32 asic_type) { u16 i; acp_dma_dscr_transfer_t dmadscr[NUM_DSCRS_PER_CHANNEL]; @@ -195,58 +199,58 @@ static void set_acp_sysmem_dma_descriptors(void __iomem *acp_mmio, dmadscr[i].xfer_val = 0; if (direction == SNDRV_PCM_STREAM_PLAYBACK) { dma_dscr_idx = dma_dscr_idx + i; - dmadscr[i].dest = sram_bank + (i * (size/2)); + dmadscr[i].dest = sram_bank + (i * (size / 2)); dmadscr[i].src = ACP_INTERNAL_APERTURE_WINDOW_0_ADDRESS - + (pte_offset * SZ_4K) + (i * (size/2)); + + (pte_offset * SZ_4K) + (i * (size / 2)); switch (asic_type) { case CHIP_STONEY: dmadscr[i].xfer_val |= - (ACP_DMA_ATTRIBUTES_DAGB_GARLIC_TO_SHAREDMEM << 16) | + (ACP_DMA_ATTR_DAGB_GARLIC_TO_SHAREDMEM << 16) | (size / 2); break; default: dmadscr[i].xfer_val |= - (ACP_DMA_ATTRIBUTES_DAGB_ONION_TO_SHAREDMEM << 16) | + (ACP_DMA_ATTR_DAGB_ONION_TO_SHAREDMEM << 16) | (size / 2); } } else { dma_dscr_idx = dma_dscr_idx + i; - dmadscr[i].src = sram_bank + (i * (size/2)); + dmadscr[i].src = sram_bank + (i * (size / 2)); dmadscr[i].dest = ACP_INTERNAL_APERTURE_WINDOW_0_ADDRESS + - (pte_offset * SZ_4K) + (i * (size/2)); + (pte_offset * SZ_4K) + (i * (size / 2)); switch (asic_type) { case CHIP_STONEY: dmadscr[i].xfer_val |= BIT(22) | - (ACP_DMA_ATTRIBUTES_SHARED_MEM_TO_DAGB_GARLIC << 16) | + (ACP_DMA_ATTR_SHARED_MEM_TO_DAGB_GARLIC << 16) | (size / 2); break; default: dmadscr[i].xfer_val |= BIT(22) | - (ACP_DMA_ATTRIBUTES_SHAREDMEM_TO_DAGB_ONION << 16) | + (ACP_DMA_ATTR_SHAREDMEM_TO_DAGB_ONION << 16) | (size / 2); } } config_dma_descriptor_in_sram(acp_mmio, dma_dscr_idx, - &dmadscr[i]); + &dmadscr[i]); } config_acp_dma_channel(acp_mmio, ch, - dma_dscr_idx - 1, - NUM_DSCRS_PER_CHANNEL, - ACP_DMA_PRIORITY_LEVEL_NORMAL); + dma_dscr_idx - 1, + NUM_DSCRS_PER_CHANNEL, + ACP_DMA_PRIORITY_LEVEL_NORMAL); } -/* Initialize the DMA descriptor information for transfer between +/* + * Initialize the DMA descriptor information for transfer between * ACP SRAM <-> I2S */ static void set_acp_to_i2s_dma_descriptors(void __iomem *acp_mmio, u32 size, - int direction, u32 sram_bank, - u16 destination, u16 ch, - u16 dma_dscr_idx, u32 asic_type) + int direction, u32 sram_bank, + u16 destination, u16 ch, + u16 dma_dscr_idx, u32 asic_type) { - u16 i; acp_dma_dscr_transfer_t dmadscr[NUM_DSCRS_PER_CHANNEL]; @@ -254,7 +258,7 @@ static void set_acp_to_i2s_dma_descriptors(void __iomem *acp_mmio, u32 size, dmadscr[i].xfer_val = 0; if (direction == SNDRV_PCM_STREAM_PLAYBACK) { dma_dscr_idx = dma_dscr_idx + i; - dmadscr[i].src = sram_bank + (i * (size/2)); + dmadscr[i].src = sram_bank + (i * (size / 2)); /* dmadscr[i].dest is unused by hardware. */ dmadscr[i].dest = 0; dmadscr[i].xfer_val |= BIT(22) | (destination << 16) | @@ -269,12 +273,12 @@ static void set_acp_to_i2s_dma_descriptors(void __iomem *acp_mmio, u32 size, (destination << 16) | (size / 2); } config_dma_descriptor_in_sram(acp_mmio, dma_dscr_idx, - &dmadscr[i]); + &dmadscr[i]); } /* Configure the DMA channel with the above descriptore */ config_acp_dma_channel(acp_mmio, ch, dma_dscr_idx - 1, - NUM_DSCRS_PER_CHANNEL, - ACP_DMA_PRIORITY_LEVEL_NORMAL); + NUM_DSCRS_PER_CHANNEL, + ACP_DMA_PRIORITY_LEVEL_NORMAL); } /* Create page table entries in ACP SRAM for the allocated memory */ @@ -291,7 +295,7 @@ static void acp_pte_config(void __iomem *acp_mmio, struct page *pg, for (page_idx = 0; page_idx < (num_of_pages); page_idx++) { /* Load the low address of page int ACP SRAM through SRBM */ acp_reg_write((offset + (page_idx * 8)), - acp_mmio, mmACP_SRBM_Targ_Idx_Addr); + acp_mmio, mmACP_SRBM_Targ_Idx_Addr); addr = page_to_phys(pg); low = lower_32_bits(addr); @@ -301,7 +305,7 @@ static void acp_pte_config(void __iomem *acp_mmio, struct page *pg, /* Load the High address of page int ACP SRAM through SRBM */ acp_reg_write((offset + (page_idx * 8) + 4), - acp_mmio, mmACP_SRBM_Targ_Idx_Addr); + acp_mmio, mmACP_SRBM_Targ_Idx_Addr); /* page enable in ACP */ high |= BIT(31); @@ -313,8 +317,8 @@ static void acp_pte_config(void __iomem *acp_mmio, struct page *pg, } static void config_acp_dma(void __iomem *acp_mmio, - struct audio_substream_data *audio_config, - u32 asic_type) + struct audio_substream_data *audio_config, + u32 asic_type) { u32 pte_offset, sram_bank; u16 ch1, ch2, destination, dma_dscr_idx; @@ -341,7 +345,7 @@ static void config_acp_dma(void __iomem *acp_mmio, } acp_pte_config(acp_mmio, audio_config->pg, audio_config->num_of_pages, - pte_offset); + pte_offset); if (audio_config->direction == SNDRV_PCM_STREAM_PLAYBACK) dma_dscr_idx = PLAYBACK_START_DMA_DESCR_CH12; else @@ -349,8 +353,8 @@ static void config_acp_dma(void __iomem *acp_mmio, /* Configure System memory <-> ACP SRAM DMA descriptors */ set_acp_sysmem_dma_descriptors(acp_mmio, audio_config->size, - audio_config->direction, pte_offset, - ch1, sram_bank, dma_dscr_idx, asic_type); + audio_config->direction, pte_offset, ch1, + sram_bank, dma_dscr_idx, asic_type); if (audio_config->direction == SNDRV_PCM_STREAM_PLAYBACK) dma_dscr_idx = PLAYBACK_START_DMA_DESCR_CH13; @@ -358,14 +362,14 @@ static void config_acp_dma(void __iomem *acp_mmio, dma_dscr_idx = CAPTURE_START_DMA_DESCR_CH15; /* Configure ACP SRAM <-> I2S DMA descriptors */ set_acp_to_i2s_dma_descriptors(acp_mmio, audio_config->size, - audio_config->direction, sram_bank, - destination, ch2, dma_dscr_idx, - asic_type); + audio_config->direction, sram_bank, + destination, ch2, dma_dscr_idx, + asic_type); } /* Start a given DMA channel transfer */ static void acp_dma_start(void __iomem *acp_mmio, - u16 ch_num, bool is_circular) + u16 ch_num, bool is_circular) { u32 dma_ctrl; @@ -375,7 +379,8 @@ static void acp_dma_start(void __iomem *acp_mmio, /* Invalidating the DAGB cache */ acp_reg_write(1, acp_mmio, mmACP_DAGB_ATU_CTRL); - /* configure the DMA channel and start the DMA transfer + /* + * configure the DMA channel and start the DMA transfer * set dmachrun bit to start the transfer and enable the * interrupt on completion of the dma transfer */ @@ -410,9 +415,10 @@ static int acp_dma_stop(void __iomem *acp_mmio, u8 ch_num) dma_ctrl = acp_reg_read(acp_mmio, mmACP_DMA_CNTL_0 + ch_num); - /* clear the dma control register fields before writing zero + /* + * clear the dma control register fields before writing zero * in reset bit - */ + */ dma_ctrl &= ~ACP_DMA_CNTL_0__DMAChRun_MASK; dma_ctrl &= ~ACP_DMA_CNTL_0__DMAChIOCEn_MASK; @@ -420,9 +426,10 @@ static int acp_dma_stop(void __iomem *acp_mmio, u8 ch_num) dma_ch_sts = acp_reg_read(acp_mmio, mmACP_DMA_CH_STS); if (dma_ch_sts & BIT(ch_num)) { - /* set the reset bit for this channel to stop the dma - * transfer - */ + /* + * set the reset bit for this channel to stop the dma + * transfer + */ dma_ctrl |= ACP_DMA_CNTL_0__DMAChRst_MASK; acp_reg_write(dma_ctrl, acp_mmio, mmACP_DMA_CNTL_0 + ch_num); } @@ -431,13 +438,14 @@ static int acp_dma_stop(void __iomem *acp_mmio, u8 ch_num) while (true) { dma_ch_sts = acp_reg_read(acp_mmio, mmACP_DMA_CH_STS); if (!(dma_ch_sts & BIT(ch_num))) { - /* clear the reset flag after successfully stopping - * the dma transfer and break from the loop - */ + /* + * clear the reset flag after successfully stopping + * the dma transfer and break from the loop + */ dma_ctrl &= ~ACP_DMA_CNTL_0__DMAChRst_MASK; acp_reg_write(dma_ctrl, acp_mmio, mmACP_DMA_CNTL_0 - + ch_num); + + ch_num); break; } if (--count == 0) { @@ -450,7 +458,7 @@ static int acp_dma_stop(void __iomem *acp_mmio, u8 ch_num) } static void acp_set_sram_bank_state(void __iomem *acp_mmio, u16 bank, - bool power_on) + bool power_on) { u32 val, req_reg, sts_reg, sts_reg_mask; u32 loops = 1000; @@ -530,7 +538,7 @@ static int acp_init(void __iomem *acp_mmio, u32 asic_type) while (true) { val = acp_reg_read(acp_mmio, mmACP_STATUS); - if (val & (u32) 0x1) + if (val & (u32)0x1) break; if (--count == 0) { pr_err("Failed to reset ACP\n"); @@ -546,11 +554,11 @@ static int acp_init(void __iomem *acp_mmio, u32 asic_type) /* initiailize Onion control DAGB register */ acp_reg_write(ACP_ONION_CNTL_DEFAULT, acp_mmio, - mmACP_AXI2DAGB_ONION_CNTL); + mmACP_AXI2DAGB_ONION_CNTL); /* initiailize Garlic control DAGB registers */ acp_reg_write(ACP_GARLIC_CNTL_DEFAULT, acp_mmio, - mmACP_AXI2DAGB_GARLIC_CNTL); + mmACP_AXI2DAGB_GARLIC_CNTL); sram_pte_offset = ACP_DAGB_GRP_SRAM_BASE_ADDRESS | ACP_DAGB_BASE_ADDR_GRP_1__AXI2DAGBSnoopSel_MASK | @@ -558,17 +566,18 @@ static int acp_init(void __iomem *acp_mmio, u32 asic_type) ACP_DAGB_BASE_ADDR_GRP_1__AXI2DAGBGrpEnable_MASK; acp_reg_write(sram_pte_offset, acp_mmio, mmACP_DAGB_BASE_ADDR_GRP_1); acp_reg_write(ACP_PAGE_SIZE_4K_ENABLE, acp_mmio, - mmACP_DAGB_PAGE_SIZE_GRP_1); + mmACP_DAGB_PAGE_SIZE_GRP_1); acp_reg_write(ACP_SRAM_BASE_ADDRESS, acp_mmio, - mmACP_DMA_DESC_BASE_ADDR); + mmACP_DMA_DESC_BASE_ADDR); /* Num of descriptiors in SRAM 0x4, means 256 descriptors;(64 * 4) */ acp_reg_write(0x4, acp_mmio, mmACP_DMA_DESC_MAX_NUM_DSCR); acp_reg_write(ACP_EXTERNAL_INTR_CNTL__DMAIOCMask_MASK, - acp_mmio, mmACP_EXTERNAL_INTR_CNTL); + acp_mmio, mmACP_EXTERNAL_INTR_CNTL); - /* When ACP_TILE_P1 is turned on, all SRAM banks get turned on. + /* + * When ACP_TILE_P1 is turned on, all SRAM banks get turned on. * Now, turn off all of them. This can't be done in 'poweron' of * ACP pm domain, as this requires ACP to be initialized. * For Stoney, Memory gating is disabled,i.e SRAM Banks @@ -606,7 +615,7 @@ static int acp_deinit(void __iomem *acp_mmio) } udelay(100); } - /** Disable ACP clock */ + /* Disable ACP clock */ val = acp_reg_read(acp_mmio, mmACP_CONTROL); val &= ~ACP_CONTROL__ClkEn_MASK; acp_reg_write(val, acp_mmio, mmACP_CONTROL); @@ -615,7 +624,7 @@ static int acp_deinit(void __iomem *acp_mmio) while (true) { val = acp_reg_read(acp_mmio, mmACP_STATUS); - if (!(val & (u32) 0x1)) + if (!(val & (u32)0x1)) break; if (--count == 0) { pr_err("Failed to reset ACP\n"); @@ -658,7 +667,7 @@ static irqreturn_t dma_irq_handler(int irq, void *arg) snd_pcm_period_elapsed(irq_data->play_i2ssp_stream); acp_reg_write((intr_flag & BIT(ACP_TO_I2S_DMA_CH_NUM)) << 16, - acp_mmio, mmACP_EXTERNAL_INTR_STAT); + acp_mmio, mmACP_EXTERNAL_INTR_STAT); } if ((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) != 0) { @@ -673,14 +682,14 @@ static irqreturn_t dma_irq_handler(int irq, void *arg) acp_dma_start(acp_mmio, ACP_TO_SYSRAM_CH_NUM, false); acp_reg_write((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) << 16, - acp_mmio, mmACP_EXTERNAL_INTR_STAT); + acp_mmio, mmACP_EXTERNAL_INTR_STAT); } if ((intr_flag & BIT(ACP_TO_SYSRAM_CH_NUM)) != 0) { valid_irq = true; snd_pcm_period_elapsed(irq_data->capture_i2ssp_stream); acp_reg_write((intr_flag & BIT(ACP_TO_SYSRAM_CH_NUM)) << 16, - acp_mmio, mmACP_EXTERNAL_INTR_STAT); + acp_mmio, mmACP_EXTERNAL_INTR_STAT); } if (valid_irq) @@ -695,11 +704,12 @@ static int acp_dma_open(struct snd_pcm_substream *substream) int ret = 0; struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *prtd = substream->private_data; - struct snd_soc_component *component = snd_soc_rtdcom_lookup(prtd, DRV_NAME); + struct snd_soc_component *component = snd_soc_rtdcom_lookup(prtd, + DRV_NAME); struct audio_drv_data *intr_data = dev_get_drvdata(component->dev); struct audio_substream_data *adata = kzalloc(sizeof(struct audio_substream_data), GFP_KERNEL); - if (adata == NULL) + if (!adata) return -ENOMEM; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { @@ -731,17 +741,19 @@ static int acp_dma_open(struct snd_pcm_substream *substream) adata->acp_mmio = intr_data->acp_mmio; runtime->private_data = adata; - /* Enable ACP irq, when neither playback or capture streams are + /* + * Enable ACP irq, when neither playback or capture streams are * active by the time when a new stream is being opened. * This enablement is not required for another stream, if current * stream is not closed - */ + */ if (!intr_data->play_i2ssp_stream && !intr_data->capture_i2ssp_stream) acp_reg_write(1, adata->acp_mmio, mmACP_EXTERNAL_INTR_ENB); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { intr_data->play_i2ssp_stream = substream; - /* For Stoney, Memory gating is disabled,i.e SRAM Banks + /* + * For Stoney, Memory gating is disabled,i.e SRAM Banks * won't be turned off. The default state for SRAM banks is ON. * Setting SRAM bank state code skipped for STONEY platform. */ @@ -772,7 +784,8 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime; struct audio_substream_data *rtd; struct snd_soc_pcm_runtime *prtd = substream->private_data; - struct snd_soc_component *component = snd_soc_rtdcom_lookup(prtd, DRV_NAME); + struct snd_soc_component *component = snd_soc_rtdcom_lookup(prtd, + DRV_NAME); struct audio_drv_data *adata = dev_get_drvdata(component->dev); runtime = substream->runtime; @@ -782,12 +795,14 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, return -EINVAL; if (adata->asic_type == CHIP_STONEY) { - val = acp_reg_read(adata->acp_mmio, mmACP_I2S_16BIT_RESOLUTION_EN); + val = acp_reg_read(adata->acp_mmio, + mmACP_I2S_16BIT_RESOLUTION_EN); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) val |= ACP_I2S_SP_16BIT_RESOLUTION_EN; else val |= ACP_I2S_MIC_16BIT_RESOLUTION_EN; - acp_reg_write(val, adata->acp_mmio, mmACP_I2S_16BIT_RESOLUTION_EN); + acp_reg_write(val, adata->acp_mmio, + mmACP_I2S_16BIT_RESOLUTION_EN); } size = params_buffer_bytes(params); status = snd_pcm_lib_malloc_pages(substream, size); @@ -797,7 +812,7 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, memset(substream->runtime->dma_area, 0, params_buffer_bytes(params)); pg = virt_to_page(substream->dma_buffer.area); - if (pg != NULL) { + if (pg) { acp_set_sram_bank_state(rtd->acp_mmio, 0, true); /* Save for runtime private data */ rtd->pg = pg; @@ -885,18 +900,18 @@ static int acp_dma_prepare(struct snd_pcm_substream *substream) return -EINVAL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { config_acp_dma_channel(rtd->acp_mmio, SYSRAM_TO_ACP_CH_NUM, - PLAYBACK_START_DMA_DESCR_CH12, - NUM_DSCRS_PER_CHANNEL, 0); + PLAYBACK_START_DMA_DESCR_CH12, + NUM_DSCRS_PER_CHANNEL, 0); config_acp_dma_channel(rtd->acp_mmio, ACP_TO_I2S_DMA_CH_NUM, - PLAYBACK_START_DMA_DESCR_CH13, - NUM_DSCRS_PER_CHANNEL, 0); + PLAYBACK_START_DMA_DESCR_CH13, + NUM_DSCRS_PER_CHANNEL, 0); } else { config_acp_dma_channel(rtd->acp_mmio, ACP_TO_SYSRAM_CH_NUM, - CAPTURE_START_DMA_DESCR_CH14, - NUM_DSCRS_PER_CHANNEL, 0); + CAPTURE_START_DMA_DESCR_CH14, + NUM_DSCRS_PER_CHANNEL, 0); config_acp_dma_channel(rtd->acp_mmio, I2S_TO_ACP_DMA_CH_NUM, - CAPTURE_START_DMA_DESCR_CH15, - NUM_DSCRS_PER_CHANNEL, 0); + CAPTURE_START_DMA_DESCR_CH15, + NUM_DSCRS_PER_CHANNEL, 0); } return 0; } @@ -910,7 +925,8 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *prtd = substream->private_data; struct audio_substream_data *rtd = runtime->private_data; - struct snd_soc_component *component = snd_soc_rtdcom_lookup(prtd, DRV_NAME); + struct snd_soc_component *component = snd_soc_rtdcom_lookup(prtd, + DRV_NAME); if (!rtd) return -EINVAL; @@ -924,7 +940,7 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) if (rtd->i2ssp_renderbytescount == 0) rtd->i2ssp_renderbytescount = bytescount; acp_dma_start(rtd->acp_mmio, - SYSRAM_TO_ACP_CH_NUM, false); + SYSRAM_TO_ACP_CH_NUM, false); while (acp_reg_read(rtd->acp_mmio, mmACP_DMA_CH_STS) & BIT(SYSRAM_TO_ACP_CH_NUM)) { if (!loops--) { @@ -936,41 +952,41 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) } acp_dma_start(rtd->acp_mmio, - ACP_TO_I2S_DMA_CH_NUM, true); + ACP_TO_I2S_DMA_CH_NUM, true); } else { if (rtd->i2ssp_capturebytescount == 0) rtd->i2ssp_capturebytescount = bytescount; acp_dma_start(rtd->acp_mmio, - I2S_TO_ACP_DMA_CH_NUM, true); + I2S_TO_ACP_DMA_CH_NUM, true); } ret = 0; break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: - /* Need to stop only circular DMA channels : + /* + * Need to stop only circular DMA channels : * ACP_TO_I2S_DMA_CH_NUM / I2S_TO_ACP_DMA_CH_NUM. Non-circular * channels will stopped automatically after its transfer * completes : SYSRAM_TO_ACP_CH_NUM / ACP_TO_SYSRAM_CH_NUM */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = acp_dma_stop(rtd->acp_mmio, - SYSRAM_TO_ACP_CH_NUM); + SYSRAM_TO_ACP_CH_NUM); ret = acp_dma_stop(rtd->acp_mmio, - ACP_TO_I2S_DMA_CH_NUM); + ACP_TO_I2S_DMA_CH_NUM); rtd->i2ssp_renderbytescount = 0; } else { ret = acp_dma_stop(rtd->acp_mmio, - I2S_TO_ACP_DMA_CH_NUM); + I2S_TO_ACP_DMA_CH_NUM); ret = acp_dma_stop(rtd->acp_mmio, - ACP_TO_SYSRAM_CH_NUM); + ACP_TO_SYSRAM_CH_NUM); rtd->i2ssp_capturebytescount = 0; } break; default: ret = -EINVAL; - } return ret; } @@ -978,26 +994,27 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) static int acp_dma_new(struct snd_soc_pcm_runtime *rtd) { int ret; - struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, DRV_NAME); + struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, + DRV_NAME); struct audio_drv_data *adata = dev_get_drvdata(component->dev); switch (adata->asic_type) { case CHIP_STONEY: ret = snd_pcm_lib_preallocate_pages_for_all(rtd->pcm, - SNDRV_DMA_TYPE_DEV, - NULL, ST_MIN_BUFFER, - ST_MAX_BUFFER); + SNDRV_DMA_TYPE_DEV, + NULL, ST_MIN_BUFFER, + ST_MAX_BUFFER); break; default: ret = snd_pcm_lib_preallocate_pages_for_all(rtd->pcm, - SNDRV_DMA_TYPE_DEV, - NULL, MIN_BUFFER, - MAX_BUFFER); + SNDRV_DMA_TYPE_DEV, + NULL, MIN_BUFFER, + MAX_BUFFER); break; } if (ret < 0) dev_err(component->dev, - "buffer preallocation failer error:%d\n", ret); + "buffer preallocation failer error:%d\n", ret); return ret; } @@ -1007,14 +1024,16 @@ static int acp_dma_close(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct audio_substream_data *rtd = runtime->private_data; struct snd_soc_pcm_runtime *prtd = substream->private_data; - struct snd_soc_component *component = snd_soc_rtdcom_lookup(prtd, DRV_NAME); + struct snd_soc_component *component = snd_soc_rtdcom_lookup(prtd, + DRV_NAME); struct audio_drv_data *adata = dev_get_drvdata(component->dev); kfree(rtd); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { adata->play_i2ssp_stream = NULL; - /* For Stoney, Memory gating is disabled,i.e SRAM Banks + /* + * For Stoney, Memory gating is disabled,i.e SRAM Banks * won't be turned off. The default state for SRAM banks is ON. * Setting SRAM bank state code skipped for STONEY platform. * added condition checks for Carrizo platform only @@ -1022,20 +1041,21 @@ static int acp_dma_close(struct snd_pcm_substream *substream) if (adata->asic_type != CHIP_STONEY) { for (bank = 1; bank <= 4; bank++) acp_set_sram_bank_state(adata->acp_mmio, bank, - false); + false); } } else { adata->capture_i2ssp_stream = NULL; if (adata->asic_type != CHIP_STONEY) { for (bank = 5; bank <= 8; bank++) acp_set_sram_bank_state(adata->acp_mmio, bank, - false); + false); } } - /* Disable ACP irq, when the current stream is being closed and + /* + * Disable ACP irq, when the current stream is being closed and * another stream is also not active. - */ + */ if (!adata->play_i2ssp_stream && !adata->capture_i2ssp_stream) acp_reg_write(0, adata->acp_mmio, mmACP_EXTERNAL_INTR_ENB); @@ -1054,7 +1074,7 @@ static const struct snd_pcm_ops acp_dma_ops = { .prepare = acp_dma_prepare, }; -static struct snd_soc_component_driver acp_asoc_platform = { +static const struct snd_soc_component_driver acp_asoc_platform = { .name = DRV_NAME, .ops = &acp_dma_ops, .pcm_new = acp_dma_new, @@ -1073,8 +1093,8 @@ static int acp_audio_probe(struct platform_device *pdev) } audio_drv_data = devm_kzalloc(&pdev->dev, sizeof(struct audio_drv_data), - GFP_KERNEL); - if (audio_drv_data == NULL) + GFP_KERNEL); + if (!audio_drv_data) return -ENOMEM; res = platform_get_resource(pdev, IORESOURCE_MEM, 0); @@ -1082,7 +1102,8 @@ static int acp_audio_probe(struct platform_device *pdev) if (IS_ERR(audio_drv_data->acp_mmio)) return PTR_ERR(audio_drv_data->acp_mmio); - /* The following members gets populated in device 'open' + /* + * The following members gets populated in device 'open' * function. Till then interrupts are disabled in 'acp_init' * and device doesn't generate any interrupts. */ @@ -1099,7 +1120,7 @@ static int acp_audio_probe(struct platform_device *pdev) } status = devm_request_irq(&pdev->dev, res->start, dma_irq_handler, - 0, "ACP_IRQ", &pdev->dev); + 0, "ACP_IRQ", &pdev->dev); if (status) { dev_err(&pdev->dev, "ACP IRQ request failed\n"); return status; @@ -1115,7 +1136,7 @@ static int acp_audio_probe(struct platform_device *pdev) } status = devm_snd_soc_register_component(&pdev->dev, - &acp_asoc_platform, NULL, 0); + &acp_asoc_platform, NULL, 0); if (status != 0) { dev_err(&pdev->dev, "Fail to register ALSA platform device\n"); return status; @@ -1154,28 +1175,30 @@ static int acp_pcm_resume(struct device *dev) } if (adata->play_i2ssp_stream && adata->play_i2ssp_stream->runtime) { - /* For Stoney, Memory gating is disabled,i.e SRAM Banks + /* + * For Stoney, Memory gating is disabled,i.e SRAM Banks * won't be turned off. The default state for SRAM banks is ON. * Setting SRAM bank state code skipped for STONEY platform. */ if (adata->asic_type != CHIP_STONEY) { for (bank = 1; bank <= 4; bank++) acp_set_sram_bank_state(adata->acp_mmio, bank, - true); + true); } config_acp_dma(adata->acp_mmio, - adata->play_i2ssp_stream->runtime->private_data, - adata->asic_type); + adata->play_i2ssp_stream->runtime->private_data, + adata->asic_type); } - if (adata->capture_i2ssp_stream && adata->capture_i2ssp_stream->runtime) { + if (adata->capture_i2ssp_stream && + adata->capture_i2ssp_stream->runtime) { if (adata->asic_type != CHIP_STONEY) { for (bank = 5; bank <= 8; bank++) acp_set_sram_bank_state(adata->acp_mmio, bank, - true); + true); } config_acp_dma(adata->acp_mmio, - adata->capture_i2ssp_stream->runtime->private_data, - adata->asic_type); + adata->capture_i2ssp_stream->runtime->private_data, + adata->asic_type); } acp_reg_write(1, adata->acp_mmio, mmACP_EXTERNAL_INTR_ENB); return 0; diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index ba01510..0e6089b 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -115,23 +115,25 @@ enum { }; enum { - ACP_DMA_ATTRIBUTES_SHAREDMEM_TO_DAGB_ONION = 0x0, - ACP_DMA_ATTRIBUTES_SHARED_MEM_TO_DAGB_GARLIC = 0x1, - ACP_DMA_ATTRIBUTES_DAGB_ONION_TO_SHAREDMEM = 0x8, - ACP_DMA_ATTRIBUTES_DAGB_GARLIC_TO_SHAREDMEM = 0x9, - ACP_DMA_ATTRIBUTES_FORCE_SIZE = 0xF + ACP_DMA_ATTR_SHAREDMEM_TO_DAGB_ONION = 0x0, + ACP_DMA_ATTR_SHARED_MEM_TO_DAGB_GARLIC = 0x1, + ACP_DMA_ATTR_DAGB_ONION_TO_SHAREDMEM = 0x8, + ACP_DMA_ATTR_DAGB_GARLIC_TO_SHAREDMEM = 0x9, + ACP_DMA_ATTR_FORCE_SIZE = 0xF }; typedef struct acp_dma_dscr_transfer { /* Specifies the source memory location for the DMA data transfer. */ u32 src; - /* Specifies the destination memory location to where the data will + /* + * Specifies the destination memory location to where the data will * be transferred. - */ + */ u32 dest; - /* Specifies the number of bytes need to be transferred - * from source to destination memory.Transfer direction & IOC enable - */ + /* + * Specifies the number of bytes need to be transferred + * from source to destination memory.Transfer direction & IOC enable + */ u32 xfer_val; /* Reserved for future use */ u32 reserved; -- 2.7.4

2 2

[alsa-devel] [PATCH] ASoC: nau8824: user configuration of key detection
by John Hsu 17 Apr '18

17 Apr '18

The SAR ADC of key press detection varies depending on headset. We can't make a set of common threshold values for every case. Therefore, the driver provides configuration for user and they can set up values by UCM configuration. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8824.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 637e952..76502c0 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -409,6 +409,15 @@ static const struct snd_kcontrol_new nau8824_snd_controls[] = { SOC_SINGLE("DACL LR Mix", NAU8824_REG_DAC_MUTE_CTRL, 0, 1, 0), SOC_SINGLE("DACR LR Mix", NAU8824_REG_DAC_MUTE_CTRL, 1, 1, 0), + + SOC_SINGLE("THD for key media", + NAU8824_REG_VDET_THRESHOLD_1, 8, 0xff, 0), + SOC_SINGLE("THD for key voice command", + NAU8824_REG_VDET_THRESHOLD_1, 0, 0xff, 0), + SOC_SINGLE("THD for key volume up", + NAU8824_REG_VDET_THRESHOLD_2, 8, 0xff, 0), + SOC_SINGLE("THD for key volume down", + NAU8824_REG_VDET_THRESHOLD_2, 0, 0xff, 0), }; static int nau8824_output_dac_event(struct snd_soc_dapm_widget *w, -- 2.6.4

2 4

[alsa-devel] [PATCH] ASoC: core: Add name prefix for machines with different topologies
by Liam Girdwood 17 Apr '18

17 Apr '18

Provide a mechanism for drivers to modify there card name with a prefix so that userspace can load a different configuration file to default. This can be useful when topology has changed the PCMs and kcontrols for a machine. This will be used by SOF to differentiate sound card names for UCM etc. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 8 ++++++++ 2 files changed, 10 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 4e368c308f93..99b77feab364 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1015,6 +1015,7 @@ struct snd_soc_platform_driver { /* this platform uses topology and ignore machine driver FEs */ const char *ignore_machine; + const char *topology_name_prefix; int (*be_hw_params_fixup)(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params); bool use_dai_pcm_id; /* use the DAI link PCM ID as PCM device number */ @@ -1167,6 +1168,7 @@ struct snd_soc_card { const char *long_name; const char *driver_name; char dmi_longname[80]; + char topology_shortname[32]; struct device *dev; struct snd_card *snd_card; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 33840e5bc24e..4ab2b4ac8522 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2144,6 +2144,14 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) if (!dai_link->stream_name) dai_link->stream_name = dai_link->name; } + + /* Inform userspace we are using alternate topology */ + if (platform->driver->topology_name_prefix) { + snprintf(card->topology_shortname, 32, "%s-%s", + platform->driver->topology_name_prefix, + card->name); + card->name = card->topology_shortname; + } } } -- 2.14.1

2 3

[alsa-devel] [PATCH] ASoC: topology: Add support for compressed PCMs
by Liam Girdwood 17 Apr '18

17 Apr '18

Register a compressed PCM if topology defines one. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/soc-topology.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index dd2b118a35b3..f941d37e825b 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1768,6 +1768,9 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg, set_stream_info(stream, caps); } + if (pcm->compress) + dai_drv->compress_new = snd_soc_new_compress; + /* pass control to component driver for optional further init */ ret = soc_tplg_dai_load(tplg, dai_drv, pcm, NULL); if (ret < 0) { -- 2.14.1

2 1

[alsa-devel] [PATCH 4/4] ASoC: amd: enabling bt i2s config after acp reset
by Vijendar Mukunda 17 Apr '18

17 Apr '18

On ST/CZ based platforms, for specific platform bt uart mux to be defined for bt i2s. By default, these pins will be used for uart. After acp reset , it requires to reprogram bt i2s config mux pins to enable bt i2s instance. added bt i2s enablement sequence during acp init. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp-da7219-max98357a.c | 2 ++ sound/soc/amd/acp-pcm-dma.c | 9 +++++++++ sound/soc/amd/acp.h | 1 + 3 files changed, 12 insertions(+) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index b205c78..6dad0cb 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -44,6 +44,7 @@ static struct snd_soc_jack cz_jack; struct clk *da7219_dai_clk; +extern int bt_pad_enable; static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) { @@ -81,6 +82,7 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) } da7219_aad_jack_det(component, &cz_jack); + bt_pad_enable = device_property_read_bool(&pdev->dev, "bt-pad-enable"); return 0; } diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 7c392fe..b52c660 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -46,6 +46,8 @@ #define DRV_NAME "acp_audio_dma" +bool bt_pad_enable = false; +EXPORT_SYMBOL(bt_pad_enable); static const struct snd_pcm_hardware acp_pcm_hardware_playback = { .info = SNDRV_PCM_INFO_INTERLEAVED | @@ -525,6 +527,13 @@ static int acp_init(void __iomem *acp_mmio, u32 asic_type) val &= ~ACP_SOFT_RESET__SoftResetAud_MASK; acp_reg_write(val, acp_mmio, mmACP_SOFT_RESET); + /* For BT instance change pins from UART to BT */ + if (bt_pad_enable) { + val = acp_reg_read(acp_mmio, mmACP_BT_UART_PAD_SEL); + val |= ACP_BT_UART_PAD_SELECT_MASK; + acp_reg_write(val, acp_mmio, mmACP_BT_UART_PAD_SEL); + } + /* initiailize Onion control DAGB register */ acp_reg_write(ACP_ONION_CNTL_DEFAULT, acp_mmio, mmACP_AXI2DAGB_ONION_CNTL); diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index 460365c..6b43144 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -107,6 +107,7 @@ #define ACP_I2S_MIC_16BIT_RESOLUTION_EN 0x01 #define ACP_I2S_SP_16BIT_RESOLUTION_EN 0x02 #define ACP_I2S_BT_16BIT_RESOLUTION_EN 0x04 +#define ACP_BT_UART_PAD_SELECT_MASK 0x1 enum acp_dma_priority_level { /* 0x0 Specifies the DMA channel is given normal priority */ -- 2.7.4

5 6

[alsa-devel] [PATCH v3 00/13] soundwire: Add stream support
by Vinod Koul 17 Apr '18

17 Apr '18

This series adds support in SoundWire subsystem for: - Documentation for stream support - stream management - data port management - DAI ops in cadence and Intel drivers - ASoC API to propagate SDW stream Updates in v3: - Remove stream state transition checks and redefine stream states - Update bansk switch routines - Other nitpicks pointed out by Pierre. Updates in v2: - Make ASoC API inlined - Make stream states as states and not action - Update the direction enum - Fix some typos and comment updates Sanyog Kale (7): Documentation: soundwire: Add more documentation soundwire: Add support for SoundWire stream management soundwire: Add support for port management soundwire: Add Master and Slave port programming soundwire: Add helpers for ports operations soundwire: Add bank switch routine soundwire: Add stream configuration APIs Shreyas NC (2): ASoC: Add SoundWire stream programming interface soundwire: Remove cdns_master_ops Vinod Koul (4): soundwire: cdns: Add port routines soundwire: cdns: Add stream routines soundwire: intel: Add stream initialization soundwire: intel: Add audio DAI ops .../driver-api/soundwire/error_handling.rst | 65 + Documentation/driver-api/soundwire/index.rst | 3 + Documentation/driver-api/soundwire/locking.rst | 106 ++ Documentation/driver-api/soundwire/stream.rst | 372 +++++ drivers/soundwire/Kconfig | 2 +- drivers/soundwire/Makefile | 2 +- drivers/soundwire/bus.c | 43 + drivers/soundwire/bus.h | 72 + drivers/soundwire/cadence_master.c | 453 +++++- drivers/soundwire/cadence_master.h | 151 ++ drivers/soundwire/intel.c | 528 ++++++- drivers/soundwire/intel.h | 4 + drivers/soundwire/intel_init.c | 3 + drivers/soundwire/stream.c | 1471 ++++++++++++++++++++ include/linux/soundwire/sdw.h | 332 ++++- include/linux/soundwire/sdw_intel.h | 14 + include/sound/soc-dai.h | 21 + 17 files changed, 3628 insertions(+), 14 deletions(-) create mode 100644 Documentation/driver-api/soundwire/error_handling.rst create mode 100644 Documentation/driver-api/soundwire/locking.rst create mode 100644 Documentation/driver-api/soundwire/stream.rst create mode 100644 drivers/soundwire/stream.c -- 2.7.4

3 21

[alsa-devel] [PATCH RESEND] xen/sndif: Sync up with the canonical definition in Xen
by Oleksandr Andrushchenko 17 Apr '18

17 Apr '18

This is the sync up with the canonical definition of the sound protocol in Xen: 1. Protocol version was referenced in the protocol description, but missed its definition. Fixed by adding a constant for current protocol version. 2. Some of the request descriptions have "reserved" fields missed: fixed by adding corresponding entries. 3. Extend the size of the requests and responses to 64 octets. Bump protocol version to 2. 4. Add explicit back and front synchronization In order to provide explicit synchronization between backend and frontend the following changes are introduced in the protocol: - add new ring buffer for sending asynchronous events from backend to frontend to report number of bytes played by the frontend (XENSND_EVT_CUR_POS) - introduce trigger events for playback control: start/stop/pause/resume - add "req-" prefix to event-channel and ring-ref to unify naming of the Xen event channels for requests and events 5. Add explicit back and front parameter negotiation In order to provide explicit stream parameter negotiation between backend and frontend the following changes are introduced in the protocol: add XENSND_OP_HW_PARAM_QUERY request to read/update configuration space for the parameters given: request passes desired parameter's intervals/masks and the response to this request returns allowed min/max intervals/masks to be used. Signed-off-by: Oleksandr Andrushchenko <oleksandr_andrushchenko(a)epam.com> Signed-off-by: Oleksandr Grytsov <oleksandr_grytsov(a)epam.com> Cc: Konrad Rzeszutek Wilk <konrad.wilk(a)oracle.com> Cc: Takashi Iwai <tiwai(a)suse.de> --- include/xen/interface/io/sndif.h | 322 +++++++++++++++++++++++++++++-- 1 file changed, 306 insertions(+), 16 deletions(-) diff --git a/include/xen/interface/io/sndif.h b/include/xen/interface/io/sndif.h index 5c918276835e..78bb5d9f8d83 100644 --- a/include/xen/interface/io/sndif.h +++ b/include/xen/interface/io/sndif.h @@ -36,6 +36,13 @@ #include "ring.h" #include "../grant_table.h" +/* + ****************************************************************************** + * Protocol version + ****************************************************************************** + */ +#define XENSND_PROTOCOL_VERSION 2 + /* ****************************************************************************** * Feature and Parameter Negotiation @@ -106,6 +113,8 @@ * * /local/domain/1/device/vsnd/0/0/0/ring-ref = "386" * /local/domain/1/device/vsnd/0/0/0/event-channel = "15" + * /local/domain/1/device/vsnd/0/0/0/evt-ring-ref = "1386" + * /local/domain/1/device/vsnd/0/0/0/evt-event-channel = "215" * *------------------------------ Stream 1, capture ---------------------------- * @@ -115,6 +124,8 @@ * * /local/domain/1/device/vsnd/0/0/1/ring-ref = "384" * /local/domain/1/device/vsnd/0/0/1/event-channel = "13" + * /local/domain/1/device/vsnd/0/0/1/evt-ring-ref = "1384" + * /local/domain/1/device/vsnd/0/0/1/evt-event-channel = "213" * *------------------------------- PCM device 1 -------------------------------- * @@ -128,6 +139,8 @@ * * /local/domain/1/device/vsnd/0/1/0/ring-ref = "387" * /local/domain/1/device/vsnd/0/1/0/event-channel = "151" + * /local/domain/1/device/vsnd/0/1/0/evt-ring-ref = "1387" + * /local/domain/1/device/vsnd/0/1/0/evt-event-channel = "351" * *------------------------------- PCM device 2 -------------------------------- * @@ -140,6 +153,8 @@ * * /local/domain/1/device/vsnd/0/2/0/ring-ref = "389" * /local/domain/1/device/vsnd/0/2/0/event-channel = "152" + * /local/domain/1/device/vsnd/0/2/0/evt-ring-ref = "1389" + * /local/domain/1/device/vsnd/0/2/0/evt-event-channel = "452" * ****************************************************************************** * Backend XenBus Nodes @@ -285,6 +300,23 @@ * The Xen grant reference granting permission for the backend to map * a sole page in a single page sized ring buffer. * + *--------------------- Stream Event Transport Parameters --------------------- + * + * This communication path is used to deliver asynchronous events from backend + * to frontend, set up per stream. + * + * evt-event-channel + * Values: <uint32_t> + * + * The identifier of the Xen event channel used to signal activity + * in the ring buffer. + * + * evt-ring-ref + * Values: <uint32_t> + * + * The Xen grant reference granting permission for the backend to map + * a sole page in a single page sized ring buffer. + * ****************************************************************************** * STATE DIAGRAMS ****************************************************************************** @@ -432,6 +464,20 @@ #define XENSND_OP_GET_VOLUME 5 #define XENSND_OP_MUTE 6 #define XENSND_OP_UNMUTE 7 +#define XENSND_OP_TRIGGER 8 +#define XENSND_OP_HW_PARAM_QUERY 9 + +#define XENSND_OP_TRIGGER_START 0 +#define XENSND_OP_TRIGGER_PAUSE 1 +#define XENSND_OP_TRIGGER_STOP 2 +#define XENSND_OP_TRIGGER_RESUME 3 + +/* + ****************************************************************************** + * EVENT CODES + ****************************************************************************** + */ +#define XENSND_EVT_CUR_POS 0 /* ****************************************************************************** @@ -448,6 +494,8 @@ #define XENSND_FIELD_VCARD_LONG_NAME "long-name" #define XENSND_FIELD_RING_REF "ring-ref" #define XENSND_FIELD_EVT_CHNL "event-channel" +#define XENSND_FIELD_EVT_RING_REF "evt-ring-ref" +#define XENSND_FIELD_EVT_EVT_CHNL "evt-event-channel" #define XENSND_FIELD_DEVICE_NAME "name" #define XENSND_FIELD_TYPE "type" #define XENSND_FIELD_STREAM_UNIQUE_ID "unique-id" @@ -526,7 +574,7 @@ * *---------------------------------- Requests --------------------------------- * - * All request packets have the same length (32 octets) + * All request packets have the same length (64 octets) * All request packets have common header: * 0 1 2 3 octet * +----------------+----------------+----------------+----------------+ @@ -559,11 +607,13 @@ * +----------------+----------------+----------------+----------------+ * | gref_directory | 24 * +----------------+----------------+----------------+----------------+ - * | reserved | 28 + * | period_sz | 28 + * +----------------+----------------+----------------+----------------+ + * | reserved | 32 * +----------------+----------------+----------------+----------------+ * |/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/| * +----------------+----------------+----------------+----------------+ - * | reserved | 32 + * | reserved | 64 * +----------------+----------------+----------------+----------------+ * * pcm_rate - uint32_t, stream data rate, Hz @@ -571,6 +621,14 @@ * pcm_channels - uint8_t, number of channels of this stream, * [channels-min; channels-max] * buffer_sz - uint32_t, buffer size to be allocated, octets + * period_sz - uint32_t, event period size, octets + * This is the requested value of the period at which frontend would + * like to receive XENSND_EVT_CUR_POS notifications from the backend when + * stream position advances during playback/capture. + * It shows how many octets are expected to be played/captured before + * sending such an event. + * If set to 0 no XENSND_EVT_CUR_POS events are sent by the backend. + * * gref_directory - grant_ref_t, a reference to the first shared page * describing shared buffer references. At least one page exists. If shared * buffer size (buffer_sz) exceeds what can be addressed by this single page, @@ -585,6 +643,7 @@ struct xensnd_open_req { uint16_t reserved; uint32_t buffer_sz; grant_ref_t gref_directory; + uint32_t period_sz; }; /* @@ -632,7 +691,7 @@ struct xensnd_page_directory { * +----------------+----------------+----------------+----------------+ * |/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/| * +----------------+----------------+----------------+----------------+ - * | reserved | 32 + * | reserved | 64 * +----------------+----------------+----------------+----------------+ * * Request read/write - used for read (for capture) or write (for playback): @@ -650,7 +709,7 @@ struct xensnd_page_directory { * +----------------+----------------+----------------+----------------+ * |/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/| * +----------------+----------------+----------------+----------------+ - * | reserved | 32 + * | reserved | 64 * +----------------+----------------+----------------+----------------+ * * operation - XENSND_OP_READ for read or XENSND_OP_WRITE for write @@ -673,9 +732,11 @@ struct xensnd_rw_req { * +----------------+----------------+----------------+----------------+ * | length | 16 * +----------------+----------------+----------------+----------------+ + * | reserved | 20 + * +----------------+----------------+----------------+----------------+ * |/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/| * +----------------+----------------+----------------+----------------+ - * | reserved | 32 + * | reserved | 64 * +----------------+----------------+----------------+----------------+ * * operation - XENSND_OP_SET_VOLUME for volume set @@ -713,9 +774,11 @@ struct xensnd_rw_req { * +----------------+----------------+----------------+----------------+ * | length | 16 * +----------------+----------------+----------------+----------------+ + * | reserved | 20 + * +----------------+----------------+----------------+----------------+ * |/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/| * +----------------+----------------+----------------+----------------+ - * | reserved | 32 + * | reserved | 64 * +----------------+----------------+----------------+----------------+ * * operation - XENSND_OP_MUTE for mute or XENSND_OP_UNMUTE for unmute @@ -743,32 +806,213 @@ struct xensnd_rw_req { * * The 'struct xensnd_rw_req' is also used for XENSND_OP_SET_VOLUME, * XENSND_OP_GET_VOLUME, XENSND_OP_MUTE, XENSND_OP_UNMUTE. + * + * Request stream running state change - trigger PCM stream running state + * to start, stop, pause or resume: + * + * 0 1 2 3 octet + * +----------------+----------------+----------------+----------------+ + * | id | _OP_TRIGGER | reserved | 4 + * +----------------+----------------+----------------+----------------+ + * | reserved | 8 + * +----------------+----------------+----------------+----------------+ + * | type | reserved | 12 + * +----------------+----------------+----------------+----------------+ + * | reserved | 16 + * +----------------+----------------+----------------+----------------+ + * |/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/| + * +----------------+----------------+----------------+----------------+ + * | reserved | 64 + * +----------------+----------------+----------------+----------------+ + * + * type - uint8_t, XENSND_OP_TRIGGER_XXX value */ +struct xensnd_trigger_req { + uint8_t type; +}; + /* - *---------------------------------- Responses -------------------------------- + * Request stream parameter ranges: request intervals and + * masks of supported ranges for stream configuration values. * - * All response packets have the same length (32 octets) + * Sound device configuration for a particular stream is a limited subset + * of the multidimensional configuration available on XenStore, e.g. + * once the frame rate has been selected there is a limited supported range + * for sample rates becomes available (which might be the same set configured + * on XenStore or less). For example, selecting 96kHz sample rate may limit + * number of channels available for such configuration from 4 to 2, etc. + * Thus, each call to XENSND_OP_HW_PARAM_QUERY may reduce configuration + * space making it possible to iteratively get the final stream configuration, + * used in XENSND_OP_OPEN request. + * + * See response format for this request. * - * Response for all requests: * 0 1 2 3 octet * +----------------+----------------+----------------+----------------+ - * | id | operation | reserved | 4 + * | id | _HW_PARAM_QUERY| reserved | 4 * +----------------+----------------+----------------+----------------+ - * | status | 8 + * | reserved | 8 + * +----------------+----------------+----------------+----------------+ + * | formats mask low 32-bit | 12 + * +----------------+----------------+----------------+----------------+ + * | formats mask high 32-bit | 16 * +----------------+----------------+----------------+----------------+ - * | reserved | 12 + * | min rate | 20 + * +----------------+----------------+----------------+----------------+ + * | max rate | 24 + * +----------------+----------------+----------------+----------------+ + * | min channels | 28 + * +----------------+----------------+----------------+----------------+ + * | max channels | 32 + * +----------------+----------------+----------------+----------------+ + * | min buffer frames | 36 + * +----------------+----------------+----------------+----------------+ + * | max buffer frames | 40 + * +----------------+----------------+----------------+----------------+ + * | min period frames | 44 + * +----------------+----------------+----------------+----------------+ + * | max period frames | 48 + * +----------------+----------------+----------------+----------------+ + * | reserved | 52 * +----------------+----------------+----------------+----------------+ * |/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/| * +----------------+----------------+----------------+----------------+ - * | reserved | 32 + * | reserved | 64 + * +----------------+----------------+----------------+----------------+ + * + * formats - uint64_t, bit mask representing values of the parameter + * made as bitwise OR of (1 << XENSND_PCM_FORMAT_XXX) values + * + * For interval parameters: + * min - uint32_t, minimum value of the parameter + * max - uint32_t, maximum value of the parameter + * + * Frame is defined as a product of the number of channels by the + * number of octets per one sample. + */ + +struct xensnd_query_hw_param { + uint64_t formats; + struct { + uint32_t min; + uint32_t max; + } rates; + struct { + uint32_t min; + uint32_t max; + } channels; + struct { + uint32_t min; + uint32_t max; + } buffer; + struct { + uint32_t min; + uint32_t max; + } period; +}; + +/* + *---------------------------------- Responses -------------------------------- + * + * All response packets have the same length (64 octets) + * + * All response packets have common header: + * 0 1 2 3 octet + * +----------------+----------------+----------------+----------------+ + * | id | operation | reserved | 4 + * +----------------+----------------+----------------+----------------+ + * | status | 8 * +----------------+----------------+----------------+----------------+ * * id - uint16_t, copied from the request * operation - uint8_t, XENSND_OP_* - copied from request * status - int32_t, response status, zero on success and -XEN_EXX on failure + * + * + * HW parameter query response - response for XENSND_OP_HW_PARAM_QUERY: + * 0 1 2 3 octet + * +----------------+----------------+----------------+----------------+ + * | id | operation | reserved | 4 + * +----------------+----------------+----------------+----------------+ + * | status | 8 + * +----------------+----------------+----------------+----------------+ + * | formats mask low 32-bit | 12 + * +----------------+----------------+----------------+----------------+ + * | formats mask high 32-bit | 16 + * +----------------+----------------+----------------+----------------+ + * | min rate | 20 + * +----------------+----------------+----------------+----------------+ + * | max rate | 24 + * +----------------+----------------+----------------+----------------+ + * | min channels | 28 + * +----------------+----------------+----------------+----------------+ + * | max channels | 32 + * +----------------+----------------+----------------+----------------+ + * | min buffer frames | 36 + * +----------------+----------------+----------------+----------------+ + * | max buffer frames | 40 + * +----------------+----------------+----------------+----------------+ + * | min period frames | 44 + * +----------------+----------------+----------------+----------------+ + * | max period frames | 48 + * +----------------+----------------+----------------+----------------+ + * | reserved | 52 + * +----------------+----------------+----------------+----------------+ + * |/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/| + * +----------------+----------------+----------------+----------------+ + * | reserved | 64 + * +----------------+----------------+----------------+----------------+ + * + * Meaning of the values in this response is the same as for + * XENSND_OP_HW_PARAM_QUERY request. + */ + +/* + *----------------------------------- Events ---------------------------------- + * + * Events are sent via shared page allocated by the front and propagated by + * evt-event-channel/evt-ring-ref XenStore entries + * All event packets have the same length (64 octets) + * All event packets have common header: + * 0 1 2 3 octet + * +----------------+----------------+----------------+----------------+ + * | id | type | reserved | 4 + * +----------------+----------------+----------------+----------------+ + * | reserved | 8 + * +----------------+----------------+----------------+----------------+ + * + * id - uint16_t, event id, may be used by front + * type - uint8_t, type of the event + * + * + * Current stream position - event from back to front when stream's + * playback/capture position has advanced: + * 0 1 2 3 octet + * +----------------+----------------+----------------+----------------+ + * | id | _EVT_CUR_POS | reserved | 4 + * +----------------+----------------+----------------+----------------+ + * | reserved | 8 + * +----------------+----------------+----------------+----------------+ + * | position low 32-bit | 12 + * +----------------+----------------+----------------+----------------+ + * | position high 32-bit | 16 + * +----------------+----------------+----------------+----------------+ + * | reserved | 20 + * +----------------+----------------+----------------+----------------+ + * |/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/\/| + * +----------------+----------------+----------------+----------------+ + * | reserved | 64 + * +----------------+----------------+----------------+----------------+ + * + * position - current value of stream's playback/capture position, octets + * */ +struct xensnd_cur_pos_evt { + uint64_t position; +}; + struct xensnd_req { uint16_t id; uint8_t operation; @@ -776,7 +1020,9 @@ struct xensnd_req { union { struct xensnd_open_req open; struct xensnd_rw_req rw; - uint8_t reserved[24]; + struct xensnd_trigger_req trigger; + struct xensnd_query_hw_param hw_param; + uint8_t reserved[56]; } op; }; @@ -785,9 +1031,53 @@ struct xensnd_resp { uint8_t operation; uint8_t reserved; int32_t status; - uint8_t reserved1[24]; + union { + struct xensnd_query_hw_param hw_param; + uint8_t reserved1[56]; + } resp; +}; + +struct xensnd_evt { + uint16_t id; + uint8_t type; + uint8_t reserved[5]; + union { + struct xensnd_cur_pos_evt cur_pos; + uint8_t reserved[56]; + } op; }; DEFINE_RING_TYPES(xen_sndif, struct xensnd_req, struct xensnd_resp); +/* + ****************************************************************************** + * Back to front events delivery + ****************************************************************************** + * In order to deliver asynchronous events from back to front a shared page is + * allocated by front and its granted reference propagated to back via + * XenStore entries (evt-ring-ref/evt-event-channel). + * This page has a common header used by both front and back to synchronize + * access and control event's ring buffer, while back being a producer of the + * events and front being a consumer. The rest of the page after the header + * is used for event packets. + * + * Upon reception of an event(s) front may confirm its reception + * for either each event, group of events or none. + */ + +struct xensnd_event_page { + uint32_t in_cons; + uint32_t in_prod; + uint8_t reserved[56]; +}; + +#define XENSND_EVENT_PAGE_SIZE XEN_PAGE_SIZE +#define XENSND_IN_RING_OFFS (sizeof(struct xensnd_event_page)) +#define XENSND_IN_RING_SIZE (XENSND_EVENT_PAGE_SIZE - XENSND_IN_RING_OFFS) +#define XENSND_IN_RING_LEN (XENSND_IN_RING_SIZE / sizeof(struct xensnd_evt)) +#define XENSND_IN_RING(page) \ + ((struct xensnd_evt *)((char *)(page) + XENSND_IN_RING_OFFS)) +#define XENSND_IN_RING_REF(page, idx) \ + (XENSND_IN_RING((page))[(idx) % XENSND_IN_RING_LEN]) + #endif /* __XEN_PUBLIC_IO_SNDIF_H__ */ -- 2.17.0

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[alsa-devel] [PATCH] ASoC: rt1305: fix ACPI_PTR compile error
by shumingf＠realtek.com 17 Apr '18

17 Apr '18

From: Shuming Fan <shumingf(a)realtek.com> This patch added <linux/acpi.h> header Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1305.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt1305.c b/sound/soc/codecs/rt1305.c index 90baab9..8404306 100644 --- a/sound/soc/codecs/rt1305.c +++ b/sound/soc/codecs/rt1305.c @@ -14,6 +14,7 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> +#include <linux/acpi.h> #include <linux/gpio.h> #include <linux/i2c.h> #include <linux/regmap.h> -- 2.7.4

2 3

[alsa-devel] [asoc:for-4.18 35/37] sound/soc/codecs/rt1305.c:1179:23: error: implicit declaration of function 'ACPI_PTR'; did you mean 'ERR_PTR'?
by kbuild test robot 17 Apr '18

17 Apr '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.18 head: fbeabd09236664f34ea3e4a9f7dcf5a0cdb7fc47 commit: 29bc643ddd7efb741d07c8b2d9a4c3dd9228865b [35/37] ASoC: rt1305: Add RT1305/RT1306 amplifier driver config: i386-allmodconfig (attached as .config) compiler: gcc-7 (Debian 7.3.0-1) 7.3.0 reproduce: git checkout 29bc643ddd7efb741d07c8b2d9a4c3dd9228865b # save the attached .config to linux build tree make ARCH=i386 All error/warnings (new ones prefixed by >>): >> sound/soc/codecs/rt1305.c:1179:23: error: implicit declaration of function 'ACPI_PTR'; did you mean 'ERR_PTR'? [-Werror=implicit-function-declaration] .acpi_match_table = ACPI_PTR(rt1305_acpi_match) ^~~~~~~~ ERR_PTR >> sound/soc/codecs/rt1305.c:1179:23: warning: initialization makes pointer from integer without a cast [-Wint-conversion] sound/soc/codecs/rt1305.c:1179:23: note: (near initialization for 'rt1305_i2c_driver.driver.acpi_match_table') >> sound/soc/codecs/rt1305.c:1179:23: error: initializer element is not constant sound/soc/codecs/rt1305.c:1179:23: note: (near initialization for 'rt1305_i2c_driver.driver.acpi_match_table') cc1: some warnings being treated as errors coccinelle warnings: (new ones prefixed by >>) >> sound/soc/codecs/rt1305.c:1174:3-8: No need to set .owner here. The core will do it. Please review and possibly fold the followup patch. vim +1179 sound/soc/codecs/rt1305.c 1169 1170 1171 static struct i2c_driver rt1305_i2c_driver = { 1172 .driver = { 1173 .name = "rt1305", > 1174 .owner = THIS_MODULE, 1175 #if defined(CONFIG_OF) 1176 .of_match_table = rt1305_of_match, 1177 #endif 1178 #if defined(CONFIG_ACPI) > 1179 .acpi_match_table = ACPI_PTR(rt1305_acpi_match) 1180 #endif 1181 }, 1182 .probe = rt1305_i2c_probe, 1183 .remove = rt1305_i2c_remove, 1184 .shutdown = rt1305_i2c_shutdown, 1185 .id_table = rt1305_i2c_id, 1186 }; 1187 module_i2c_driver(rt1305_i2c_driver); 1188 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH 0/3] ASoC: Intel: Skylake: Driver updates for new DSP FW
by Sriram Periyasamy 17 Apr '18

17 Apr '18

The series contains driver changes which are needed to be compatible with the new DSP Firmware. Divya Prakash (1): ASoC: Intel: Skylake: Reset DSP pipe in skl_pcm_hw_free Pardha Saradhi K (1): ASoC: Intel: Skylake: Reset DSP Pipelines in prepare Pradeep Tewani (1): ASoC: Intel: Skylake: Unify the fw ops for SKL and KBL sound/soc/intel/skylake/skl-messages.c | 2 +- sound/soc/intel/skylake/skl-pcm.c | 36 ++++++++++++++++++++++++++++++---- sound/soc/intel/skylake/skl-sst-dsp.h | 3 --- sound/soc/intel/skylake/skl-sst.c | 34 ++------------------------------ 4 files changed, 35 insertions(+), 40 deletions(-) -- 2.7.4

2 6

[alsa-devel] [PATCH] ASoC: Intel: Skylake: Reset stream to link mapping
by Sanyog Kale 17 Apr '18

17 Apr '18

From: Rakesh Ughreja <rakesh.a.ughreja(a)intel.com> By default all the streams are mapped to all links after controller is reset which causes stream to be broadcast on all the links. This patch resets the stream-link mapping after controller reset. The stream is mapped later to the appropriate link as part of stream setup. Tested-by: Abhijeet Kumar <abhijeet.kumar(a)intel.com> Signed-off-by: Rakesh Ughreja <rakesh.a.ughreja(a)intel.com> Signed-off-by: Sanyog Kale <sanyog.r.kale(a)intel.com> --- sound/soc/intel/skylake/skl.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index abf324747b29..f0d9793f872a 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -127,10 +127,17 @@ static void skl_clock_power_gating(struct device *dev, bool enable) */ static int skl_init_chip(struct hdac_bus *bus, bool full_reset) { + struct hdac_ext_bus *ebus = hbus_to_ebus(bus); + struct hdac_ext_link *hlink; int ret; skl_enable_miscbdcge(bus->dev, false); ret = snd_hdac_bus_init_chip(bus, full_reset); + + /* Reset stream-to-link mapping */ + list_for_each_entry(hlink, &ebus->hlink_list, list) + bus->io_ops->reg_writel(0, hlink->ml_addr + AZX_REG_ML_LOSIDV); + skl_enable_miscbdcge(bus->dev, true); return ret; -- 2.13.0

2 1

[alsa-devel] [PATCH 0/3] PCM cleanups
by Takashi Iwai 17 Apr '18

17 Apr '18

Hi, this is a patch series for minor cleanups of PCM core codes. Mostly to unify the direction-separated PCM callbacks to unified ones. Takashi === Takashi Iwai (3): ALSA: pcm: Clean up with snd_pcm_avail() and snd_pcm_hw_avail() helpers ALSA: pcm: Unify playback and capture poll callbacks ALSA: pcm: Unify delay calculation in snd_pcm_status() and snd_pcm_delay() sound/core/pcm_compat.c | 10 +-- sound/core/pcm_lib.c | 15 +---- sound/core/pcm_local.h | 18 +++++ sound/core/pcm_native.c | 175 ++++++++++++------------------------------------ 4 files changed, 67 insertions(+), 151 deletions(-) -- 2.16.3

2 7

[alsa-devel] [PATCH v3 1/3] dell-led: Change dell-led.h to dell-common.h
by Kai-Heng Feng 17 Apr '18

17 Apr '18

This header will be used for more than just led. Change it to a more generic name. Cc: Mario Limonciello <mario.limonciello(a)dell.com> Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- v3: Simplify dell_switchable_gfx_is_enabled() by returning bool instead of error code. Use DMI_DEV_TYPE_OEM_STRING to match Dell System. v2: Mario suggested to squash the HDA part into the same series. drivers/platform/x86/dell-laptop.c | 2 +- include/linux/{dell-led.h => dell-common.h} | 4 ++-- sound/pci/hda/dell_wmi_helper.c | 2 +- 3 files changed, 4 insertions(+), 4 deletions(-) rename include/linux/{dell-led.h => dell-common.h} (61%) diff --git a/drivers/platform/x86/dell-laptop.c b/drivers/platform/x86/dell-laptop.c index c52c6723374b..8ba820e6c3d0 100644 --- a/drivers/platform/x86/dell-laptop.c +++ b/drivers/platform/x86/dell-laptop.c @@ -29,7 +29,7 @@ #include <linux/mm.h> #include <linux/i8042.h> #include <linux/debugfs.h> -#include <linux/dell-led.h> +#include <linux/dell-common.h> #include <linux/seq_file.h> #include <acpi/video.h> #include "dell-rbtn.h" diff --git a/include/linux/dell-led.h b/include/linux/dell-common.h similarity index 61% rename from include/linux/dell-led.h rename to include/linux/dell-common.h index 92521471517f..37e4b614dd74 100644 --- a/include/linux/dell-led.h +++ b/include/linux/dell-common.h @@ -1,6 +1,6 @@ /* SPDX-License-Identifier: GPL-2.0 */ -#ifndef __DELL_LED_H__ -#define __DELL_LED_H__ +#ifndef __DELL_COMMON_H__ +#define __DELL_COMMON_H__ int dell_micmute_led_set(int on); diff --git a/sound/pci/hda/dell_wmi_helper.c b/sound/pci/hda/dell_wmi_helper.c index 1b48a8c19d28..56050cc3c0ee 100644 --- a/sound/pci/hda/dell_wmi_helper.c +++ b/sound/pci/hda/dell_wmi_helper.c @@ -4,7 +4,7 @@ */ #if IS_ENABLED(CONFIG_DELL_LAPTOP) -#include <linux/dell-led.h> +#include <linux/dell-common.h> enum { MICMUTE_LED_ON, -- 2.17.0

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[alsa-devel] [PATCH V2] ASoC: amd: dma driver changes for BT I2S controller instance
by Vijendar Mukunda 17 Apr '18

17 Apr '18

With in ACP, There are three I2S controllers can be configured/enabled ( I2S SP, I2S MICSP, I2S BT). Default enabled I2S controller instance is I2S SP. This patch provides required changes to support I2S BT controller Instance. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> Reviewed-by: Alex Deucher <alexander.deucher(a)amd.com> v1->v2: fixed kbuild errors --- sound/soc/amd/acp-pcm-dma.c | 116 +++++++++++++++++++++++++++++++++++--------- sound/soc/amd/acp.h | 17 +++++++ 2 files changed, 111 insertions(+), 22 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 540088d..cad995e 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -697,6 +697,9 @@ static int acp_dma_open(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *prtd = substream->private_data; struct snd_soc_component *component = snd_soc_rtdcom_lookup(prtd, DRV_NAME); struct audio_drv_data *intr_data = dev_get_drvdata(component->dev); +#ifdef CONFIG_SND_DESIGNWARE_PCM + struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(prtd->cpu_dai); +#endif struct audio_substream_data *adata = kzalloc(sizeof(struct audio_substream_data), GFP_KERNEL); if (adata == NULL) @@ -710,7 +713,29 @@ static int acp_dma_open(struct snd_pcm_substream *substream) default: runtime->hw = acp_pcm_hardware_playback; } +#ifdef CONFIG_SND_DESIGNWARE_PCM + adata->i2s_play_instance = dev->i2s_instance; +#else + adata->i2s_play_instance = I2S_SP_INSTANCE; +#endif + if (adata->i2s_play_instance == I2S_SP_INSTANCE) + adata->i2ssp_renderbytescount = 0; + else if (adata->i2s_play_instance == I2S_BT_INSTANCE) + adata->i2sbt_renderbytescount = 0; + else + return -EINVAL; } else { +#ifdef CONFIG_SND_DESIGNWARE_PCM + adata->i2s_capture_instance = dev->i2s_instance; +#else + adata->i2s_capture_instance = I2S_SP_INSTANCE; +#endif + if (adata->i2s_capture_instance == I2S_SP_INSTANCE) + adata->i2ssp_capturebytescount = 0; + else if (adata->i2s_capture_instance == I2S_BT_INSTANCE) + adata->i2sbt_capturebytescount = 0; + else + return -EINVAL; switch (intr_data->asic_type) { case CHIP_STONEY: runtime->hw = acp_st_pcm_hardware_capture; @@ -736,11 +761,20 @@ static int acp_dma_open(struct snd_pcm_substream *substream) * This enablement is not required for another stream, if current * stream is not closed */ - if (!intr_data->play_i2ssp_stream && !intr_data->capture_i2ssp_stream) + if (!intr_data->play_i2ssp_stream && !intr_data->capture_i2ssp_stream && + !intr_data->play_i2sbt_stream && + !intr_data->capture_i2sbt_stream) acp_reg_write(1, adata->acp_mmio, mmACP_EXTERNAL_INTR_ENB); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - intr_data->play_i2ssp_stream = substream; + switch (adata->i2s_play_instance) { + case I2S_BT_INSTANCE: + intr_data->play_i2sbt_stream = substream; + break; + case I2S_SP_INSTANCE: + default: + intr_data->play_i2ssp_stream = substream; + } /* For Stoney, Memory gating is disabled,i.e SRAM Banks * won't be turned off. The default state for SRAM banks is ON. * Setting SRAM bank state code skipped for STONEY platform. @@ -751,7 +785,14 @@ static int acp_dma_open(struct snd_pcm_substream *substream) bank, true); } } else { - intr_data->capture_i2ssp_stream = substream; + switch (adata->i2s_capture_instance) { + case I2S_BT_INSTANCE: + intr_data->capture_i2sbt_stream = substream; + break; + case I2S_SP_INSTANCE: + default: + intr_data->capture_i2ssp_stream = substream; + } if (intr_data->asic_type != CHIP_STONEY) { for (bank = 5; bank <= 8; bank++) acp_set_sram_bank_state(intr_data->acp_mmio, @@ -1010,34 +1051,49 @@ static int acp_dma_close(struct snd_pcm_substream *substream) struct snd_soc_component *component = snd_soc_rtdcom_lookup(prtd, DRV_NAME); struct audio_drv_data *adata = dev_get_drvdata(component->dev); - kfree(rtd); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - adata->play_i2ssp_stream = NULL; - /* For Stoney, Memory gating is disabled,i.e SRAM Banks - * won't be turned off. The default state for SRAM banks is ON. - * Setting SRAM bank state code skipped for STONEY platform. - * added condition checks for Carrizo platform only - */ - if (adata->asic_type != CHIP_STONEY) { - for (bank = 1; bank <= 4; bank++) - acp_set_sram_bank_state(adata->acp_mmio, bank, - false); + switch (rtd->i2s_play_instance) { + case I2S_BT_INSTANCE: + adata->play_i2sbt_stream = NULL; + break; + case I2S_SP_INSTANCE: + default: + adata->play_i2ssp_stream = NULL; + /* For Stoney, Memory gating is disabled,i.e SRAM Banks + * won't be turned off. The default state for SRAM banks + * is ON.Setting SRAM bank state code skipped for STONEY + * platform.Added condition checks for Carrizo platform + * only. + */ + if (adata->asic_type != CHIP_STONEY) { + for (bank = 1; bank <= 4; bank++) + acp_set_sram_bank_state(adata->acp_mmio, + bank, false); + } } } else { - adata->capture_i2ssp_stream = NULL; - if (adata->asic_type != CHIP_STONEY) { - for (bank = 5; bank <= 8; bank++) - acp_set_sram_bank_state(adata->acp_mmio, bank, - false); + switch (rtd->i2s_capture_instance) { + case I2S_BT_INSTANCE: + adata->capture_i2sbt_stream = NULL; + break; + case I2S_SP_INSTANCE: + default: + adata->capture_i2ssp_stream = NULL; + if (adata->asic_type != CHIP_STONEY) { + for (bank = 5; bank <= 8; bank++) + acp_set_sram_bank_state(adata->acp_mmio, + bank, false); + } } } /* Disable ACP irq, when the current stream is being closed and * another stream is also not active. - */ - if (!adata->play_i2ssp_stream && !adata->capture_i2ssp_stream) + */ + if (!adata->play_i2ssp_stream && !adata->capture_i2ssp_stream && + !adata->play_i2sbt_stream && !adata->capture_i2sbt_stream) acp_reg_write(0, adata->acp_mmio, mmACP_EXTERNAL_INTR_ENB); + kfree(rtd); return 0; } @@ -1089,6 +1145,8 @@ static int acp_audio_probe(struct platform_device *pdev) audio_drv_data->play_i2ssp_stream = NULL; audio_drv_data->capture_i2ssp_stream = NULL; + audio_drv_data->play_i2sbt_stream = NULL; + audio_drv_data->capture_i2sbt_stream = NULL; audio_drv_data->asic_type = *pdata; @@ -1177,6 +1235,20 @@ static int acp_pcm_resume(struct device *dev) adata->capture_i2ssp_stream->runtime->private_data, adata->asic_type); } + if (adata->asic_type != CHIP_CARRIZO) { + if (adata->play_i2sbt_stream && + adata->play_i2sbt_stream->runtime) { + config_acp_dma(adata->acp_mmio, + adata->play_i2sbt_stream->runtime->private_data, + adata->asic_type); + } + if (adata->capture_i2sbt_stream && + adata->capture_i2sbt_stream->runtime) { + config_acp_dma(adata->acp_mmio, + adata->capture_i2sbt_stream->runtime->private_data, + adata->asic_type); + } + } acp_reg_write(1, adata->acp_mmio, mmACP_EXTERNAL_INTR_ENB); return 0; } diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index ba01510..7ad165b 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -4,6 +4,17 @@ #include "include/acp_2_2_d.h" #include "include/acp_2_2_sh_mask.h" +#ifdef CONFIG_SND_DESIGNWARE_PCM +#include "../dwc/local.h" +#endif + +#ifndef I2S_SP_INSTANCE +#define I2S_SP_INSTANCE 0x01 +#endif + +#ifndef I2S_BT_INSTANCE +#define I2S_BT_INSTANCE 0x02 +#endif #define ACP_PAGE_SIZE_4K_ENABLE 0x02 @@ -88,12 +99,18 @@ struct audio_substream_data { uint64_t size; u64 i2ssp_renderbytescount; u64 i2ssp_capturebytescount; + u64 i2sbt_renderbytescount; + u64 i2sbt_capturebytescount; void __iomem *acp_mmio; + u16 i2s_play_instance; + u16 i2s_capture_instance; }; struct audio_drv_data { struct snd_pcm_substream *play_i2ssp_stream; struct snd_pcm_substream *capture_i2ssp_stream; + struct snd_pcm_substream *play_i2sbt_stream; + struct snd_pcm_substream *capture_i2sbt_stream; void __iomem *acp_mmio; u32 asic_type; }; -- 2.7.4

3 2

[alsa-devel] [PATCH 3/4] ASoC: amd: dma driver changes for BT I2S instance
by Vijendar Mukunda 17 Apr '18

17 Apr '18

With in ACP, There are three I2S controllers can be configured/enabled ( I2S SP, I2S MICSP, I2S BT). Default enabled I2S controller instance is I2S SP. This patch provides required changes to support I2S BT controller Instance. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/Kconfig | 1 + sound/soc/amd/acp-pcm-dma.c | 388 +++++++++++++++++++++++++++----------------- sound/soc/amd/acp.h | 64 +++++++- 3 files changed, 295 insertions(+), 158 deletions(-) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 6cbf9cf..6b7c620 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -1,5 +1,6 @@ config SND_SOC_AMD_ACP tristate "AMD Audio Coprocessor support" + select SND_DESIGNWARE_PCM help This option enables ACP DMA support on AMD platform. diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 5ffe2ef..7c392fe 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -23,6 +23,8 @@ #include <drm/amd_asic_type.h> #include "acp.h" +#include "../dwc/local.h" + #define DRV_NAME "acp_audio_dma" #define PLAYBACK_MIN_NUM_PERIODS 2 @@ -37,13 +39,14 @@ #define MAX_BUFFER (PLAYBACK_MAX_PERIOD_SIZE * PLAYBACK_MAX_NUM_PERIODS) #define MIN_BUFFER MAX_BUFFER -#define ST_PLAYBACK_MAX_PERIOD_SIZE 8192 +#define ST_PLAYBACK_MAX_PERIOD_SIZE 4096 #define ST_CAPTURE_MAX_PERIOD_SIZE ST_PLAYBACK_MAX_PERIOD_SIZE #define ST_MAX_BUFFER (ST_PLAYBACK_MAX_PERIOD_SIZE * PLAYBACK_MAX_NUM_PERIODS) #define ST_MIN_BUFFER ST_MAX_BUFFER #define DRV_NAME "acp_audio_dma" + static const struct snd_pcm_hardware acp_pcm_hardware_playback = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | @@ -317,54 +320,21 @@ static void acp_pte_config(void __iomem *acp_mmio, struct page *pg, } static void config_acp_dma(void __iomem *acp_mmio, - struct audio_substream_data *audio_config, + struct audio_substream_data *rtd, u32 asic_type) { - u32 pte_offset, sram_bank; - u16 ch1, ch2, destination, dma_dscr_idx; - - if (audio_config->direction == SNDRV_PCM_STREAM_PLAYBACK) { - pte_offset = ACP_PLAYBACK_PTE_OFFSET; - ch1 = SYSRAM_TO_ACP_CH_NUM; - ch2 = ACP_TO_I2S_DMA_CH_NUM; - sram_bank = ACP_SHARED_RAM_BANK_1_ADDRESS; - destination = TO_ACP_I2S_1; - - } else { - pte_offset = ACP_CAPTURE_PTE_OFFSET; - ch1 = SYSRAM_TO_ACP_CH_NUM; - ch2 = ACP_TO_I2S_DMA_CH_NUM; - switch (asic_type) { - case CHIP_STONEY: - sram_bank = ACP_SHARED_RAM_BANK_3_ADDRESS; - break; - default: - sram_bank = ACP_SHARED_RAM_BANK_5_ADDRESS; - } - destination = FROM_ACP_I2S_1; - } - - acp_pte_config(acp_mmio, audio_config->pg, audio_config->num_of_pages, - pte_offset); - if (audio_config->direction == SNDRV_PCM_STREAM_PLAYBACK) - dma_dscr_idx = PLAYBACK_START_DMA_DESCR_CH12; - else - dma_dscr_idx = CAPTURE_START_DMA_DESCR_CH14; - + acp_pte_config(acp_mmio, rtd->pg, rtd->num_of_pages, + rtd->pte_offset); /* Configure System memory <-> ACP SRAM DMA descriptors */ - set_acp_sysmem_dma_descriptors(acp_mmio, audio_config->size, - audio_config->direction, pte_offset, ch1, - sram_bank, dma_dscr_idx, asic_type); - - if (audio_config->direction == SNDRV_PCM_STREAM_PLAYBACK) - dma_dscr_idx = PLAYBACK_START_DMA_DESCR_CH13; - else - dma_dscr_idx = CAPTURE_START_DMA_DESCR_CH15; + set_acp_sysmem_dma_descriptors(acp_mmio, rtd->size, + rtd->direction, rtd->pte_offset, + rtd->ch1, rtd->sram_bank, + rtd->dma_dscr_idx_1, asic_type); /* Configure ACP SRAM <-> I2S DMA descriptors */ - set_acp_to_i2s_dma_descriptors(acp_mmio, audio_config->size, - audio_config->direction, sram_bank, - destination, ch2, dma_dscr_idx, - asic_type); + set_acp_to_i2s_dma_descriptors(acp_mmio, rtd->size, + rtd->direction, rtd->sram_bank, + rtd->destination, rtd->ch2, + rtd->dma_dscr_idx_2, asic_type); } /* Start a given DMA channel transfer */ @@ -390,6 +360,9 @@ static void acp_dma_start(void __iomem *acp_mmio, case ACP_TO_I2S_DMA_CH_NUM: case ACP_TO_SYSRAM_CH_NUM: case I2S_TO_ACP_DMA_CH_NUM: + case ACP_TO_I2S_DMA_BT_INSTANCE_CH_NUM: + case ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM: + case I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM: dma_ctrl |= ACP_DMA_CNTL_0__DMAChIOCEn_MASK; break; default: @@ -670,6 +643,24 @@ static irqreturn_t dma_irq_handler(int irq, void *arg) acp_mmio, mmACP_EXTERNAL_INTR_STAT); } + if ((intr_flag & BIT(ACP_TO_I2S_DMA_BT_INSTANCE_CH_NUM)) != 0) { + valid_irq = true; + if (acp_reg_read(acp_mmio, mmACP_DMA_CUR_DSCR_9) == + PLAYBACK_START_DMA_DESCR_CH9) + dscr_idx = PLAYBACK_END_DMA_DESCR_CH8; + else + dscr_idx = PLAYBACK_START_DMA_DESCR_CH8; + config_acp_dma_channel(acp_mmio, + SYSRAM_TO_ACP_BT_INSTANCE_CH_NUM, + dscr_idx, 1, 0); + acp_dma_start(acp_mmio, SYSRAM_TO_ACP_BT_INSTANCE_CH_NUM, + false); + snd_pcm_period_elapsed(irq_data->play_i2sbt_stream); + acp_reg_write((intr_flag & + BIT(ACP_TO_I2S_DMA_BT_INSTANCE_CH_NUM)) << 16, + acp_mmio, mmACP_EXTERNAL_INTR_STAT); + } + if ((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) != 0) { valid_irq = true; if (acp_reg_read(acp_mmio, mmACP_DMA_CUR_DSCR_15) == @@ -692,6 +683,31 @@ static irqreturn_t dma_irq_handler(int irq, void *arg) acp_mmio, mmACP_EXTERNAL_INTR_STAT); } + if ((intr_flag & BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) != 0) { + valid_irq = true; + if (acp_reg_read(acp_mmio, mmACP_DMA_CUR_DSCR_11) == + CAPTURE_START_DMA_DESCR_CH11) + dscr_idx = CAPTURE_END_DMA_DESCR_CH10; + else + dscr_idx = CAPTURE_START_DMA_DESCR_CH10; + config_acp_dma_channel(acp_mmio, + ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM, + dscr_idx, 1, 0); + acp_dma_start(acp_mmio, ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM, + false); + acp_reg_write((intr_flag & + BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) << 16, + acp_mmio, mmACP_EXTERNAL_INTR_STAT); + } + + if ((intr_flag & BIT(ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM)) != 0) { + valid_irq = true; + snd_pcm_period_elapsed(irq_data->capture_i2sbt_stream); + acp_reg_write((intr_flag & + BIT(ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM)) << 16, + acp_mmio, mmACP_EXTERNAL_INTR_STAT); + } + if (valid_irq) return IRQ_HANDLED; else @@ -707,6 +723,7 @@ static int acp_dma_open(struct snd_pcm_substream *substream) struct snd_soc_component *component = snd_soc_rtdcom_lookup(prtd, DRV_NAME); struct audio_drv_data *intr_data = dev_get_drvdata(component->dev); + struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(prtd->cpu_dai); struct audio_substream_data *adata = kzalloc(sizeof(struct audio_substream_data), GFP_KERNEL); if (!adata) @@ -720,6 +737,8 @@ static int acp_dma_open(struct snd_pcm_substream *substream) default: runtime->hw = acp_pcm_hardware_playback; } + adata->i2s_play_instance = dev->i2s_instance; + adata->bytescount = 0; } else { switch (intr_data->asic_type) { case CHIP_STONEY: @@ -728,6 +747,8 @@ static int acp_dma_open(struct snd_pcm_substream *substream) default: runtime->hw = acp_pcm_hardware_capture; } + adata->i2s_capture_instance = dev->i2s_instance; + adata->bytescount = 0; } ret = snd_pcm_hw_constraint_integer(runtime, @@ -747,11 +768,46 @@ static int acp_dma_open(struct snd_pcm_substream *substream) * This enablement is not required for another stream, if current * stream is not closed */ - if (!intr_data->play_i2ssp_stream && !intr_data->capture_i2ssp_stream) + if (!intr_data->play_i2ssp_stream && !intr_data->capture_i2ssp_stream && + !intr_data->play_i2sbt_stream && !intr_data->capture_i2sbt_stream) acp_reg_write(1, adata->acp_mmio, mmACP_EXTERNAL_INTR_ENB); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - intr_data->play_i2ssp_stream = substream; + switch (adata->i2s_play_instance) { + case I2S_BT_INSTANCE: + adata->pte_offset = ACP_ST_BT_PLAYBACK_PTE_OFFSET; + adata->ch1 = SYSRAM_TO_ACP_BT_INSTANCE_CH_NUM; + adata->ch2 = ACP_TO_I2S_DMA_BT_INSTANCE_CH_NUM; + adata->sram_bank = ACP_SRAM_BANK_3_ADDRESS; + adata->destination = TO_BLUETOOTH; + adata->dma_dscr_idx_1 = PLAYBACK_START_DMA_DESCR_CH8; + adata->dma_dscr_idx_2 = PLAYBACK_START_DMA_DESCR_CH9; + adata->byte_cnt_high = + mmACP_I2S_BT_TRANSMIT_BYTE_CNT_HIGH; + adata->byte_cnt_low = + mmACP_I2S_BT_TRANSMIT_BYTE_CNT_LOW; + intr_data->play_i2sbt_stream = substream; + break; + case I2S_SP_INSTANCE: + default: + switch (intr_data->asic_type) { + case CHIP_STONEY: + adata->pte_offset = ACP_ST_PLAYBACK_PTE_OFFSET; + break; + default: + adata->pte_offset = ACP_PLAYBACK_PTE_OFFSET; + } + adata->ch1 = SYSRAM_TO_ACP_CH_NUM; + adata->ch2 = ACP_TO_I2S_DMA_CH_NUM; + adata->sram_bank = ACP_SRAM_BANK_1_ADDRESS; + adata->destination = TO_ACP_I2S_1; + adata->dma_dscr_idx_1 = PLAYBACK_START_DMA_DESCR_CH12; + adata->dma_dscr_idx_2 = PLAYBACK_START_DMA_DESCR_CH13; + adata->byte_cnt_high = mmACP_I2S_TRANSMIT_BYTE_CNT_HIGH; + adata->byte_cnt_low = mmACP_I2S_TRANSMIT_BYTE_CNT_LOW; + intr_data->play_i2ssp_stream = substream; + break; + } /* * For Stoney, Memory gating is disabled,i.e SRAM Banks * won't be turned off. The default state for SRAM banks is ON. @@ -763,7 +819,40 @@ static int acp_dma_open(struct snd_pcm_substream *substream) bank, true); } } else { - intr_data->capture_i2ssp_stream = substream; + switch (adata->i2s_capture_instance) { + case I2S_BT_INSTANCE: + adata->pte_offset = ACP_ST_BT_CAPTURE_PTE_OFFSET; + adata->ch1 = ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM; + adata->ch2 = I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM; + adata->sram_bank = ACP_SRAM_BANK_4_ADDRESS; + adata->destination = FROM_BLUETOOTH; + adata->dma_dscr_idx_1 = CAPTURE_START_DMA_DESCR_CH10; + adata->dma_dscr_idx_2 = CAPTURE_START_DMA_DESCR_CH11; + adata->byte_cnt_high = + mmACP_I2S_BT_RECEIVE_BYTE_CNT_HIGH; + adata->byte_cnt_low = + mmACP_I2S_BT_RECEIVE_BYTE_CNT_LOW; + intr_data->capture_i2sbt_stream = substream; + break; + case I2S_SP_INSTANCE: + default: + adata->pte_offset = ACP_CAPTURE_PTE_OFFSET; + adata->ch1 = SYSRAM_TO_ACP_CH_NUM; + adata->ch2 = ACP_TO_I2S_DMA_CH_NUM; + switch (intr_data->asic_type) { + case CHIP_STONEY: + adata->sram_bank = ACP_SRAM_BANK_2_ADDRESS; + break; + default: + adata->sram_bank = ACP_SRAM_BANK_5_ADDRESS; + } + adata->destination = FROM_ACP_I2S_1; + adata->dma_dscr_idx_1 = CAPTURE_START_DMA_DESCR_CH14; + adata->dma_dscr_idx_2 = CAPTURE_START_DMA_DESCR_CH15; + adata->byte_cnt_high = mmACP_I2S_RECEIVED_BYTE_CNT_HIGH; + adata->byte_cnt_low = mmACP_I2S_RECEIVED_BYTE_CNT_LOW; + intr_data->capture_i2ssp_stream = substream; + } if (intr_data->asic_type != CHIP_STONEY) { for (bank = 5; bank <= 8; bank++) acp_set_sram_bank_state(intr_data->acp_mmio, @@ -797,10 +886,25 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, if (adata->asic_type == CHIP_STONEY) { val = acp_reg_read(adata->acp_mmio, mmACP_I2S_16BIT_RESOLUTION_EN); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - val |= ACP_I2S_SP_16BIT_RESOLUTION_EN; - else - val |= ACP_I2S_MIC_16BIT_RESOLUTION_EN; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + switch (rtd->i2s_play_instance) { + case I2S_BT_INSTANCE: + val |= ACP_I2S_BT_16BIT_RESOLUTION_EN; + break; + case I2S_SP_INSTANCE: + default: + val |= ACP_I2S_SP_16BIT_RESOLUTION_EN; + } + } else { + switch (rtd->i2s_capture_instance) { + case I2S_BT_INSTANCE: + val |= ACP_I2S_BT_16BIT_RESOLUTION_EN; + break; + case I2S_SP_INSTANCE: + default: + val |= ACP_I2S_MIC_16BIT_RESOLUTION_EN; + } + } acp_reg_write(val, adata->acp_mmio, mmACP_I2S_16BIT_RESOLUTION_EN); } @@ -837,26 +941,15 @@ static int acp_dma_hw_free(struct snd_pcm_substream *substream) return snd_pcm_lib_free_pages(substream); } -static u64 acp_get_byte_count(void __iomem *acp_mmio, int stream) +static u64 acp_get_byte_count(struct audio_substream_data *rtd) { - union acp_dma_count playback_dma_count; - union acp_dma_count capture_dma_count; - u64 bytescount = 0; + union acp_dma_count byte_count; - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - playback_dma_count.bcount.high = acp_reg_read(acp_mmio, - mmACP_I2S_TRANSMIT_BYTE_CNT_HIGH); - playback_dma_count.bcount.low = acp_reg_read(acp_mmio, - mmACP_I2S_TRANSMIT_BYTE_CNT_LOW); - bytescount = playback_dma_count.bytescount; - } else { - capture_dma_count.bcount.high = acp_reg_read(acp_mmio, - mmACP_I2S_RECEIVED_BYTE_CNT_HIGH); - capture_dma_count.bcount.low = acp_reg_read(acp_mmio, - mmACP_I2S_RECEIVED_BYTE_CNT_LOW); - bytescount = capture_dma_count.bytescount; - } - return bytescount; + byte_count.bcount.high = acp_reg_read(rtd->acp_mmio, + rtd->byte_cnt_high); + byte_count.bcount.low = acp_reg_read(rtd->acp_mmio, + rtd->byte_cnt_low); + return byte_count.bytescount; } static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) @@ -872,15 +965,10 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) return -EINVAL; buffersize = frames_to_bytes(runtime, runtime->buffer_size); - bytescount = acp_get_byte_count(rtd->acp_mmio, substream->stream); + bytescount = acp_get_byte_count(rtd); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (bytescount > rtd->i2ssp_renderbytescount) - bytescount = bytescount - rtd->i2ssp_renderbytescount; - } else { - if (bytescount > rtd->i2ssp_capturebytescount) - bytescount = bytescount - rtd->i2ssp_capturebytescount; - } + if (bytescount > rtd->bytescount) + bytescount = bytescount - rtd->bytescount; pos = do_div(bytescount, buffersize); return bytes_to_frames(runtime, pos); } @@ -898,21 +986,14 @@ static int acp_dma_prepare(struct snd_pcm_substream *substream) if (!rtd) return -EINVAL; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - config_acp_dma_channel(rtd->acp_mmio, SYSRAM_TO_ACP_CH_NUM, - PLAYBACK_START_DMA_DESCR_CH12, - NUM_DSCRS_PER_CHANNEL, 0); - config_acp_dma_channel(rtd->acp_mmio, ACP_TO_I2S_DMA_CH_NUM, - PLAYBACK_START_DMA_DESCR_CH13, - NUM_DSCRS_PER_CHANNEL, 0); - } else { - config_acp_dma_channel(rtd->acp_mmio, ACP_TO_SYSRAM_CH_NUM, - CAPTURE_START_DMA_DESCR_CH14, - NUM_DSCRS_PER_CHANNEL, 0); - config_acp_dma_channel(rtd->acp_mmio, I2S_TO_ACP_DMA_CH_NUM, - CAPTURE_START_DMA_DESCR_CH15, - NUM_DSCRS_PER_CHANNEL, 0); - } + config_acp_dma_channel(rtd->acp_mmio, + rtd->ch1, + rtd->dma_dscr_idx_1, + NUM_DSCRS_PER_CHANNEL, 0); + config_acp_dma_channel(rtd->acp_mmio, + rtd->ch2, + rtd->dma_dscr_idx_2, + NUM_DSCRS_PER_CHANNEL, 0); return 0; } @@ -934,15 +1015,13 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: - bytescount = acp_get_byte_count(rtd->acp_mmio, - substream->stream); + bytescount = acp_get_byte_count(rtd); + if (rtd->bytescount == 0) + rtd->bytescount = bytescount; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (rtd->i2ssp_renderbytescount == 0) - rtd->i2ssp_renderbytescount = bytescount; - acp_dma_start(rtd->acp_mmio, - SYSRAM_TO_ACP_CH_NUM, false); + acp_dma_start(rtd->acp_mmio, rtd->ch1, false); while (acp_reg_read(rtd->acp_mmio, mmACP_DMA_CH_STS) & - BIT(SYSRAM_TO_ACP_CH_NUM)) { + BIT(rtd->ch1)) { if (!loops--) { dev_err(component->dev, "acp dma start timeout\n"); @@ -950,40 +1029,21 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) } cpu_relax(); } - - acp_dma_start(rtd->acp_mmio, - ACP_TO_I2S_DMA_CH_NUM, true); - - } else { - if (rtd->i2ssp_capturebytescount == 0) - rtd->i2ssp_capturebytescount = bytescount; - acp_dma_start(rtd->acp_mmio, - I2S_TO_ACP_DMA_CH_NUM, true); } + acp_dma_start(rtd->acp_mmio, rtd->ch2, true); ret = 0; break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: - /* - * Need to stop only circular DMA channels : - * ACP_TO_I2S_DMA_CH_NUM / I2S_TO_ACP_DMA_CH_NUM. Non-circular - * channels will stopped automatically after its transfer - * completes : SYSRAM_TO_ACP_CH_NUM / ACP_TO_SYSRAM_CH_NUM - */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - ret = acp_dma_stop(rtd->acp_mmio, - SYSRAM_TO_ACP_CH_NUM); - ret = acp_dma_stop(rtd->acp_mmio, - ACP_TO_I2S_DMA_CH_NUM); - rtd->i2ssp_renderbytescount = 0; + acp_dma_stop(rtd->acp_mmio, rtd->ch1); + ret = acp_dma_stop(rtd->acp_mmio, rtd->ch2); } else { - ret = acp_dma_stop(rtd->acp_mmio, - I2S_TO_ACP_DMA_CH_NUM); - ret = acp_dma_stop(rtd->acp_mmio, - ACP_TO_SYSRAM_CH_NUM); - rtd->i2ssp_capturebytescount = 0; + acp_dma_stop(rtd->acp_mmio, rtd->ch2); + ret = acp_dma_stop(rtd->acp_mmio, rtd->ch1); } + rtd->bytescount = 0; break; default: ret = -EINVAL; @@ -1028,27 +1088,40 @@ static int acp_dma_close(struct snd_pcm_substream *substream) DRV_NAME); struct audio_drv_data *adata = dev_get_drvdata(component->dev); - kfree(rtd); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - adata->play_i2ssp_stream = NULL; - /* - * For Stoney, Memory gating is disabled,i.e SRAM Banks - * won't be turned off. The default state for SRAM banks is ON. - * Setting SRAM bank state code skipped for STONEY platform. - * added condition checks for Carrizo platform only - */ - if (adata->asic_type != CHIP_STONEY) { - for (bank = 1; bank <= 4; bank++) - acp_set_sram_bank_state(adata->acp_mmio, bank, - false); + switch (rtd->i2s_play_instance) { + case I2S_BT_INSTANCE: + adata->play_i2sbt_stream = NULL; + break; + case I2S_SP_INSTANCE: + default: + adata->play_i2ssp_stream = NULL; + /* + * For Stoney, Memory gating is disabled,i.e SRAM Banks + * won't be turned off. The default state for SRAM banks + * is ON.Setting SRAM bank state code skipped for STONEY + * platform. Added condition checks for Carrizo platform + * only. + */ + if (adata->asic_type != CHIP_STONEY) { + for (bank = 1; bank <= 4; bank++) + acp_set_sram_bank_state(adata->acp_mmio, + bank, false); + } } } else { - adata->capture_i2ssp_stream = NULL; - if (adata->asic_type != CHIP_STONEY) { - for (bank = 5; bank <= 8; bank++) - acp_set_sram_bank_state(adata->acp_mmio, bank, - false); + switch (rtd->i2s_capture_instance) { + case I2S_BT_INSTANCE: + adata->capture_i2sbt_stream = NULL; + break; + case I2S_SP_INSTANCE: + default: + adata->capture_i2ssp_stream = NULL; + if (adata->asic_type != CHIP_STONEY) { + for (bank = 5; bank <= 8; bank++) + acp_set_sram_bank_state(adata->acp_mmio, + bank, false); + } } } @@ -1056,8 +1129,10 @@ static int acp_dma_close(struct snd_pcm_substream *substream) * Disable ACP irq, when the current stream is being closed and * another stream is also not active. */ - if (!adata->play_i2ssp_stream && !adata->capture_i2ssp_stream) + if (!adata->play_i2ssp_stream && !adata->capture_i2ssp_stream && + !adata->play_i2sbt_stream && !adata->capture_i2sbt_stream) acp_reg_write(0, adata->acp_mmio, mmACP_EXTERNAL_INTR_ENB); + kfree(rtd); return 0; } @@ -1110,6 +1185,8 @@ static int acp_audio_probe(struct platform_device *pdev) audio_drv_data->play_i2ssp_stream = NULL; audio_drv_data->capture_i2ssp_stream = NULL; + audio_drv_data->play_i2sbt_stream = NULL; + audio_drv_data->capture_i2sbt_stream = NULL; audio_drv_data->asic_type = *pdata; @@ -1166,6 +1243,7 @@ static int acp_pcm_resume(struct device *dev) { u16 bank; int status; + struct audio_substream_data *rtd; struct audio_drv_data *adata = dev_get_drvdata(dev); status = acp_init(adata->acp_mmio, adata->asic_type); @@ -1185,9 +1263,8 @@ static int acp_pcm_resume(struct device *dev) acp_set_sram_bank_state(adata->acp_mmio, bank, true); } - config_acp_dma(adata->acp_mmio, - adata->play_i2ssp_stream->runtime->private_data, - adata->asic_type); + rtd = adata->play_i2ssp_stream->runtime->private_data; + config_acp_dma(adata->acp_mmio, rtd, adata->asic_type); } if (adata->capture_i2ssp_stream && adata->capture_i2ssp_stream->runtime) { @@ -1196,9 +1273,20 @@ static int acp_pcm_resume(struct device *dev) acp_set_sram_bank_state(adata->acp_mmio, bank, true); } - config_acp_dma(adata->acp_mmio, - adata->capture_i2ssp_stream->runtime->private_data, - adata->asic_type); + rtd = adata->capture_i2ssp_stream->runtime->private_data; + config_acp_dma(adata->acp_mmio, rtd, adata->asic_type); + } + if (adata->asic_type != CHIP_CARRIZO) { + if (adata->play_i2sbt_stream && + adata->play_i2sbt_stream->runtime) { + rtd = adata->play_i2sbt_stream->runtime->private_data; + config_acp_dma(adata->acp_mmio, rtd, adata->asic_type); + } + if (adata->capture_i2sbt_stream && + adata->capture_i2sbt_stream->runtime) { + rtd = adata->capture_i2sbt_stream->runtime->private_data; + config_acp_dma(adata->acp_mmio, rtd, adata->asic_type); + } } acp_reg_write(1, adata->acp_mmio, mmACP_EXTERNAL_INTR_ENB); return 0; diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index 0e6089b..460365c 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -10,17 +10,30 @@ #define ACP_PLAYBACK_PTE_OFFSET 10 #define ACP_CAPTURE_PTE_OFFSET 0 +/* Playback and Capture Offset for Stoney */ +#define ACP_ST_PLAYBACK_PTE_OFFSET 0x04 +#define ACP_ST_CAPTURE_PTE_OFFSET 0x00 +#define ACP_ST_BT_PLAYBACK_PTE_OFFSET 0x08 +#define ACP_ST_BT_CAPTURE_PTE_OFFSET 0x0c + #define ACP_GARLIC_CNTL_DEFAULT 0x00000FB4 #define ACP_ONION_CNTL_DEFAULT 0x00000FB4 #define ACP_PHYSICAL_BASE 0x14000 -/* Playback SRAM address (as a destination in dma descriptor) */ -#define ACP_SHARED_RAM_BANK_1_ADDRESS 0x4002000 - -/* Capture SRAM address (as a source in dma descriptor) */ -#define ACP_SHARED_RAM_BANK_5_ADDRESS 0x400A000 -#define ACP_SHARED_RAM_BANK_3_ADDRESS 0x4006000 +/* + * In case of I2S SP controller instance, Stoney uses SRAM bank 1 for + * playback and SRAM Bank 2 for capture where as in case of BT I2S + * Instance, Stoney uses SRAM Bank 3 for playback & SRAM Bank 4 will + * be used for capture. Carrizo uses I2S SP controller instance. SRAM Banks + * 1, 2, 3, 4 will be used for playback & SRAM Banks 5, 6, 7, 8 will be used + * for capture scenario. + */ +#define ACP_SRAM_BANK_1_ADDRESS 0x4002000 +#define ACP_SRAM_BANK_2_ADDRESS 0x4004000 +#define ACP_SRAM_BANK_3_ADDRESS 0x4006000 +#define ACP_SRAM_BANK_4_ADDRESS 0x4008000 +#define ACP_SRAM_BANK_5_ADDRESS 0x400A000 #define ACP_DMA_RESET_TIME 10000 #define ACP_CLOCK_EN_TIME_OUT_VALUE 0x000000FF @@ -35,8 +48,10 @@ #define TO_ACP_I2S_1 0x2 #define TO_ACP_I2S_2 0x4 +#define TO_BLUETOOTH 0x3 #define FROM_ACP_I2S_1 0xa #define FROM_ACP_I2S_2 0xb +#define FROM_BLUETOOTH 0xb #define ACP_TILE_ON_MASK 0x03 #define ACP_TILE_OFF_MASK 0x02 @@ -57,6 +72,14 @@ #define ACP_TO_SYSRAM_CH_NUM 14 #define I2S_TO_ACP_DMA_CH_NUM 15 +/* Playback DMA Channels for I2S BT instance */ +#define SYSRAM_TO_ACP_BT_INSTANCE_CH_NUM 8 +#define ACP_TO_I2S_DMA_BT_INSTANCE_CH_NUM 9 + +/* Capture DMA Channels for I2S BT Instance */ +#define ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM 10 +#define I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM 11 + #define NUM_DSCRS_PER_CHANNEL 2 #define PLAYBACK_START_DMA_DESCR_CH12 0 @@ -69,9 +92,22 @@ #define CAPTURE_START_DMA_DESCR_CH15 6 #define CAPTURE_END_DMA_DESCR_CH15 7 +/* I2S BT Instance DMA Descriptors */ +#define PLAYBACK_START_DMA_DESCR_CH8 8 +#define PLAYBACK_END_DMA_DESCR_CH8 9 +#define PLAYBACK_START_DMA_DESCR_CH9 10 +#define PLAYBACK_END_DMA_DESCR_CH9 11 + +#define CAPTURE_START_DMA_DESCR_CH10 12 +#define CAPTURE_END_DMA_DESCR_CH10 13 +#define CAPTURE_START_DMA_DESCR_CH11 14 +#define CAPTURE_END_DMA_DESCR_CH11 15 + #define mmACP_I2S_16BIT_RESOLUTION_EN 0x5209 #define ACP_I2S_MIC_16BIT_RESOLUTION_EN 0x01 #define ACP_I2S_SP_16BIT_RESOLUTION_EN 0x02 +#define ACP_I2S_BT_16BIT_RESOLUTION_EN 0x04 + enum acp_dma_priority_level { /* 0x0 Specifies the DMA channel is given normal priority */ ACP_DMA_PRIORITY_LEVEL_NORMAL = 0x0, @@ -84,16 +120,28 @@ struct audio_substream_data { struct page *pg; unsigned int order; u16 num_of_pages; + u16 i2s_play_instance; + u16 i2s_capture_instance; u16 direction; + u16 ch1; + u16 ch2; + u16 destination; + u16 dma_dscr_idx_1; + u16 dma_dscr_idx_2; + u32 pte_offset; + u32 sram_bank; + u32 byte_cnt_high; + u32 byte_cnt_low; uint64_t size; - u64 i2ssp_renderbytescount; - u64 i2ssp_capturebytescount; + u64 bytescount; void __iomem *acp_mmio; }; struct audio_drv_data { struct snd_pcm_substream *play_i2ssp_stream; struct snd_pcm_substream *capture_i2ssp_stream; + struct snd_pcm_substream *play_i2sbt_stream; + struct snd_pcm_substream *capture_i2sbt_stream; void __iomem *acp_mmio; u32 asic_type; }; -- 2.7.4

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[alsa-devel] alsa-lib: add-on configs directory changed to /etc/alsa/conf.d
by Jaroslav Kysela 17 Apr '18

17 Apr '18

Hi, I pushed two commits to the alsa-lib package which changes the default location for the add-on config files to /etc/alsa/conf.d from /usr/share/alsa/alsa.conf.d . The reason is to follow the scheme like in other packages. Also, the users might want to change or disable contents in those 'default' files. Example paths from other packages: # find /etc -type d -name conf.d /etc/fonts/conf.d /etc/NetworkManager/conf.d /etc/libblockdev/conf.d /etc/sssd/conf.d /etc/httpd/conf.d Commmits: http://git.alsa-project.org/?p=alsa-lib.git;a=commit;h=93e03bdc2a3dcd5d1251… http://git.alsa-project.org/?p=alsa-lib.git;a=commit;h=78505dccd23546cc77e5… Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

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[alsa-devel] [asoc:for-4.18 35/37] sound/soc/codecs/rt1305.c:1068:15: sparse: constant 562949953421312 is so big it is long
by kbuild test robot 16 Apr '18

16 Apr '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.18 head: fbeabd09236664f34ea3e4a9f7dcf5a0cdb7fc47 commit: 29bc643ddd7efb741d07c8b2d9a4c3dd9228865b [35/37] ASoC: rt1305: Add RT1305/RT1306 amplifier driver reproduce: # apt-get install sparse git checkout 29bc643ddd7efb741d07c8b2d9a4c3dd9228865b make ARCH=x86_64 allmodconfig make C=1 CF=-D__CHECK_ENDIAN__ sparse warnings: (new ones prefixed by >>) >> sound/soc/codecs/rt1305.c:1068:15: sparse: constant 562949953421312 is so big it is long sound/soc/codecs/rt1305.c:1085:15: sparse: constant 562949953421312 is so big it is long sound/soc/codecs/rt1305.c:1179:37: sparse: undefined identifier 'ACPI_PTR' sound/soc/codecs/rt1305.c:1179:23: error: implicit declaration of function 'ACPI_PTR'; did you mean 'ERR_PTR'? [-Werror=implicit-function-declaration] .acpi_match_table = ACPI_PTR(rt1305_acpi_match) ^~~~~~~~ ERR_PTR sound/soc/codecs/rt1305.c:1179:23: warning: initialization makes pointer from integer without a cast [-Wint-conversion] sound/soc/codecs/rt1305.c:1179:23: note: (near initialization for 'rt1305_i2c_driver.driver.acpi_match_table') sound/soc/codecs/rt1305.c:1179:23: error: initializer element is not constant sound/soc/codecs/rt1305.c:1179:23: note: (near initialization for 'rt1305_i2c_driver.driver.acpi_match_table') cc1: some warnings being treated as errors vim +1068 sound/soc/codecs/rt1305.c 992 993 static void rt1305_calibrate(struct rt1305_priv *rt1305) 994 { 995 unsigned int valmsb, vallsb, offsetl, offsetr; 996 unsigned int rh, rl, rhl, r0ohm; 997 u64 r0l, r0r; 998 999 regcache_cache_bypass(rt1305->regmap, true); 1000 1001 rt1305_reset(rt1305->regmap); 1002 regmap_write(rt1305->regmap, RT1305_ADC_SET_3, 0x0219); 1003 regmap_write(rt1305->regmap, RT1305_PR_BASE + 0xcf, 0x5548); 1004 regmap_write(rt1305->regmap, RT1305_PR_BASE + 0xc1, 0x0320); 1005 regmap_write(rt1305->regmap, RT1305_CLOCK_DETECT, 0x1000); 1006 regmap_write(rt1305->regmap, RT1305_CLK_1, 0x0600); 1007 regmap_write(rt1305->regmap, RT1305_POWER_CTRL_3, 0xffd0); 1008 regmap_write(rt1305->regmap, RT1305_EFUSE_1, 0x0080); 1009 regmap_write(rt1305->regmap, RT1305_EFUSE_1, 0x0880); 1010 regmap_write(rt1305->regmap, RT1305_POWER_CTRL_1, 0x0dfe); 1011 1012 /* Sin Gen */ 1013 regmap_write(rt1305->regmap, RT1305_PR_BASE + 0x5d, 0x0442); 1014 1015 regmap_write(rt1305->regmap, RT1305_CAL_EFUSE_CLOCK, 0xb000); 1016 regmap_write(rt1305->regmap, RT1305_PR_BASE + 0xc3, 0xd4a0); 1017 regmap_write(rt1305->regmap, RT1305_PR_BASE + 0xcc, 0x00cc); 1018 regmap_write(rt1305->regmap, RT1305_PR_BASE + 0xc1, 0x0320); 1019 regmap_write(rt1305->regmap, RT1305_POWER_STATUS, 0x0000); 1020 regmap_write(rt1305->regmap, RT1305_POWER_CTRL_2, 0xffff); 1021 regmap_write(rt1305->regmap, RT1305_POWER_CTRL_3, 0xfc20); 1022 regmap_write(rt1305->regmap, RT1305_PR_BASE + 0x06, 0x00c0); 1023 regmap_write(rt1305->regmap, RT1305_POWER_CTRL_3, 0xfca0); 1024 regmap_write(rt1305->regmap, RT1305_POWER_CTRL_3, 0xfce0); 1025 regmap_write(rt1305->regmap, RT1305_POWER_CTRL_3, 0xfcf0); 1026 1027 /* EFUSE read */ 1028 regmap_write(rt1305->regmap, RT1305_EFUSE_1, 0x0080); 1029 regmap_write(rt1305->regmap, RT1305_EFUSE_1, 0x0880); 1030 regmap_write(rt1305->regmap, RT1305_EFUSE_1, 0x0880); 1031 regmap_write(rt1305->regmap, RT1305_POWER_CTRL_3, 0xfce0); 1032 regmap_write(rt1305->regmap, RT1305_POWER_CTRL_3, 0xfca0); 1033 regmap_write(rt1305->regmap, RT1305_POWER_CTRL_3, 0xfc20); 1034 regmap_write(rt1305->regmap, RT1305_PR_BASE + 0x06, 0x0000); 1035 regmap_write(rt1305->regmap, RT1305_EFUSE_1, 0x0000); 1036 1037 regmap_read(rt1305->regmap, RT1305_DAC_OFFSET_5, &valmsb); 1038 regmap_read(rt1305->regmap, RT1305_DAC_OFFSET_6, &vallsb); 1039 offsetl = valmsb << 16 | vallsb; 1040 regmap_read(rt1305->regmap, RT1305_DAC_OFFSET_7, &valmsb); 1041 regmap_read(rt1305->regmap, RT1305_DAC_OFFSET_8, &vallsb); 1042 offsetr = valmsb << 16 | vallsb; 1043 pr_info("DC offsetl=0x%x, offsetr=0x%x\n", offsetl, offsetr); 1044 1045 /* R0 calibration */ 1046 regmap_write(rt1305->regmap, RT1305_PR_BASE + 0x5d, 0x9542); 1047 regmap_write(rt1305->regmap, RT1305_POWER_CTRL_3, 0xfcf0); 1048 regmap_write(rt1305->regmap, RT1305_POWER_CTRL_2, 0xffff); 1049 regmap_write(rt1305->regmap, RT1305_POWER_CTRL_1, 0x1dfe); 1050 regmap_write(rt1305->regmap, RT1305_SILENCE_DETECT, 0x0e13); 1051 regmap_write(rt1305->regmap, RT1305_CLK_1, 0x0650); 1052 1053 regmap_write(rt1305->regmap, RT1305_PR_BASE + 0x50, 0x0064); 1054 regmap_write(rt1305->regmap, RT1305_PR_BASE + 0x51, 0x0770); 1055 regmap_write(rt1305->regmap, RT1305_PR_BASE + 0x52, 0xc30c); 1056 regmap_write(rt1305->regmap, RT1305_SPK_TEMP_PROTECTION_1, 0x8200); 1057 regmap_write(rt1305->regmap, RT1305_PR_BASE + 0xd4, 0xfb00); 1058 regmap_write(rt1305->regmap, RT1305_PR_BASE + 0xd4, 0xff80); 1059 msleep(2000); 1060 regmap_read(rt1305->regmap, RT1305_PR_BASE + 0x55, &rh); 1061 regmap_read(rt1305->regmap, RT1305_PR_BASE + 0x56, &rl); 1062 rhl = (rh << 16) | rl; 1063 r0ohm = (rhl*10) / 33554432; 1064 1065 pr_debug("Left_rhl = 0x%x rh=0x%x rl=0x%x\n", rhl, rh, rl); 1066 pr_info("Left channel %d.%dohm\n", (r0ohm/10), (r0ohm%10)); 1067 > 1068 r0l = 562949953421312; 1069 if (rhl != 0) 1070 do_div(r0l, rhl); 1071 pr_debug("Left_r0 = 0x%llx\n", r0l); 1072 1073 regmap_write(rt1305->regmap, RT1305_SPK_TEMP_PROTECTION_1, 0x9200); 1074 regmap_write(rt1305->regmap, RT1305_PR_BASE + 0xd4, 0xfb00); 1075 regmap_write(rt1305->regmap, RT1305_PR_BASE + 0xd4, 0xff80); 1076 msleep(2000); 1077 regmap_read(rt1305->regmap, RT1305_PR_BASE + 0x55, &rh); 1078 regmap_read(rt1305->regmap, RT1305_PR_BASE + 0x56, &rl); 1079 rhl = (rh << 16) | rl; 1080 r0ohm = (rhl*10) / 33554432; 1081 1082 pr_debug("Right_rhl = 0x%x rh=0x%x rl=0x%x\n", rhl, rh, rl); 1083 pr_info("Right channel %d.%dohm\n", (r0ohm/10), (r0ohm%10)); 1084 1085 r0r = 562949953421312; 1086 if (rhl != 0) 1087 do_div(r0r, rhl); 1088 pr_debug("Right_r0 = 0x%llx\n", r0r); 1089 1090 regmap_write(rt1305->regmap, RT1305_SPK_TEMP_PROTECTION_1, 0xc2ec); 1091 1092 if ((r0l > R0_UPPER) && (r0l < R0_LOWER) && 1093 (r0r > R0_UPPER) && (r0r < R0_LOWER)) { 1094 regmap_write(rt1305->regmap, RT1305_PR_BASE + 0x4e, 1095 (r0l >> 16) & 0xffff); 1096 regmap_write(rt1305->regmap, RT1305_PR_BASE + 0x4f, 1097 r0l & 0xffff); 1098 regmap_write(rt1305->regmap, RT1305_PR_BASE + 0xfe, 1099 ((r0r >> 16) & 0xffff) | 0xf800); 1100 regmap_write(rt1305->regmap, RT1305_PR_BASE + 0xfd, 1101 r0r & 0xffff); 1102 } else { 1103 pr_err("R0 calibration failed\n"); 1104 } 1105 1106 /* restore some registers */ 1107 regmap_write(rt1305->regmap, RT1305_POWER_CTRL_1, 0x0dfe); 1108 usleep_range(200000, 400000); 1109 regmap_write(rt1305->regmap, RT1305_PR_BASE + 0x5d, 0x0442); 1110 regmap_write(rt1305->regmap, RT1305_CLOCK_DETECT, 0x3000); 1111 regmap_write(rt1305->regmap, RT1305_CLK_1, 0x0400); 1112 regmap_write(rt1305->regmap, RT1305_POWER_CTRL_1, 0x0000); 1113 regmap_write(rt1305->regmap, RT1305_CAL_EFUSE_CLOCK, 0x8000); 1114 regmap_write(rt1305->regmap, RT1305_POWER_CTRL_2, 0x1020); 1115 regmap_write(rt1305->regmap, RT1305_POWER_CTRL_3, 0x0000); 1116 1117 regcache_cache_bypass(rt1305->regmap, false); 1118 } 1119 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH v4 0/3] ASoC: topology: Improve hw_configs
by Kirill Marinushkin 16 Apr '18

16 Apr '18

Hello Mark, This patch series waits since 20th of February, because it modifies UAPI headers and should be done in sync with ALSA. I see that previously there was no clear understanding, how to do this in sync. Two days ago I discussed with Takashi, and he suggested the following steps: Step 1. A person from Intel, who knows the topic (Pierre), adds his "Reviewed-by" or "Acked-by" tags Step 2. I clarify with you to apply the Linux part (this patch series) Step 3. After that, I clarify with Takashi to apply the ALSA part (it is ready, but not sent untill we apply the Linux part) The step 1 is done. Old patches are reviewed by Pierre, so they have his "Reviewed-by" tag. New patches are suggested by him, and we agreed on the implementation, so they have his "Acked-by" tag. Now we are on the step 2. If you don't have anything against this patch series, please apply it to Linux. Best Regards, Kirill Kirill Marinushkin (3): ASoC: topology: Fix bclk and fsync inversion in set_link_hw_format() ASoC: topology: Add missing clock gating parameter when parsing hw_configs ASoC: topology: Add definitions for mclk_direction values include/uapi/sound/asoc.h | 29 +++++++++++++++++++++++++---- sound/soc/soc-topology.c | 19 ++++++++++++++----- 2 files changed, 39 insertions(+), 9 deletions(-) -- 2.13.6

3 8

[alsa-devel] [PATCH] [PATCH] ASoC: rt1305: Add RT1305/RT1306 amplifier driver
by shumingf＠realtek.com 16 Apr '18

16 Apr '18

From: Shuming Fan <shumingf(a)realtek.com> This is the initial amplifier driver for rt1305/rt1306. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt1305.c | 1191 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt1305.h | 276 +++++++++++ 4 files changed, 1475 insertions(+) create mode 100644 sound/soc/codecs/rt1305.c create mode 100644 sound/soc/codecs/rt1305.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9548f63..665edb5 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -126,6 +126,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_RT274 if I2C select SND_SOC_RT286 if I2C select SND_SOC_RT298 if I2C + select SND_SOC_RT1305 if I2C select SND_SOC_RT5514 if I2C select SND_SOC_RT5616 if I2C select SND_SOC_RT5631 if I2C @@ -772,6 +773,7 @@ config SND_SOC_RL6231 default y if SND_SOC_RT5665=y default y if SND_SOC_RT5670=y default y if SND_SOC_RT5677=y + default y if SND_SOC_RT1305=y default m if SND_SOC_RT5514=m default m if SND_SOC_RT5616=m default m if SND_SOC_RT5640=m @@ -783,6 +785,7 @@ config SND_SOC_RL6231 default m if SND_SOC_RT5665=m default m if SND_SOC_RT5670=m default m if SND_SOC_RT5677=m + default m if SND_SOC_RT1305=m config SND_SOC_RL6347A tristate @@ -805,6 +808,9 @@ config SND_SOC_RT298 tristate depends on I2C +config SND_SOC_RT1305 + tristate + config SND_SOC_RT5514 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index e849d14..cccd774 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -126,6 +126,7 @@ snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o snd-soc-pcm512x-spi-objs := pcm512x-spi.o snd-soc-rl6231-objs := rl6231.o snd-soc-rl6347a-objs := rl6347a.o +snd-soc-rt1305-objs := rt1305.o snd-soc-rt274-objs := rt274.o snd-soc-rt286-objs := rt286.o snd-soc-rt298-objs := rt298.o @@ -379,6 +380,7 @@ obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o obj-$(CONFIG_SND_SOC_RL6231) += snd-soc-rl6231.o obj-$(CONFIG_SND_SOC_RL6347A) += snd-soc-rl6347a.o +obj-$(CONFIG_SND_SOC_RT1305) += snd-soc-rt1305.o obj-$(CONFIG_SND_SOC_RT274) += snd-soc-rt274.o obj-$(CONFIG_SND_SOC_RT286) += snd-soc-rt286.o obj-$(CONFIG_SND_SOC_RT298) += snd-soc-rt298.o diff --git a/sound/soc/codecs/rt1305.c b/sound/soc/codecs/rt1305.c new file mode 100644 index 0000000..90baab9 --- /dev/null +++ b/sound/soc/codecs/rt1305.c @@ -0,0 +1,1191 @@ +/* + * rt1305.c -- RT1305 ALSA SoC amplifier component driver + * + * Copyright 2018 Realtek Semiconductor Corp. + * Author: Shuming Fan <shumingf(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/gpio.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/of_gpio.h> +#include <linux/platform_device.h> +#include <linux/firmware.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "rl6231.h" +#include "rt1305.h" + + +#define RT1305_PR_RANGE_BASE (0xff + 1) +#define RT1305_PR_SPACING 0x100 + +#define RT1305_PR_BASE (RT1305_PR_RANGE_BASE + (0 * RT1305_PR_SPACING)) + + +static const struct regmap_range_cfg rt1305_ranges[] = { + { + .name = "PR", + .range_min = RT1305_PR_BASE, + .range_max = RT1305_PR_BASE + 0xff, + .selector_reg = RT1305_PRIV_INDEX, + .selector_mask = 0xff, + .selector_shift = 0x0, + .window_start = RT1305_PRIV_DATA, + .window_len = 0x1, + }, +}; + + +static const struct reg_sequence init_list[] = { + + { RT1305_PR_BASE + 0xcf, 0x5548 }, + { RT1305_PR_BASE + 0x5d, 0x0442 }, + { RT1305_PR_BASE + 0xc1, 0x0320 }, + + { RT1305_POWER_STATUS, 0x0000 }, + + { RT1305_SPK_TEMP_PROTECTION_1, 0xd6de }, + { RT1305_SPK_TEMP_PROTECTION_2, 0x0707 }, + { RT1305_SPK_TEMP_PROTECTION_3, 0x4090 }, + + { RT1305_DAC_SET_1, 0xdfdf }, /* 4 ohm 2W */ + { RT1305_ADC_SET_3, 0x0219 }, + { RT1305_ADC_SET_1, 0x170f }, /* 0.2 ohm RSense*/ + +}; +#define RT1305_INIT_REG_LEN ARRAY_SIZE(init_list) + +struct rt1305_priv { + struct snd_soc_component *component; + struct regmap *regmap; + + int sysclk; + int sysclk_src; + int lrck; + int bclk; + int master; + + int pll_src; + int pll_in; + int pll_out; +}; + +static const struct reg_default rt1305_reg[] = { + + { 0x04, 0x0400 }, + { 0x05, 0x0880 }, + { 0x06, 0x0000 }, + { 0x07, 0x3100 }, + { 0x08, 0x8000 }, + { 0x09, 0x0000 }, + { 0x0a, 0x087e }, + { 0x0b, 0x0020 }, + { 0x0c, 0x0802 }, + { 0x0d, 0x0020 }, + { 0x10, 0x1d1d }, + { 0x11, 0x1d1d }, + { 0x12, 0xffff }, + { 0x14, 0x000c }, + { 0x16, 0x1717 }, + { 0x17, 0x4000 }, + { 0x18, 0x0019 }, + { 0x20, 0x0000 }, + { 0x22, 0x0000 }, + { 0x24, 0x0000 }, + { 0x26, 0x0000 }, + { 0x28, 0x0000 }, + { 0x2a, 0x4000 }, + { 0x2b, 0x3000 }, + { 0x2d, 0x6000 }, + { 0x2e, 0x0000 }, + { 0x2f, 0x8000 }, + { 0x32, 0x0000 }, + { 0x39, 0x0001 }, + { 0x3a, 0x0000 }, + { 0x3b, 0x1020 }, + { 0x3c, 0x0000 }, + { 0x3d, 0x0000 }, + { 0x3e, 0x4c00 }, + { 0x3f, 0x3000 }, + { 0x40, 0x000c }, + { 0x42, 0x0400 }, + { 0x46, 0xc22c }, + { 0x47, 0x0000 }, + { 0x4b, 0x0000 }, + { 0x4c, 0x0300 }, + { 0x4f, 0xf000 }, + { 0x50, 0xc200 }, + { 0x51, 0x1f1f }, + { 0x52, 0x01f0 }, + { 0x53, 0x407f }, + { 0x54, 0xffff }, + { 0x58, 0x4005 }, + { 0x5e, 0x0000 }, + { 0x5f, 0x0000 }, + { 0x60, 0xee13 }, + { 0x62, 0x0000 }, + { 0x63, 0x5f5f }, + { 0x64, 0x0040 }, + { 0x65, 0x4000 }, + { 0x66, 0x4004 }, + { 0x67, 0x0306 }, + { 0x68, 0x8c04 }, + { 0x69, 0xe021 }, + { 0x6a, 0x0000 }, + { 0x6c, 0xaaaa }, + { 0x70, 0x0333 }, + { 0x71, 0x3330 }, + { 0x72, 0x3333 }, + { 0x73, 0x3300 }, + { 0x74, 0x0000 }, + { 0x75, 0x0000 }, + { 0x76, 0x0000 }, + { 0x7a, 0x0003 }, + { 0x7c, 0x10ec }, + { 0x7e, 0x6251 }, + { 0x80, 0x0800 }, + { 0x81, 0x4000 }, + { 0x82, 0x0000 }, + { 0x90, 0x7a01 }, + { 0x91, 0x8431 }, + { 0x92, 0x0180 }, + { 0x93, 0x0000 }, + { 0x94, 0x0000 }, + { 0x95, 0x0000 }, + { 0x96, 0x0000 }, + { 0x97, 0x0000 }, + { 0x98, 0x0000 }, + { 0x99, 0x0000 }, + { 0x9a, 0x0000 }, + { 0x9b, 0x0000 }, + { 0x9c, 0x0000 }, + { 0x9d, 0x0000 }, + { 0x9e, 0x0000 }, + { 0x9f, 0x0000 }, + { 0xa0, 0x0000 }, + { 0xb0, 0x8200 }, + { 0xb1, 0x00ff }, + { 0xb2, 0x0008 }, + { 0xc0, 0x0200 }, + { 0xc1, 0x0000 }, + { 0xc2, 0x0000 }, + { 0xc3, 0x0000 }, + { 0xc4, 0x0000 }, + { 0xc5, 0x0000 }, + { 0xc6, 0x0000 }, + { 0xc7, 0x0000 }, + { 0xc8, 0x0000 }, + { 0xc9, 0x0000 }, + { 0xca, 0x0200 }, + { 0xcb, 0x0000 }, + { 0xcc, 0x0000 }, + { 0xcd, 0x0000 }, + { 0xce, 0x0000 }, + { 0xcf, 0x0000 }, + { 0xd0, 0x0000 }, + { 0xd1, 0x0000 }, + { 0xd2, 0x0000 }, + { 0xd3, 0x0000 }, + { 0xd4, 0x0200 }, + { 0xd5, 0x0000 }, + { 0xd6, 0x0000 }, + { 0xd7, 0x0000 }, + { 0xd8, 0x0000 }, + { 0xd9, 0x0000 }, + { 0xda, 0x0000 }, + { 0xdb, 0x0000 }, + { 0xdc, 0x0000 }, + { 0xdd, 0x0000 }, + { 0xde, 0x0200 }, + { 0xdf, 0x0000 }, + { 0xe0, 0x0000 }, + { 0xe1, 0x0000 }, + { 0xe2, 0x0000 }, + { 0xe3, 0x0000 }, + { 0xe4, 0x0000 }, + { 0xe5, 0x0000 }, + { 0xe6, 0x0000 }, + { 0xe7, 0x0000 }, + { 0xe8, 0x0200 }, + { 0xe9, 0x0000 }, + { 0xea, 0x0000 }, + { 0xeb, 0x0000 }, + { 0xec, 0x0000 }, + { 0xed, 0x0000 }, + { 0xee, 0x0000 }, + { 0xef, 0x0000 }, + { 0xf0, 0x0000 }, + { 0xf1, 0x0000 }, + { 0xf2, 0x0200 }, + { 0xf3, 0x0000 }, + { 0xf4, 0x0000 }, + { 0xf5, 0x0000 }, + { 0xf6, 0x0000 }, + { 0xf7, 0x0000 }, + { 0xf8, 0x0000 }, + { 0xf9, 0x0000 }, + { 0xfa, 0x0000 }, + { 0xfb, 0x0000 }, +}; + +static int rt1305_reg_init(struct snd_soc_component *component) +{ + struct rt1305_priv *rt1305 = snd_soc_component_get_drvdata(component); + + regmap_multi_reg_write(rt1305->regmap, init_list, RT1305_INIT_REG_LEN); + return 0; +} + +static bool rt1305_volatile_register(struct device *dev, unsigned int reg) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rt1305_ranges); i++) { + if (reg >= rt1305_ranges[i].range_min && + reg <= rt1305_ranges[i].range_max) { + return true; + } + } + + switch (reg) { + case RT1305_RESET: + case RT1305_SPDIF_IN_SET_1: + case RT1305_SPDIF_IN_SET_2: + case RT1305_SPDIF_IN_SET_3: + case RT1305_POWER_CTRL_2: + case RT1305_CLOCK_DETECT: + case RT1305_BIQUAD_SET_1: + case RT1305_BIQUAD_SET_2: + case RT1305_EQ_SET_2: + case RT1305_SPK_TEMP_PROTECTION_0: + case RT1305_SPK_TEMP_PROTECTION_2: + case RT1305_SPK_DC_DETECT_1: + case RT1305_SILENCE_DETECT: + case RT1305_VERSION_ID: + case RT1305_VENDOR_ID: + case RT1305_DEVICE_ID: + case RT1305_EFUSE_1: + case RT1305_EFUSE_3: + case RT1305_DC_CALIB_1: + case RT1305_DC_CALIB_3: + case RT1305_DAC_OFFSET_1: + case RT1305_DAC_OFFSET_2: + case RT1305_DAC_OFFSET_3: + case RT1305_DAC_OFFSET_4: + case RT1305_DAC_OFFSET_5: + case RT1305_DAC_OFFSET_6: + case RT1305_DAC_OFFSET_7: + case RT1305_DAC_OFFSET_8: + case RT1305_DAC_OFFSET_9: + case RT1305_DAC_OFFSET_10: + case RT1305_DAC_OFFSET_11: + case RT1305_TRIM_1: + case RT1305_TRIM_2: + return true; + + default: + return false; + } +} + +static bool rt1305_readable_register(struct device *dev, unsigned int reg) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rt1305_ranges); i++) { + if (reg >= rt1305_ranges[i].range_min && + reg <= rt1305_ranges[i].range_max) { + return true; + } + } + + switch (reg) { + case RT1305_RESET: + case RT1305_CLK_1 ... RT1305_CAL_EFUSE_CLOCK: + case RT1305_PLL0_1 ... RT1305_PLL1_2: + case RT1305_MIXER_CTRL_1: + case RT1305_MIXER_CTRL_2: + case RT1305_DAC_SET_1: + case RT1305_DAC_SET_2: + case RT1305_ADC_SET_1: + case RT1305_ADC_SET_2: + case RT1305_ADC_SET_3: + case RT1305_PATH_SET: + case RT1305_SPDIF_IN_SET_1: + case RT1305_SPDIF_IN_SET_2: + case RT1305_SPDIF_IN_SET_3: + case RT1305_SPDIF_OUT_SET_1: + case RT1305_SPDIF_OUT_SET_2: + case RT1305_SPDIF_OUT_SET_3: + case RT1305_I2S_SET_1: + case RT1305_I2S_SET_2: + case RT1305_PBTL_MONO_MODE_SRC: + case RT1305_MANUALLY_I2C_DEVICE: + case RT1305_POWER_STATUS: + case RT1305_POWER_CTRL_1: + case RT1305_POWER_CTRL_2: + case RT1305_POWER_CTRL_3: + case RT1305_POWER_CTRL_4: + case RT1305_POWER_CTRL_5: + case RT1305_CLOCK_DETECT: + case RT1305_BIQUAD_SET_1: + case RT1305_BIQUAD_SET_2: + case RT1305_ADJUSTED_HPF_1: + case RT1305_ADJUSTED_HPF_2: + case RT1305_EQ_SET_1: + case RT1305_EQ_SET_2: + case RT1305_SPK_TEMP_PROTECTION_0: + case RT1305_SPK_TEMP_PROTECTION_1: + case RT1305_SPK_TEMP_PROTECTION_2: + case RT1305_SPK_TEMP_PROTECTION_3: + case RT1305_SPK_DC_DETECT_1: + case RT1305_SPK_DC_DETECT_2: + case RT1305_LOUDNESS: + case RT1305_THERMAL_FOLD_BACK_1: + case RT1305_THERMAL_FOLD_BACK_2: + case RT1305_SILENCE_DETECT ... RT1305_SPK_EXCURSION_LIMITER_7: + case RT1305_VERSION_ID: + case RT1305_VENDOR_ID: + case RT1305_DEVICE_ID: + case RT1305_EFUSE_1: + case RT1305_EFUSE_2: + case RT1305_EFUSE_3: + case RT1305_DC_CALIB_1: + case RT1305_DC_CALIB_2: + case RT1305_DC_CALIB_3: + case RT1305_DAC_OFFSET_1 ... RT1305_DAC_OFFSET_14: + case RT1305_TRIM_1: + case RT1305_TRIM_2: + case RT1305_TUNE_INTERNAL_OSC: + case RT1305_BIQUAD1_H0_L_28_16 ... RT1305_BIQUAD3_A2_R_15_0: + return true; + default: + return false; + } +} + +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -9435, 37, 0); + +static const char * const rt1305_rx_data_ch_select[] = { + "LR", + "RL", + "Copy L", + "Copy R", +}; + +static SOC_ENUM_SINGLE_DECL(rt1305_rx_data_ch_enum, RT1305_I2S_SET_2, 2, + rt1305_rx_data_ch_select); + +static void rt1305_reset(struct regmap *regmap) +{ + regmap_write(regmap, RT1305_RESET, 0); +} + +static const struct snd_kcontrol_new rt1305_snd_controls[] = { + SOC_DOUBLE_TLV("DAC Playback Volume", RT1305_DAC_SET_1, + 8, 0, 0xff, 0, dac_vol_tlv), + + /* I2S Data Channel Selection */ + SOC_ENUM("RX Channel Select", rt1305_rx_data_ch_enum), +}; + +static int rt1305_is_rc_clk_from_pll(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(source->dapm); + struct rt1305_priv *rt1305 = snd_soc_component_get_drvdata(component); + unsigned int val; + + snd_soc_component_read(component, RT1305_CLK_1, &val); + + if (rt1305->sysclk_src == RT1305_FS_SYS_PRE_S_PLL1 && + (val & RT1305_SEL_PLL_SRC_2_RCCLK)) + return 1; + else + return 0; +} + +static int rt1305_is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(source->dapm); + struct rt1305_priv *rt1305 = snd_soc_component_get_drvdata(component); + + if (rt1305->sysclk_src == RT1305_FS_SYS_PRE_S_PLL1) + return 1; + else + return 0; +} + +static int rt1305_classd_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_component_update_bits(component, RT1305_POWER_CTRL_1, + RT1305_POW_PDB_JD_MASK, RT1305_POW_PDB_JD); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_component_update_bits(component, RT1305_POWER_CTRL_1, + RT1305_POW_PDB_JD_MASK, 0); + usleep_range(150000, 200000); + break; + + default: + return 0; + } + + return 0; +} + +static const struct snd_kcontrol_new rt1305_sto_dac_l = + SOC_DAPM_SINGLE("Switch", RT1305_DAC_SET_2, + RT1305_DVOL_MUTE_L_EN_SFT, 1, 1); + +static const struct snd_kcontrol_new rt1305_sto_dac_r = + SOC_DAPM_SINGLE("Switch", RT1305_DAC_SET_2, + RT1305_DVOL_MUTE_R_EN_SFT, 1, 1); + +static const struct snd_soc_dapm_widget rt1305_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("PLL0", RT1305_POWER_CTRL_1, + RT1305_POW_PLL0_EN_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL1", RT1305_POWER_CTRL_1, + RT1305_POW_PLL1_EN_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MBIAS", RT1305_POWER_CTRL_1, + RT1305_POW_MBIAS_LV_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("BG MBIAS", RT1305_POWER_CTRL_1, + RT1305_POW_BG_MBIAS_LV_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("LDO2", RT1305_POWER_CTRL_1, + RT1305_POW_LDO2_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("BG2", RT1305_POWER_CTRL_1, + RT1305_POW_BG2_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("LDO2 IB2", RT1305_POWER_CTRL_1, + RT1305_POW_LDO2_IB2_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("VREF", RT1305_POWER_CTRL_1, + RT1305_POW_VREF_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("VREF1", RT1305_POWER_CTRL_1, + RT1305_POW_VREF1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("VREF2", RT1305_POWER_CTRL_1, + RT1305_POW_VREF2_BIT, 0, NULL, 0), + + + SND_SOC_DAPM_SUPPLY("DISC VREF", RT1305_POWER_CTRL_2, + RT1305_POW_DISC_VREF_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("FASTB VREF", RT1305_POWER_CTRL_2, + RT1305_POW_FASTB_VREF_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ULTRA FAST VREF", RT1305_POWER_CTRL_2, + RT1305_POW_ULTRA_FAST_VREF_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("CHOP DAC", RT1305_POWER_CTRL_2, + RT1305_POW_CKXEN_DAC_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("CKGEN DAC", RT1305_POWER_CTRL_2, + RT1305_POW_EN_CKGEN_DAC_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("CLAMP", RT1305_POWER_CTRL_2, + RT1305_POW_CLAMP_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("BUFL", RT1305_POWER_CTRL_2, + RT1305_POW_BUFL_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("BUFR", RT1305_POWER_CTRL_2, + RT1305_POW_BUFR_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("CKGEN ADC", RT1305_POWER_CTRL_2, + RT1305_POW_EN_CKGEN_ADC_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC3 L", RT1305_POWER_CTRL_2, + RT1305_POW_ADC3_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC3 R", RT1305_POWER_CTRL_2, + RT1305_POW_ADC3_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("TRIOSC", RT1305_POWER_CTRL_2, + RT1305_POW_TRIOSC_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("AVDD1", RT1305_POWER_CTRL_2, + RT1305_POR_AVDD1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("AVDD2", RT1305_POWER_CTRL_2, + RT1305_POR_AVDD2_BIT, 0, NULL, 0), + + + SND_SOC_DAPM_SUPPLY("VSENSE R", RT1305_POWER_CTRL_3, + RT1305_POW_VSENSE_RCH_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("VSENSE L", RT1305_POWER_CTRL_3, + RT1305_POW_VSENSE_LCH_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ISENSE R", RT1305_POWER_CTRL_3, + RT1305_POW_ISENSE_RCH_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ISENSE L", RT1305_POWER_CTRL_3, + RT1305_POW_ISENSE_LCH_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("POR AVDD1", RT1305_POWER_CTRL_3, + RT1305_POW_POR_AVDD1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("POR AVDD2", RT1305_POWER_CTRL_3, + RT1305_POW_POR_AVDD2_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("VCM 6172", RT1305_POWER_CTRL_3, + RT1305_EN_VCM_6172_BIT, 0, NULL, 0), + + + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + + /* Digital Interface */ + SND_SOC_DAPM_SUPPLY("DAC L Power", RT1305_POWER_CTRL_2, + RT1305_POW_DAC1_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC R Power", RT1305_POWER_CTRL_2, + RT1305_POW_DAC1_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_DAC("DAC", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SWITCH("DAC L", SND_SOC_NOPM, 0, 0, &rt1305_sto_dac_l), + SND_SOC_DAPM_SWITCH("DAC R", SND_SOC_NOPM, 0, 0, &rt1305_sto_dac_r), + + /* Output Lines */ + SND_SOC_DAPM_PGA_E("CLASS D", SND_SOC_NOPM, 0, 0, NULL, 0, + rt1305_classd_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_OUTPUT("SPOL"), + SND_SOC_DAPM_OUTPUT("SPOR"), +}; + +static const struct snd_soc_dapm_route rt1305_dapm_routes[] = { + + { "DAC", NULL, "AIF1RX" }, + + { "DAC", NULL, "PLL0", rt1305_is_rc_clk_from_pll }, + { "DAC", NULL, "PLL1", rt1305_is_sys_clk_from_pll }, + + { "DAC", NULL, "MBIAS" }, + { "DAC", NULL, "BG MBIAS" }, + { "DAC", NULL, "LDO2" }, + { "DAC", NULL, "BG2" }, + { "DAC", NULL, "LDO2 IB2" }, + { "DAC", NULL, "VREF" }, + { "DAC", NULL, "VREF1" }, + { "DAC", NULL, "VREF2" }, + + { "DAC", NULL, "DISC VREF" }, + { "DAC", NULL, "FASTB VREF" }, + { "DAC", NULL, "ULTRA FAST VREF" }, + { "DAC", NULL, "CHOP DAC" }, + { "DAC", NULL, "CKGEN DAC" }, + { "DAC", NULL, "CLAMP" }, + { "DAC", NULL, "CKGEN ADC" }, + { "DAC", NULL, "TRIOSC" }, + { "DAC", NULL, "AVDD1" }, + { "DAC", NULL, "AVDD2" }, + + { "DAC", NULL, "POR AVDD1" }, + { "DAC", NULL, "POR AVDD2" }, + { "DAC", NULL, "VCM 6172" }, + + { "DAC L", "Switch", "DAC" }, + { "DAC R", "Switch", "DAC" }, + + { "DAC R", NULL, "VSENSE R" }, + { "DAC L", NULL, "VSENSE L" }, + { "DAC R", NULL, "ISENSE R" }, + { "DAC L", NULL, "ISENSE L" }, + { "DAC L", NULL, "ADC3 L" }, + { "DAC R", NULL, "ADC3 R" }, + { "DAC L", NULL, "BUFL" }, + { "DAC R", NULL, "BUFR" }, + { "DAC L", NULL, "DAC L Power" }, + { "DAC R", NULL, "DAC R Power" }, + + { "CLASS D", NULL, "DAC L" }, + { "CLASS D", NULL, "DAC R" }, + + { "SPOL", NULL, "CLASS D" }, + { "SPOR", NULL, "CLASS D" }, +}; + +static int rt1305_get_clk_info(int sclk, int rate) +{ + int i, pd[] = {1, 2, 3, 4, 6, 8, 12, 16}; + + if (sclk <= 0 || rate <= 0) + return -EINVAL; + + rate = rate << 8; + for (i = 0; i < ARRAY_SIZE(pd); i++) + if (sclk == rate * pd[i]) + return i; + + return -EINVAL; +} + +static int rt1305_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct rt1305_priv *rt1305 = snd_soc_component_get_drvdata(component); + unsigned int val_len = 0, val_clk, mask_clk; + int pre_div, bclk_ms, frame_size; + + rt1305->lrck = params_rate(params); + pre_div = rt1305_get_clk_info(rt1305->sysclk, rt1305->lrck); + if (pre_div < 0) { + dev_warn(component->dev, "Force using PLL "); + snd_soc_dai_set_pll(dai, 0, RT1305_PLL1_S_BCLK, + rt1305->lrck * 64, rt1305->lrck * 256); + snd_soc_dai_set_sysclk(dai, RT1305_FS_SYS_PRE_S_PLL1, + rt1305->lrck * 256, SND_SOC_CLOCK_IN); + pre_div = 0; + } + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) { + dev_err(component->dev, "Unsupported frame size: %d\n", + frame_size); + return -EINVAL; + } + + bclk_ms = frame_size > 32; + rt1305->bclk = rt1305->lrck * (32 << bclk_ms); + + dev_dbg(component->dev, "bclk_ms is %d and pre_div is %d for iis %d\n", + bclk_ms, pre_div, dai->id); + + dev_dbg(component->dev, "lrck is %dHz and pre_div is %d for iis %d\n", + rt1305->lrck, pre_div, dai->id); + + switch (params_width(params)) { + case 16: + val_len |= RT1305_I2S_DL_SEL_16B; + break; + case 20: + val_len |= RT1305_I2S_DL_SEL_20B; + break; + case 24: + val_len |= RT1305_I2S_DL_SEL_24B; + break; + case 8: + val_len |= RT1305_I2S_DL_SEL_8B; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case RT1305_AIF1: + mask_clk = RT1305_DIV_FS_SYS_MASK; + val_clk = pre_div << RT1305_DIV_FS_SYS_SFT; + snd_soc_component_update_bits(component, RT1305_I2S_SET_2, + RT1305_I2S_DL_SEL_MASK, + val_len); + break; + default: + dev_err(component->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + + snd_soc_component_update_bits(component, RT1305_CLK_2, + mask_clk, val_clk); + + return 0; +} + +static int rt1305_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *component = dai->component; + struct rt1305_priv *rt1305 = snd_soc_component_get_drvdata(component); + unsigned int reg_val = 0, reg1_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + reg_val |= RT1305_SEL_I2S_OUT_MODE_M; + rt1305->master = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + reg_val |= RT1305_SEL_I2S_OUT_MODE_S; + rt1305->master = 0; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + reg1_val |= RT1305_I2S_BCLK_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_LEFT_J: + reg1_val |= RT1305_I2S_DF_SEL_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + reg1_val |= RT1305_I2S_DF_SEL_PCM_A; + break; + case SND_SOC_DAIFMT_DSP_B: + reg1_val |= RT1305_I2S_DF_SEL_PCM_B; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case RT1305_AIF1: + snd_soc_component_update_bits(component, RT1305_I2S_SET_1, + RT1305_SEL_I2S_OUT_MODE_MASK, reg_val); + snd_soc_component_update_bits(component, RT1305_I2S_SET_2, + RT1305_I2S_DF_SEL_MASK | RT1305_I2S_BCLK_MASK, + reg1_val); + break; + default: + dev_err(component->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + return 0; +} + +static int rt1305_set_component_sysclk(struct snd_soc_component *component, + int clk_id, int source, unsigned int freq, int dir) +{ + struct rt1305_priv *rt1305 = snd_soc_component_get_drvdata(component); + unsigned int reg_val = 0; + + if (freq == rt1305->sysclk && clk_id == rt1305->sysclk_src) + return 0; + + switch (clk_id) { + case RT1305_FS_SYS_PRE_S_MCLK: + reg_val |= RT1305_SEL_FS_SYS_PRE_MCLK; + snd_soc_component_update_bits(component, + RT1305_CLOCK_DETECT, RT1305_SEL_CLK_DET_SRC_MASK, + RT1305_SEL_CLK_DET_SRC_MCLK); + break; + case RT1305_FS_SYS_PRE_S_PLL1: + reg_val |= RT1305_SEL_FS_SYS_PRE_PLL; + break; + case RT1305_FS_SYS_PRE_S_RCCLK: + reg_val |= RT1305_SEL_FS_SYS_PRE_RCCLK; + break; + default: + dev_err(component->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + snd_soc_component_update_bits(component, RT1305_CLK_1, + RT1305_SEL_FS_SYS_PRE_MASK, reg_val); + rt1305->sysclk = freq; + rt1305->sysclk_src = clk_id; + + dev_dbg(component->dev, "Sysclk is %dHz and clock id is %d\n", + freq, clk_id); + + return 0; +} + +static int rt1305_set_component_pll(struct snd_soc_component *component, + int pll_id, int source, unsigned int freq_in, + unsigned int freq_out) +{ + struct rt1305_priv *rt1305 = snd_soc_component_get_drvdata(component); + struct rl6231_pll_code pll_code; + int ret; + + if (source == rt1305->pll_src && freq_in == rt1305->pll_in && + freq_out == rt1305->pll_out) + return 0; + + if (!freq_in || !freq_out) { + dev_dbg(component->dev, "PLL disabled\n"); + + rt1305->pll_in = 0; + rt1305->pll_out = 0; + snd_soc_component_update_bits(component, RT1305_CLK_1, + RT1305_SEL_FS_SYS_PRE_MASK | RT1305_SEL_PLL_SRC_1_MASK, + RT1305_SEL_FS_SYS_PRE_PLL | RT1305_SEL_PLL_SRC_1_BCLK); + return 0; + } + + switch (source) { + case RT1305_PLL2_S_MCLK: + snd_soc_component_update_bits(component, RT1305_CLK_1, + RT1305_SEL_PLL_SRC_2_MASK | RT1305_SEL_PLL_SRC_1_MASK | + RT1305_DIV_PLL_SRC_2_MASK, + RT1305_SEL_PLL_SRC_2_MCLK | RT1305_SEL_PLL_SRC_1_PLL2); + snd_soc_component_update_bits(component, + RT1305_CLOCK_DETECT, RT1305_SEL_CLK_DET_SRC_MASK, + RT1305_SEL_CLK_DET_SRC_MCLK); + break; + case RT1305_PLL1_S_BCLK: + snd_soc_component_update_bits(component, + RT1305_CLK_1, RT1305_SEL_PLL_SRC_1_MASK, + RT1305_SEL_PLL_SRC_1_BCLK); + break; + case RT1305_PLL2_S_RCCLK: + snd_soc_component_update_bits(component, RT1305_CLK_1, + RT1305_SEL_PLL_SRC_2_MASK | RT1305_SEL_PLL_SRC_1_MASK | + RT1305_DIV_PLL_SRC_2_MASK, + RT1305_SEL_PLL_SRC_2_RCCLK | RT1305_SEL_PLL_SRC_1_PLL2); + freq_in = 98304000; + break; + default: + dev_err(component->dev, "Unknown PLL Source %d\n", source); + return -EINVAL; + } + + ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); + if (ret < 0) { + dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + return ret; + } + + dev_dbg(component->dev, "bypass=%d m=%d n=%d k=%d\n", + pll_code.m_bp, (pll_code.m_bp ? 0 : pll_code.m_code), + pll_code.n_code, pll_code.k_code); + + snd_soc_component_write(component, RT1305_PLL1_1, + (pll_code.m_bp ? 0 : pll_code.m_code) << RT1305_PLL_1_M_SFT | + pll_code.m_bp << RT1305_PLL_1_M_BYPASS_SFT | + pll_code.n_code); + snd_soc_component_write(component, RT1305_PLL1_2, + pll_code.k_code); + + rt1305->pll_in = freq_in; + rt1305->pll_out = freq_out; + rt1305->pll_src = source; + + return 0; +} + +static int rt1305_probe(struct snd_soc_component *component) +{ + struct rt1305_priv *rt1305 = snd_soc_component_get_drvdata(component); + + rt1305->component = component; + + /* initial settings */ + rt1305_reg_init(component); + + return 0; +} + +static void rt1305_remove(struct snd_soc_component *component) +{ + struct rt1305_priv *rt1305 = snd_soc_component_get_drvdata(component); + + rt1305_reset(rt1305->regmap); +} + +#ifdef CONFIG_PM +static int rt1305_suspend(struct snd_soc_component *component) +{ + struct rt1305_priv *rt1305 = snd_soc_component_get_drvdata(component); + + regcache_cache_only(rt1305->regmap, true); + regcache_mark_dirty(rt1305->regmap); + + return 0; +} + +static int rt1305_resume(struct snd_soc_component *component) +{ + struct rt1305_priv *rt1305 = snd_soc_component_get_drvdata(component); + + regcache_cache_only(rt1305->regmap, false); + regcache_sync(rt1305->regmap); + + return 0; +} +#else +#define rt1305_suspend NULL +#define rt1305_resume NULL +#endif + +#define RT1305_STEREO_RATES SNDRV_PCM_RATE_8000_192000 +#define RT1305_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +static const struct snd_soc_dai_ops rt1305_aif_dai_ops = { + .hw_params = rt1305_hw_params, + .set_fmt = rt1305_set_dai_fmt, +}; + +static struct snd_soc_dai_driver rt1305_dai[] = { + { + .name = "rt1305-aif", + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT1305_STEREO_RATES, + .formats = RT1305_FORMATS, + }, + .ops = &rt1305_aif_dai_ops, + }, +}; + +static const struct snd_soc_component_driver soc_component_dev_rt1305 = { + .probe = rt1305_probe, + .remove = rt1305_remove, + .suspend = rt1305_suspend, + .resume = rt1305_resume, + .controls = rt1305_snd_controls, + .num_controls = ARRAY_SIZE(rt1305_snd_controls), + .dapm_widgets = rt1305_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt1305_dapm_widgets), + .dapm_routes = rt1305_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt1305_dapm_routes), + .set_sysclk = rt1305_set_component_sysclk, + .set_pll = rt1305_set_component_pll, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static const struct regmap_config rt1305_regmap = { + .reg_bits = 8, + .val_bits = 16, + .max_register = RT1305_MAX_REG + 1 + (ARRAY_SIZE(rt1305_ranges) * + RT1305_PR_SPACING), + .volatile_reg = rt1305_volatile_register, + .readable_reg = rt1305_readable_register, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt1305_reg, + .num_reg_defaults = ARRAY_SIZE(rt1305_reg), + .ranges = rt1305_ranges, + .num_ranges = ARRAY_SIZE(rt1305_ranges), + .use_single_rw = true, +}; + +#if defined(CONFIG_OF) +static const struct of_device_id rt1305_of_match[] = { + { .compatible = "realtek,rt1305", }, + { .compatible = "realtek,rt1306", }, + {}, +}; +MODULE_DEVICE_TABLE(of, rt1305_of_match); +#endif + +#ifdef CONFIG_ACPI +static struct acpi_device_id rt1305_acpi_match[] = { + {"10EC1305", 0,}, + {"10EC1306", 0,}, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt1305_acpi_match); +#endif + +static const struct i2c_device_id rt1305_i2c_id[] = { + { "rt1305", 0 }, + { "rt1306", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, rt1305_i2c_id); + +static void rt1305_calibrate(struct rt1305_priv *rt1305) +{ + unsigned int valmsb, vallsb, offsetl, offsetr; + unsigned int rh, rl, rhl, r0ohm; + u64 r0l, r0r; + + regcache_cache_bypass(rt1305->regmap, true); + + rt1305_reset(rt1305->regmap); + regmap_write(rt1305->regmap, RT1305_ADC_SET_3, 0x0219); + regmap_write(rt1305->regmap, RT1305_PR_BASE + 0xcf, 0x5548); + regmap_write(rt1305->regmap, RT1305_PR_BASE + 0xc1, 0x0320); + regmap_write(rt1305->regmap, RT1305_CLOCK_DETECT, 0x1000); + regmap_write(rt1305->regmap, RT1305_CLK_1, 0x0600); + regmap_write(rt1305->regmap, RT1305_POWER_CTRL_3, 0xffd0); + regmap_write(rt1305->regmap, RT1305_EFUSE_1, 0x0080); + regmap_write(rt1305->regmap, RT1305_EFUSE_1, 0x0880); + regmap_write(rt1305->regmap, RT1305_POWER_CTRL_1, 0x0dfe); + + /* Sin Gen */ + regmap_write(rt1305->regmap, RT1305_PR_BASE + 0x5d, 0x0442); + + regmap_write(rt1305->regmap, RT1305_CAL_EFUSE_CLOCK, 0xb000); + regmap_write(rt1305->regmap, RT1305_PR_BASE + 0xc3, 0xd4a0); + regmap_write(rt1305->regmap, RT1305_PR_BASE + 0xcc, 0x00cc); + regmap_write(rt1305->regmap, RT1305_PR_BASE + 0xc1, 0x0320); + regmap_write(rt1305->regmap, RT1305_POWER_STATUS, 0x0000); + regmap_write(rt1305->regmap, RT1305_POWER_CTRL_2, 0xffff); + regmap_write(rt1305->regmap, RT1305_POWER_CTRL_3, 0xfc20); + regmap_write(rt1305->regmap, RT1305_PR_BASE + 0x06, 0x00c0); + regmap_write(rt1305->regmap, RT1305_POWER_CTRL_3, 0xfca0); + regmap_write(rt1305->regmap, RT1305_POWER_CTRL_3, 0xfce0); + regmap_write(rt1305->regmap, RT1305_POWER_CTRL_3, 0xfcf0); + + /* EFUSE read */ + regmap_write(rt1305->regmap, RT1305_EFUSE_1, 0x0080); + regmap_write(rt1305->regmap, RT1305_EFUSE_1, 0x0880); + regmap_write(rt1305->regmap, RT1305_EFUSE_1, 0x0880); + regmap_write(rt1305->regmap, RT1305_POWER_CTRL_3, 0xfce0); + regmap_write(rt1305->regmap, RT1305_POWER_CTRL_3, 0xfca0); + regmap_write(rt1305->regmap, RT1305_POWER_CTRL_3, 0xfc20); + regmap_write(rt1305->regmap, RT1305_PR_BASE + 0x06, 0x0000); + regmap_write(rt1305->regmap, RT1305_EFUSE_1, 0x0000); + + regmap_read(rt1305->regmap, RT1305_DAC_OFFSET_5, &valmsb); + regmap_read(rt1305->regmap, RT1305_DAC_OFFSET_6, &vallsb); + offsetl = valmsb << 16 | vallsb; + regmap_read(rt1305->regmap, RT1305_DAC_OFFSET_7, &valmsb); + regmap_read(rt1305->regmap, RT1305_DAC_OFFSET_8, &vallsb); + offsetr = valmsb << 16 | vallsb; + pr_info("DC offsetl=0x%x, offsetr=0x%x\n", offsetl, offsetr); + + /* R0 calibration */ + regmap_write(rt1305->regmap, RT1305_PR_BASE + 0x5d, 0x9542); + regmap_write(rt1305->regmap, RT1305_POWER_CTRL_3, 0xfcf0); + regmap_write(rt1305->regmap, RT1305_POWER_CTRL_2, 0xffff); + regmap_write(rt1305->regmap, RT1305_POWER_CTRL_1, 0x1dfe); + regmap_write(rt1305->regmap, RT1305_SILENCE_DETECT, 0x0e13); + regmap_write(rt1305->regmap, RT1305_CLK_1, 0x0650); + + regmap_write(rt1305->regmap, RT1305_PR_BASE + 0x50, 0x0064); + regmap_write(rt1305->regmap, RT1305_PR_BASE + 0x51, 0x0770); + regmap_write(rt1305->regmap, RT1305_PR_BASE + 0x52, 0xc30c); + regmap_write(rt1305->regmap, RT1305_SPK_TEMP_PROTECTION_1, 0x8200); + regmap_write(rt1305->regmap, RT1305_PR_BASE + 0xd4, 0xfb00); + regmap_write(rt1305->regmap, RT1305_PR_BASE + 0xd4, 0xff80); + msleep(2000); + regmap_read(rt1305->regmap, RT1305_PR_BASE + 0x55, &rh); + regmap_read(rt1305->regmap, RT1305_PR_BASE + 0x56, &rl); + rhl = (rh << 16) | rl; + r0ohm = (rhl*10) / 33554432; + + pr_debug("Left_rhl = 0x%x rh=0x%x rl=0x%x\n", rhl, rh, rl); + pr_info("Left channel %d.%dohm\n", (r0ohm/10), (r0ohm%10)); + + r0l = 562949953421312; + if (rhl != 0) + do_div(r0l, rhl); + pr_debug("Left_r0 = 0x%llx\n", r0l); + + regmap_write(rt1305->regmap, RT1305_SPK_TEMP_PROTECTION_1, 0x9200); + regmap_write(rt1305->regmap, RT1305_PR_BASE + 0xd4, 0xfb00); + regmap_write(rt1305->regmap, RT1305_PR_BASE + 0xd4, 0xff80); + msleep(2000); + regmap_read(rt1305->regmap, RT1305_PR_BASE + 0x55, &rh); + regmap_read(rt1305->regmap, RT1305_PR_BASE + 0x56, &rl); + rhl = (rh << 16) | rl; + r0ohm = (rhl*10) / 33554432; + + pr_debug("Right_rhl = 0x%x rh=0x%x rl=0x%x\n", rhl, rh, rl); + pr_info("Right channel %d.%dohm\n", (r0ohm/10), (r0ohm%10)); + + r0r = 562949953421312; + if (rhl != 0) + do_div(r0r, rhl); + pr_debug("Right_r0 = 0x%llx\n", r0r); + + regmap_write(rt1305->regmap, RT1305_SPK_TEMP_PROTECTION_1, 0xc2ec); + + if ((r0l > R0_UPPER) && (r0l < R0_LOWER) && + (r0r > R0_UPPER) && (r0r < R0_LOWER)) { + regmap_write(rt1305->regmap, RT1305_PR_BASE + 0x4e, + (r0l >> 16) & 0xffff); + regmap_write(rt1305->regmap, RT1305_PR_BASE + 0x4f, + r0l & 0xffff); + regmap_write(rt1305->regmap, RT1305_PR_BASE + 0xfe, + ((r0r >> 16) & 0xffff) | 0xf800); + regmap_write(rt1305->regmap, RT1305_PR_BASE + 0xfd, + r0r & 0xffff); + } else { + pr_err("R0 calibration failed\n"); + } + + /* restore some registers */ + regmap_write(rt1305->regmap, RT1305_POWER_CTRL_1, 0x0dfe); + usleep_range(200000, 400000); + regmap_write(rt1305->regmap, RT1305_PR_BASE + 0x5d, 0x0442); + regmap_write(rt1305->regmap, RT1305_CLOCK_DETECT, 0x3000); + regmap_write(rt1305->regmap, RT1305_CLK_1, 0x0400); + regmap_write(rt1305->regmap, RT1305_POWER_CTRL_1, 0x0000); + regmap_write(rt1305->regmap, RT1305_CAL_EFUSE_CLOCK, 0x8000); + regmap_write(rt1305->regmap, RT1305_POWER_CTRL_2, 0x1020); + regmap_write(rt1305->regmap, RT1305_POWER_CTRL_3, 0x0000); + + regcache_cache_bypass(rt1305->regmap, false); +} + +static int rt1305_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt1305_priv *rt1305; + int ret; + unsigned int val; + + rt1305 = devm_kzalloc(&i2c->dev, sizeof(struct rt1305_priv), + GFP_KERNEL); + if (rt1305 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, rt1305); + + rt1305->regmap = devm_regmap_init_i2c(i2c, &rt1305_regmap); + if (IS_ERR(rt1305->regmap)) { + ret = PTR_ERR(rt1305->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + regmap_read(rt1305->regmap, RT1305_DEVICE_ID, &val); + if (val != RT1305_DEVICE_ID_NUM) { + dev_err(&i2c->dev, + "Device with ID register %x is not rt1305\n", val); + return -ENODEV; + } + + rt1305_reset(rt1305->regmap); + rt1305_calibrate(rt1305); + + return snd_soc_register_component(&i2c->dev, &soc_component_dev_rt1305, + rt1305_dai, ARRAY_SIZE(rt1305_dai)); +} + +static int rt1305_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_component(&i2c->dev); + + return 0; +} + +static void rt1305_i2c_shutdown(struct i2c_client *client) +{ + struct rt1305_priv *rt1305 = i2c_get_clientdata(client); + + rt1305_reset(rt1305->regmap); +} + + +static struct i2c_driver rt1305_i2c_driver = { + .driver = { + .name = "rt1305", + .owner = THIS_MODULE, +#if defined(CONFIG_OF) + .of_match_table = rt1305_of_match, +#endif +#if defined(CONFIG_ACPI) + .acpi_match_table = ACPI_PTR(rt1305_acpi_match) +#endif + }, + .probe = rt1305_i2c_probe, + .remove = rt1305_i2c_remove, + .shutdown = rt1305_i2c_shutdown, + .id_table = rt1305_i2c_id, +}; +module_i2c_driver(rt1305_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT1305 amplifier driver"); +MODULE_AUTHOR("Shuming Fan <shumingf(a)realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt1305.h b/sound/soc/codecs/rt1305.h new file mode 100644 index 0000000..bde86f9 --- /dev/null +++ b/sound/soc/codecs/rt1305.h @@ -0,0 +1,276 @@ +/* + * RT1305.h -- RT1305 ALSA SoC amplifier component driver + * + * Copyright 2018 Realtek Semiconductor Corp. + * Author: Shuming Fan <shumingf(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _RT1305_H_ +#define _RT1305_H_ + +#define RT1305_DEVICE_ID_NUM 0x6251 + +#define RT1305_RESET 0x00 +#define RT1305_CLK_1 0x04 +#define RT1305_CLK_2 0x05 +#define RT1305_CLK_3 0x06 +#define RT1305_DFLL_REG 0x07 +#define RT1305_CAL_EFUSE_CLOCK 0x08 +#define RT1305_PLL0_1 0x0a +#define RT1305_PLL0_2 0x0b +#define RT1305_PLL1_1 0x0c +#define RT1305_PLL1_2 0x0d +#define RT1305_MIXER_CTRL_1 0x10 +#define RT1305_MIXER_CTRL_2 0x11 +#define RT1305_DAC_SET_1 0x12 +#define RT1305_DAC_SET_2 0x14 +#define RT1305_ADC_SET_1 0x16 +#define RT1305_ADC_SET_2 0x17 +#define RT1305_ADC_SET_3 0x18 +#define RT1305_PATH_SET 0x20 +#define RT1305_SPDIF_IN_SET_1 0x22 +#define RT1305_SPDIF_IN_SET_2 0x24 +#define RT1305_SPDIF_IN_SET_3 0x26 +#define RT1305_SPDIF_OUT_SET_1 0x28 +#define RT1305_SPDIF_OUT_SET_2 0x2a +#define RT1305_SPDIF_OUT_SET_3 0x2b +#define RT1305_I2S_SET_1 0x2d +#define RT1305_I2S_SET_2 0x2e +#define RT1305_PBTL_MONO_MODE_SRC 0x2f +#define RT1305_MANUALLY_I2C_DEVICE 0x32 +#define RT1305_POWER_STATUS 0x39 +#define RT1305_POWER_CTRL_1 0x3a +#define RT1305_POWER_CTRL_2 0x3b +#define RT1305_POWER_CTRL_3 0x3c +#define RT1305_POWER_CTRL_4 0x3d +#define RT1305_POWER_CTRL_5 0x3e +#define RT1305_CLOCK_DETECT 0x3f +#define RT1305_BIQUAD_SET_1 0x40 +#define RT1305_BIQUAD_SET_2 0x42 +#define RT1305_ADJUSTED_HPF_1 0x46 +#define RT1305_ADJUSTED_HPF_2 0x47 +#define RT1305_EQ_SET_1 0x4b +#define RT1305_EQ_SET_2 0x4c +#define RT1305_SPK_TEMP_PROTECTION_0 0x4f +#define RT1305_SPK_TEMP_PROTECTION_1 0x50 +#define RT1305_SPK_TEMP_PROTECTION_2 0x51 +#define RT1305_SPK_TEMP_PROTECTION_3 0x52 +#define RT1305_SPK_DC_DETECT_1 0x53 +#define RT1305_SPK_DC_DETECT_2 0x54 +#define RT1305_LOUDNESS 0x58 +#define RT1305_THERMAL_FOLD_BACK_1 0x5e +#define RT1305_THERMAL_FOLD_BACK_2 0x5f +#define RT1305_SILENCE_DETECT 0x60 +#define RT1305_ALC_DRC_1 0x62 +#define RT1305_ALC_DRC_2 0x63 +#define RT1305_ALC_DRC_3 0x64 +#define RT1305_ALC_DRC_4 0x65 +#define RT1305_PRIV_INDEX 0x6a +#define RT1305_PRIV_DATA 0x6c +#define RT1305_SPK_EXCURSION_LIMITER_7 0x76 +#define RT1305_VERSION_ID 0x7a +#define RT1305_VENDOR_ID 0x7c +#define RT1305_DEVICE_ID 0x7e +#define RT1305_EFUSE_1 0x80 +#define RT1305_EFUSE_2 0x81 +#define RT1305_EFUSE_3 0x82 +#define RT1305_DC_CALIB_1 0x90 +#define RT1305_DC_CALIB_2 0x91 +#define RT1305_DC_CALIB_3 0x92 +#define RT1305_DAC_OFFSET_1 0x93 +#define RT1305_DAC_OFFSET_2 0x94 +#define RT1305_DAC_OFFSET_3 0x95 +#define RT1305_DAC_OFFSET_4 0x96 +#define RT1305_DAC_OFFSET_5 0x97 +#define RT1305_DAC_OFFSET_6 0x98 +#define RT1305_DAC_OFFSET_7 0x99 +#define RT1305_DAC_OFFSET_8 0x9a +#define RT1305_DAC_OFFSET_9 0x9b +#define RT1305_DAC_OFFSET_10 0x9c +#define RT1305_DAC_OFFSET_11 0x9d +#define RT1305_DAC_OFFSET_12 0x9e +#define RT1305_DAC_OFFSET_13 0x9f +#define RT1305_DAC_OFFSET_14 0xa0 +#define RT1305_TRIM_1 0xb0 +#define RT1305_TRIM_2 0xb1 +#define RT1305_TUNE_INTERNAL_OSC 0xb2 +#define RT1305_BIQUAD1_H0_L_28_16 0xc0 +#define RT1305_BIQUAD3_A2_R_15_0 0xfb +#define RT1305_MAX_REG 0xff + +/* CLOCK-1 (0x04) */ +#define RT1305_SEL_PLL_SRC_2_MASK (0x1 << 15) +#define RT1305_SEL_PLL_SRC_2_SFT 15 +#define RT1305_SEL_PLL_SRC_2_MCLK (0x0 << 15) +#define RT1305_SEL_PLL_SRC_2_RCCLK (0x1 << 15) +#define RT1305_DIV_PLL_SRC_2_MASK (0x3 << 13) +#define RT1305_DIV_PLL_SRC_2_SFT 13 +#define RT1305_SEL_PLL_SRC_1_MASK (0x3 << 10) +#define RT1305_SEL_PLL_SRC_1_SFT 10 +#define RT1305_SEL_PLL_SRC_1_PLL2 (0x0 << 10) +#define RT1305_SEL_PLL_SRC_1_BCLK (0x1 << 10) +#define RT1305_SEL_PLL_SRC_1_DFLL (0x2 << 10) +#define RT1305_SEL_FS_SYS_PRE_MASK (0x3 << 8) +#define RT1305_SEL_FS_SYS_PRE_SFT 8 +#define RT1305_SEL_FS_SYS_PRE_MCLK (0x0 << 8) +#define RT1305_SEL_FS_SYS_PRE_PLL (0x1 << 8) +#define RT1305_SEL_FS_SYS_PRE_RCCLK (0x2 << 8) +#define RT1305_DIV_FS_SYS_MASK (0x7 << 4) +#define RT1305_DIV_FS_SYS_SFT 4 + +/* PLL1M/N/K Code-1 (0x0c) */ +#define RT1305_PLL_1_M_SFT 12 +#define RT1305_PLL_1_M_BYPASS_MASK (0x1 << 11) +#define RT1305_PLL_1_M_BYPASS_SFT 11 +#define RT1305_PLL_1_M_BYPASS (0x1 << 11) +#define RT1305_PLL_1_N_MASK (0x1ff << 0) + +/* DAC Setting (0x14) */ +#define RT1305_DVOL_MUTE_L_EN_SFT 15 +#define RT1305_DVOL_MUTE_R_EN_SFT 14 + +/* I2S Setting-1 (0x2d) */ +#define RT1305_SEL_I2S_OUT_MODE_MASK (0x1 << 15) +#define RT1305_SEL_I2S_OUT_MODE_SFT 15 +#define RT1305_SEL_I2S_OUT_MODE_S (0x0 << 15) +#define RT1305_SEL_I2S_OUT_MODE_M (0x1 << 15) + +/* I2S Setting-2 (0x2e) */ +#define RT1305_I2S_DF_SEL_MASK (0x3 << 12) +#define RT1305_I2S_DF_SEL_SFT 12 +#define RT1305_I2S_DF_SEL_I2S (0x0 << 12) +#define RT1305_I2S_DF_SEL_LEFT (0x1 << 12) +#define RT1305_I2S_DF_SEL_PCM_A (0x2 << 12) +#define RT1305_I2S_DF_SEL_PCM_B (0x3 << 12) +#define RT1305_I2S_DL_SEL_MASK (0x3 << 10) +#define RT1305_I2S_DL_SEL_SFT 10 +#define RT1305_I2S_DL_SEL_16B (0x0 << 10) +#define RT1305_I2S_DL_SEL_20B (0x1 << 10) +#define RT1305_I2S_DL_SEL_24B (0x2 << 10) +#define RT1305_I2S_DL_SEL_8B (0x3 << 10) +#define RT1305_I2S_BCLK_MASK (0x1 << 9) +#define RT1305_I2S_BCLK_SFT 9 +#define RT1305_I2S_BCLK_NORMAL (0x0 << 9) +#define RT1305_I2S_BCLK_INV (0x1 << 9) + +/* Power Control-1 (0x3a) */ +#define RT1305_POW_PDB_JD_MASK (0x1 << 12) +#define RT1305_POW_PDB_JD (0x1 << 12) +#define RT1305_POW_PDB_JD_BIT 12 +#define RT1305_POW_PLL0_EN (0x1 << 11) +#define RT1305_POW_PLL0_EN_BIT 11 +#define RT1305_POW_PLL1_EN (0x1 << 10) +#define RT1305_POW_PLL1_EN_BIT 10 +#define RT1305_POW_PDB_JD_POLARITY (0x1 << 9) +#define RT1305_POW_PDB_JD_POLARITY_BIT 9 +#define RT1305_POW_MBIAS_LV (0x1 << 8) +#define RT1305_POW_MBIAS_LV_BIT 8 +#define RT1305_POW_BG_MBIAS_LV (0x1 << 7) +#define RT1305_POW_BG_MBIAS_LV_BIT 7 +#define RT1305_POW_LDO2 (0x1 << 6) +#define RT1305_POW_LDO2_BIT 6 +#define RT1305_POW_BG2 (0x1 << 5) +#define RT1305_POW_BG2_BIT 5 +#define RT1305_POW_LDO2_IB2 (0x1 << 4) +#define RT1305_POW_LDO2_IB2_BIT 4 +#define RT1305_POW_VREF (0x1 << 3) +#define RT1305_POW_VREF_BIT 3 +#define RT1305_POW_VREF1 (0x1 << 2) +#define RT1305_POW_VREF1_BIT 2 +#define RT1305_POW_VREF2 (0x1 << 1) +#define RT1305_POW_VREF2_BIT 1 + +/* Power Control-2 (0x3b) */ +#define RT1305_POW_DISC_VREF (1 << 15) +#define RT1305_POW_DISC_VREF_BIT 15 +#define RT1305_POW_FASTB_VREF (1 << 14) +#define RT1305_POW_FASTB_VREF_BIT 14 +#define RT1305_POW_ULTRA_FAST_VREF (1 << 13) +#define RT1305_POW_ULTRA_FAST_VREF_BIT 13 +#define RT1305_POW_CKXEN_DAC (1 << 12) +#define RT1305_POW_CKXEN_DAC_BIT 12 +#define RT1305_POW_EN_CKGEN_DAC (1 << 11) +#define RT1305_POW_EN_CKGEN_DAC_BIT 11 +#define RT1305_POW_DAC1_L (1 << 10) +#define RT1305_POW_DAC1_L_BIT 10 +#define RT1305_POW_DAC1_R (1 << 9) +#define RT1305_POW_DAC1_R_BIT 9 +#define RT1305_POW_CLAMP (1 << 8) +#define RT1305_POW_CLAMP_BIT 8 +#define RT1305_POW_BUFL (1 << 7) +#define RT1305_POW_BUFL_BIT 7 +#define RT1305_POW_BUFR (1 << 6) +#define RT1305_POW_BUFR_BIT 6 +#define RT1305_POW_EN_CKGEN_ADC (1 << 5) +#define RT1305_POW_EN_CKGEN_ADC_BIT 5 +#define RT1305_POW_ADC3_L (1 << 4) +#define RT1305_POW_ADC3_L_BIT 4 +#define RT1305_POW_ADC3_R (1 << 3) +#define RT1305_POW_ADC3_R_BIT 3 +#define RT1305_POW_TRIOSC (1 << 2) +#define RT1305_POW_TRIOSC_BIT 2 +#define RT1305_POR_AVDD1 (1 << 1) +#define RT1305_POR_AVDD1_BIT 1 +#define RT1305_POR_AVDD2 (1 << 0) +#define RT1305_POR_AVDD2_BIT 0 + +/* Power Control-3 (0x3c) */ +#define RT1305_POW_VSENSE_RCH (1 << 15) +#define RT1305_POW_VSENSE_RCH_BIT 15 +#define RT1305_POW_VSENSE_LCH (1 << 14) +#define RT1305_POW_VSENSE_LCH_BIT 14 +#define RT1305_POW_ISENSE_RCH (1 << 13) +#define RT1305_POW_ISENSE_RCH_BIT 13 +#define RT1305_POW_ISENSE_LCH (1 << 12) +#define RT1305_POW_ISENSE_LCH_BIT 12 +#define RT1305_POW_POR_AVDD1 (1 << 11) +#define RT1305_POW_POR_AVDD1_BIT 11 +#define RT1305_POW_POR_AVDD2 (1 << 10) +#define RT1305_POW_POR_AVDD2_BIT 10 +#define RT1305_EN_K_HV (1 << 9) +#define RT1305_EN_K_HV_BIT 9 +#define RT1305_EN_PRE_K_HV (1 << 8) +#define RT1305_EN_PRE_K_HV_BIT 8 +#define RT1305_EN_EFUSE_1P8V (1 << 7) +#define RT1305_EN_EFUSE_1P8V_BIT 7 +#define RT1305_EN_EFUSE_5V (1 << 6) +#define RT1305_EN_EFUSE_5V_BIT 6 +#define RT1305_EN_VCM_6172 (1 << 5) +#define RT1305_EN_VCM_6172_BIT 5 +#define RT1305_POR_EFUSE (1 << 4) +#define RT1305_POR_EFUSE_BIT 4 + +/* Clock Detect (0x3f) */ +#define RT1305_SEL_CLK_DET_SRC_MASK (0x1 << 12) +#define RT1305_SEL_CLK_DET_SRC_SFT 12 +#define RT1305_SEL_CLK_DET_SRC_MCLK (0x0 << 12) +#define RT1305_SEL_CLK_DET_SRC_BCLK (0x1 << 12) + + +/* System Clock Source */ +enum { + RT1305_FS_SYS_PRE_S_MCLK, + RT1305_FS_SYS_PRE_S_PLL1, + RT1305_FS_SYS_PRE_S_RCCLK, /* 98.304M Hz */ +}; + +/* PLL Source 1/2 */ +enum { + RT1305_PLL1_S_BCLK, + RT1305_PLL2_S_MCLK, + RT1305_PLL2_S_RCCLK, /* 98.304M Hz */ +}; + +enum { + RT1305_AIF1, + RT1305_AIFS +}; + +#define R0_UPPER 0x2E8BA2 //5.5 ohm +#define R0_LOWER 0x666666 //2.5 ohm + +#endif /* end of _RT1305_H_ */ -- 2.7.4

2 1

[alsa-devel] Applied "ASoC: rt5668: add rt5668B codec driver" to the asoc tree
by Mark Brown 16 Apr '18

16 Apr '18

The patch ASoC: rt5668: add rt5668B codec driver has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From d59fb2856223219ccaa73bd2e96021f02ea5c266 Mon Sep 17 00:00:00 2001 From: Bard Liao <bardliao(a)realtek.com> Date: Thu, 22 Mar 2018 14:12:33 +0800 Subject: [PATCH] ASoC: rt5668: add rt5668B codec driver This is the initial codec driver for rt5668b. Signed-off-by: Bard Liao <bardliao(a)realtek.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../devicetree/bindings/sound/rt5668.txt | 50 + include/sound/rt5668.h | 40 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt5668.c | 2639 +++++++++++++++++ sound/soc/codecs/rt5668.h | 1318 ++++++++ 6 files changed, 4055 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rt5668.txt create mode 100644 include/sound/rt5668.h create mode 100644 sound/soc/codecs/rt5668.c create mode 100644 sound/soc/codecs/rt5668.h diff --git a/Documentation/devicetree/bindings/sound/rt5668.txt b/Documentation/devicetree/bindings/sound/rt5668.txt new file mode 100644 index 000000000000..c88b96e7764b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5668.txt @@ -0,0 +1,50 @@ +RT5668B audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt5668b" + +- reg : The I2C address of the device. + +Optional properties: + +- interrupts : The CODEC's interrupt output. + +- realtek,dmic1-data-pin + 0: dmic1 is not used + 1: using GPIO2 pin as dmic1 data pin + 2: using GPIO5 pin as dmic1 data pin + +- realtek,dmic1-clk-pin + 0: using GPIO1 pin as dmic1 clock pin + 1: using GPIO3 pin as dmic1 clock pin + +- realtek,jd-src + 0: No JD is used + 1: using JD1 as JD source + +- realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin. + +Pins on the device (for linking into audio routes) for RT5668B: + + * DMIC L1 + * DMIC R1 + * IN1P + * HPOL + * HPOR + +Example: + +rt5668 { + compatible = "realtek,rt5668b"; + reg = <0x1a>; + interrupt-parent = <&gpio>; + interrupts = <TEGRA_GPIO(U, 6) GPIO_ACTIVE_HIGH>; + realtek,ldo1-en-gpios = + <&gpio TEGRA_GPIO(R, 2) GPIO_ACTIVE_HIGH>; + realtek,dmic1-data-pin = <1>; + realtek,dmic1-clk-pin = <1>; + realtek,jd-src = <1>; +}; diff --git a/include/sound/rt5668.h b/include/sound/rt5668.h new file mode 100644 index 000000000000..f907b78696cf --- /dev/null +++ b/include/sound/rt5668.h @@ -0,0 +1,40 @@ +/* + * linux/sound/rt5668.h -- Platform data for RT5668 + * + * Copyright 2018 Realtek Microelectronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __LINUX_SND_RT5668_H +#define __LINUX_SND_RT5668_H + +enum rt5668_dmic1_data_pin { + RT5668_DMIC1_NULL, + RT5668_DMIC1_DATA_GPIO2, + RT5668_DMIC1_DATA_GPIO5, +}; + +enum rt5668_dmic1_clk_pin { + RT5668_DMIC1_CLK_GPIO1, + RT5668_DMIC1_CLK_GPIO3, +}; + +enum rt5668_jd_src { + RT5668_JD_NULL, + RT5668_JD1, +}; + +struct rt5668_platform_data { + + int ldo1_en; /* GPIO for LDO1_EN */ + + enum rt5668_dmic1_data_pin dmic1_data_pin; + enum rt5668_dmic1_clk_pin dmic1_clk_pin; + enum rt5668_jd_src jd_src; +}; + +#endif + diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 665edb5b77ff..251e67f180fe 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -137,6 +137,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_RT5660 if I2C select SND_SOC_RT5663 if I2C select SND_SOC_RT5665 if I2C + select SND_SOC_RT5668 if I2C select SND_SOC_RT5670 if I2C select SND_SOC_RT5677 if I2C && SPI_MASTER select SND_SOC_SGTL5000 if I2C @@ -771,6 +772,7 @@ config SND_SOC_RL6231 default y if SND_SOC_RT5660=y default y if SND_SOC_RT5663=y default y if SND_SOC_RT5665=y + default y if SND_SOC_RT5668=y default y if SND_SOC_RT5670=y default y if SND_SOC_RT5677=y default y if SND_SOC_RT1305=y @@ -783,6 +785,7 @@ config SND_SOC_RL6231 default m if SND_SOC_RT5660=m default m if SND_SOC_RT5663=m default m if SND_SOC_RT5665=m + default m if SND_SOC_RT5668=m default m if SND_SOC_RT5670=m default m if SND_SOC_RT5677=m default m if SND_SOC_RT1305=m @@ -850,6 +853,9 @@ config SND_SOC_RT5663 config SND_SOC_RT5665 tristate +config SND_SOC_RT5668 + tristate + config SND_SOC_RT5670 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index cccd7749e319..d3b73021a401 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -141,6 +141,7 @@ snd-soc-rt5659-objs := rt5659.o snd-soc-rt5660-objs := rt5660.o snd-soc-rt5663-objs := rt5663.o snd-soc-rt5665-objs := rt5665.o +snd-soc-rt5668-objs := rt5668.o snd-soc-rt5670-objs := rt5670.o snd-soc-rt5677-objs := rt5677.o snd-soc-rt5677-spi-objs := rt5677-spi.o @@ -396,6 +397,7 @@ obj-$(CONFIG_SND_SOC_RT5659) += snd-soc-rt5659.o obj-$(CONFIG_SND_SOC_RT5660) += snd-soc-rt5660.o obj-$(CONFIG_SND_SOC_RT5663) += snd-soc-rt5663.o obj-$(CONFIG_SND_SOC_RT5665) += snd-soc-rt5665.o +obj-$(CONFIG_SND_SOC_RT5668) += snd-soc-rt5668.o obj-$(CONFIG_SND_SOC_RT5670) += snd-soc-rt5670.o obj-$(CONFIG_SND_SOC_RT5677) += snd-soc-rt5677.o obj-$(CONFIG_SND_SOC_RT5677_SPI) += snd-soc-rt5677-spi.o diff --git a/sound/soc/codecs/rt5668.c b/sound/soc/codecs/rt5668.c new file mode 100644 index 000000000000..52a343f96eb2 --- /dev/null +++ b/sound/soc/codecs/rt5668.c @@ -0,0 +1,2639 @@ +/* + * rt5668.c -- RT5668B ALSA SoC audio component driver + * + * Copyright 2018 Realtek Semiconductor Corp. + * Author: Bard Liao <bardliao(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <linux/acpi.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <linux/regulator/consumer.h> +#include <linux/mutex.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/jack.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/rt5668.h> + +#include "rl6231.h" +#include "rt5668.h" + +#define RT5668_NUM_SUPPLIES 3 + +static const char *rt5668_supply_names[RT5668_NUM_SUPPLIES] = { + "AVDD", + "MICVDD", + "VBAT", +}; + +struct rt5668_priv { + struct snd_soc_component *component; + struct rt5668_platform_data pdata; + struct regmap *regmap; + struct snd_soc_jack *hs_jack; + struct regulator_bulk_data supplies[RT5668_NUM_SUPPLIES]; + struct delayed_work jack_detect_work; + struct delayed_work jd_check_work; + struct mutex calibrate_mutex; + + int sysclk; + int sysclk_src; + int lrck[RT5668_AIFS]; + int bclk[RT5668_AIFS]; + int master[RT5668_AIFS]; + + int pll_src; + int pll_in; + int pll_out; + + int jack_type; +}; + +static const struct reg_default rt5668_reg[] = { + {0x0002, 0x8080}, + {0x0003, 0x8000}, + {0x0005, 0x0000}, + {0x0006, 0x0000}, + {0x0008, 0x800f}, + {0x000b, 0x0000}, + {0x0010, 0x4040}, + {0x0011, 0x0000}, + {0x0012, 0x1404}, + {0x0013, 0x1000}, + {0x0014, 0xa00a}, + {0x0015, 0x0404}, + {0x0016, 0x0404}, + {0x0019, 0xafaf}, + {0x001c, 0x2f2f}, + {0x001f, 0x0000}, + {0x0022, 0x5757}, + {0x0023, 0x0039}, + {0x0024, 0x000b}, + {0x0026, 0xc0c4}, + {0x0029, 0x8080}, + {0x002a, 0xa0a0}, + {0x002b, 0x0300}, + {0x0030, 0x0000}, + {0x003c, 0x0080}, + {0x0044, 0x0c0c}, + {0x0049, 0x0000}, + {0x0061, 0x0000}, + {0x0062, 0x0000}, + {0x0063, 0x003f}, + {0x0064, 0x0000}, + {0x0065, 0x0000}, + {0x0066, 0x0030}, + {0x0067, 0x0000}, + {0x006b, 0x0000}, + {0x006c, 0x0000}, + {0x006d, 0x2200}, + {0x006e, 0x0a10}, + {0x0070, 0x8000}, + {0x0071, 0x8000}, + {0x0073, 0x0000}, + {0x0074, 0x0000}, + {0x0075, 0x0002}, + {0x0076, 0x0001}, + {0x0079, 0x0000}, + {0x007a, 0x0000}, + {0x007b, 0x0000}, + {0x007c, 0x0100}, + {0x007e, 0x0000}, + {0x0080, 0x0000}, + {0x0081, 0x0000}, + {0x0082, 0x0000}, + {0x0083, 0x0000}, + {0x0084, 0x0000}, + {0x0085, 0x0000}, + {0x0086, 0x0005}, + {0x0087, 0x0000}, + {0x0088, 0x0000}, + {0x008c, 0x0003}, + {0x008d, 0x0000}, + {0x008e, 0x0060}, + {0x008f, 0x1000}, + {0x0091, 0x0c26}, + {0x0092, 0x0073}, + {0x0093, 0x0000}, + {0x0094, 0x0080}, + {0x0098, 0x0000}, + {0x009a, 0x0000}, + {0x009b, 0x0000}, + {0x009c, 0x0000}, + {0x009d, 0x0000}, + {0x009e, 0x100c}, + {0x009f, 0x0000}, + {0x00a0, 0x0000}, + {0x00a3, 0x0002}, + {0x00a4, 0x0001}, + {0x00ae, 0x2040}, + {0x00af, 0x0000}, + {0x00b6, 0x0000}, + {0x00b7, 0x0000}, + {0x00b8, 0x0000}, + {0x00b9, 0x0002}, + {0x00be, 0x0000}, + {0x00c0, 0x0160}, + {0x00c1, 0x82a0}, + {0x00c2, 0x0000}, + {0x00d0, 0x0000}, + {0x00d1, 0x2244}, + {0x00d2, 0x3300}, + {0x00d3, 0x2200}, + {0x00d4, 0x0000}, + {0x00d9, 0x0009}, + {0x00da, 0x0000}, + {0x00db, 0x0000}, + {0x00dc, 0x00c0}, + {0x00dd, 0x2220}, + {0x00de, 0x3131}, + {0x00df, 0x3131}, + {0x00e0, 0x3131}, + {0x00e2, 0x0000}, + {0x00e3, 0x4000}, + {0x00e4, 0x0aa0}, + {0x00e5, 0x3131}, + {0x00e6, 0x3131}, + {0x00e7, 0x3131}, + {0x00e8, 0x3131}, + {0x00ea, 0xb320}, + {0x00eb, 0x0000}, + {0x00f0, 0x0000}, + {0x00f1, 0x00d0}, + {0x00f2, 0x00d0}, + {0x00f6, 0x0000}, + {0x00fa, 0x0000}, + {0x00fb, 0x0000}, + {0x00fc, 0x0000}, + {0x00fd, 0x0000}, + {0x00fe, 0x10ec}, + {0x00ff, 0x6530}, + {0x0100, 0xa0a0}, + {0x010b, 0x0000}, + {0x010c, 0xae00}, + {0x010d, 0xaaa0}, + {0x010e, 0x8aa2}, + {0x010f, 0x02a2}, + {0x0110, 0xc000}, + {0x0111, 0x04a2}, + {0x0112, 0x2800}, + {0x0113, 0x0000}, + {0x0117, 0x0100}, + {0x0125, 0x0410}, + {0x0132, 0x6026}, + {0x0136, 0x5555}, + {0x0138, 0x3700}, + {0x013a, 0x2000}, + {0x013b, 0x2000}, + {0x013c, 0x2005}, + {0x013f, 0x0000}, + {0x0142, 0x0000}, + {0x0145, 0x0002}, + {0x0146, 0x0000}, + {0x0147, 0x0000}, + {0x0148, 0x0000}, + {0x0149, 0x0000}, + {0x0150, 0x79a1}, + {0x0151, 0x0000}, + {0x0160, 0x4ec0}, + {0x0161, 0x0080}, + {0x0162, 0x0200}, + {0x0163, 0x0800}, + {0x0164, 0x0000}, + {0x0165, 0x0000}, + {0x0166, 0x0000}, + {0x0167, 0x000f}, + {0x0168, 0x000f}, + {0x0169, 0x0021}, + {0x0190, 0x413d}, + {0x0194, 0x0000}, + {0x0195, 0x0000}, + {0x0197, 0x0022}, + {0x0198, 0x0000}, + {0x0199, 0x0000}, + {0x01af, 0x0000}, + {0x01b0, 0x0400}, + {0x01b1, 0x0000}, + {0x01b2, 0x0000}, + {0x01b3, 0x0000}, + {0x01b4, 0x0000}, + {0x01b5, 0x0000}, + {0x01b6, 0x01c3}, + {0x01b7, 0x02a0}, + {0x01b8, 0x03e9}, + {0x01b9, 0x1389}, + {0x01ba, 0xc351}, + {0x01bb, 0x0009}, + {0x01bc, 0x0018}, + {0x01bd, 0x002a}, + {0x01be, 0x004c}, + {0x01bf, 0x0097}, + {0x01c0, 0x433d}, + {0x01c1, 0x2800}, + {0x01c2, 0x0000}, + {0x01c3, 0x0000}, + {0x01c4, 0x0000}, + {0x01c5, 0x0000}, + {0x01c6, 0x0000}, + {0x01c7, 0x0000}, + {0x01c8, 0x40af}, + {0x01c9, 0x0702}, + {0x01ca, 0x0000}, + {0x01cb, 0x0000}, + {0x01cc, 0x5757}, + {0x01cd, 0x5757}, + {0x01ce, 0x5757}, + {0x01cf, 0x5757}, + {0x01d0, 0x5757}, + {0x01d1, 0x5757}, + {0x01d2, 0x5757}, + {0x01d3, 0x5757}, + {0x01d4, 0x5757}, + {0x01d5, 0x5757}, + {0x01d6, 0x0000}, + {0x01d7, 0x0008}, + {0x01d8, 0x0029}, + {0x01d9, 0x3333}, + {0x01da, 0x0000}, + {0x01db, 0x0004}, + {0x01dc, 0x0000}, + {0x01de, 0x7c00}, + {0x01df, 0x0320}, + {0x01e0, 0x06a1}, + {0x01e1, 0x0000}, + {0x01e2, 0x0000}, + {0x01e3, 0x0000}, + {0x01e4, 0x0000}, + {0x01e6, 0x0001}, + {0x01e7, 0x0000}, + {0x01e8, 0x0000}, + {0x01ea, 0x0000}, + {0x01eb, 0x0000}, + {0x01ec, 0x0000}, + {0x01ed, 0x0000}, + {0x01ee, 0x0000}, + {0x01ef, 0x0000}, + {0x01f0, 0x0000}, + {0x01f1, 0x0000}, + {0x01f2, 0x0000}, + {0x01f3, 0x0000}, + {0x01f4, 0x0000}, + {0x0210, 0x6297}, + {0x0211, 0xa005}, + {0x0212, 0x824c}, + {0x0213, 0xf7ff}, + {0x0214, 0xf24c}, + {0x0215, 0x0102}, + {0x0216, 0x00a3}, + {0x0217, 0x0048}, + {0x0218, 0xa2c0}, + {0x0219, 0x0400}, + {0x021a, 0x00c8}, + {0x021b, 0x00c0}, + {0x021c, 0x0000}, + {0x0250, 0x4500}, + {0x0251, 0x40b3}, + {0x0252, 0x0000}, + {0x0253, 0x0000}, + {0x0254, 0x0000}, + {0x0255, 0x0000}, + {0x0256, 0x0000}, + {0x0257, 0x0000}, + {0x0258, 0x0000}, + {0x0259, 0x0000}, + {0x025a, 0x0005}, + {0x0270, 0x0000}, + {0x02ff, 0x0110}, + {0x0300, 0x001f}, + {0x0301, 0x032c}, + {0x0302, 0x5f21}, + {0x0303, 0x4000}, + {0x0304, 0x4000}, + {0x0305, 0x06d5}, + {0x0306, 0x8000}, + {0x0307, 0x0700}, + {0x0310, 0x4560}, + {0x0311, 0xa4a8}, + {0x0312, 0x7418}, + {0x0313, 0x0000}, + {0x0314, 0x0006}, + {0x0315, 0xffff}, + {0x0316, 0xc400}, + {0x0317, 0x0000}, + {0x03c0, 0x7e00}, + {0x03c1, 0x8000}, + {0x03c2, 0x8000}, + {0x03c3, 0x8000}, + {0x03c4, 0x8000}, + {0x03c5, 0x8000}, + {0x03c6, 0x8000}, + {0x03c7, 0x8000}, + {0x03c8, 0x8000}, + {0x03c9, 0x8000}, + {0x03ca, 0x8000}, + {0x03cb, 0x8000}, + {0x03cc, 0x8000}, + {0x03d0, 0x0000}, + {0x03d1, 0x0000}, + {0x03d2, 0x0000}, + {0x03d3, 0x0000}, + {0x03d4, 0x2000}, + {0x03d5, 0x2000}, + {0x03d6, 0x0000}, + {0x03d7, 0x0000}, + {0x03d8, 0x2000}, + {0x03d9, 0x2000}, + {0x03da, 0x2000}, + {0x03db, 0x2000}, + {0x03dc, 0x0000}, + {0x03dd, 0x0000}, + {0x03de, 0x0000}, + {0x03df, 0x2000}, + {0x03e0, 0x0000}, + {0x03e1, 0x0000}, + {0x03e2, 0x0000}, + {0x03e3, 0x0000}, + {0x03e4, 0x0000}, + {0x03e5, 0x0000}, + {0x03e6, 0x0000}, + {0x03e7, 0x0000}, + {0x03e8, 0x0000}, + {0x03e9, 0x0000}, + {0x03ea, 0x0000}, + {0x03eb, 0x0000}, + {0x03ec, 0x0000}, + {0x03ed, 0x0000}, + {0x03ee, 0x0000}, + {0x03ef, 0x0000}, + {0x03f0, 0x0800}, + {0x03f1, 0x0800}, + {0x03f2, 0x0800}, + {0x03f3, 0x0800}, +}; + +static bool rt5668_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT5668_RESET: + case RT5668_CBJ_CTRL_2: + case RT5668_INT_ST_1: + case RT5668_4BTN_IL_CMD_1: + case RT5668_AJD1_CTRL: + case RT5668_HP_CALIB_CTRL_1: + case RT5668_DEVICE_ID: + case RT5668_I2C_MODE: + case RT5668_HP_CALIB_CTRL_10: + case RT5668_EFUSE_CTRL_2: + case RT5668_JD_TOP_VC_VTRL: + case RT5668_HP_IMP_SENS_CTRL_19: + case RT5668_IL_CMD_1: + case RT5668_SAR_IL_CMD_2: + case RT5668_SAR_IL_CMD_4: + case RT5668_SAR_IL_CMD_10: + case RT5668_SAR_IL_CMD_11: + case RT5668_EFUSE_CTRL_6...RT5668_EFUSE_CTRL_11: + case RT5668_HP_CALIB_STA_1...RT5668_HP_CALIB_STA_11: + return true; + default: + return false; + } +} + +static bool rt5668_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT5668_RESET: + case RT5668_VERSION_ID: + case RT5668_VENDOR_ID: + case RT5668_DEVICE_ID: + case RT5668_HP_CTRL_1: + case RT5668_HP_CTRL_2: + case RT5668_HPL_GAIN: + case RT5668_HPR_GAIN: + case RT5668_I2C_CTRL: + case RT5668_CBJ_BST_CTRL: + case RT5668_CBJ_CTRL_1: + case RT5668_CBJ_CTRL_2: + case RT5668_CBJ_CTRL_3: + case RT5668_CBJ_CTRL_4: + case RT5668_CBJ_CTRL_5: + case RT5668_CBJ_CTRL_6: + case RT5668_CBJ_CTRL_7: + case RT5668_DAC1_DIG_VOL: + case RT5668_STO1_ADC_DIG_VOL: + case RT5668_STO1_ADC_BOOST: + case RT5668_HP_IMP_GAIN_1: + case RT5668_HP_IMP_GAIN_2: + case RT5668_SIDETONE_CTRL: + case RT5668_STO1_ADC_MIXER: + case RT5668_AD_DA_MIXER: + case RT5668_STO1_DAC_MIXER: + case RT5668_A_DAC1_MUX: + case RT5668_DIG_INF2_DATA: + case RT5668_REC_MIXER: + case RT5668_CAL_REC: + case RT5668_ALC_BACK_GAIN: + case RT5668_PWR_DIG_1: + case RT5668_PWR_DIG_2: + case RT5668_PWR_ANLG_1: + case RT5668_PWR_ANLG_2: + case RT5668_PWR_ANLG_3: + case RT5668_PWR_MIXER: + case RT5668_PWR_VOL: + case RT5668_CLK_DET: + case RT5668_RESET_LPF_CTRL: + case RT5668_RESET_HPF_CTRL: + case RT5668_DMIC_CTRL_1: + case RT5668_I2S1_SDP: + case RT5668_I2S2_SDP: + case RT5668_ADDA_CLK_1: + case RT5668_ADDA_CLK_2: + case RT5668_I2S1_F_DIV_CTRL_1: + case RT5668_I2S1_F_DIV_CTRL_2: + case RT5668_TDM_CTRL: + case RT5668_TDM_ADDA_CTRL_1: + case RT5668_TDM_ADDA_CTRL_2: + case RT5668_DATA_SEL_CTRL_1: + case RT5668_TDM_TCON_CTRL: + case RT5668_GLB_CLK: + case RT5668_PLL_CTRL_1: + case RT5668_PLL_CTRL_2: + case RT5668_PLL_TRACK_1: + case RT5668_PLL_TRACK_2: + case RT5668_PLL_TRACK_3: + case RT5668_PLL_TRACK_4: + case RT5668_PLL_TRACK_5: + case RT5668_PLL_TRACK_6: + case RT5668_PLL_TRACK_11: + case RT5668_SDW_REF_CLK: + case RT5668_DEPOP_1: + case RT5668_DEPOP_2: + case RT5668_HP_CHARGE_PUMP_1: + case RT5668_HP_CHARGE_PUMP_2: + case RT5668_MICBIAS_1: + case RT5668_MICBIAS_2: + case RT5668_PLL_TRACK_12: + case RT5668_PLL_TRACK_14: + case RT5668_PLL2_CTRL_1: + case RT5668_PLL2_CTRL_2: + case RT5668_PLL2_CTRL_3: + case RT5668_PLL2_CTRL_4: + case RT5668_RC_CLK_CTRL: + case RT5668_I2S_M_CLK_CTRL_1: + case RT5668_I2S2_F_DIV_CTRL_1: + case RT5668_I2S2_F_DIV_CTRL_2: + case RT5668_EQ_CTRL_1: + case RT5668_EQ_CTRL_2: + case RT5668_IRQ_CTRL_1: + case RT5668_IRQ_CTRL_2: + case RT5668_IRQ_CTRL_3: + case RT5668_IRQ_CTRL_4: + case RT5668_INT_ST_1: + case RT5668_GPIO_CTRL_1: + case RT5668_GPIO_CTRL_2: + case RT5668_GPIO_CTRL_3: + case RT5668_HP_AMP_DET_CTRL_1: + case RT5668_HP_AMP_DET_CTRL_2: + case RT5668_MID_HP_AMP_DET: + case RT5668_LOW_HP_AMP_DET: + case RT5668_DELAY_BUF_CTRL: + case RT5668_SV_ZCD_1: + case RT5668_SV_ZCD_2: + case RT5668_IL_CMD_1: + case RT5668_IL_CMD_2: + case RT5668_IL_CMD_3: + case RT5668_IL_CMD_4: + case RT5668_IL_CMD_5: + case RT5668_IL_CMD_6: + case RT5668_4BTN_IL_CMD_1: + case RT5668_4BTN_IL_CMD_2: + case RT5668_4BTN_IL_CMD_3: + case RT5668_4BTN_IL_CMD_4: + case RT5668_4BTN_IL_CMD_5: + case RT5668_4BTN_IL_CMD_6: + case RT5668_4BTN_IL_CMD_7: + case RT5668_ADC_STO1_HP_CTRL_1: + case RT5668_ADC_STO1_HP_CTRL_2: + case RT5668_AJD1_CTRL: + case RT5668_JD1_THD: + case RT5668_JD2_THD: + case RT5668_JD_CTRL_1: + case RT5668_DUMMY_1: + case RT5668_DUMMY_2: + case RT5668_DUMMY_3: + case RT5668_DAC_ADC_DIG_VOL1: + case RT5668_BIAS_CUR_CTRL_2: + case RT5668_BIAS_CUR_CTRL_3: + case RT5668_BIAS_CUR_CTRL_4: + case RT5668_BIAS_CUR_CTRL_5: + case RT5668_BIAS_CUR_CTRL_6: + case RT5668_BIAS_CUR_CTRL_7: + case RT5668_BIAS_CUR_CTRL_8: + case RT5668_BIAS_CUR_CTRL_9: + case RT5668_BIAS_CUR_CTRL_10: + case RT5668_VREF_REC_OP_FB_CAP_CTRL: + case RT5668_CHARGE_PUMP_1: + case RT5668_DIG_IN_CTRL_1: + case RT5668_PAD_DRIVING_CTRL: + case RT5668_SOFT_RAMP_DEPOP: + case RT5668_CHOP_DAC: + case RT5668_CHOP_ADC: + case RT5668_CALIB_ADC_CTRL: + case RT5668_VOL_TEST: + case RT5668_SPKVDD_DET_STA: + case RT5668_TEST_MODE_CTRL_1: + case RT5668_TEST_MODE_CTRL_2: + case RT5668_TEST_MODE_CTRL_3: + case RT5668_TEST_MODE_CTRL_4: + case RT5668_TEST_MODE_CTRL_5: + case RT5668_PLL1_INTERNAL: + case RT5668_PLL2_INTERNAL: + case RT5668_STO_NG2_CTRL_1: + case RT5668_STO_NG2_CTRL_2: + case RT5668_STO_NG2_CTRL_3: + case RT5668_STO_NG2_CTRL_4: + case RT5668_STO_NG2_CTRL_5: + case RT5668_STO_NG2_CTRL_6: + case RT5668_STO_NG2_CTRL_7: + case RT5668_STO_NG2_CTRL_8: + case RT5668_STO_NG2_CTRL_9: + case RT5668_STO_NG2_CTRL_10: + case RT5668_STO1_DAC_SIL_DET: + case RT5668_SIL_PSV_CTRL1: + case RT5668_SIL_PSV_CTRL2: + case RT5668_SIL_PSV_CTRL3: + case RT5668_SIL_PSV_CTRL4: + case RT5668_SIL_PSV_CTRL5: + case RT5668_HP_IMP_SENS_CTRL_01: + case RT5668_HP_IMP_SENS_CTRL_02: + case RT5668_HP_IMP_SENS_CTRL_03: + case RT5668_HP_IMP_SENS_CTRL_04: + case RT5668_HP_IMP_SENS_CTRL_05: + case RT5668_HP_IMP_SENS_CTRL_06: + case RT5668_HP_IMP_SENS_CTRL_07: + case RT5668_HP_IMP_SENS_CTRL_08: + case RT5668_HP_IMP_SENS_CTRL_09: + case RT5668_HP_IMP_SENS_CTRL_10: + case RT5668_HP_IMP_SENS_CTRL_11: + case RT5668_HP_IMP_SENS_CTRL_12: + case RT5668_HP_IMP_SENS_CTRL_13: + case RT5668_HP_IMP_SENS_CTRL_14: + case RT5668_HP_IMP_SENS_CTRL_15: + case RT5668_HP_IMP_SENS_CTRL_16: + case RT5668_HP_IMP_SENS_CTRL_17: + case RT5668_HP_IMP_SENS_CTRL_18: + case RT5668_HP_IMP_SENS_CTRL_19: + case RT5668_HP_IMP_SENS_CTRL_20: + case RT5668_HP_IMP_SENS_CTRL_21: + case RT5668_HP_IMP_SENS_CTRL_22: + case RT5668_HP_IMP_SENS_CTRL_23: + case RT5668_HP_IMP_SENS_CTRL_24: + case RT5668_HP_IMP_SENS_CTRL_25: + case RT5668_HP_IMP_SENS_CTRL_26: + case RT5668_HP_IMP_SENS_CTRL_27: + case RT5668_HP_IMP_SENS_CTRL_28: + case RT5668_HP_IMP_SENS_CTRL_29: + case RT5668_HP_IMP_SENS_CTRL_30: + case RT5668_HP_IMP_SENS_CTRL_31: + case RT5668_HP_IMP_SENS_CTRL_32: + case RT5668_HP_IMP_SENS_CTRL_33: + case RT5668_HP_IMP_SENS_CTRL_34: + case RT5668_HP_IMP_SENS_CTRL_35: + case RT5668_HP_IMP_SENS_CTRL_36: + case RT5668_HP_IMP_SENS_CTRL_37: + case RT5668_HP_IMP_SENS_CTRL_38: + case RT5668_HP_IMP_SENS_CTRL_39: + case RT5668_HP_IMP_SENS_CTRL_40: + case RT5668_HP_IMP_SENS_CTRL_41: + case RT5668_HP_IMP_SENS_CTRL_42: + case RT5668_HP_IMP_SENS_CTRL_43: + case RT5668_HP_LOGIC_CTRL_1: + case RT5668_HP_LOGIC_CTRL_2: + case RT5668_HP_LOGIC_CTRL_3: + case RT5668_HP_CALIB_CTRL_1: + case RT5668_HP_CALIB_CTRL_2: + case RT5668_HP_CALIB_CTRL_3: + case RT5668_HP_CALIB_CTRL_4: + case RT5668_HP_CALIB_CTRL_5: + case RT5668_HP_CALIB_CTRL_6: + case RT5668_HP_CALIB_CTRL_7: + case RT5668_HP_CALIB_CTRL_9: + case RT5668_HP_CALIB_CTRL_10: + case RT5668_HP_CALIB_CTRL_11: + case RT5668_HP_CALIB_STA_1: + case RT5668_HP_CALIB_STA_2: + case RT5668_HP_CALIB_STA_3: + case RT5668_HP_CALIB_STA_4: + case RT5668_HP_CALIB_STA_5: + case RT5668_HP_CALIB_STA_6: + case RT5668_HP_CALIB_STA_7: + case RT5668_HP_CALIB_STA_8: + case RT5668_HP_CALIB_STA_9: + case RT5668_HP_CALIB_STA_10: + case RT5668_HP_CALIB_STA_11: + case RT5668_SAR_IL_CMD_1: + case RT5668_SAR_IL_CMD_2: + case RT5668_SAR_IL_CMD_3: + case RT5668_SAR_IL_CMD_4: + case RT5668_SAR_IL_CMD_5: + case RT5668_SAR_IL_CMD_6: + case RT5668_SAR_IL_CMD_7: + case RT5668_SAR_IL_CMD_8: + case RT5668_SAR_IL_CMD_9: + case RT5668_SAR_IL_CMD_10: + case RT5668_SAR_IL_CMD_11: + case RT5668_SAR_IL_CMD_12: + case RT5668_SAR_IL_CMD_13: + case RT5668_EFUSE_CTRL_1: + case RT5668_EFUSE_CTRL_2: + case RT5668_EFUSE_CTRL_3: + case RT5668_EFUSE_CTRL_4: + case RT5668_EFUSE_CTRL_5: + case RT5668_EFUSE_CTRL_6: + case RT5668_EFUSE_CTRL_7: + case RT5668_EFUSE_CTRL_8: + case RT5668_EFUSE_CTRL_9: + case RT5668_EFUSE_CTRL_10: + case RT5668_EFUSE_CTRL_11: + case RT5668_JD_TOP_VC_VTRL: + case RT5668_DRC1_CTRL_0: + case RT5668_DRC1_CTRL_1: + case RT5668_DRC1_CTRL_2: + case RT5668_DRC1_CTRL_3: + case RT5668_DRC1_CTRL_4: + case RT5668_DRC1_CTRL_5: + case RT5668_DRC1_CTRL_6: + case RT5668_DRC1_HARD_LMT_CTRL_1: + case RT5668_DRC1_HARD_LMT_CTRL_2: + case RT5668_DRC1_PRIV_1: + case RT5668_DRC1_PRIV_2: + case RT5668_DRC1_PRIV_3: + case RT5668_DRC1_PRIV_4: + case RT5668_DRC1_PRIV_5: + case RT5668_DRC1_PRIV_6: + case RT5668_DRC1_PRIV_7: + case RT5668_DRC1_PRIV_8: + case RT5668_EQ_AUTO_RCV_CTRL1: + case RT5668_EQ_AUTO_RCV_CTRL2: + case RT5668_EQ_AUTO_RCV_CTRL3: + case RT5668_EQ_AUTO_RCV_CTRL4: + case RT5668_EQ_AUTO_RCV_CTRL5: + case RT5668_EQ_AUTO_RCV_CTRL6: + case RT5668_EQ_AUTO_RCV_CTRL7: + case RT5668_EQ_AUTO_RCV_CTRL8: + case RT5668_EQ_AUTO_RCV_CTRL9: + case RT5668_EQ_AUTO_RCV_CTRL10: + case RT5668_EQ_AUTO_RCV_CTRL11: + case RT5668_EQ_AUTO_RCV_CTRL12: + case RT5668_EQ_AUTO_RCV_CTRL13: + case RT5668_ADC_L_EQ_LPF1_A1: + case RT5668_R_EQ_LPF1_A1: + case RT5668_L_EQ_LPF1_H0: + case RT5668_R_EQ_LPF1_H0: + case RT5668_L_EQ_BPF1_A1: + case RT5668_R_EQ_BPF1_A1: + case RT5668_L_EQ_BPF1_A2: + case RT5668_R_EQ_BPF1_A2: + case RT5668_L_EQ_BPF1_H0: + case RT5668_R_EQ_BPF1_H0: + case RT5668_L_EQ_BPF2_A1: + case RT5668_R_EQ_BPF2_A1: + case RT5668_L_EQ_BPF2_A2: + case RT5668_R_EQ_BPF2_A2: + case RT5668_L_EQ_BPF2_H0: + case RT5668_R_EQ_BPF2_H0: + case RT5668_L_EQ_BPF3_A1: + case RT5668_R_EQ_BPF3_A1: + case RT5668_L_EQ_BPF3_A2: + case RT5668_R_EQ_BPF3_A2: + case RT5668_L_EQ_BPF3_H0: + case RT5668_R_EQ_BPF3_H0: + case RT5668_L_EQ_BPF4_A1: + case RT5668_R_EQ_BPF4_A1: + case RT5668_L_EQ_BPF4_A2: + case RT5668_R_EQ_BPF4_A2: + case RT5668_L_EQ_BPF4_H0: + case RT5668_R_EQ_BPF4_H0: + case RT5668_L_EQ_HPF1_A1: + case RT5668_R_EQ_HPF1_A1: + case RT5668_L_EQ_HPF1_H0: + case RT5668_R_EQ_HPF1_H0: + case RT5668_L_EQ_PRE_VOL: + case RT5668_R_EQ_PRE_VOL: + case RT5668_L_EQ_POST_VOL: + case RT5668_R_EQ_POST_VOL: + case RT5668_I2C_MODE: + return true; + default: + return false; + } +} + +static const DECLARE_TLV_DB_SCALE(hp_vol_tlv, -2250, 150, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); + +/* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */ +static const DECLARE_TLV_DB_RANGE(bst_tlv, + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0), + 3, 5, TLV_DB_SCALE_ITEM(3000, 500, 0), + 6, 6, TLV_DB_SCALE_ITEM(4400, 0, 0), + 7, 7, TLV_DB_SCALE_ITEM(5000, 0, 0), + 8, 8, TLV_DB_SCALE_ITEM(5200, 0, 0) +); + +/* Interface data select */ +static const char * const rt5668_data_select[] = { + "L/R", "R/L", "L/L", "R/R" +}; + +static SOC_ENUM_SINGLE_DECL(rt5668_if2_adc_enum, + RT5668_DIG_INF2_DATA, RT5668_IF2_ADC_SEL_SFT, rt5668_data_select); + +static SOC_ENUM_SINGLE_DECL(rt5668_if1_01_adc_enum, + RT5668_TDM_ADDA_CTRL_1, RT5668_IF1_ADC1_SEL_SFT, rt5668_data_select); + +static SOC_ENUM_SINGLE_DECL(rt5668_if1_23_adc_enum, + RT5668_TDM_ADDA_CTRL_1, RT5668_IF1_ADC2_SEL_SFT, rt5668_data_select); + +static SOC_ENUM_SINGLE_DECL(rt5668_if1_45_adc_enum, + RT5668_TDM_ADDA_CTRL_1, RT5668_IF1_ADC3_SEL_SFT, rt5668_data_select); + +static SOC_ENUM_SINGLE_DECL(rt5668_if1_67_adc_enum, + RT5668_TDM_ADDA_CTRL_1, RT5668_IF1_ADC4_SEL_SFT, rt5668_data_select); + +static const struct snd_kcontrol_new rt5668_if2_adc_swap_mux = + SOC_DAPM_ENUM("IF2 ADC Swap Mux", rt5668_if2_adc_enum); + +static const struct snd_kcontrol_new rt5668_if1_01_adc_swap_mux = + SOC_DAPM_ENUM("IF1 01 ADC Swap Mux", rt5668_if1_01_adc_enum); + +static const struct snd_kcontrol_new rt5668_if1_23_adc_swap_mux = + SOC_DAPM_ENUM("IF1 23 ADC Swap Mux", rt5668_if1_23_adc_enum); + +static const struct snd_kcontrol_new rt5668_if1_45_adc_swap_mux = + SOC_DAPM_ENUM("IF1 45 ADC Swap Mux", rt5668_if1_45_adc_enum); + +static const struct snd_kcontrol_new rt5668_if1_67_adc_swap_mux = + SOC_DAPM_ENUM("IF1 67 ADC Swap Mux", rt5668_if1_67_adc_enum); + +static void rt5668_reset(struct regmap *regmap) +{ + regmap_write(regmap, RT5668_RESET, 0); + regmap_write(regmap, RT5668_I2C_MODE, 1); +} +/** + * rt5668_sel_asrc_clk_src - select ASRC clock source for a set of filters + * @component: SoC audio component device. + * @filter_mask: mask of filters. + * @clk_src: clock source + * + * The ASRC function is for asynchronous MCLK and LRCK. Also, since RT5668 can + * only support standard 32fs or 64fs i2s format, ASRC should be enabled to + * support special i2s clock format such as Intel's 100fs(100 * sampling rate). + * ASRC function will track i2s clock and generate a corresponding system clock + * for codec. This function provides an API to select the clock source for a + * set of filters specified by the mask. And the component driver will turn on + * ASRC for these filters if ASRC is selected as their clock source. + */ +int rt5668_sel_asrc_clk_src(struct snd_soc_component *component, + unsigned int filter_mask, unsigned int clk_src) +{ + + switch (clk_src) { + case RT5668_CLK_SEL_SYS: + case RT5668_CLK_SEL_I2S1_ASRC: + case RT5668_CLK_SEL_I2S2_ASRC: + break; + + default: + return -EINVAL; + } + + if (filter_mask & RT5668_DA_STEREO1_FILTER) { + snd_soc_component_update_bits(component, RT5668_PLL_TRACK_2, + RT5668_FILTER_CLK_SEL_MASK, + clk_src << RT5668_FILTER_CLK_SEL_SFT); + } + + if (filter_mask & RT5668_AD_STEREO1_FILTER) { + snd_soc_component_update_bits(component, RT5668_PLL_TRACK_3, + RT5668_FILTER_CLK_SEL_MASK, + clk_src << RT5668_FILTER_CLK_SEL_SFT); + } + + return 0; +} +EXPORT_SYMBOL_GPL(rt5668_sel_asrc_clk_src); + +static int rt5668_button_detect(struct snd_soc_component *component) +{ + int btn_type, val; + + val = snd_soc_component_read32(component, RT5668_4BTN_IL_CMD_1); + btn_type = val & 0xfff0; + snd_soc_component_write(component, RT5668_4BTN_IL_CMD_1, val); + pr_debug("%s btn_type=%x\n", __func__, btn_type); + + return btn_type; +} + +static void rt5668_enable_push_button_irq(struct snd_soc_component *component, + bool enable) +{ + if (enable) { + snd_soc_component_update_bits(component, RT5668_SAR_IL_CMD_1, + RT5668_SAR_BUTT_DET_MASK, RT5668_SAR_BUTT_DET_EN); + snd_soc_component_update_bits(component, RT5668_SAR_IL_CMD_13, + RT5668_SAR_SOUR_MASK, RT5668_SAR_SOUR_BTN); + snd_soc_component_write(component, RT5668_IL_CMD_1, 0x0040); + snd_soc_component_update_bits(component, RT5668_4BTN_IL_CMD_2, + RT5668_4BTN_IL_MASK | RT5668_4BTN_IL_RST_MASK, + RT5668_4BTN_IL_EN | RT5668_4BTN_IL_NOR); + snd_soc_component_update_bits(component, RT5668_IRQ_CTRL_3, + RT5668_IL_IRQ_MASK, RT5668_IL_IRQ_EN); + } else { + snd_soc_component_update_bits(component, RT5668_IRQ_CTRL_3, + RT5668_IL_IRQ_MASK, RT5668_IL_IRQ_DIS); + snd_soc_component_update_bits(component, RT5668_SAR_IL_CMD_1, + RT5668_SAR_BUTT_DET_MASK, RT5668_SAR_BUTT_DET_DIS); + snd_soc_component_update_bits(component, RT5668_4BTN_IL_CMD_2, + RT5668_4BTN_IL_MASK, RT5668_4BTN_IL_DIS); + snd_soc_component_update_bits(component, RT5668_4BTN_IL_CMD_2, + RT5668_4BTN_IL_RST_MASK, RT5668_4BTN_IL_RST); + snd_soc_component_update_bits(component, RT5668_SAR_IL_CMD_13, + RT5668_SAR_SOUR_MASK, RT5668_SAR_SOUR_TYPE); + } +} + +/** + * rt5668_headset_detect - Detect headset. + * @component: SoC audio component device. + * @jack_insert: Jack insert or not. + * + * Detect whether is headset or not when jack inserted. + * + * Returns detect status. + */ +static int rt5668_headset_detect(struct snd_soc_component *component, + int jack_insert) +{ + struct rt5668_priv *rt5668 = snd_soc_component_get_drvdata(component); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + unsigned int val, count; + + if (jack_insert) { + snd_soc_dapm_force_enable_pin(dapm, "CBJ Power"); + snd_soc_dapm_sync(dapm); + snd_soc_component_update_bits(component, RT5668_CBJ_CTRL_1, + RT5668_TRIG_JD_MASK, RT5668_TRIG_JD_HIGH); + + count = 0; + val = snd_soc_component_read32(component, RT5668_CBJ_CTRL_2) + & RT5668_JACK_TYPE_MASK; + while (val == 0 && count < 50) { + usleep_range(10000, 15000); + val = snd_soc_component_read32(component, + RT5668_CBJ_CTRL_2) & RT5668_JACK_TYPE_MASK; + count++; + } + + switch (val) { + case 0x1: + case 0x2: + rt5668->jack_type = SND_JACK_HEADSET; + rt5668_enable_push_button_irq(component, true); + break; + default: + rt5668->jack_type = SND_JACK_HEADPHONE; + } + + } else { + rt5668_enable_push_button_irq(component, false); + snd_soc_component_update_bits(component, RT5668_CBJ_CTRL_1, + RT5668_TRIG_JD_MASK, RT5668_TRIG_JD_LOW); + snd_soc_dapm_disable_pin(dapm, "CBJ Power"); + snd_soc_dapm_sync(dapm); + + rt5668->jack_type = 0; + } + + dev_dbg(component->dev, "jack_type = %d\n", rt5668->jack_type); + return rt5668->jack_type; +} + +static irqreturn_t rt5668_irq(int irq, void *data) +{ + struct rt5668_priv *rt5668 = data; + + mod_delayed_work(system_power_efficient_wq, + &rt5668->jack_detect_work, msecs_to_jiffies(250)); + + return IRQ_HANDLED; +} + +static void rt5668_jd_check_handler(struct work_struct *work) +{ + struct rt5668_priv *rt5668 = container_of(work, struct rt5668_priv, + jd_check_work.work); + + if (snd_soc_component_read32(rt5668->component, RT5668_AJD1_CTRL) + & RT5668_JDH_RS_MASK) { + /* jack out */ + rt5668->jack_type = rt5668_headset_detect(rt5668->component, 0); + + snd_soc_jack_report(rt5668->hs_jack, rt5668->jack_type, + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); + } else { + schedule_delayed_work(&rt5668->jd_check_work, 500); + } +} + +static int rt5668_set_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *hs_jack, void *data) +{ + struct rt5668_priv *rt5668 = snd_soc_component_get_drvdata(component); + + switch (rt5668->pdata.jd_src) { + case RT5668_JD1: + snd_soc_component_update_bits(component, RT5668_CBJ_CTRL_2, + RT5668_EXT_JD_SRC, RT5668_EXT_JD_SRC_MANUAL); + snd_soc_component_write(component, RT5668_CBJ_CTRL_1, 0xd002); + snd_soc_component_update_bits(component, RT5668_CBJ_CTRL_3, + RT5668_CBJ_IN_BUF_EN, RT5668_CBJ_IN_BUF_EN); + snd_soc_component_update_bits(component, RT5668_SAR_IL_CMD_1, + RT5668_SAR_POW_MASK, RT5668_SAR_POW_EN); + regmap_update_bits(rt5668->regmap, RT5668_GPIO_CTRL_1, + RT5668_GP1_PIN_MASK, RT5668_GP1_PIN_IRQ); + regmap_update_bits(rt5668->regmap, RT5668_RC_CLK_CTRL, + RT5668_POW_IRQ | RT5668_POW_JDH | + RT5668_POW_ANA, RT5668_POW_IRQ | + RT5668_POW_JDH | RT5668_POW_ANA); + regmap_update_bits(rt5668->regmap, RT5668_PWR_ANLG_2, + RT5668_PWR_JDH | RT5668_PWR_JDL, + RT5668_PWR_JDH | RT5668_PWR_JDL); + regmap_update_bits(rt5668->regmap, RT5668_IRQ_CTRL_2, + RT5668_JD1_EN_MASK | RT5668_JD1_POL_MASK, + RT5668_JD1_EN | RT5668_JD1_POL_NOR); + mod_delayed_work(system_power_efficient_wq, + &rt5668->jack_detect_work, msecs_to_jiffies(250)); + break; + + case RT5668_JD_NULL: + regmap_update_bits(rt5668->regmap, RT5668_IRQ_CTRL_2, + RT5668_JD1_EN_MASK, RT5668_JD1_DIS); + regmap_update_bits(rt5668->regmap, RT5668_RC_CLK_CTRL, + RT5668_POW_JDH | RT5668_POW_JDL, 0); + break; + + default: + dev_warn(component->dev, "Wrong JD source\n"); + break; + } + + rt5668->hs_jack = hs_jack; + + return 0; +} + +static void rt5668_jack_detect_handler(struct work_struct *work) +{ + struct rt5668_priv *rt5668 = + container_of(work, struct rt5668_priv, jack_detect_work.work); + int val, btn_type; + + while (!rt5668->component) + usleep_range(10000, 15000); + + while (!rt5668->component->card->instantiated) + usleep_range(10000, 15000); + + mutex_lock(&rt5668->calibrate_mutex); + + val = snd_soc_component_read32(rt5668->component, RT5668_AJD1_CTRL) + & RT5668_JDH_RS_MASK; + if (!val) { + /* jack in */ + if (rt5668->jack_type == 0) { + /* jack was out, report jack type */ + rt5668->jack_type = + rt5668_headset_detect(rt5668->component, 1); + } else { + /* jack is already in, report button event */ + rt5668->jack_type = SND_JACK_HEADSET; + btn_type = rt5668_button_detect(rt5668->component); + /** + * rt5668 can report three kinds of button behavior, + * one click, double click and hold. However, + * currently we will report button pressed/released + * event. So all the three button behaviors are + * treated as button pressed. + */ + switch (btn_type) { + case 0x8000: + case 0x4000: + case 0x2000: + rt5668->jack_type |= SND_JACK_BTN_0; + break; + case 0x1000: + case 0x0800: + case 0x0400: + rt5668->jack_type |= SND_JACK_BTN_1; + break; + case 0x0200: + case 0x0100: + case 0x0080: + rt5668->jack_type |= SND_JACK_BTN_2; + break; + case 0x0040: + case 0x0020: + case 0x0010: + rt5668->jack_type |= SND_JACK_BTN_3; + break; + case 0x0000: /* unpressed */ + break; + default: + btn_type = 0; + dev_err(rt5668->component->dev, + "Unexpected button code 0x%04x\n", + btn_type); + break; + } + } + } else { + /* jack out */ + rt5668->jack_type = rt5668_headset_detect(rt5668->component, 0); + } + + snd_soc_jack_report(rt5668->hs_jack, rt5668->jack_type, + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); + + if (rt5668->jack_type & (SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3)) + schedule_delayed_work(&rt5668->jd_check_work, 0); + else + cancel_delayed_work_sync(&rt5668->jd_check_work); + + mutex_unlock(&rt5668->calibrate_mutex); +} + +static const struct snd_kcontrol_new rt5668_snd_controls[] = { + /* Headphone Output Volume */ + SOC_DOUBLE_R_TLV("Headphone Playback Volume", RT5668_HPL_GAIN, + RT5668_HPR_GAIN, RT5668_G_HP_SFT, 15, 1, hp_vol_tlv), + + /* DAC Digital Volume */ + SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5668_DAC1_DIG_VOL, + RT5668_L_VOL_SFT, RT5668_R_VOL_SFT, 175, 0, dac_vol_tlv), + + /* IN Boost Volume */ + SOC_SINGLE_TLV("CBJ Boost Volume", RT5668_CBJ_BST_CTRL, + RT5668_BST_CBJ_SFT, 8, 0, bst_tlv), + + /* ADC Digital Volume Control */ + SOC_DOUBLE("STO1 ADC Capture Switch", RT5668_STO1_ADC_DIG_VOL, + RT5668_L_MUTE_SFT, RT5668_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_TLV("STO1 ADC Capture Volume", RT5668_STO1_ADC_DIG_VOL, + RT5668_L_VOL_SFT, RT5668_R_VOL_SFT, 127, 0, adc_vol_tlv), + + /* ADC Boost Volume Control */ + SOC_DOUBLE_TLV("STO1 ADC Boost Gain Volume", RT5668_STO1_ADC_BOOST, + RT5668_STO1_ADC_L_BST_SFT, RT5668_STO1_ADC_R_BST_SFT, + 3, 0, adc_bst_tlv), +}; + + +static int rt5668_div_sel(struct rt5668_priv *rt5668, + int target, const int div[], int size) +{ + int i; + + if (rt5668->sysclk < target) { + pr_err("sysclk rate %d is too low\n", + rt5668->sysclk); + return 0; + } + + for (i = 0; i < size - 1; i++) { + pr_info("div[%d]=%d\n", i, div[i]); + if (target * div[i] == rt5668->sysclk) + return i; + if (target * div[i + 1] > rt5668->sysclk) { + pr_err("can't find div for sysclk %d\n", + rt5668->sysclk); + return i; + } + } + + if (target * div[i] < rt5668->sysclk) + pr_err("sysclk rate %d is too high\n", + rt5668->sysclk); + + return size - 1; + +} + +/** + * set_dmic_clk - Set parameter of dmic. + * + * @w: DAPM widget. + * @kcontrol: The kcontrol of this widget. + * @event: Event id. + * + * Choose dmic clock between 1MHz and 3MHz. + * It is better for clock to approximate 3MHz. + */ +static int set_dmic_clk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt5668_priv *rt5668 = snd_soc_component_get_drvdata(component); + int idx = -EINVAL; + static const int div[] = {2, 4, 6, 8, 12, 16, 24, 32, 48, 64, 96, 128}; + + idx = rt5668_div_sel(rt5668, 1500000, div, ARRAY_SIZE(div)); + + snd_soc_component_update_bits(component, RT5668_DMIC_CTRL_1, + RT5668_DMIC_CLK_MASK, idx << RT5668_DMIC_CLK_SFT); + + return 0; +} + +static int set_filter_clk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt5668_priv *rt5668 = snd_soc_component_get_drvdata(component); + int ref, val, reg, idx = -EINVAL; + static const int div[] = {1, 2, 3, 4, 6, 8, 12, 16, 24, 32, 48}; + + val = snd_soc_component_read32(component, RT5668_GPIO_CTRL_1) && + RT5668_GP4_PIN_MASK; + if (w->shift == RT5668_PWR_ADC_S1F_BIT && + val == RT5668_GP4_PIN_ADCDAT2) + ref = 256 * rt5668->lrck[RT5668_AIF2]; + else + ref = 256 * rt5668->lrck[RT5668_AIF1]; + + idx = rt5668_div_sel(rt5668, ref, div, ARRAY_SIZE(div)); + + if (w->shift == RT5668_PWR_ADC_S1F_BIT) + reg = RT5668_PLL_TRACK_3; + else + reg = RT5668_PLL_TRACK_2; + + snd_soc_component_update_bits(component, reg, + RT5668_FILTER_CLK_SEL_MASK, idx << RT5668_FILTER_CLK_SEL_SFT); + + return 0; +} + +static int is_sys_clk_from_pll1(struct snd_soc_dapm_widget *w, + struct snd_soc_dapm_widget *sink) +{ + unsigned int val; + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + + val = snd_soc_component_read32(component, RT5668_GLB_CLK); + val &= RT5668_SCLK_SRC_MASK; + if (val == RT5668_SCLK_SRC_PLL1) + return 1; + else + return 0; +} + +static int is_using_asrc(struct snd_soc_dapm_widget *w, + struct snd_soc_dapm_widget *sink) +{ + unsigned int reg, shift, val; + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + + switch (w->shift) { + case RT5668_ADC_STO1_ASRC_SFT: + reg = RT5668_PLL_TRACK_3; + shift = RT5668_FILTER_CLK_SEL_SFT; + break; + case RT5668_DAC_STO1_ASRC_SFT: + reg = RT5668_PLL_TRACK_2; + shift = RT5668_FILTER_CLK_SEL_SFT; + break; + default: + return 0; + } + + val = (snd_soc_component_read32(component, reg) >> shift) & 0xf; + switch (val) { + case RT5668_CLK_SEL_I2S1_ASRC: + case RT5668_CLK_SEL_I2S2_ASRC: + return 1; + default: + return 0; + } + +} + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt5668_sto1_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5668_STO1_ADC_MIXER, + RT5668_M_STO1_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5668_STO1_ADC_MIXER, + RT5668_M_STO1_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5668_sto1_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5668_STO1_ADC_MIXER, + RT5668_M_STO1_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5668_STO1_ADC_MIXER, + RT5668_M_STO1_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5668_dac_l_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5668_AD_DA_MIXER, + RT5668_M_ADCMIX_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 Switch", RT5668_AD_DA_MIXER, + RT5668_M_DAC1_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5668_dac_r_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5668_AD_DA_MIXER, + RT5668_M_ADCMIX_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 Switch", RT5668_AD_DA_MIXER, + RT5668_M_DAC1_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5668_sto1_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5668_STO1_DAC_MIXER, + RT5668_M_DAC_L1_STO_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5668_STO1_DAC_MIXER, + RT5668_M_DAC_R1_STO_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5668_sto1_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5668_STO1_DAC_MIXER, + RT5668_M_DAC_L1_STO_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5668_STO1_DAC_MIXER, + RT5668_M_DAC_R1_STO_R_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt5668_rec1_l_mix[] = { + SOC_DAPM_SINGLE("CBJ Switch", RT5668_REC_MIXER, + RT5668_M_CBJ_RM1_L_SFT, 1, 1), +}; + +/* STO1 ADC1 Source */ +/* MX-26 [13] [5] */ +static const char * const rt5668_sto1_adc1_src[] = { + "DAC MIX", "ADC" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5668_sto1_adc1l_enum, RT5668_STO1_ADC_MIXER, + RT5668_STO1_ADC1L_SRC_SFT, rt5668_sto1_adc1_src); + +static const struct snd_kcontrol_new rt5668_sto1_adc1l_mux = + SOC_DAPM_ENUM("Stereo1 ADC1L Source", rt5668_sto1_adc1l_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5668_sto1_adc1r_enum, RT5668_STO1_ADC_MIXER, + RT5668_STO1_ADC1R_SRC_SFT, rt5668_sto1_adc1_src); + +static const struct snd_kcontrol_new rt5668_sto1_adc1r_mux = + SOC_DAPM_ENUM("Stereo1 ADC1L Source", rt5668_sto1_adc1r_enum); + +/* STO1 ADC Source */ +/* MX-26 [11:10] [3:2] */ +static const char * const rt5668_sto1_adc_src[] = { + "ADC1 L", "ADC1 R" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5668_sto1_adcl_enum, RT5668_STO1_ADC_MIXER, + RT5668_STO1_ADCL_SRC_SFT, rt5668_sto1_adc_src); + +static const struct snd_kcontrol_new rt5668_sto1_adcl_mux = + SOC_DAPM_ENUM("Stereo1 ADCL Source", rt5668_sto1_adcl_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5668_sto1_adcr_enum, RT5668_STO1_ADC_MIXER, + RT5668_STO1_ADCR_SRC_SFT, rt5668_sto1_adc_src); + +static const struct snd_kcontrol_new rt5668_sto1_adcr_mux = + SOC_DAPM_ENUM("Stereo1 ADCR Source", rt5668_sto1_adcr_enum); + +/* STO1 ADC2 Source */ +/* MX-26 [12] [4] */ +static const char * const rt5668_sto1_adc2_src[] = { + "DAC MIX", "DMIC" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5668_sto1_adc2l_enum, RT5668_STO1_ADC_MIXER, + RT5668_STO1_ADC2L_SRC_SFT, rt5668_sto1_adc2_src); + +static const struct snd_kcontrol_new rt5668_sto1_adc2l_mux = + SOC_DAPM_ENUM("Stereo1 ADC2L Source", rt5668_sto1_adc2l_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5668_sto1_adc2r_enum, RT5668_STO1_ADC_MIXER, + RT5668_STO1_ADC2R_SRC_SFT, rt5668_sto1_adc2_src); + +static const struct snd_kcontrol_new rt5668_sto1_adc2r_mux = + SOC_DAPM_ENUM("Stereo1 ADC2R Source", rt5668_sto1_adc2r_enum); + +/* MX-79 [6:4] I2S1 ADC data location */ +static const unsigned int rt5668_if1_adc_slot_values[] = { + 0, + 2, + 4, + 6, +}; + +static const char * const rt5668_if1_adc_slot_src[] = { + "Slot 0", "Slot 2", "Slot 4", "Slot 6" +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(rt5668_if1_adc_slot_enum, + RT5668_TDM_CTRL, RT5668_TDM_ADC_LCA_SFT, RT5668_TDM_ADC_LCA_MASK, + rt5668_if1_adc_slot_src, rt5668_if1_adc_slot_values); + +static const struct snd_kcontrol_new rt5668_if1_adc_slot_mux = + SOC_DAPM_ENUM("IF1 ADC Slot location", rt5668_if1_adc_slot_enum); + +/* Analog DAC L1 Source, Analog DAC R1 Source*/ +/* MX-2B [4], MX-2B [0]*/ +static const char * const rt5668_alg_dac1_src[] = { + "Stereo1 DAC Mixer", "DAC1" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5668_alg_dac_l1_enum, RT5668_A_DAC1_MUX, + RT5668_A_DACL1_SFT, rt5668_alg_dac1_src); + +static const struct snd_kcontrol_new rt5668_alg_dac_l1_mux = + SOC_DAPM_ENUM("Analog DAC L1 Source", rt5668_alg_dac_l1_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5668_alg_dac_r1_enum, RT5668_A_DAC1_MUX, + RT5668_A_DACR1_SFT, rt5668_alg_dac1_src); + +static const struct snd_kcontrol_new rt5668_alg_dac_r1_mux = + SOC_DAPM_ENUM("Analog DAC R1 Source", rt5668_alg_dac_r1_enum); + +/* Out Switch */ +static const struct snd_kcontrol_new hpol_switch = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5668_HP_CTRL_1, + RT5668_L_MUTE_SFT, 1, 1); +static const struct snd_kcontrol_new hpor_switch = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5668_HP_CTRL_1, + RT5668_R_MUTE_SFT, 1, 1); + +static int rt5668_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_component_write(component, + RT5668_HP_LOGIC_CTRL_2, 0x0012); + snd_soc_component_write(component, + RT5668_HP_CTRL_2, 0x6000); + snd_soc_component_update_bits(component, RT5668_STO_NG2_CTRL_1, + RT5668_NG2_EN_MASK, RT5668_NG2_EN); + snd_soc_component_update_bits(component, + RT5668_DEPOP_1, 0x60, 0x60); + break; + + case SND_SOC_DAPM_POST_PMD: + snd_soc_component_update_bits(component, + RT5668_DEPOP_1, 0x60, 0x0); + snd_soc_component_write(component, + RT5668_HP_CTRL_2, 0x0000); + break; + + default: + return 0; + } + + return 0; + +} + +static int set_dmic_power(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /*Add delay to avoid pop noise*/ + msleep(150); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5655_set_verf(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + switch (w->shift) { + case RT5668_PWR_VREF1_BIT: + snd_soc_component_update_bits(component, + RT5668_PWR_ANLG_1, RT5668_PWR_FV1, 0); + break; + + case RT5668_PWR_VREF2_BIT: + snd_soc_component_update_bits(component, + RT5668_PWR_ANLG_1, RT5668_PWR_FV2, 0); + break; + + default: + break; + } + break; + + case SND_SOC_DAPM_POST_PMU: + usleep_range(15000, 20000); + switch (w->shift) { + case RT5668_PWR_VREF1_BIT: + snd_soc_component_update_bits(component, + RT5668_PWR_ANLG_1, RT5668_PWR_FV1, + RT5668_PWR_FV1); + break; + + case RT5668_PWR_VREF2_BIT: + snd_soc_component_update_bits(component, + RT5668_PWR_ANLG_1, RT5668_PWR_FV2, + RT5668_PWR_FV2); + break; + + default: + break; + } + break; + + default: + return 0; + } + + return 0; +} + +static const unsigned int rt5668_adcdat_pin_values[] = { + 1, + 3, +}; + +static const char * const rt5668_adcdat_pin_select[] = { + "ADCDAT1", + "ADCDAT2", +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(rt5668_adcdat_pin_enum, + RT5668_GPIO_CTRL_1, RT5668_GP4_PIN_SFT, RT5668_GP4_PIN_MASK, + rt5668_adcdat_pin_select, rt5668_adcdat_pin_values); + +static const struct snd_kcontrol_new rt5668_adcdat_pin_ctrl = + SOC_DAPM_ENUM("ADCDAT", rt5668_adcdat_pin_enum); + +static const struct snd_soc_dapm_widget rt5668_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("LDO2", RT5668_PWR_ANLG_3, RT5668_PWR_LDO2_BIT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL1", RT5668_PWR_ANLG_3, RT5668_PWR_PLL_BIT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL2B", RT5668_PWR_ANLG_3, RT5668_PWR_PLL2B_BIT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL2F", RT5668_PWR_ANLG_3, RT5668_PWR_PLL2F_BIT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Vref1", RT5668_PWR_ANLG_1, RT5668_PWR_VREF1_BIT, 0, + rt5655_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SUPPLY("Vref2", RT5668_PWR_ANLG_1, RT5668_PWR_VREF2_BIT, 0, + rt5655_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + + /* ASRC */ + SND_SOC_DAPM_SUPPLY_S("DAC STO1 ASRC", 1, RT5668_PLL_TRACK_1, + RT5668_DAC_STO1_ASRC_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC STO1 ASRC", 1, RT5668_PLL_TRACK_1, + RT5668_ADC_STO1_ASRC_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("AD ASRC", 1, RT5668_PLL_TRACK_1, + RT5668_AD_ASRC_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DA ASRC", 1, RT5668_PLL_TRACK_1, + RT5668_DA_ASRC_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DMIC ASRC", 1, RT5668_PLL_TRACK_1, + RT5668_DMIC_ASRC_SFT, 0, NULL, 0), + + /* Input Side */ + SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5668_PWR_ANLG_2, RT5668_PWR_MB1_BIT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS2", RT5668_PWR_ANLG_2, RT5668_PWR_MB2_BIT, + 0, NULL, 0), + + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC L1"), + SND_SOC_DAPM_INPUT("DMIC R1"), + + SND_SOC_DAPM_INPUT("IN1P"), + + SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0, + set_dmic_clk, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY("DMIC1 Power", RT5668_DMIC_CTRL_1, + RT5668_DMIC_1_EN_SFT, 0, set_dmic_power, SND_SOC_DAPM_POST_PMU), + + /* Boost */ + SND_SOC_DAPM_PGA("BST1 CBJ", SND_SOC_NOPM, + 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("CBJ Power", RT5668_PWR_ANLG_3, + RT5668_PWR_CBJ_BIT, 0, NULL, 0), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIX1L", SND_SOC_NOPM, 0, 0, rt5668_rec1_l_mix, + ARRAY_SIZE(rt5668_rec1_l_mix)), + SND_SOC_DAPM_SUPPLY("RECMIX1L Power", RT5668_PWR_ANLG_2, + RT5668_PWR_RM1_L_BIT, 0, NULL, 0), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC1 L", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC1 R", NULL, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_SUPPLY("ADC1 L Power", RT5668_PWR_DIG_1, + RT5668_PWR_ADC_L1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC1 R Power", RT5668_PWR_DIG_1, + RT5668_PWR_ADC_R1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC1 clock", RT5668_CHOP_ADC, + RT5668_CKGEN_ADC1_SFT, 0, NULL, 0), + + /* ADC Mux */ + SND_SOC_DAPM_MUX("Stereo1 ADC L1 Mux", SND_SOC_NOPM, 0, 0, + &rt5668_sto1_adc1l_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC R1 Mux", SND_SOC_NOPM, 0, 0, + &rt5668_sto1_adc1r_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5668_sto1_adc2l_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5668_sto1_adc2r_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC L Mux", SND_SOC_NOPM, 0, 0, + &rt5668_sto1_adcl_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC R Mux", SND_SOC_NOPM, 0, 0, + &rt5668_sto1_adcr_mux), + SND_SOC_DAPM_MUX("IF1_ADC Mux", SND_SOC_NOPM, 0, 0, + &rt5668_if1_adc_slot_mux), + + /* ADC Mixer */ + SND_SOC_DAPM_SUPPLY("ADC Stereo1 Filter", RT5668_PWR_DIG_2, + RT5668_PWR_ADC_S1F_BIT, 0, set_filter_clk, + SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_MIXER("Stereo1 ADC MIXL", RT5668_STO1_ADC_DIG_VOL, + RT5668_L_MUTE_SFT, 1, rt5668_sto1_adc_l_mix, + ARRAY_SIZE(rt5668_sto1_adc_l_mix)), + SND_SOC_DAPM_MIXER("Stereo1 ADC MIXR", RT5668_STO1_ADC_DIG_VOL, + RT5668_R_MUTE_SFT, 1, rt5668_sto1_adc_r_mix, + ARRAY_SIZE(rt5668_sto1_adc_r_mix)), + + /* ADC PGA */ + SND_SOC_DAPM_PGA("Stereo1 ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Digital Interface */ + SND_SOC_DAPM_SUPPLY("I2S1", RT5668_PWR_DIG_1, RT5668_PWR_I2S1_BIT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S2", RT5668_PWR_DIG_1, RT5668_PWR_I2S2_BIT, + 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1 L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1 R", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Digital Interface Select */ + SND_SOC_DAPM_MUX("IF1 01 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5668_if1_01_adc_swap_mux), + SND_SOC_DAPM_MUX("IF1 23 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5668_if1_23_adc_swap_mux), + SND_SOC_DAPM_MUX("IF1 45 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5668_if1_45_adc_swap_mux), + SND_SOC_DAPM_MUX("IF1 67 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5668_if1_67_adc_swap_mux), + SND_SOC_DAPM_MUX("IF2 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5668_if2_adc_swap_mux), + + SND_SOC_DAPM_MUX("ADCDAT Mux", SND_SOC_NOPM, 0, 0, + &rt5668_adcdat_pin_ctrl), + + /* Audio Interface */ + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, + RT5668_I2S1_SDP, RT5668_SEL_ADCDAT_SFT, 1), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, + RT5668_I2S2_SDP, RT5668_I2S2_PIN_CFG_SFT, 1), + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + + /* Output Side */ + /* DAC mixer before sound effect */ + SND_SOC_DAPM_MIXER("DAC1 MIXL", SND_SOC_NOPM, 0, 0, + rt5668_dac_l_mix, ARRAY_SIZE(rt5668_dac_l_mix)), + SND_SOC_DAPM_MIXER("DAC1 MIXR", SND_SOC_NOPM, 0, 0, + rt5668_dac_r_mix, ARRAY_SIZE(rt5668_dac_r_mix)), + + /* DAC channel Mux */ + SND_SOC_DAPM_MUX("DAC L1 Source", SND_SOC_NOPM, 0, 0, + &rt5668_alg_dac_l1_mux), + SND_SOC_DAPM_MUX("DAC R1 Source", SND_SOC_NOPM, 0, 0, + &rt5668_alg_dac_r1_mux), + + /* DAC Mixer */ + SND_SOC_DAPM_SUPPLY("DAC Stereo1 Filter", RT5668_PWR_DIG_2, + RT5668_PWR_DAC_S1F_BIT, 0, set_filter_clk, + SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_MIXER("Stereo1 DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5668_sto1_dac_l_mix, ARRAY_SIZE(rt5668_sto1_dac_l_mix)), + SND_SOC_DAPM_MIXER("Stereo1 DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5668_sto1_dac_r_mix, ARRAY_SIZE(rt5668_sto1_dac_r_mix)), + + /* DACs */ + SND_SOC_DAPM_DAC("DAC L1", NULL, RT5668_PWR_DIG_1, + RT5668_PWR_DAC_L1_BIT, 0), + SND_SOC_DAPM_DAC("DAC R1", NULL, RT5668_PWR_DIG_1, + RT5668_PWR_DAC_R1_BIT, 0), + SND_SOC_DAPM_SUPPLY_S("DAC 1 Clock", 3, RT5668_CHOP_DAC, + RT5668_CKGEN_DAC1_SFT, 0, NULL, 0), + + /* HPO */ + SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, rt5668_hp_event, + SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_PRE_PMU), + + SND_SOC_DAPM_SUPPLY("HP Amp L", RT5668_PWR_ANLG_1, + RT5668_PWR_HA_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("HP Amp R", RT5668_PWR_ANLG_1, + RT5668_PWR_HA_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("Charge Pump", 1, RT5668_DEPOP_1, + RT5668_PUMP_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("Capless", 2, RT5668_DEPOP_1, + RT5668_CAPLESS_EN_SFT, 0, NULL, 0), + + SND_SOC_DAPM_SWITCH("HPOL Playback", SND_SOC_NOPM, 0, 0, + &hpol_switch), + SND_SOC_DAPM_SWITCH("HPOR Playback", SND_SOC_NOPM, 0, 0, + &hpor_switch), + + /* CLK DET */ + SND_SOC_DAPM_SUPPLY("CLKDET SYS", RT5668_CLK_DET, + RT5668_SYS_CLK_DET_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("CLKDET PLL1", RT5668_CLK_DET, + RT5668_PLL1_CLK_DET_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("CLKDET PLL2", RT5668_CLK_DET, + RT5668_PLL2_CLK_DET_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("CLKDET", RT5668_CLK_DET, + RT5668_POW_CLK_DET_SFT, 0, NULL, 0), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), + +}; + +static const struct snd_soc_dapm_route rt5668_dapm_routes[] = { + /*PLL*/ + {"ADC Stereo1 Filter", NULL, "PLL1", is_sys_clk_from_pll1}, + {"DAC Stereo1 Filter", NULL, "PLL1", is_sys_clk_from_pll1}, + + /*ASRC*/ + {"ADC Stereo1 Filter", NULL, "ADC STO1 ASRC", is_using_asrc}, + {"DAC Stereo1 Filter", NULL, "DAC STO1 ASRC", is_using_asrc}, + {"ADC STO1 ASRC", NULL, "AD ASRC"}, + {"DAC STO1 ASRC", NULL, "DA ASRC"}, + + /*Vref*/ + {"MICBIAS1", NULL, "Vref1"}, + {"MICBIAS1", NULL, "Vref2"}, + {"MICBIAS2", NULL, "Vref1"}, + {"MICBIAS2", NULL, "Vref2"}, + + {"CLKDET SYS", NULL, "CLKDET"}, + + {"IN1P", NULL, "LDO2"}, + + {"BST1 CBJ", NULL, "IN1P"}, + {"BST1 CBJ", NULL, "CBJ Power"}, + {"CBJ Power", NULL, "Vref2"}, + + {"RECMIX1L", "CBJ Switch", "BST1 CBJ"}, + {"RECMIX1L", NULL, "RECMIX1L Power"}, + + {"ADC1 L", NULL, "RECMIX1L"}, + {"ADC1 L", NULL, "ADC1 L Power"}, + {"ADC1 L", NULL, "ADC1 clock"}, + + {"DMIC L1", NULL, "DMIC CLK"}, + {"DMIC L1", NULL, "DMIC1 Power"}, + {"DMIC R1", NULL, "DMIC CLK"}, + {"DMIC R1", NULL, "DMIC1 Power"}, + {"DMIC CLK", NULL, "DMIC ASRC"}, + + {"Stereo1 ADC L Mux", "ADC1 L", "ADC1 L"}, + {"Stereo1 ADC L Mux", "ADC1 R", "ADC1 R"}, + {"Stereo1 ADC R Mux", "ADC1 L", "ADC1 L"}, + {"Stereo1 ADC R Mux", "ADC1 R", "ADC1 R"}, + + {"Stereo1 ADC L1 Mux", "ADC", "Stereo1 ADC L Mux"}, + {"Stereo1 ADC L1 Mux", "DAC MIX", "Stereo1 DAC MIXL"}, + {"Stereo1 ADC L2 Mux", "DMIC", "DMIC L1"}, + {"Stereo1 ADC L2 Mux", "DAC MIX", "Stereo1 DAC MIXL"}, + + {"Stereo1 ADC R1 Mux", "ADC", "Stereo1 ADC R Mux"}, + {"Stereo1 ADC R1 Mux", "DAC MIX", "Stereo1 DAC MIXR"}, + {"Stereo1 ADC R2 Mux", "DMIC", "DMIC R1"}, + {"Stereo1 ADC R2 Mux", "DAC MIX", "Stereo1 DAC MIXR"}, + + {"Stereo1 ADC MIXL", "ADC1 Switch", "Stereo1 ADC L1 Mux"}, + {"Stereo1 ADC MIXL", "ADC2 Switch", "Stereo1 ADC L2 Mux"}, + {"Stereo1 ADC MIXL", NULL, "ADC Stereo1 Filter"}, + + {"Stereo1 ADC MIXR", "ADC1 Switch", "Stereo1 ADC R1 Mux"}, + {"Stereo1 ADC MIXR", "ADC2 Switch", "Stereo1 ADC R2 Mux"}, + {"Stereo1 ADC MIXR", NULL, "ADC Stereo1 Filter"}, + + {"Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXL"}, + {"Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXR"}, + + {"IF1 01 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, + {"IF1 01 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"}, + {"IF1 01 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, + {"IF1 01 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"}, + {"IF1 23 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, + {"IF1 23 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, + {"IF1 23 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"}, + {"IF1 23 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"}, + {"IF1 45 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, + {"IF1 45 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, + {"IF1 45 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"}, + {"IF1 45 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"}, + {"IF1 67 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, + {"IF1 67 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, + {"IF1 67 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"}, + {"IF1 67 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"}, + + {"IF1_ADC Mux", "Slot 0", "IF1 01 ADC Swap Mux"}, + {"IF1_ADC Mux", "Slot 2", "IF1 23 ADC Swap Mux"}, + {"IF1_ADC Mux", "Slot 4", "IF1 45 ADC Swap Mux"}, + {"IF1_ADC Mux", "Slot 6", "IF1 67 ADC Swap Mux"}, + {"IF1_ADC Mux", NULL, "I2S1"}, + {"ADCDAT Mux", "ADCDAT1", "IF1_ADC Mux"}, + {"AIF1TX", NULL, "ADCDAT Mux"}, + {"IF2 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, + {"IF2 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, + {"IF2 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"}, + {"IF2 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"}, + {"ADCDAT Mux", "ADCDAT2", "IF2 ADC Swap Mux"}, + {"AIF2TX", NULL, "ADCDAT Mux"}, + + {"IF1 DAC1 L", NULL, "AIF1RX"}, + {"IF1 DAC1 L", NULL, "I2S1"}, + {"IF1 DAC1 L", NULL, "DAC Stereo1 Filter"}, + {"IF1 DAC1 R", NULL, "AIF1RX"}, + {"IF1 DAC1 R", NULL, "I2S1"}, + {"IF1 DAC1 R", NULL, "DAC Stereo1 Filter"}, + + {"DAC1 MIXL", "Stereo ADC Switch", "Stereo1 ADC MIXL"}, + {"DAC1 MIXL", "DAC1 Switch", "IF1 DAC1 L"}, + {"DAC1 MIXR", "Stereo ADC Switch", "Stereo1 ADC MIXR"}, + {"DAC1 MIXR", "DAC1 Switch", "IF1 DAC1 R"}, + + {"Stereo1 DAC MIXL", "DAC L1 Switch", "DAC1 MIXL"}, + {"Stereo1 DAC MIXL", "DAC R1 Switch", "DAC1 MIXR"}, + + {"Stereo1 DAC MIXR", "DAC R1 Switch", "DAC1 MIXR"}, + {"Stereo1 DAC MIXR", "DAC L1 Switch", "DAC1 MIXL"}, + + {"DAC L1 Source", "DAC1", "DAC1 MIXL"}, + {"DAC L1 Source", "Stereo1 DAC Mixer", "Stereo1 DAC MIXL"}, + {"DAC R1 Source", "DAC1", "DAC1 MIXR"}, + {"DAC R1 Source", "Stereo1 DAC Mixer", "Stereo1 DAC MIXR"}, + + {"DAC L1", NULL, "DAC L1 Source"}, + {"DAC R1", NULL, "DAC R1 Source"}, + + {"DAC L1", NULL, "DAC 1 Clock"}, + {"DAC R1", NULL, "DAC 1 Clock"}, + + {"HP Amp", NULL, "DAC L1"}, + {"HP Amp", NULL, "DAC R1"}, + {"HP Amp", NULL, "HP Amp L"}, + {"HP Amp", NULL, "HP Amp R"}, + {"HP Amp", NULL, "Capless"}, + {"HP Amp", NULL, "Charge Pump"}, + {"HP Amp", NULL, "CLKDET SYS"}, + {"HP Amp", NULL, "CBJ Power"}, + {"HP Amp", NULL, "Vref2"}, + {"HPOL Playback", "Switch", "HP Amp"}, + {"HPOR Playback", "Switch", "HP Amp"}, + {"HPOL", NULL, "HPOL Playback"}, + {"HPOR", NULL, "HPOR Playback"}, +}; + +static int rt5668_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_component *component = dai->component; + unsigned int val = 0; + + switch (slots) { + case 4: + val |= RT5668_TDM_TX_CH_4; + val |= RT5668_TDM_RX_CH_4; + break; + case 6: + val |= RT5668_TDM_TX_CH_6; + val |= RT5668_TDM_RX_CH_6; + break; + case 8: + val |= RT5668_TDM_TX_CH_8; + val |= RT5668_TDM_RX_CH_8; + break; + case 2: + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, RT5668_TDM_CTRL, + RT5668_TDM_TX_CH_MASK | RT5668_TDM_RX_CH_MASK, val); + + switch (slot_width) { + case 16: + val = RT5668_TDM_CL_16; + break; + case 20: + val = RT5668_TDM_CL_20; + break; + case 24: + val = RT5668_TDM_CL_24; + break; + case 32: + val = RT5668_TDM_CL_32; + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, RT5668_TDM_TCON_CTRL, + RT5668_TDM_CL_MASK, val); + + return 0; +} + + +static int rt5668_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct rt5668_priv *rt5668 = snd_soc_component_get_drvdata(component); + unsigned int len_1 = 0, len_2 = 0; + int pre_div, frame_size; + + rt5668->lrck[dai->id] = params_rate(params); + pre_div = rl6231_get_clk_info(rt5668->sysclk, rt5668->lrck[dai->id]); + + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) { + dev_err(component->dev, "Unsupported frame size: %d\n", + frame_size); + return -EINVAL; + } + + dev_dbg(dai->dev, "lrck is %dHz and pre_div is %d for iis %d\n", + rt5668->lrck[dai->id], pre_div, dai->id); + + switch (params_width(params)) { + case 16: + break; + case 20: + len_1 |= RT5668_I2S1_DL_20; + len_2 |= RT5668_I2S2_DL_20; + break; + case 24: + len_1 |= RT5668_I2S1_DL_24; + len_2 |= RT5668_I2S2_DL_24; + break; + case 32: + len_1 |= RT5668_I2S1_DL_32; + len_2 |= RT5668_I2S2_DL_24; + break; + case 8: + len_1 |= RT5668_I2S2_DL_8; + len_2 |= RT5668_I2S2_DL_8; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case RT5668_AIF1: + snd_soc_component_update_bits(component, RT5668_I2S1_SDP, + RT5668_I2S1_DL_MASK, len_1); + if (rt5668->master[RT5668_AIF1]) { + snd_soc_component_update_bits(component, + RT5668_ADDA_CLK_1, RT5668_I2S_M_DIV_MASK, + pre_div << RT5668_I2S_M_DIV_SFT); + } + if (params_channels(params) == 1) /* mono mode */ + snd_soc_component_update_bits(component, + RT5668_I2S1_SDP, RT5668_I2S1_MONO_MASK, + RT5668_I2S1_MONO_EN); + else + snd_soc_component_update_bits(component, + RT5668_I2S1_SDP, RT5668_I2S1_MONO_MASK, + RT5668_I2S1_MONO_DIS); + break; + case RT5668_AIF2: + snd_soc_component_update_bits(component, RT5668_I2S2_SDP, + RT5668_I2S2_DL_MASK, len_2); + if (rt5668->master[RT5668_AIF2]) { + snd_soc_component_update_bits(component, + RT5668_I2S_M_CLK_CTRL_1, RT5668_I2S2_M_PD_MASK, + pre_div << RT5668_I2S2_M_PD_SFT); + } + if (params_channels(params) == 1) /* mono mode */ + snd_soc_component_update_bits(component, + RT5668_I2S2_SDP, RT5668_I2S2_MONO_MASK, + RT5668_I2S2_MONO_EN); + else + snd_soc_component_update_bits(component, + RT5668_I2S2_SDP, RT5668_I2S2_MONO_MASK, + RT5668_I2S2_MONO_DIS); + break; + default: + dev_err(component->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + + return 0; +} + +static int rt5668_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *component = dai->component; + struct rt5668_priv *rt5668 = snd_soc_component_get_drvdata(component); + unsigned int reg_val = 0, tdm_ctrl = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + rt5668->master[dai->id] = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + rt5668->master[dai->id] = 0; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + reg_val |= RT5668_I2S_BP_INV; + tdm_ctrl |= RT5668_TDM_S_BP_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + if (dai->id == RT5668_AIF1) + tdm_ctrl |= RT5668_TDM_S_LP_INV | RT5668_TDM_M_BP_INV; + else + return -EINVAL; + break; + case SND_SOC_DAIFMT_IB_IF: + if (dai->id == RT5668_AIF1) + tdm_ctrl |= RT5668_TDM_S_BP_INV | RT5668_TDM_S_LP_INV | + RT5668_TDM_M_BP_INV | RT5668_TDM_M_LP_INV; + else + return -EINVAL; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_LEFT_J: + reg_val |= RT5668_I2S_DF_LEFT; + tdm_ctrl |= RT5668_TDM_DF_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + reg_val |= RT5668_I2S_DF_PCM_A; + tdm_ctrl |= RT5668_TDM_DF_PCM_A; + break; + case SND_SOC_DAIFMT_DSP_B: + reg_val |= RT5668_I2S_DF_PCM_B; + tdm_ctrl |= RT5668_TDM_DF_PCM_B; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case RT5668_AIF1: + snd_soc_component_update_bits(component, RT5668_I2S1_SDP, + RT5668_I2S_DF_MASK, reg_val); + snd_soc_component_update_bits(component, RT5668_TDM_TCON_CTRL, + RT5668_TDM_MS_MASK | RT5668_TDM_S_BP_MASK | + RT5668_TDM_DF_MASK | RT5668_TDM_M_BP_MASK | + RT5668_TDM_M_LP_MASK | RT5668_TDM_S_LP_MASK, + tdm_ctrl | rt5668->master[dai->id]); + break; + case RT5668_AIF2: + if (rt5668->master[dai->id] == 0) + reg_val |= RT5668_I2S2_MS_S; + snd_soc_component_update_bits(component, RT5668_I2S2_SDP, + RT5668_I2S2_MS_MASK | RT5668_I2S_BP_MASK | + RT5668_I2S_DF_MASK, reg_val); + break; + default: + dev_err(component->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + return 0; +} + +static int rt5668_set_component_sysclk(struct snd_soc_component *component, + int clk_id, int source, unsigned int freq, int dir) +{ + struct rt5668_priv *rt5668 = snd_soc_component_get_drvdata(component); + unsigned int reg_val = 0, src = 0; + + if (freq == rt5668->sysclk && clk_id == rt5668->sysclk_src) + return 0; + + switch (clk_id) { + case RT5668_SCLK_S_MCLK: + reg_val |= RT5668_SCLK_SRC_MCLK; + src = RT5668_CLK_SRC_MCLK; + break; + case RT5668_SCLK_S_PLL1: + reg_val |= RT5668_SCLK_SRC_PLL1; + src = RT5668_CLK_SRC_PLL1; + break; + case RT5668_SCLK_S_PLL2: + reg_val |= RT5668_SCLK_SRC_PLL2; + src = RT5668_CLK_SRC_PLL2; + break; + case RT5668_SCLK_S_RCCLK: + reg_val |= RT5668_SCLK_SRC_RCCLK; + src = RT5668_CLK_SRC_RCCLK; + break; + default: + dev_err(component->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + snd_soc_component_update_bits(component, RT5668_GLB_CLK, + RT5668_SCLK_SRC_MASK, reg_val); + + if (rt5668->master[RT5668_AIF2]) { + snd_soc_component_update_bits(component, + RT5668_I2S_M_CLK_CTRL_1, RT5668_I2S2_SRC_MASK, + src << RT5668_I2S2_SRC_SFT); + } + + rt5668->sysclk = freq; + rt5668->sysclk_src = clk_id; + + dev_dbg(component->dev, "Sysclk is %dHz and clock id is %d\n", + freq, clk_id); + + return 0; +} + +static int rt5668_set_component_pll(struct snd_soc_component *component, + int pll_id, int source, unsigned int freq_in, + unsigned int freq_out) +{ + struct rt5668_priv *rt5668 = snd_soc_component_get_drvdata(component); + struct rl6231_pll_code pll_code; + int ret; + + if (source == rt5668->pll_src && freq_in == rt5668->pll_in && + freq_out == rt5668->pll_out) + return 0; + + if (!freq_in || !freq_out) { + dev_dbg(component->dev, "PLL disabled\n"); + + rt5668->pll_in = 0; + rt5668->pll_out = 0; + snd_soc_component_update_bits(component, RT5668_GLB_CLK, + RT5668_SCLK_SRC_MASK, RT5668_SCLK_SRC_MCLK); + return 0; + } + + switch (source) { + case RT5668_PLL1_S_MCLK: + snd_soc_component_update_bits(component, RT5668_GLB_CLK, + RT5668_PLL1_SRC_MASK, RT5668_PLL1_SRC_MCLK); + break; + case RT5668_PLL1_S_BCLK1: + snd_soc_component_update_bits(component, RT5668_GLB_CLK, + RT5668_PLL1_SRC_MASK, RT5668_PLL1_SRC_BCLK1); + break; + default: + dev_err(component->dev, "Unknown PLL Source %d\n", source); + return -EINVAL; + } + + ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); + if (ret < 0) { + dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + return ret; + } + + dev_dbg(component->dev, "bypass=%d m=%d n=%d k=%d\n", + pll_code.m_bp, (pll_code.m_bp ? 0 : pll_code.m_code), + pll_code.n_code, pll_code.k_code); + + snd_soc_component_write(component, RT5668_PLL_CTRL_1, + pll_code.n_code << RT5668_PLL_N_SFT | pll_code.k_code); + snd_soc_component_write(component, RT5668_PLL_CTRL_2, + (pll_code.m_bp ? 0 : pll_code.m_code) << RT5668_PLL_M_SFT | + pll_code.m_bp << RT5668_PLL_M_BP_SFT); + + rt5668->pll_in = freq_in; + rt5668->pll_out = freq_out; + rt5668->pll_src = source; + + return 0; +} + +static int rt5668_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_component *component = dai->component; + struct rt5668_priv *rt5668 = snd_soc_component_get_drvdata(component); + + rt5668->bclk[dai->id] = ratio; + + switch (ratio) { + case 64: + snd_soc_component_update_bits(component, RT5668_ADDA_CLK_2, + RT5668_I2S2_BCLK_MS2_MASK, + RT5668_I2S2_BCLK_MS2_64); + break; + case 32: + snd_soc_component_update_bits(component, RT5668_ADDA_CLK_2, + RT5668_I2S2_BCLK_MS2_MASK, + RT5668_I2S2_BCLK_MS2_32); + break; + default: + dev_err(dai->dev, "Invalid bclk ratio %d\n", ratio); + return -EINVAL; + } + + return 0; +} + +static int rt5668_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct rt5668_priv *rt5668 = snd_soc_component_get_drvdata(component); + + switch (level) { + case SND_SOC_BIAS_PREPARE: + regmap_update_bits(rt5668->regmap, RT5668_PWR_ANLG_1, + RT5668_PWR_MB | RT5668_PWR_BG, + RT5668_PWR_MB | RT5668_PWR_BG); + regmap_update_bits(rt5668->regmap, RT5668_PWR_DIG_1, + RT5668_DIG_GATE_CTRL | RT5668_PWR_LDO, + RT5668_DIG_GATE_CTRL | RT5668_PWR_LDO); + break; + + case SND_SOC_BIAS_STANDBY: + regmap_update_bits(rt5668->regmap, RT5668_PWR_ANLG_1, + RT5668_PWR_MB, RT5668_PWR_MB); + regmap_update_bits(rt5668->regmap, RT5668_PWR_DIG_1, + RT5668_DIG_GATE_CTRL, RT5668_DIG_GATE_CTRL); + break; + case SND_SOC_BIAS_OFF: + regmap_update_bits(rt5668->regmap, RT5668_PWR_DIG_1, + RT5668_DIG_GATE_CTRL | RT5668_PWR_LDO, 0); + regmap_update_bits(rt5668->regmap, RT5668_PWR_ANLG_1, + RT5668_PWR_MB | RT5668_PWR_BG, 0); + break; + + default: + break; + } + + return 0; +} + +static int rt5668_probe(struct snd_soc_component *component) +{ + struct rt5668_priv *rt5668 = snd_soc_component_get_drvdata(component); + + rt5668->component = component; + + return 0; +} + +static void rt5668_remove(struct snd_soc_component *component) +{ + struct rt5668_priv *rt5668 = snd_soc_component_get_drvdata(component); + + rt5668_reset(rt5668->regmap); +} + +#ifdef CONFIG_PM +static int rt5668_suspend(struct snd_soc_component *component) +{ + struct rt5668_priv *rt5668 = snd_soc_component_get_drvdata(component); + + regcache_cache_only(rt5668->regmap, true); + regcache_mark_dirty(rt5668->regmap); + return 0; +} + +static int rt5668_resume(struct snd_soc_component *component) +{ + struct rt5668_priv *rt5668 = snd_soc_component_get_drvdata(component); + + regcache_cache_only(rt5668->regmap, false); + regcache_sync(rt5668->regmap); + + return 0; +} +#else +#define rt5668_suspend NULL +#define rt5668_resume NULL +#endif + +#define RT5668_STEREO_RATES SNDRV_PCM_RATE_8000_192000 +#define RT5668_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +static const struct snd_soc_dai_ops rt5668_aif1_dai_ops = { + .hw_params = rt5668_hw_params, + .set_fmt = rt5668_set_dai_fmt, + .set_tdm_slot = rt5668_set_tdm_slot, +}; + +static const struct snd_soc_dai_ops rt5668_aif2_dai_ops = { + .hw_params = rt5668_hw_params, + .set_fmt = rt5668_set_dai_fmt, + .set_bclk_ratio = rt5668_set_bclk_ratio, +}; + +static struct snd_soc_dai_driver rt5668_dai[] = { + { + .name = "rt5668-aif1", + .id = RT5668_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5668_STEREO_RATES, + .formats = RT5668_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5668_STEREO_RATES, + .formats = RT5668_FORMATS, + }, + .ops = &rt5668_aif1_dai_ops, + }, + { + .name = "rt5668-aif2", + .id = RT5668_AIF2, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5668_STEREO_RATES, + .formats = RT5668_FORMATS, + }, + .ops = &rt5668_aif2_dai_ops, + }, +}; + +static const struct snd_soc_component_driver soc_component_dev_rt5668 = { + .probe = rt5668_probe, + .remove = rt5668_remove, + .suspend = rt5668_suspend, + .resume = rt5668_resume, + .set_bias_level = rt5668_set_bias_level, + .controls = rt5668_snd_controls, + .num_controls = ARRAY_SIZE(rt5668_snd_controls), + .dapm_widgets = rt5668_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt5668_dapm_widgets), + .dapm_routes = rt5668_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt5668_dapm_routes), + .set_sysclk = rt5668_set_component_sysclk, + .set_pll = rt5668_set_component_pll, + .set_jack = rt5668_set_jack_detect, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static const struct regmap_config rt5668_regmap = { + .reg_bits = 16, + .val_bits = 16, + .max_register = RT5668_I2C_MODE, + .volatile_reg = rt5668_volatile_register, + .readable_reg = rt5668_readable_register, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt5668_reg, + .num_reg_defaults = ARRAY_SIZE(rt5668_reg), + .use_single_rw = true, +}; + +static const struct i2c_device_id rt5668_i2c_id[] = { + {"rt5668b", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, rt5668_i2c_id); + +static int rt5668_parse_dt(struct rt5668_priv *rt5668, struct device *dev) +{ + + of_property_read_u32(dev->of_node, "realtek,dmic1-data-pin", + &rt5668->pdata.dmic1_data_pin); + of_property_read_u32(dev->of_node, "realtek,dmic1-clk-pin", + &rt5668->pdata.dmic1_clk_pin); + of_property_read_u32(dev->of_node, "realtek,jd-src", + &rt5668->pdata.jd_src); + + rt5668->pdata.ldo1_en = of_get_named_gpio(dev->of_node, + "realtek,ldo1-en-gpios", 0); + + return 0; +} + +static void rt5668_calibrate(struct rt5668_priv *rt5668) +{ + int value, count; + + mutex_lock(&rt5668->calibrate_mutex); + + rt5668_reset(rt5668->regmap); + regmap_write(rt5668->regmap, RT5668_PWR_ANLG_1, 0xa2bf); + usleep_range(15000, 20000); + regmap_write(rt5668->regmap, RT5668_PWR_ANLG_1, 0xf2bf); + regmap_write(rt5668->regmap, RT5668_MICBIAS_2, 0x0380); + regmap_write(rt5668->regmap, RT5668_PWR_DIG_1, 0x8001); + regmap_write(rt5668->regmap, RT5668_TEST_MODE_CTRL_1, 0x0000); + regmap_write(rt5668->regmap, RT5668_STO1_DAC_MIXER, 0x2080); + regmap_write(rt5668->regmap, RT5668_STO1_ADC_MIXER, 0x4040); + regmap_write(rt5668->regmap, RT5668_DEPOP_1, 0x0069); + regmap_write(rt5668->regmap, RT5668_CHOP_DAC, 0x3000); + regmap_write(rt5668->regmap, RT5668_HP_CTRL_2, 0x6000); + regmap_write(rt5668->regmap, RT5668_HP_CHARGE_PUMP_1, 0x0f26); + regmap_write(rt5668->regmap, RT5668_CALIB_ADC_CTRL, 0x7f05); + regmap_write(rt5668->regmap, RT5668_STO1_ADC_MIXER, 0x686c); + regmap_write(rt5668->regmap, RT5668_CAL_REC, 0x0d0d); + regmap_write(rt5668->regmap, RT5668_HP_CALIB_CTRL_9, 0x000f); + regmap_write(rt5668->regmap, RT5668_PWR_DIG_1, 0x8d01); + regmap_write(rt5668->regmap, RT5668_HP_CALIB_CTRL_2, 0x0321); + regmap_write(rt5668->regmap, RT5668_HP_LOGIC_CTRL_2, 0x0004); + regmap_write(rt5668->regmap, RT5668_HP_CALIB_CTRL_1, 0x7c00); + regmap_write(rt5668->regmap, RT5668_HP_CALIB_CTRL_3, 0x06a1); + regmap_write(rt5668->regmap, RT5668_A_DAC1_MUX, 0x0311); + regmap_write(rt5668->regmap, RT5668_RESET_HPF_CTRL, 0x0000); + regmap_write(rt5668->regmap, RT5668_ADC_STO1_HP_CTRL_1, 0x3320); + + regmap_write(rt5668->regmap, RT5668_HP_CALIB_CTRL_1, 0xfc00); + + for (count = 0; count < 60; count++) { + regmap_read(rt5668->regmap, RT5668_HP_CALIB_STA_1, &value); + if (!(value & 0x8000)) + break; + + usleep_range(10000, 10005); + } + + if (count >= 60) + pr_err("HP Calibration Failure\n"); + + /* restore settings */ + regmap_write(rt5668->regmap, RT5668_STO1_ADC_MIXER, 0xc0c4); + regmap_write(rt5668->regmap, RT5668_PWR_DIG_1, 0x0000); + + mutex_unlock(&rt5668->calibrate_mutex); + +} + +static int rt5668_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt5668_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt5668_priv *rt5668; + int i, ret; + unsigned int val; + + rt5668 = devm_kzalloc(&i2c->dev, sizeof(struct rt5668_priv), + GFP_KERNEL); + + if (rt5668 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, rt5668); + + if (pdata) + rt5668->pdata = *pdata; + else + rt5668_parse_dt(rt5668, &i2c->dev); + + rt5668->regmap = devm_regmap_init_i2c(i2c, &rt5668_regmap); + if (IS_ERR(rt5668->regmap)) { + ret = PTR_ERR(rt5668->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + for (i = 0; i < ARRAY_SIZE(rt5668->supplies); i++) + rt5668->supplies[i].supply = rt5668_supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c->dev, ARRAY_SIZE(rt5668->supplies), + rt5668->supplies); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(rt5668->supplies), + rt5668->supplies); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + + if (gpio_is_valid(rt5668->pdata.ldo1_en)) { + if (devm_gpio_request_one(&i2c->dev, rt5668->pdata.ldo1_en, + GPIOF_OUT_INIT_HIGH, "rt5668")) + dev_err(&i2c->dev, "Fail gpio_request gpio_ldo\n"); + } + + /* Sleep for 300 ms miniumum */ + usleep_range(300000, 350000); + + regmap_write(rt5668->regmap, RT5668_I2C_MODE, 0x1); + usleep_range(10000, 15000); + + regmap_read(rt5668->regmap, RT5668_DEVICE_ID, &val); + if (val != DEVICE_ID) { + pr_err("Device with ID register %x is not rt5668\n", val); + return -ENODEV; + } + + rt5668_reset(rt5668->regmap); + + rt5668_calibrate(rt5668); + + regmap_write(rt5668->regmap, RT5668_DEPOP_1, 0x0000); + + /* DMIC pin*/ + if (rt5668->pdata.dmic1_data_pin != RT5668_DMIC1_NULL) { + switch (rt5668->pdata.dmic1_data_pin) { + case RT5668_DMIC1_DATA_GPIO2: /* share with LRCK2 */ + regmap_update_bits(rt5668->regmap, RT5668_DMIC_CTRL_1, + RT5668_DMIC_1_DP_MASK, RT5668_DMIC_1_DP_GPIO2); + regmap_update_bits(rt5668->regmap, RT5668_GPIO_CTRL_1, + RT5668_GP2_PIN_MASK, RT5668_GP2_PIN_DMIC_SDA); + break; + + case RT5668_DMIC1_DATA_GPIO5: /* share with DACDAT1 */ + regmap_update_bits(rt5668->regmap, RT5668_DMIC_CTRL_1, + RT5668_DMIC_1_DP_MASK, RT5668_DMIC_1_DP_GPIO5); + regmap_update_bits(rt5668->regmap, RT5668_GPIO_CTRL_1, + RT5668_GP5_PIN_MASK, RT5668_GP5_PIN_DMIC_SDA); + break; + + default: + dev_dbg(&i2c->dev, "invalid DMIC_DAT pin\n"); + break; + } + + switch (rt5668->pdata.dmic1_clk_pin) { + case RT5668_DMIC1_CLK_GPIO1: /* share with IRQ */ + regmap_update_bits(rt5668->regmap, RT5668_GPIO_CTRL_1, + RT5668_GP1_PIN_MASK, RT5668_GP1_PIN_DMIC_CLK); + break; + + case RT5668_DMIC1_CLK_GPIO3: /* share with BCLK2 */ + regmap_update_bits(rt5668->regmap, RT5668_GPIO_CTRL_1, + RT5668_GP3_PIN_MASK, RT5668_GP3_PIN_DMIC_CLK); + break; + + default: + dev_dbg(&i2c->dev, "invalid DMIC_CLK pin\n"); + break; + } + } + + regmap_update_bits(rt5668->regmap, RT5668_PWR_ANLG_1, + RT5668_LDO1_DVO_MASK | RT5668_HP_DRIVER_MASK, + RT5668_LDO1_DVO_14 | RT5668_HP_DRIVER_5X); + regmap_write(rt5668->regmap, RT5668_MICBIAS_2, 0x0380); + regmap_update_bits(rt5668->regmap, RT5668_GPIO_CTRL_1, + RT5668_GP4_PIN_MASK | RT5668_GP5_PIN_MASK, + RT5668_GP4_PIN_ADCDAT1 | RT5668_GP5_PIN_DACDAT1); + regmap_write(rt5668->regmap, RT5668_TEST_MODE_CTRL_1, 0x0000); + + INIT_DELAYED_WORK(&rt5668->jack_detect_work, + rt5668_jack_detect_handler); + INIT_DELAYED_WORK(&rt5668->jd_check_work, + rt5668_jd_check_handler); + + mutex_init(&rt5668->calibrate_mutex); + + if (i2c->irq) { + ret = devm_request_threaded_irq(&i2c->dev, i2c->irq, NULL, + rt5668_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING + | IRQF_ONESHOT, "rt5668", rt5668); + if (ret) + dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); + + } + + return snd_soc_register_component(&i2c->dev, &soc_component_dev_rt5668, + rt5668_dai, ARRAY_SIZE(rt5668_dai)); +} + +static int rt5668_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_component(&i2c->dev); + + return 0; +} + +static void rt5668_i2c_shutdown(struct i2c_client *client) +{ + struct rt5668_priv *rt5668 = i2c_get_clientdata(client); + + rt5668_reset(rt5668->regmap); +} + +#ifdef CONFIG_OF +static const struct of_device_id rt5668_of_match[] = { + {.compatible = "realtek,rt5668b"}, + {}, +}; +MODULE_DEVICE_TABLE(of, rt5668_of_match); +#endif + +#ifdef CONFIG_ACPI +static const struct acpi_device_id rt5668_acpi_match[] = { + {"10EC5668", 0,}, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt5668_acpi_match); +#endif + +static struct i2c_driver rt5668_i2c_driver = { + .driver = { + .name = "rt5668b", + .of_match_table = of_match_ptr(rt5668_of_match), + .acpi_match_table = ACPI_PTR(rt5668_acpi_match), + }, + .probe = rt5668_i2c_probe, + .remove = rt5668_i2c_remove, + .shutdown = rt5668_i2c_shutdown, + .id_table = rt5668_i2c_id, +}; +module_i2c_driver(rt5668_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT5668B driver"); +MODULE_AUTHOR("Bard Liao <bardliao(a)realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt5668.h b/sound/soc/codecs/rt5668.h new file mode 100644 index 000000000000..3e7bcfd569ec --- /dev/null +++ b/sound/soc/codecs/rt5668.h @@ -0,0 +1,1318 @@ +/* + * rt5668.h -- RT5668/RT5658 ALSA SoC audio driver + * + * Copyright 2018 Realtek Microelectronics + * Author: Bard Liao <bardliao(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __RT5668_H__ +#define __RT5668_H__ + +#include <sound/rt5668.h> + +#define DEVICE_ID 0x6530 + +/* Info */ +#define RT5668_RESET 0x0000 +#define RT5668_VERSION_ID 0x00fd +#define RT5668_VENDOR_ID 0x00fe +#define RT5668_DEVICE_ID 0x00ff +/* I/O - Output */ +#define RT5668_HP_CTRL_1 0x0002 +#define RT5668_HP_CTRL_2 0x0003 +#define RT5668_HPL_GAIN 0x0005 +#define RT5668_HPR_GAIN 0x0006 + +#define RT5668_I2C_CTRL 0x0008 + +/* I/O - Input */ +#define RT5668_CBJ_BST_CTRL 0x000b +#define RT5668_CBJ_CTRL_1 0x0010 +#define RT5668_CBJ_CTRL_2 0x0011 +#define RT5668_CBJ_CTRL_3 0x0012 +#define RT5668_CBJ_CTRL_4 0x0013 +#define RT5668_CBJ_CTRL_5 0x0014 +#define RT5668_CBJ_CTRL_6 0x0015 +#define RT5668_CBJ_CTRL_7 0x0016 +/* I/O - ADC/DAC/DMIC */ +#define RT5668_DAC1_DIG_VOL 0x0019 +#define RT5668_STO1_ADC_DIG_VOL 0x001c +#define RT5668_STO1_ADC_BOOST 0x001f +#define RT5668_HP_IMP_GAIN_1 0x0022 +#define RT5668_HP_IMP_GAIN_2 0x0023 +/* Mixer - D-D */ +#define RT5668_SIDETONE_CTRL 0x0024 +#define RT5668_STO1_ADC_MIXER 0x0026 +#define RT5668_AD_DA_MIXER 0x0029 +#define RT5668_STO1_DAC_MIXER 0x002a +#define RT5668_A_DAC1_MUX 0x002b +#define RT5668_DIG_INF2_DATA 0x0030 +/* Mixer - ADC */ +#define RT5668_REC_MIXER 0x003c +#define RT5668_CAL_REC 0x0044 +#define RT5668_ALC_BACK_GAIN 0x0049 +/* Power */ +#define RT5668_PWR_DIG_1 0x0061 +#define RT5668_PWR_DIG_2 0x0062 +#define RT5668_PWR_ANLG_1 0x0063 +#define RT5668_PWR_ANLG_2 0x0064 +#define RT5668_PWR_ANLG_3 0x0065 +#define RT5668_PWR_MIXER 0x0066 +#define RT5668_PWR_VOL 0x0067 +/* Clock Detect */ +#define RT5668_CLK_DET 0x006b +/* Filter Auto Reset */ +#define RT5668_RESET_LPF_CTRL 0x006c +#define RT5668_RESET_HPF_CTRL 0x006d +/* DMIC */ +#define RT5668_DMIC_CTRL_1 0x006e +/* Format - ADC/DAC */ +#define RT5668_I2S1_SDP 0x0070 +#define RT5668_I2S2_SDP 0x0071 +#define RT5668_ADDA_CLK_1 0x0073 +#define RT5668_ADDA_CLK_2 0x0074 +#define RT5668_I2S1_F_DIV_CTRL_1 0x0075 +#define RT5668_I2S1_F_DIV_CTRL_2 0x0076 +/* Format - TDM Control */ +#define RT5668_TDM_CTRL 0x0079 +#define RT5668_TDM_ADDA_CTRL_1 0x007a +#define RT5668_TDM_ADDA_CTRL_2 0x007b +#define RT5668_DATA_SEL_CTRL_1 0x007c +#define RT5668_TDM_TCON_CTRL 0x007e +/* Function - Analog */ +#define RT5668_GLB_CLK 0x0080 +#define RT5668_PLL_CTRL_1 0x0081 +#define RT5668_PLL_CTRL_2 0x0082 +#define RT5668_PLL_TRACK_1 0x0083 +#define RT5668_PLL_TRACK_2 0x0084 +#define RT5668_PLL_TRACK_3 0x0085 +#define RT5668_PLL_TRACK_4 0x0086 +#define RT5668_PLL_TRACK_5 0x0087 +#define RT5668_PLL_TRACK_6 0x0088 +#define RT5668_PLL_TRACK_11 0x008c +#define RT5668_SDW_REF_CLK 0x008d +#define RT5668_DEPOP_1 0x008e +#define RT5668_DEPOP_2 0x008f +#define RT5668_HP_CHARGE_PUMP_1 0x0091 +#define RT5668_HP_CHARGE_PUMP_2 0x0092 +#define RT5668_MICBIAS_1 0x0093 +#define RT5668_MICBIAS_2 0x0094 +#define RT5668_PLL_TRACK_12 0x0098 +#define RT5668_PLL_TRACK_14 0x009a +#define RT5668_PLL2_CTRL_1 0x009b +#define RT5668_PLL2_CTRL_2 0x009c +#define RT5668_PLL2_CTRL_3 0x009d +#define RT5668_PLL2_CTRL_4 0x009e +#define RT5668_RC_CLK_CTRL 0x009f +#define RT5668_I2S_M_CLK_CTRL_1 0x00a0 +#define RT5668_I2S2_F_DIV_CTRL_1 0x00a3 +#define RT5668_I2S2_F_DIV_CTRL_2 0x00a4 +/* Function - Digital */ +#define RT5668_EQ_CTRL_1 0x00ae +#define RT5668_EQ_CTRL_2 0x00af +#define RT5668_IRQ_CTRL_1 0x00b6 +#define RT5668_IRQ_CTRL_2 0x00b7 +#define RT5668_IRQ_CTRL_3 0x00b8 +#define RT5668_IRQ_CTRL_4 0x00b9 +#define RT5668_INT_ST_1 0x00be +#define RT5668_GPIO_CTRL_1 0x00c0 +#define RT5668_GPIO_CTRL_2 0x00c1 +#define RT5668_GPIO_CTRL_3 0x00c2 +#define RT5668_HP_AMP_DET_CTRL_1 0x00d0 +#define RT5668_HP_AMP_DET_CTRL_2 0x00d1 +#define RT5668_MID_HP_AMP_DET 0x00d2 +#define RT5668_LOW_HP_AMP_DET 0x00d3 +#define RT5668_DELAY_BUF_CTRL 0x00d4 +#define RT5668_SV_ZCD_1 0x00d9 +#define RT5668_SV_ZCD_2 0x00da +#define RT5668_IL_CMD_1 0x00db +#define RT5668_IL_CMD_2 0x00dc +#define RT5668_IL_CMD_3 0x00dd +#define RT5668_IL_CMD_4 0x00de +#define RT5668_IL_CMD_5 0x00df +#define RT5668_IL_CMD_6 0x00e0 +#define RT5668_4BTN_IL_CMD_1 0x00e2 +#define RT5668_4BTN_IL_CMD_2 0x00e3 +#define RT5668_4BTN_IL_CMD_3 0x00e4 +#define RT5668_4BTN_IL_CMD_4 0x00e5 +#define RT5668_4BTN_IL_CMD_5 0x00e6 +#define RT5668_4BTN_IL_CMD_6 0x00e7 +#define RT5668_4BTN_IL_CMD_7 0x00e8 + +#define RT5668_ADC_STO1_HP_CTRL_1 0x00ea +#define RT5668_ADC_STO1_HP_CTRL_2 0x00eb +#define RT5668_AJD1_CTRL 0x00f0 +#define RT5668_JD1_THD 0x00f1 +#define RT5668_JD2_THD 0x00f2 +#define RT5668_JD_CTRL_1 0x00f6 +/* General Control */ +#define RT5668_DUMMY_1 0x00fa +#define RT5668_DUMMY_2 0x00fb +#define RT5668_DUMMY_3 0x00fc + +#define RT5668_DAC_ADC_DIG_VOL1 0x0100 +#define RT5668_BIAS_CUR_CTRL_2 0x010b +#define RT5668_BIAS_CUR_CTRL_3 0x010c +#define RT5668_BIAS_CUR_CTRL_4 0x010d +#define RT5668_BIAS_CUR_CTRL_5 0x010e +#define RT5668_BIAS_CUR_CTRL_6 0x010f +#define RT5668_BIAS_CUR_CTRL_7 0x0110 +#define RT5668_BIAS_CUR_CTRL_8 0x0111 +#define RT5668_BIAS_CUR_CTRL_9 0x0112 +#define RT5668_BIAS_CUR_CTRL_10 0x0113 +#define RT5668_VREF_REC_OP_FB_CAP_CTRL 0x0117 +#define RT5668_CHARGE_PUMP_1 0x0125 +#define RT5668_DIG_IN_CTRL_1 0x0132 +#define RT5668_PAD_DRIVING_CTRL 0x0136 +#define RT5668_SOFT_RAMP_DEPOP 0x0138 +#define RT5668_CHOP_DAC 0x013a +#define RT5668_CHOP_ADC 0x013b +#define RT5668_CALIB_ADC_CTRL 0x013c +#define RT5668_VOL_TEST 0x013f +#define RT5668_SPKVDD_DET_STA 0x0142 +#define RT5668_TEST_MODE_CTRL_1 0x0145 +#define RT5668_TEST_MODE_CTRL_2 0x0146 +#define RT5668_TEST_MODE_CTRL_3 0x0147 +#define RT5668_TEST_MODE_CTRL_4 0x0148 +#define RT5668_TEST_MODE_CTRL_5 0x0149 +#define RT5668_PLL1_INTERNAL 0x0150 +#define RT5668_PLL2_INTERNAL 0x0151 +#define RT5668_STO_NG2_CTRL_1 0x0160 +#define RT5668_STO_NG2_CTRL_2 0x0161 +#define RT5668_STO_NG2_CTRL_3 0x0162 +#define RT5668_STO_NG2_CTRL_4 0x0163 +#define RT5668_STO_NG2_CTRL_5 0x0164 +#define RT5668_STO_NG2_CTRL_6 0x0165 +#define RT5668_STO_NG2_CTRL_7 0x0166 +#define RT5668_STO_NG2_CTRL_8 0x0167 +#define RT5668_STO_NG2_CTRL_9 0x0168 +#define RT5668_STO_NG2_CTRL_10 0x0169 +#define RT5668_STO1_DAC_SIL_DET 0x0190 +#define RT5668_SIL_PSV_CTRL1 0x0194 +#define RT5668_SIL_PSV_CTRL2 0x0195 +#define RT5668_SIL_PSV_CTRL3 0x0197 +#define RT5668_SIL_PSV_CTRL4 0x0198 +#define RT5668_SIL_PSV_CTRL5 0x0199 +#define RT5668_HP_IMP_SENS_CTRL_01 0x01af +#define RT5668_HP_IMP_SENS_CTRL_02 0x01b0 +#define RT5668_HP_IMP_SENS_CTRL_03 0x01b1 +#define RT5668_HP_IMP_SENS_CTRL_04 0x01b2 +#define RT5668_HP_IMP_SENS_CTRL_05 0x01b3 +#define RT5668_HP_IMP_SENS_CTRL_06 0x01b4 +#define RT5668_HP_IMP_SENS_CTRL_07 0x01b5 +#define RT5668_HP_IMP_SENS_CTRL_08 0x01b6 +#define RT5668_HP_IMP_SENS_CTRL_09 0x01b7 +#define RT5668_HP_IMP_SENS_CTRL_10 0x01b8 +#define RT5668_HP_IMP_SENS_CTRL_11 0x01b9 +#define RT5668_HP_IMP_SENS_CTRL_12 0x01ba +#define RT5668_HP_IMP_SENS_CTRL_13 0x01bb +#define RT5668_HP_IMP_SENS_CTRL_14 0x01bc +#define RT5668_HP_IMP_SENS_CTRL_15 0x01bd +#define RT5668_HP_IMP_SENS_CTRL_16 0x01be +#define RT5668_HP_IMP_SENS_CTRL_17 0x01bf +#define RT5668_HP_IMP_SENS_CTRL_18 0x01c0 +#define RT5668_HP_IMP_SENS_CTRL_19 0x01c1 +#define RT5668_HP_IMP_SENS_CTRL_20 0x01c2 +#define RT5668_HP_IMP_SENS_CTRL_21 0x01c3 +#define RT5668_HP_IMP_SENS_CTRL_22 0x01c4 +#define RT5668_HP_IMP_SENS_CTRL_23 0x01c5 +#define RT5668_HP_IMP_SENS_CTRL_24 0x01c6 +#define RT5668_HP_IMP_SENS_CTRL_25 0x01c7 +#define RT5668_HP_IMP_SENS_CTRL_26 0x01c8 +#define RT5668_HP_IMP_SENS_CTRL_27 0x01c9 +#define RT5668_HP_IMP_SENS_CTRL_28 0x01ca +#define RT5668_HP_IMP_SENS_CTRL_29 0x01cb +#define RT5668_HP_IMP_SENS_CTRL_30 0x01cc +#define RT5668_HP_IMP_SENS_CTRL_31 0x01cd +#define RT5668_HP_IMP_SENS_CTRL_32 0x01ce +#define RT5668_HP_IMP_SENS_CTRL_33 0x01cf +#define RT5668_HP_IMP_SENS_CTRL_34 0x01d0 +#define RT5668_HP_IMP_SENS_CTRL_35 0x01d1 +#define RT5668_HP_IMP_SENS_CTRL_36 0x01d2 +#define RT5668_HP_IMP_SENS_CTRL_37 0x01d3 +#define RT5668_HP_IMP_SENS_CTRL_38 0x01d4 +#define RT5668_HP_IMP_SENS_CTRL_39 0x01d5 +#define RT5668_HP_IMP_SENS_CTRL_40 0x01d6 +#define RT5668_HP_IMP_SENS_CTRL_41 0x01d7 +#define RT5668_HP_IMP_SENS_CTRL_42 0x01d8 +#define RT5668_HP_IMP_SENS_CTRL_43 0x01d9 +#define RT5668_HP_LOGIC_CTRL_1 0x01da +#define RT5668_HP_LOGIC_CTRL_2 0x01db +#define RT5668_HP_LOGIC_CTRL_3 0x01dc +#define RT5668_HP_CALIB_CTRL_1 0x01de +#define RT5668_HP_CALIB_CTRL_2 0x01df +#define RT5668_HP_CALIB_CTRL_3 0x01e0 +#define RT5668_HP_CALIB_CTRL_4 0x01e1 +#define RT5668_HP_CALIB_CTRL_5 0x01e2 +#define RT5668_HP_CALIB_CTRL_6 0x01e3 +#define RT5668_HP_CALIB_CTRL_7 0x01e4 +#define RT5668_HP_CALIB_CTRL_9 0x01e6 +#define RT5668_HP_CALIB_CTRL_10 0x01e7 +#define RT5668_HP_CALIB_CTRL_11 0x01e8 +#define RT5668_HP_CALIB_STA_1 0x01ea +#define RT5668_HP_CALIB_STA_2 0x01eb +#define RT5668_HP_CALIB_STA_3 0x01ec +#define RT5668_HP_CALIB_STA_4 0x01ed +#define RT5668_HP_CALIB_STA_5 0x01ee +#define RT5668_HP_CALIB_STA_6 0x01ef +#define RT5668_HP_CALIB_STA_7 0x01f0 +#define RT5668_HP_CALIB_STA_8 0x01f1 +#define RT5668_HP_CALIB_STA_9 0x01f2 +#define RT5668_HP_CALIB_STA_10 0x01f3 +#define RT5668_HP_CALIB_STA_11 0x01f4 +#define RT5668_SAR_IL_CMD_1 0x0210 +#define RT5668_SAR_IL_CMD_2 0x0211 +#define RT5668_SAR_IL_CMD_3 0x0212 +#define RT5668_SAR_IL_CMD_4 0x0213 +#define RT5668_SAR_IL_CMD_5 0x0214 +#define RT5668_SAR_IL_CMD_6 0x0215 +#define RT5668_SAR_IL_CMD_7 0x0216 +#define RT5668_SAR_IL_CMD_8 0x0217 +#define RT5668_SAR_IL_CMD_9 0x0218 +#define RT5668_SAR_IL_CMD_10 0x0219 +#define RT5668_SAR_IL_CMD_11 0x021a +#define RT5668_SAR_IL_CMD_12 0x021b +#define RT5668_SAR_IL_CMD_13 0x021c +#define RT5668_EFUSE_CTRL_1 0x0250 +#define RT5668_EFUSE_CTRL_2 0x0251 +#define RT5668_EFUSE_CTRL_3 0x0252 +#define RT5668_EFUSE_CTRL_4 0x0253 +#define RT5668_EFUSE_CTRL_5 0x0254 +#define RT5668_EFUSE_CTRL_6 0x0255 +#define RT5668_EFUSE_CTRL_7 0x0256 +#define RT5668_EFUSE_CTRL_8 0x0257 +#define RT5668_EFUSE_CTRL_9 0x0258 +#define RT5668_EFUSE_CTRL_10 0x0259 +#define RT5668_EFUSE_CTRL_11 0x025a +#define RT5668_JD_TOP_VC_VTRL 0x0270 +#define RT5668_DRC1_CTRL_0 0x02ff +#define RT5668_DRC1_CTRL_1 0x0300 +#define RT5668_DRC1_CTRL_2 0x0301 +#define RT5668_DRC1_CTRL_3 0x0302 +#define RT5668_DRC1_CTRL_4 0x0303 +#define RT5668_DRC1_CTRL_5 0x0304 +#define RT5668_DRC1_CTRL_6 0x0305 +#define RT5668_DRC1_HARD_LMT_CTRL_1 0x0306 +#define RT5668_DRC1_HARD_LMT_CTRL_2 0x0307 +#define RT5668_DRC1_PRIV_1 0x0310 +#define RT5668_DRC1_PRIV_2 0x0311 +#define RT5668_DRC1_PRIV_3 0x0312 +#define RT5668_DRC1_PRIV_4 0x0313 +#define RT5668_DRC1_PRIV_5 0x0314 +#define RT5668_DRC1_PRIV_6 0x0315 +#define RT5668_DRC1_PRIV_7 0x0316 +#define RT5668_DRC1_PRIV_8 0x0317 +#define RT5668_EQ_AUTO_RCV_CTRL1 0x03c0 +#define RT5668_EQ_AUTO_RCV_CTRL2 0x03c1 +#define RT5668_EQ_AUTO_RCV_CTRL3 0x03c2 +#define RT5668_EQ_AUTO_RCV_CTRL4 0x03c3 +#define RT5668_EQ_AUTO_RCV_CTRL5 0x03c4 +#define RT5668_EQ_AUTO_RCV_CTRL6 0x03c5 +#define RT5668_EQ_AUTO_RCV_CTRL7 0x03c6 +#define RT5668_EQ_AUTO_RCV_CTRL8 0x03c7 +#define RT5668_EQ_AUTO_RCV_CTRL9 0x03c8 +#define RT5668_EQ_AUTO_RCV_CTRL10 0x03c9 +#define RT5668_EQ_AUTO_RCV_CTRL11 0x03ca +#define RT5668_EQ_AUTO_RCV_CTRL12 0x03cb +#define RT5668_EQ_AUTO_RCV_CTRL13 0x03cc +#define RT5668_ADC_L_EQ_LPF1_A1 0x03d0 +#define RT5668_R_EQ_LPF1_A1 0x03d1 +#define RT5668_L_EQ_LPF1_H0 0x03d2 +#define RT5668_R_EQ_LPF1_H0 0x03d3 +#define RT5668_L_EQ_BPF1_A1 0x03d4 +#define RT5668_R_EQ_BPF1_A1 0x03d5 +#define RT5668_L_EQ_BPF1_A2 0x03d6 +#define RT5668_R_EQ_BPF1_A2 0x03d7 +#define RT5668_L_EQ_BPF1_H0 0x03d8 +#define RT5668_R_EQ_BPF1_H0 0x03d9 +#define RT5668_L_EQ_BPF2_A1 0x03da +#define RT5668_R_EQ_BPF2_A1 0x03db +#define RT5668_L_EQ_BPF2_A2 0x03dc +#define RT5668_R_EQ_BPF2_A2 0x03dd +#define RT5668_L_EQ_BPF2_H0 0x03de +#define RT5668_R_EQ_BPF2_H0 0x03df +#define RT5668_L_EQ_BPF3_A1 0x03e0 +#define RT5668_R_EQ_BPF3_A1 0x03e1 +#define RT5668_L_EQ_BPF3_A2 0x03e2 +#define RT5668_R_EQ_BPF3_A2 0x03e3 +#define RT5668_L_EQ_BPF3_H0 0x03e4 +#define RT5668_R_EQ_BPF3_H0 0x03e5 +#define RT5668_L_EQ_BPF4_A1 0x03e6 +#define RT5668_R_EQ_BPF4_A1 0x03e7 +#define RT5668_L_EQ_BPF4_A2 0x03e8 +#define RT5668_R_EQ_BPF4_A2 0x03e9 +#define RT5668_L_EQ_BPF4_H0 0x03ea +#define RT5668_R_EQ_BPF4_H0 0x03eb +#define RT5668_L_EQ_HPF1_A1 0x03ec +#define RT5668_R_EQ_HPF1_A1 0x03ed +#define RT5668_L_EQ_HPF1_H0 0x03ee +#define RT5668_R_EQ_HPF1_H0 0x03ef +#define RT5668_L_EQ_PRE_VOL 0x03f0 +#define RT5668_R_EQ_PRE_VOL 0x03f1 +#define RT5668_L_EQ_POST_VOL 0x03f2 +#define RT5668_R_EQ_POST_VOL 0x03f3 +#define RT5668_I2C_MODE 0xffff + + +/* global definition */ +#define RT5668_L_MUTE (0x1 << 15) +#define RT5668_L_MUTE_SFT 15 +#define RT5668_VOL_L_MUTE (0x1 << 14) +#define RT5668_VOL_L_SFT 14 +#define RT5668_R_MUTE (0x1 << 7) +#define RT5668_R_MUTE_SFT 7 +#define RT5668_VOL_R_MUTE (0x1 << 6) +#define RT5668_VOL_R_SFT 6 +#define RT5668_L_VOL_MASK (0x3f << 8) +#define RT5668_L_VOL_SFT 8 +#define RT5668_R_VOL_MASK (0x3f) +#define RT5668_R_VOL_SFT 0 + +/*Headphone Amp L/R Analog Gain and Digital NG2 Gain Control (0x0005 0x0006)*/ +#define RT5668_G_HP (0xf << 8) +#define RT5668_G_HP_SFT 8 +#define RT5668_G_STO_DA_DMIX (0xf) +#define RT5668_G_STO_DA_SFT 0 + +/* CBJ Control (0x000b) */ +#define RT5668_BST_CBJ_MASK (0xf << 8) +#define RT5668_BST_CBJ_SFT 8 + +/* Embeeded Jack and Type Detection Control 1 (0x0010) */ +#define RT5668_EMB_JD_EN (0x1 << 15) +#define RT5668_EMB_JD_EN_SFT 15 +#define RT5668_EMB_JD_RST (0x1 << 14) +#define RT5668_JD_MODE (0x1 << 13) +#define RT5668_JD_MODE_SFT 13 +#define RT5668_DET_TYPE (0x1 << 12) +#define RT5668_DET_TYPE_SFT 12 +#define RT5668_POLA_EXT_JD_MASK (0x1 << 11) +#define RT5668_POLA_EXT_JD_LOW (0x1 << 11) +#define RT5668_POLA_EXT_JD_HIGH (0x0 << 11) +#define RT5668_EXT_JD_DIG (0x1 << 9) +#define RT5668_POL_FAST_OFF_MASK (0x1 << 8) +#define RT5668_POL_FAST_OFF_HIGH (0x1 << 8) +#define RT5668_POL_FAST_OFF_LOW (0x0 << 8) +#define RT5668_FAST_OFF_MASK (0x1 << 7) +#define RT5668_FAST_OFF_EN (0x1 << 7) +#define RT5668_FAST_OFF_DIS (0x0 << 7) +#define RT5668_VREF_POW_MASK (0x1 << 6) +#define RT5668_VREF_POW_FSM (0x0 << 6) +#define RT5668_VREF_POW_REG (0x1 << 6) +#define RT5668_MB1_PATH_MASK (0x1 << 5) +#define RT5668_CTRL_MB1_REG (0x1 << 5) +#define RT5668_CTRL_MB1_FSM (0x0 << 5) +#define RT5668_MB2_PATH_MASK (0x1 << 4) +#define RT5668_CTRL_MB2_REG (0x1 << 4) +#define RT5668_CTRL_MB2_FSM (0x0 << 4) +#define RT5668_TRIG_JD_MASK (0x1 << 3) +#define RT5668_TRIG_JD_HIGH (0x1 << 3) +#define RT5668_TRIG_JD_LOW (0x0 << 3) +#define RT5668_MIC_CAP_MASK (0x1 << 1) +#define RT5668_MIC_CAP_HS (0x1 << 1) +#define RT5668_MIC_CAP_HP (0x0 << 1) +#define RT5668_MIC_CAP_SRC_MASK (0x1) +#define RT5668_MIC_CAP_SRC_REG (0x1) +#define RT5668_MIC_CAP_SRC_ANA (0x0) + +/* Embeeded Jack and Type Detection Control 2 (0x0011) */ +#define RT5668_EXT_JD_SRC (0x7 << 4) +#define RT5668_EXT_JD_SRC_SFT 4 +#define RT5668_EXT_JD_SRC_GPIO_JD1 (0x0 << 4) +#define RT5668_EXT_JD_SRC_GPIO_JD2 (0x1 << 4) +#define RT5668_EXT_JD_SRC_JDH (0x2 << 4) +#define RT5668_EXT_JD_SRC_JDL (0x3 << 4) +#define RT5668_EXT_JD_SRC_MANUAL (0x4 << 4) +#define RT5668_JACK_TYPE_MASK (0x3) + +/* Combo Jack and Type Detection Control 3 (0x0012) */ +#define RT5668_CBJ_IN_BUF_EN (0x1 << 7) + +/* Combo Jack and Type Detection Control 4 (0x0013) */ +#define RT5668_SEL_SHT_MID_TON_MASK (0x3 << 12) +#define RT5668_SEL_SHT_MID_TON_2 (0x0 << 12) +#define RT5668_SEL_SHT_MID_TON_3 (0x1 << 12) +#define RT5668_CBJ_JD_TEST_MASK (0x1 << 6) +#define RT5668_CBJ_JD_TEST_NORM (0x0 << 6) +#define RT5668_CBJ_JD_TEST_MODE (0x1 << 6) + +/* DAC1 Digital Volume (0x0019) */ +#define RT5668_DAC_L1_VOL_MASK (0xff << 8) +#define RT5668_DAC_L1_VOL_SFT 8 +#define RT5668_DAC_R1_VOL_MASK (0xff) +#define RT5668_DAC_R1_VOL_SFT 0 + +/* ADC Digital Volume Control (0x001c) */ +#define RT5668_ADC_L_VOL_MASK (0x7f << 8) +#define RT5668_ADC_L_VOL_SFT 8 +#define RT5668_ADC_R_VOL_MASK (0x7f) +#define RT5668_ADC_R_VOL_SFT 0 + +/* Stereo1 ADC Boost Gain Control (0x001f) */ +#define RT5668_STO1_ADC_L_BST_MASK (0x3 << 14) +#define RT5668_STO1_ADC_L_BST_SFT 14 +#define RT5668_STO1_ADC_R_BST_MASK (0x3 << 12) +#define RT5668_STO1_ADC_R_BST_SFT 12 + +/* Sidetone Control (0x0024) */ +#define RT5668_ST_SRC_SEL (0x1 << 8) +#define RT5668_ST_SRC_SFT 8 +#define RT5668_ST_EN_MASK (0x1 << 6) +#define RT5668_ST_DIS (0x0 << 6) +#define RT5668_ST_EN (0x1 << 6) +#define RT5668_ST_EN_SFT 6 + +/* Stereo1 ADC Mixer Control (0x0026) */ +#define RT5668_M_STO1_ADC_L1 (0x1 << 15) +#define RT5668_M_STO1_ADC_L1_SFT 15 +#define RT5668_M_STO1_ADC_L2 (0x1 << 14) +#define RT5668_M_STO1_ADC_L2_SFT 14 +#define RT5668_STO1_ADC1L_SRC_MASK (0x1 << 13) +#define RT5668_STO1_ADC1L_SRC_SFT 13 +#define RT5668_STO1_ADC1_SRC_ADC (0x1 << 13) +#define RT5668_STO1_ADC1_SRC_DACMIX (0x0 << 13) +#define RT5668_STO1_ADC2L_SRC_MASK (0x1 << 12) +#define RT5668_STO1_ADC2L_SRC_SFT 12 +#define RT5668_STO1_ADCL_SRC_MASK (0x3 << 10) +#define RT5668_STO1_ADCL_SRC_SFT 10 +#define RT5668_STO1_DD_L_SRC_MASK (0x1 << 9) +#define RT5668_STO1_DD_L_SRC_SFT 9 +#define RT5668_STO1_DMIC_SRC_MASK (0x1 << 8) +#define RT5668_STO1_DMIC_SRC_SFT 8 +#define RT5668_STO1_DMIC_SRC_DMIC2 (0x1 << 8) +#define RT5668_STO1_DMIC_SRC_DMIC1 (0x0 << 8) +#define RT5668_M_STO1_ADC_R1 (0x1 << 7) +#define RT5668_M_STO1_ADC_R1_SFT 7 +#define RT5668_M_STO1_ADC_R2 (0x1 << 6) +#define RT5668_M_STO1_ADC_R2_SFT 6 +#define RT5668_STO1_ADC1R_SRC_MASK (0x1 << 5) +#define RT5668_STO1_ADC1R_SRC_SFT 5 +#define RT5668_STO1_ADC2R_SRC_MASK (0x1 << 4) +#define RT5668_STO1_ADC2R_SRC_SFT 4 +#define RT5668_STO1_ADCR_SRC_MASK (0x3 << 2) +#define RT5668_STO1_ADCR_SRC_SFT 2 + +/* ADC Mixer to DAC Mixer Control (0x0029) */ +#define RT5668_M_ADCMIX_L (0x1 << 15) +#define RT5668_M_ADCMIX_L_SFT 15 +#define RT5668_M_DAC1_L (0x1 << 14) +#define RT5668_M_DAC1_L_SFT 14 +#define RT5668_DAC1_R_SEL_MASK (0x1 << 10) +#define RT5668_DAC1_R_SEL_SFT 10 +#define RT5668_DAC1_L_SEL_MASK (0x1 << 8) +#define RT5668_DAC1_L_SEL_SFT 8 +#define RT5668_M_ADCMIX_R (0x1 << 7) +#define RT5668_M_ADCMIX_R_SFT 7 +#define RT5668_M_DAC1_R (0x1 << 6) +#define RT5668_M_DAC1_R_SFT 6 + +/* Stereo1 DAC Mixer Control (0x002a) */ +#define RT5668_M_DAC_L1_STO_L (0x1 << 15) +#define RT5668_M_DAC_L1_STO_L_SFT 15 +#define RT5668_G_DAC_L1_STO_L_MASK (0x1 << 14) +#define RT5668_G_DAC_L1_STO_L_SFT 14 +#define RT5668_M_DAC_R1_STO_L (0x1 << 13) +#define RT5668_M_DAC_R1_STO_L_SFT 13 +#define RT5668_G_DAC_R1_STO_L_MASK (0x1 << 12) +#define RT5668_G_DAC_R1_STO_L_SFT 12 +#define RT5668_M_DAC_L1_STO_R (0x1 << 7) +#define RT5668_M_DAC_L1_STO_R_SFT 7 +#define RT5668_G_DAC_L1_STO_R_MASK (0x1 << 6) +#define RT5668_G_DAC_L1_STO_R_SFT 6 +#define RT5668_M_DAC_R1_STO_R (0x1 << 5) +#define RT5668_M_DAC_R1_STO_R_SFT 5 +#define RT5668_G_DAC_R1_STO_R_MASK (0x1 << 4) +#define RT5668_G_DAC_R1_STO_R_SFT 4 + +/* Analog DAC1 Input Source Control (0x002b) */ +#define RT5668_M_ST_STO_L (0x1 << 9) +#define RT5668_M_ST_STO_L_SFT 9 +#define RT5668_M_ST_STO_R (0x1 << 8) +#define RT5668_M_ST_STO_R_SFT 8 +#define RT5668_DAC_L1_SRC_MASK (0x3 << 4) +#define RT5668_A_DACL1_SFT 4 +#define RT5668_DAC_R1_SRC_MASK (0x3) +#define RT5668_A_DACR1_SFT 0 + +/* Digital Interface Data Control (0x0030) */ +#define RT5668_IF2_ADC_SEL_MASK (0x3 << 0) +#define RT5668_IF2_ADC_SEL_SFT 0 + +/* REC Left Mixer Control 2 (0x003c) */ +#define RT5668_G_CBJ_RM1_L (0x7 << 10) +#define RT5668_G_CBJ_RM1_L_SFT 10 +#define RT5668_M_CBJ_RM1_L (0x1 << 7) +#define RT5668_M_CBJ_RM1_L_SFT 7 + +/* Power Management for Digital 1 (0x0061) */ +#define RT5668_PWR_I2S1 (0x1 << 15) +#define RT5668_PWR_I2S1_BIT 15 +#define RT5668_PWR_I2S2 (0x1 << 14) +#define RT5668_PWR_I2S2_BIT 14 +#define RT5668_PWR_DAC_L1 (0x1 << 11) +#define RT5668_PWR_DAC_L1_BIT 11 +#define RT5668_PWR_DAC_R1 (0x1 << 10) +#define RT5668_PWR_DAC_R1_BIT 10 +#define RT5668_PWR_LDO (0x1 << 8) +#define RT5668_PWR_LDO_BIT 8 +#define RT5668_PWR_ADC_L1 (0x1 << 4) +#define RT5668_PWR_ADC_L1_BIT 4 +#define RT5668_PWR_ADC_R1 (0x1 << 3) +#define RT5668_PWR_ADC_R1_BIT 3 +#define RT5668_DIG_GATE_CTRL (0x1 << 0) +#define RT5668_DIG_GATE_CTRL_SFT 0 + + +/* Power Management for Digital 2 (0x0062) */ +#define RT5668_PWR_ADC_S1F (0x1 << 15) +#define RT5668_PWR_ADC_S1F_BIT 15 +#define RT5668_PWR_DAC_S1F (0x1 << 10) +#define RT5668_PWR_DAC_S1F_BIT 10 + +/* Power Management for Analog 1 (0x0063) */ +#define RT5668_PWR_VREF1 (0x1 << 15) +#define RT5668_PWR_VREF1_BIT 15 +#define RT5668_PWR_FV1 (0x1 << 14) +#define RT5668_PWR_FV1_BIT 14 +#define RT5668_PWR_VREF2 (0x1 << 13) +#define RT5668_PWR_VREF2_BIT 13 +#define RT5668_PWR_FV2 (0x1 << 12) +#define RT5668_PWR_FV2_BIT 12 +#define RT5668_LDO1_DBG_MASK (0x3 << 10) +#define RT5668_PWR_MB (0x1 << 9) +#define RT5668_PWR_MB_BIT 9 +#define RT5668_PWR_BG (0x1 << 7) +#define RT5668_PWR_BG_BIT 7 +#define RT5668_LDO1_BYPASS_MASK (0x1 << 6) +#define RT5668_LDO1_BYPASS (0x1 << 6) +#define RT5668_LDO1_NOT_BYPASS (0x0 << 6) +#define RT5668_PWR_MA_BIT 6 +#define RT5668_LDO1_DVO_MASK (0x3 << 4) +#define RT5668_LDO1_DVO_09 (0x0 << 4) +#define RT5668_LDO1_DVO_10 (0x1 << 4) +#define RT5668_LDO1_DVO_12 (0x2 << 4) +#define RT5668_LDO1_DVO_14 (0x3 << 4) +#define RT5668_HP_DRIVER_MASK (0x3 << 2) +#define RT5668_HP_DRIVER_1X (0x0 << 2) +#define RT5668_HP_DRIVER_3X (0x1 << 2) +#define RT5668_HP_DRIVER_5X (0x3 << 2) +#define RT5668_PWR_HA_L (0x1 << 1) +#define RT5668_PWR_HA_L_BIT 1 +#define RT5668_PWR_HA_R (0x1 << 0) +#define RT5668_PWR_HA_R_BIT 0 + +/* Power Management for Analog 2 (0x0064) */ +#define RT5668_PWR_MB1 (0x1 << 11) +#define RT5668_PWR_MB1_PWR_DOWN (0x0 << 11) +#define RT5668_PWR_MB1_BIT 11 +#define RT5668_PWR_MB2 (0x1 << 10) +#define RT5668_PWR_MB2_PWR_DOWN (0x0 << 10) +#define RT5668_PWR_MB2_BIT 10 +#define RT5668_PWR_JDH (0x1 << 3) +#define RT5668_PWR_JDH_BIT 3 +#define RT5668_PWR_JDL (0x1 << 2) +#define RT5668_PWR_JDL_BIT 2 +#define RT5668_PWR_RM1_L (0x1 << 1) +#define RT5668_PWR_RM1_L_BIT 1 + +/* Power Management for Analog 3 (0x0065) */ +#define RT5668_PWR_CBJ (0x1 << 9) +#define RT5668_PWR_CBJ_BIT 9 +#define RT5668_PWR_PLL (0x1 << 6) +#define RT5668_PWR_PLL_BIT 6 +#define RT5668_PWR_PLL2B (0x1 << 5) +#define RT5668_PWR_PLL2B_BIT 5 +#define RT5668_PWR_PLL2F (0x1 << 4) +#define RT5668_PWR_PLL2F_BIT 4 +#define RT5668_PWR_LDO2 (0x1 << 2) +#define RT5668_PWR_LDO2_BIT 2 +#define RT5668_PWR_DET_SPKVDD (0x1 << 1) +#define RT5668_PWR_DET_SPKVDD_BIT 1 + +/* Power Management for Mixer (0x0066) */ +#define RT5668_PWR_STO1_DAC_L (0x1 << 5) +#define RT5668_PWR_STO1_DAC_L_BIT 5 +#define RT5668_PWR_STO1_DAC_R (0x1 << 4) +#define RT5668_PWR_STO1_DAC_R_BIT 4 + +/* MCLK and System Clock Detection Control (0x006b) */ +#define RT5668_SYS_CLK_DET (0x1 << 15) +#define RT5668_SYS_CLK_DET_SFT 15 +#define RT5668_PLL1_CLK_DET (0x1 << 14) +#define RT5668_PLL1_CLK_DET_SFT 14 +#define RT5668_PLL2_CLK_DET (0x1 << 13) +#define RT5668_PLL2_CLK_DET_SFT 13 +#define RT5668_POW_CLK_DET2_SFT 8 +#define RT5668_POW_CLK_DET_SFT 0 + +/* Digital Microphone Control 1 (0x006e) */ +#define RT5668_DMIC_1_EN_MASK (0x1 << 15) +#define RT5668_DMIC_1_EN_SFT 15 +#define RT5668_DMIC_1_DIS (0x0 << 15) +#define RT5668_DMIC_1_EN (0x1 << 15) +#define RT5668_DMIC_1_DP_MASK (0x3 << 4) +#define RT5668_DMIC_1_DP_SFT 4 +#define RT5668_DMIC_1_DP_GPIO2 (0x0 << 4) +#define RT5668_DMIC_1_DP_GPIO5 (0x1 << 4) +#define RT5668_DMIC_CLK_MASK (0xf << 0) +#define RT5668_DMIC_CLK_SFT 0 + +/* I2S1 Audio Serial Data Port Control (0x0070) */ +#define RT5668_SEL_ADCDAT_MASK (0x1 << 15) +#define RT5668_SEL_ADCDAT_OUT (0x0 << 15) +#define RT5668_SEL_ADCDAT_IN (0x1 << 15) +#define RT5668_SEL_ADCDAT_SFT 15 +#define RT5668_I2S1_TX_CHL_MASK (0x7 << 12) +#define RT5668_I2S1_TX_CHL_SFT 12 +#define RT5668_I2S1_TX_CHL_16 (0x0 << 12) +#define RT5668_I2S1_TX_CHL_20 (0x1 << 12) +#define RT5668_I2S1_TX_CHL_24 (0x2 << 12) +#define RT5668_I2S1_TX_CHL_32 (0x3 << 12) +#define RT5668_I2S1_TX_CHL_8 (0x4 << 12) +#define RT5668_I2S1_RX_CHL_MASK (0x7 << 8) +#define RT5668_I2S1_RX_CHL_SFT 8 +#define RT5668_I2S1_RX_CHL_16 (0x0 << 8) +#define RT5668_I2S1_RX_CHL_20 (0x1 << 8) +#define RT5668_I2S1_RX_CHL_24 (0x2 << 8) +#define RT5668_I2S1_RX_CHL_32 (0x3 << 8) +#define RT5668_I2S1_RX_CHL_8 (0x4 << 8) +#define RT5668_I2S1_MONO_MASK (0x1 << 7) +#define RT5668_I2S1_MONO_EN (0x1 << 7) +#define RT5668_I2S1_MONO_DIS (0x0 << 7) +#define RT5668_I2S2_MONO_MASK (0x1 << 6) +#define RT5668_I2S2_MONO_EN (0x1 << 6) +#define RT5668_I2S2_MONO_DIS (0x0 << 6) +#define RT5668_I2S1_DL_MASK (0x7 << 4) +#define RT5668_I2S1_DL_SFT 4 +#define RT5668_I2S1_DL_16 (0x0 << 4) +#define RT5668_I2S1_DL_20 (0x1 << 4) +#define RT5668_I2S1_DL_24 (0x2 << 4) +#define RT5668_I2S1_DL_32 (0x3 << 4) +#define RT5668_I2S1_DL_8 (0x4 << 4) + +/* I2S1/2 Audio Serial Data Port Control (0x0070)(0x0071) */ +#define RT5668_I2S2_MS_MASK (0x1 << 15) +#define RT5668_I2S2_MS_SFT 15 +#define RT5668_I2S2_MS_M (0x0 << 15) +#define RT5668_I2S2_MS_S (0x1 << 15) +#define RT5668_I2S2_PIN_CFG_MASK (0x1 << 14) +#define RT5668_I2S2_PIN_CFG_SFT 14 +#define RT5668_I2S2_CLK_SEL_MASK (0x1 << 11) +#define RT5668_I2S2_CLK_SEL_SFT 11 +#define RT5668_I2S2_OUT_MASK (0x1 << 9) +#define RT5668_I2S2_OUT_SFT 9 +#define RT5668_I2S2_OUT_UM (0x0 << 9) +#define RT5668_I2S2_OUT_M (0x1 << 9) +#define RT5668_I2S_BP_MASK (0x1 << 8) +#define RT5668_I2S_BP_SFT 8 +#define RT5668_I2S_BP_NOR (0x0 << 8) +#define RT5668_I2S_BP_INV (0x1 << 8) +#define RT5668_I2S2_MONO_EN (0x1 << 6) +#define RT5668_I2S2_MONO_DIS (0x0 << 6) +#define RT5668_I2S2_DL_MASK (0x3 << 4) +#define RT5668_I2S2_DL_SFT 4 +#define RT5668_I2S2_DL_16 (0x0 << 4) +#define RT5668_I2S2_DL_20 (0x1 << 4) +#define RT5668_I2S2_DL_24 (0x2 << 4) +#define RT5668_I2S2_DL_8 (0x3 << 4) +#define RT5668_I2S_DF_MASK (0x7) +#define RT5668_I2S_DF_SFT 0 +#define RT5668_I2S_DF_I2S (0x0) +#define RT5668_I2S_DF_LEFT (0x1) +#define RT5668_I2S_DF_PCM_A (0x2) +#define RT5668_I2S_DF_PCM_B (0x3) +#define RT5668_I2S_DF_PCM_A_N (0x6) +#define RT5668_I2S_DF_PCM_B_N (0x7) + +/* ADC/DAC Clock Control 1 (0x0073) */ +#define RT5668_ADC_OSR_MASK (0xf << 12) +#define RT5668_ADC_OSR_SFT 12 +#define RT5668_ADC_OSR_D_1 (0x0 << 12) +#define RT5668_ADC_OSR_D_2 (0x1 << 12) +#define RT5668_ADC_OSR_D_4 (0x2 << 12) +#define RT5668_ADC_OSR_D_6 (0x3 << 12) +#define RT5668_ADC_OSR_D_8 (0x4 << 12) +#define RT5668_ADC_OSR_D_12 (0x5 << 12) +#define RT5668_ADC_OSR_D_16 (0x6 << 12) +#define RT5668_ADC_OSR_D_24 (0x7 << 12) +#define RT5668_ADC_OSR_D_32 (0x8 << 12) +#define RT5668_ADC_OSR_D_48 (0x9 << 12) +#define RT5668_I2S_M_DIV_MASK (0xf << 12) +#define RT5668_I2S_M_DIV_SFT 8 +#define RT5668_I2S_M_D_1 (0x0 << 8) +#define RT5668_I2S_M_D_2 (0x1 << 8) +#define RT5668_I2S_M_D_3 (0x2 << 8) +#define RT5668_I2S_M_D_4 (0x3 << 8) +#define RT5668_I2S_M_D_6 (0x4 << 8) +#define RT5668_I2S_M_D_8 (0x5 << 8) +#define RT5668_I2S_M_D_12 (0x6 << 8) +#define RT5668_I2S_M_D_16 (0x7 << 8) +#define RT5668_I2S_M_D_24 (0x8 << 8) +#define RT5668_I2S_M_D_32 (0x9 << 8) +#define RT5668_I2S_M_D_48 (0x10 << 8) +#define RT5668_I2S_CLK_SRC_MASK (0x7 << 4) +#define RT5668_I2S_CLK_SRC_SFT 4 +#define RT5668_I2S_CLK_SRC_MCLK (0x0 << 4) +#define RT5668_I2S_CLK_SRC_PLL1 (0x1 << 4) +#define RT5668_I2S_CLK_SRC_PLL2 (0x2 << 4) +#define RT5668_I2S_CLK_SRC_SDW (0x3 << 4) +#define RT5668_I2S_CLK_SRC_RCCLK (0x4 << 4) /* 25M */ +#define RT5668_DAC_OSR_MASK (0xf << 0) +#define RT5668_DAC_OSR_SFT 0 +#define RT5668_DAC_OSR_D_1 (0x0 << 0) +#define RT5668_DAC_OSR_D_2 (0x1 << 0) +#define RT5668_DAC_OSR_D_4 (0x2 << 0) +#define RT5668_DAC_OSR_D_6 (0x3 << 0) +#define RT5668_DAC_OSR_D_8 (0x4 << 0) +#define RT5668_DAC_OSR_D_12 (0x5 << 0) +#define RT5668_DAC_OSR_D_16 (0x6 << 0) +#define RT5668_DAC_OSR_D_24 (0x7 << 0) +#define RT5668_DAC_OSR_D_32 (0x8 << 0) +#define RT5668_DAC_OSR_D_48 (0x9 << 0) + +/* ADC/DAC Clock Control 2 (0x0074) */ +#define RT5668_I2S2_BCLK_MS2_MASK (0x1 << 11) +#define RT5668_I2S2_BCLK_MS2_SFT 11 +#define RT5668_I2S2_BCLK_MS2_32 (0x0 << 11) +#define RT5668_I2S2_BCLK_MS2_64 (0x1 << 11) + + +/* TDM control 1 (0x0079) */ +#define RT5668_TDM_TX_CH_MASK (0x3 << 12) +#define RT5668_TDM_TX_CH_2 (0x0 << 12) +#define RT5668_TDM_TX_CH_4 (0x1 << 12) +#define RT5668_TDM_TX_CH_6 (0x2 << 12) +#define RT5668_TDM_TX_CH_8 (0x3 << 12) +#define RT5668_TDM_RX_CH_MASK (0x3 << 8) +#define RT5668_TDM_RX_CH_2 (0x0 << 8) +#define RT5668_TDM_RX_CH_4 (0x1 << 8) +#define RT5668_TDM_RX_CH_6 (0x2 << 8) +#define RT5668_TDM_RX_CH_8 (0x3 << 8) +#define RT5668_TDM_ADC_LCA_MASK (0xf << 4) +#define RT5668_TDM_ADC_LCA_SFT 4 +#define RT5668_TDM_ADC_DL_SFT 0 + +/* TDM control 3 (0x007a) */ +#define RT5668_IF1_ADC1_SEL_SFT 14 +#define RT5668_IF1_ADC2_SEL_SFT 12 +#define RT5668_IF1_ADC3_SEL_SFT 10 +#define RT5668_IF1_ADC4_SEL_SFT 8 +#define RT5668_TDM_ADC_SEL_SFT 4 + +/* TDM/I2S control (0x007e) */ +#define RT5668_TDM_S_BP_MASK (0x1 << 15) +#define RT5668_TDM_S_BP_SFT 15 +#define RT5668_TDM_S_BP_NOR (0x0 << 15) +#define RT5668_TDM_S_BP_INV (0x1 << 15) +#define RT5668_TDM_S_LP_MASK (0x1 << 14) +#define RT5668_TDM_S_LP_SFT 14 +#define RT5668_TDM_S_LP_NOR (0x0 << 14) +#define RT5668_TDM_S_LP_INV (0x1 << 14) +#define RT5668_TDM_DF_MASK (0x7 << 11) +#define RT5668_TDM_DF_SFT 11 +#define RT5668_TDM_DF_I2S (0x0 << 11) +#define RT5668_TDM_DF_LEFT (0x1 << 11) +#define RT5668_TDM_DF_PCM_A (0x2 << 11) +#define RT5668_TDM_DF_PCM_B (0x3 << 11) +#define RT5668_TDM_DF_PCM_A_N (0x6 << 11) +#define RT5668_TDM_DF_PCM_B_N (0x7 << 11) +#define RT5668_TDM_CL_MASK (0x3 << 4) +#define RT5668_TDM_CL_16 (0x0 << 4) +#define RT5668_TDM_CL_20 (0x1 << 4) +#define RT5668_TDM_CL_24 (0x2 << 4) +#define RT5668_TDM_CL_32 (0x3 << 4) +#define RT5668_TDM_M_BP_MASK (0x1 << 2) +#define RT5668_TDM_M_BP_SFT 2 +#define RT5668_TDM_M_BP_NOR (0x0 << 2) +#define RT5668_TDM_M_BP_INV (0x1 << 2) +#define RT5668_TDM_M_LP_MASK (0x1 << 1) +#define RT5668_TDM_M_LP_SFT 1 +#define RT5668_TDM_M_LP_NOR (0x0 << 1) +#define RT5668_TDM_M_LP_INV (0x1 << 1) +#define RT5668_TDM_MS_MASK (0x1 << 0) +#define RT5668_TDM_MS_SFT 0 +#define RT5668_TDM_MS_M (0x0 << 0) +#define RT5668_TDM_MS_S (0x1 << 0) + +/* Global Clock Control (0x0080) */ +#define RT5668_SCLK_SRC_MASK (0x7 << 13) +#define RT5668_SCLK_SRC_SFT 13 +#define RT5668_SCLK_SRC_MCLK (0x0 << 13) +#define RT5668_SCLK_SRC_PLL1 (0x1 << 13) +#define RT5668_SCLK_SRC_PLL2 (0x2 << 13) +#define RT5668_SCLK_SRC_SDW (0x3 << 13) +#define RT5668_SCLK_SRC_RCCLK (0x4 << 13) +#define RT5668_PLL1_SRC_MASK (0x3 << 10) +#define RT5668_PLL1_SRC_SFT 10 +#define RT5668_PLL1_SRC_MCLK (0x0 << 10) +#define RT5668_PLL1_SRC_BCLK1 (0x1 << 10) +#define RT5668_PLL1_SRC_SDW (0x2 << 10) +#define RT5668_PLL1_SRC_RC (0x3 << 10) +#define RT5668_PLL2_SRC_MASK (0x3 << 8) +#define RT5668_PLL2_SRC_SFT 8 +#define RT5668_PLL2_SRC_MCLK (0x0 << 8) +#define RT5668_PLL2_SRC_BCLK1 (0x1 << 8) +#define RT5668_PLL2_SRC_SDW (0x2 << 8) +#define RT5668_PLL2_SRC_RC (0x3 << 8) + + + +#define RT5668_PLL_INP_MAX 40000000 +#define RT5668_PLL_INP_MIN 256000 +/* PLL M/N/K Code Control 1 (0x0081) */ +#define RT5668_PLL_N_MAX 0x001ff +#define RT5668_PLL_N_MASK (RT5668_PLL_N_MAX << 7) +#define RT5668_PLL_N_SFT 7 +#define RT5668_PLL_K_MAX 0x001f +#define RT5668_PLL_K_MASK (RT5668_PLL_K_MAX) +#define RT5668_PLL_K_SFT 0 + +/* PLL M/N/K Code Control 2 (0x0082) */ +#define RT5668_PLL_M_MAX 0x00f +#define RT5668_PLL_M_MASK (RT5668_PLL_M_MAX << 12) +#define RT5668_PLL_M_SFT 12 +#define RT5668_PLL_M_BP (0x1 << 11) +#define RT5668_PLL_M_BP_SFT 11 +#define RT5668_PLL_K_BP (0x1 << 10) +#define RT5668_PLL_K_BP_SFT 10 + +/* PLL tracking mode 1 (0x0083) */ +#define RT5668_DA_ASRC_MASK (0x1 << 13) +#define RT5668_DA_ASRC_SFT 13 +#define RT5668_DAC_STO1_ASRC_MASK (0x1 << 12) +#define RT5668_DAC_STO1_ASRC_SFT 12 +#define RT5668_AD_ASRC_MASK (0x1 << 8) +#define RT5668_AD_ASRC_SFT 8 +#define RT5668_AD_ASRC_SEL_MASK (0x1 << 4) +#define RT5668_AD_ASRC_SEL_SFT 4 +#define RT5668_DMIC_ASRC_MASK (0x1 << 3) +#define RT5668_DMIC_ASRC_SFT 3 +#define RT5668_ADC_STO1_ASRC_MASK (0x1 << 2) +#define RT5668_ADC_STO1_ASRC_SFT 2 +#define RT5668_DA_ASRC_SEL_MASK (0x1 << 0) +#define RT5668_DA_ASRC_SEL_SFT 0 + +/* PLL tracking mode 2 3 (0x0084)(0x0085)*/ +#define RT5668_FILTER_CLK_SEL_MASK (0x7 << 12) +#define RT5668_FILTER_CLK_SEL_SFT 12 + +/* ASRC Control 4 (0x0086) */ +#define RT5668_ASRCIN_FTK_N1_MASK (0x3 << 14) +#define RT5668_ASRCIN_FTK_N1_SFT 14 +#define RT5668_ASRCIN_FTK_N2_MASK (0x3 << 12) +#define RT5668_ASRCIN_FTK_N2_SFT 12 +#define RT5668_ASRCIN_FTK_M1_MASK (0x7 << 8) +#define RT5668_ASRCIN_FTK_M1_SFT 8 +#define RT5668_ASRCIN_FTK_M2_MASK (0x7 << 4) +#define RT5668_ASRCIN_FTK_M2_SFT 4 + +/* SoundWire reference clk (0x008d) */ +#define RT5668_PLL2_OUT_MASK (0x1 << 8) +#define RT5668_PLL2_OUT_98M (0x0 << 8) +#define RT5668_PLL2_OUT_49M (0x1 << 8) +#define RT5668_SDW_REF_2_MASK (0xf << 4) +#define RT5668_SDW_REF_2_SFT 4 +#define RT5668_SDW_REF_2_48K (0x0 << 4) +#define RT5668_SDW_REF_2_96K (0x1 << 4) +#define RT5668_SDW_REF_2_192K (0x2 << 4) +#define RT5668_SDW_REF_2_32K (0x3 << 4) +#define RT5668_SDW_REF_2_24K (0x4 << 4) +#define RT5668_SDW_REF_2_16K (0x5 << 4) +#define RT5668_SDW_REF_2_12K (0x6 << 4) +#define RT5668_SDW_REF_2_8K (0x7 << 4) +#define RT5668_SDW_REF_2_44K (0x8 << 4) +#define RT5668_SDW_REF_2_88K (0x9 << 4) +#define RT5668_SDW_REF_2_176K (0xa << 4) +#define RT5668_SDW_REF_2_353K (0xb << 4) +#define RT5668_SDW_REF_2_22K (0xc << 4) +#define RT5668_SDW_REF_2_384K (0xd << 4) +#define RT5668_SDW_REF_2_11K (0xe << 4) +#define RT5668_SDW_REF_1_MASK (0xf << 0) +#define RT5668_SDW_REF_1_SFT 0 +#define RT5668_SDW_REF_1_48K (0x0 << 0) +#define RT5668_SDW_REF_1_96K (0x1 << 0) +#define RT5668_SDW_REF_1_192K (0x2 << 0) +#define RT5668_SDW_REF_1_32K (0x3 << 0) +#define RT5668_SDW_REF_1_24K (0x4 << 0) +#define RT5668_SDW_REF_1_16K (0x5 << 0) +#define RT5668_SDW_REF_1_12K (0x6 << 0) +#define RT5668_SDW_REF_1_8K (0x7 << 0) +#define RT5668_SDW_REF_1_44K (0x8 << 0) +#define RT5668_SDW_REF_1_88K (0x9 << 0) +#define RT5668_SDW_REF_1_176K (0xa << 0) +#define RT5668_SDW_REF_1_353K (0xb << 0) +#define RT5668_SDW_REF_1_22K (0xc << 0) +#define RT5668_SDW_REF_1_384K (0xd << 0) +#define RT5668_SDW_REF_1_11K (0xe << 0) + +/* Depop Mode Control 1 (0x008e) */ +#define RT5668_PUMP_EN (0x1 << 3) +#define RT5668_PUMP_EN_SFT 3 +#define RT5668_CAPLESS_EN (0x1 << 0) +#define RT5668_CAPLESS_EN_SFT 0 + +/* Depop Mode Control 2 (0x8f) */ +#define RT5668_RAMP_MASK (0x1 << 12) +#define RT5668_RAMP_SFT 12 +#define RT5668_RAMP_DIS (0x0 << 12) +#define RT5668_RAMP_EN (0x1 << 12) +#define RT5668_BPS_MASK (0x1 << 11) +#define RT5668_BPS_SFT 11 +#define RT5668_BPS_DIS (0x0 << 11) +#define RT5668_BPS_EN (0x1 << 11) +#define RT5668_FAST_UPDN_MASK (0x1 << 10) +#define RT5668_FAST_UPDN_SFT 10 +#define RT5668_FAST_UPDN_DIS (0x0 << 10) +#define RT5668_FAST_UPDN_EN (0x1 << 10) +#define RT5668_VLO_MASK (0x1 << 7) +#define RT5668_VLO_SFT 7 +#define RT5668_VLO_3V (0x0 << 7) +#define RT5668_VLO_33V (0x1 << 7) + +/* HPOUT charge pump 1 (0x0091) */ +#define RT5668_OSW_L_MASK (0x1 << 11) +#define RT5668_OSW_L_SFT 11 +#define RT5668_OSW_L_DIS (0x0 << 11) +#define RT5668_OSW_L_EN (0x1 << 11) +#define RT5668_OSW_R_MASK (0x1 << 10) +#define RT5668_OSW_R_SFT 10 +#define RT5668_OSW_R_DIS (0x0 << 10) +#define RT5668_OSW_R_EN (0x1 << 10) +#define RT5668_PM_HP_MASK (0x3 << 8) +#define RT5668_PM_HP_SFT 8 +#define RT5668_PM_HP_LV (0x0 << 8) +#define RT5668_PM_HP_MV (0x1 << 8) +#define RT5668_PM_HP_HV (0x2 << 8) +#define RT5668_IB_HP_MASK (0x3 << 6) +#define RT5668_IB_HP_SFT 6 +#define RT5668_IB_HP_125IL (0x0 << 6) +#define RT5668_IB_HP_25IL (0x1 << 6) +#define RT5668_IB_HP_5IL (0x2 << 6) +#define RT5668_IB_HP_1IL (0x3 << 6) + +/* Micbias Control1 (0x93) */ +#define RT5668_MIC1_OV_MASK (0x3 << 14) +#define RT5668_MIC1_OV_SFT 14 +#define RT5668_MIC1_OV_2V7 (0x0 << 14) +#define RT5668_MIC1_OV_2V4 (0x1 << 14) +#define RT5668_MIC1_OV_2V25 (0x3 << 14) +#define RT5668_MIC1_OV_1V8 (0x4 << 14) +#define RT5668_MIC1_CLK_MASK (0x1 << 13) +#define RT5668_MIC1_CLK_SFT 13 +#define RT5668_MIC1_CLK_DIS (0x0 << 13) +#define RT5668_MIC1_CLK_EN (0x1 << 13) +#define RT5668_MIC1_OVCD_MASK (0x1 << 12) +#define RT5668_MIC1_OVCD_SFT 12 +#define RT5668_MIC1_OVCD_DIS (0x0 << 12) +#define RT5668_MIC1_OVCD_EN (0x1 << 12) +#define RT5668_MIC1_OVTH_MASK (0x3 << 10) +#define RT5668_MIC1_OVTH_SFT 10 +#define RT5668_MIC1_OVTH_768UA (0x0 << 10) +#define RT5668_MIC1_OVTH_960UA (0x1 << 10) +#define RT5668_MIC1_OVTH_1152UA (0x2 << 10) +#define RT5668_MIC1_OVTH_1960UA (0x3 << 10) +#define RT5668_MIC2_OV_MASK (0x3 << 8) +#define RT5668_MIC2_OV_SFT 8 +#define RT5668_MIC2_OV_2V7 (0x0 << 8) +#define RT5668_MIC2_OV_2V4 (0x1 << 8) +#define RT5668_MIC2_OV_2V25 (0x3 << 8) +#define RT5668_MIC2_OV_1V8 (0x4 << 8) +#define RT5668_MIC2_CLK_MASK (0x1 << 7) +#define RT5668_MIC2_CLK_SFT 7 +#define RT5668_MIC2_CLK_DIS (0x0 << 7) +#define RT5668_MIC2_CLK_EN (0x1 << 7) +#define RT5668_MIC2_OVTH_MASK (0x3 << 4) +#define RT5668_MIC2_OVTH_SFT 4 +#define RT5668_MIC2_OVTH_768UA (0x0 << 4) +#define RT5668_MIC2_OVTH_960UA (0x1 << 4) +#define RT5668_MIC2_OVTH_1152UA (0x2 << 4) +#define RT5668_MIC2_OVTH_1960UA (0x3 << 4) +#define RT5668_PWR_MB_MASK (0x1 << 3) +#define RT5668_PWR_MB_SFT 3 +#define RT5668_PWR_MB_PD (0x0 << 3) +#define RT5668_PWR_MB_PU (0x1 << 3) + +/* Micbias Control2 (0x0094) */ +#define RT5668_PWR_CLK25M_MASK (0x1 << 9) +#define RT5668_PWR_CLK25M_SFT 9 +#define RT5668_PWR_CLK25M_PD (0x0 << 9) +#define RT5668_PWR_CLK25M_PU (0x1 << 9) +#define RT5668_PWR_CLK1M_MASK (0x1 << 8) +#define RT5668_PWR_CLK1M_SFT 8 +#define RT5668_PWR_CLK1M_PD (0x0 << 8) +#define RT5668_PWR_CLK1M_PU (0x1 << 8) + +/* RC Clock Control (0x009f) */ +#define RT5668_POW_IRQ (0x1 << 15) +#define RT5668_POW_JDH (0x1 << 14) +#define RT5668_POW_JDL (0x1 << 13) +#define RT5668_POW_ANA (0x1 << 12) + +/* I2S Master Mode Clock Control 1 (0x00a0) */ +#define RT5668_CLK_SRC_MCLK (0x0) +#define RT5668_CLK_SRC_PLL1 (0x1) +#define RT5668_CLK_SRC_PLL2 (0x2) +#define RT5668_CLK_SRC_SDW (0x3) +#define RT5668_CLK_SRC_RCCLK (0x4) +#define RT5668_I2S_PD_1 (0x0) +#define RT5668_I2S_PD_2 (0x1) +#define RT5668_I2S_PD_3 (0x2) +#define RT5668_I2S_PD_4 (0x3) +#define RT5668_I2S_PD_6 (0x4) +#define RT5668_I2S_PD_8 (0x5) +#define RT5668_I2S_PD_12 (0x6) +#define RT5668_I2S_PD_16 (0x7) +#define RT5668_I2S_PD_24 (0x8) +#define RT5668_I2S_PD_32 (0x9) +#define RT5668_I2S_PD_48 (0xa) +#define RT5668_I2S2_SRC_MASK (0x3 << 4) +#define RT5668_I2S2_SRC_SFT 4 +#define RT5668_I2S2_M_PD_MASK (0xf << 0) +#define RT5668_I2S2_M_PD_SFT 0 + +/* IRQ Control 1 (0x00b6) */ +#define RT5668_JD1_PULSE_EN_MASK (0x1 << 10) +#define RT5668_JD1_PULSE_EN_SFT 10 +#define RT5668_JD1_PULSE_DIS (0x0 << 10) +#define RT5668_JD1_PULSE_EN (0x1 << 10) + +/* IRQ Control 2 (0x00b7) */ +#define RT5668_JD1_EN_MASK (0x1 << 15) +#define RT5668_JD1_EN_SFT 15 +#define RT5668_JD1_DIS (0x0 << 15) +#define RT5668_JD1_EN (0x1 << 15) +#define RT5668_JD1_POL_MASK (0x1 << 13) +#define RT5668_JD1_POL_NOR (0x0 << 13) +#define RT5668_JD1_POL_INV (0x1 << 13) + +/* IRQ Control 3 (0x00b8) */ +#define RT5668_IL_IRQ_MASK (0x1 << 7) +#define RT5668_IL_IRQ_DIS (0x0 << 7) +#define RT5668_IL_IRQ_EN (0x1 << 7) + +/* GPIO Control 1 (0x00c0) */ +#define RT5668_GP1_PIN_MASK (0x3 << 14) +#define RT5668_GP1_PIN_SFT 14 +#define RT5668_GP1_PIN_GPIO1 (0x0 << 14) +#define RT5668_GP1_PIN_IRQ (0x1 << 14) +#define RT5668_GP1_PIN_DMIC_CLK (0x2 << 14) +#define RT5668_GP2_PIN_MASK (0x3 << 12) +#define RT5668_GP2_PIN_SFT 12 +#define RT5668_GP2_PIN_GPIO2 (0x0 << 12) +#define RT5668_GP2_PIN_LRCK2 (0x1 << 12) +#define RT5668_GP2_PIN_DMIC_SDA (0x2 << 12) +#define RT5668_GP3_PIN_MASK (0x3 << 10) +#define RT5668_GP3_PIN_SFT 10 +#define RT5668_GP3_PIN_GPIO3 (0x0 << 10) +#define RT5668_GP3_PIN_BCLK2 (0x1 << 10) +#define RT5668_GP3_PIN_DMIC_CLK (0x2 << 10) +#define RT5668_GP4_PIN_MASK (0x3 << 8) +#define RT5668_GP4_PIN_SFT 8 +#define RT5668_GP4_PIN_GPIO4 (0x0 << 8) +#define RT5668_GP4_PIN_ADCDAT1 (0x1 << 8) +#define RT5668_GP4_PIN_DMIC_CLK (0x2 << 8) +#define RT5668_GP4_PIN_ADCDAT2 (0x3 << 8) +#define RT5668_GP5_PIN_MASK (0x3 << 6) +#define RT5668_GP5_PIN_SFT 6 +#define RT5668_GP5_PIN_GPIO5 (0x0 << 6) +#define RT5668_GP5_PIN_DACDAT1 (0x1 << 6) +#define RT5668_GP5_PIN_DMIC_SDA (0x2 << 6) +#define RT5668_GP6_PIN_MASK (0x1 << 5) +#define RT5668_GP6_PIN_SFT 5 +#define RT5668_GP6_PIN_GPIO6 (0x0 << 5) +#define RT5668_GP6_PIN_LRCK1 (0x1 << 5) + +/* GPIO Control 2 (0x00c1)*/ +#define RT5668_GP1_PF_MASK (0x1 << 15) +#define RT5668_GP1_PF_IN (0x0 << 15) +#define RT5668_GP1_PF_OUT (0x1 << 15) +#define RT5668_GP1_OUT_MASK (0x1 << 14) +#define RT5668_GP1_OUT_L (0x0 << 14) +#define RT5668_GP1_OUT_H (0x1 << 14) +#define RT5668_GP2_PF_MASK (0x1 << 13) +#define RT5668_GP2_PF_IN (0x0 << 13) +#define RT5668_GP2_PF_OUT (0x1 << 13) +#define RT5668_GP2_OUT_MASK (0x1 << 12) +#define RT5668_GP2_OUT_L (0x0 << 12) +#define RT5668_GP2_OUT_H (0x1 << 12) +#define RT5668_GP3_PF_MASK (0x1 << 11) +#define RT5668_GP3_PF_IN (0x0 << 11) +#define RT5668_GP3_PF_OUT (0x1 << 11) +#define RT5668_GP3_OUT_MASK (0x1 << 10) +#define RT5668_GP3_OUT_L (0x0 << 10) +#define RT5668_GP3_OUT_H (0x1 << 10) +#define RT5668_GP4_PF_MASK (0x1 << 9) +#define RT5668_GP4_PF_IN (0x0 << 9) +#define RT5668_GP4_PF_OUT (0x1 << 9) +#define RT5668_GP4_OUT_MASK (0x1 << 8) +#define RT5668_GP4_OUT_L (0x0 << 8) +#define RT5668_GP4_OUT_H (0x1 << 8) +#define RT5668_GP5_PF_MASK (0x1 << 7) +#define RT5668_GP5_PF_IN (0x0 << 7) +#define RT5668_GP5_PF_OUT (0x1 << 7) +#define RT5668_GP5_OUT_MASK (0x1 << 6) +#define RT5668_GP5_OUT_L (0x0 << 6) +#define RT5668_GP5_OUT_H (0x1 << 6) +#define RT5668_GP6_PF_MASK (0x1 << 5) +#define RT5668_GP6_PF_IN (0x0 << 5) +#define RT5668_GP6_PF_OUT (0x1 << 5) +#define RT5668_GP6_OUT_MASK (0x1 << 4) +#define RT5668_GP6_OUT_L (0x0 << 4) +#define RT5668_GP6_OUT_H (0x1 << 4) + + +/* GPIO Status (0x00c2) */ +#define RT5668_GP6_STA (0x1 << 6) +#define RT5668_GP5_STA (0x1 << 5) +#define RT5668_GP4_STA (0x1 << 4) +#define RT5668_GP3_STA (0x1 << 3) +#define RT5668_GP2_STA (0x1 << 2) +#define RT5668_GP1_STA (0x1 << 1) + +/* Soft volume and zero cross control 1 (0x00d9) */ +#define RT5668_SV_MASK (0x1 << 15) +#define RT5668_SV_SFT 15 +#define RT5668_SV_DIS (0x0 << 15) +#define RT5668_SV_EN (0x1 << 15) +#define RT5668_ZCD_MASK (0x1 << 10) +#define RT5668_ZCD_SFT 10 +#define RT5668_ZCD_PD (0x0 << 10) +#define RT5668_ZCD_PU (0x1 << 10) +#define RT5668_SV_DLY_MASK (0xf) +#define RT5668_SV_DLY_SFT 0 + +/* Soft volume and zero cross control 2 (0x00da) */ +#define RT5668_ZCD_BST1_CBJ_MASK (0x1 << 7) +#define RT5668_ZCD_BST1_CBJ_SFT 7 +#define RT5668_ZCD_BST1_CBJ_DIS (0x0 << 7) +#define RT5668_ZCD_BST1_CBJ_EN (0x1 << 7) +#define RT5668_ZCD_RECMIX_MASK (0x1) +#define RT5668_ZCD_RECMIX_SFT 0 +#define RT5668_ZCD_RECMIX_DIS (0x0) +#define RT5668_ZCD_RECMIX_EN (0x1) + +/* 4 Button Inline Command Control 2 (0x00e3) */ +#define RT5668_4BTN_IL_MASK (0x1 << 15) +#define RT5668_4BTN_IL_EN (0x1 << 15) +#define RT5668_4BTN_IL_DIS (0x0 << 15) +#define RT5668_4BTN_IL_RST_MASK (0x1 << 14) +#define RT5668_4BTN_IL_NOR (0x1 << 14) +#define RT5668_4BTN_IL_RST (0x0 << 14) + +/* Analog JD Control (0x00f0) */ +#define RT5668_JDH_RS_MASK (0x1 << 4) +#define RT5668_JDH_NO_PLUG (0x1 << 4) +#define RT5668_JDH_PLUG (0x0 << 4) + +/* Chopper and Clock control for DAC (0x013a)*/ +#define RT5668_CKXEN_DAC1_MASK (0x1 << 13) +#define RT5668_CKXEN_DAC1_SFT 13 +#define RT5668_CKGEN_DAC1_MASK (0x1 << 12) +#define RT5668_CKGEN_DAC1_SFT 12 + +/* Chopper and Clock control for ADC (0x013b)*/ +#define RT5668_CKXEN_ADC1_MASK (0x1 << 13) +#define RT5668_CKXEN_ADC1_SFT 13 +#define RT5668_CKGEN_ADC1_MASK (0x1 << 12) +#define RT5668_CKGEN_ADC1_SFT 12 + +/* Volume test (0x013f)*/ +#define RT5668_SEL_CLK_VOL_MASK (0x1 << 15) +#define RT5668_SEL_CLK_VOL_EN (0x1 << 15) +#define RT5668_SEL_CLK_VOL_DIS (0x0 << 15) + +/* Test Mode Control 1 (0x0145) */ +#define RT5668_AD2DA_LB_MASK (0x1 << 10) +#define RT5668_AD2DA_LB_SFT 10 + +/* Stereo Noise Gate Control 1 (0x0160) */ +#define RT5668_NG2_EN_MASK (0x1 << 15) +#define RT5668_NG2_EN (0x1 << 15) +#define RT5668_NG2_DIS (0x0 << 15) + +/* Stereo1 DAC Silence Detection Control (0x0190) */ +#define RT5668_DEB_STO_DAC_MASK (0x7 << 4) +#define RT5668_DEB_80_MS (0x0 << 4) + +/* SAR ADC Inline Command Control 1 (0x0210) */ +#define RT5668_SAR_BUTT_DET_MASK (0x1 << 15) +#define RT5668_SAR_BUTT_DET_EN (0x1 << 15) +#define RT5668_SAR_BUTT_DET_DIS (0x0 << 15) +#define RT5668_SAR_BUTDET_MODE_MASK (0x1 << 14) +#define RT5668_SAR_BUTDET_POW_SAV (0x1 << 14) +#define RT5668_SAR_BUTDET_POW_NORM (0x0 << 14) +#define RT5668_SAR_BUTDET_RST_MASK (0x1 << 13) +#define RT5668_SAR_BUTDET_RST_NORMAL (0x1 << 13) +#define RT5668_SAR_BUTDET_RST (0x0 << 13) +#define RT5668_SAR_POW_MASK (0x1 << 12) +#define RT5668_SAR_POW_EN (0x1 << 12) +#define RT5668_SAR_POW_DIS (0x0 << 12) +#define RT5668_SAR_RST_MASK (0x1 << 11) +#define RT5668_SAR_RST_NORMAL (0x1 << 11) +#define RT5668_SAR_RST (0x0 << 11) +#define RT5668_SAR_BYPASS_MASK (0x1 << 10) +#define RT5668_SAR_BYPASS_EN (0x1 << 10) +#define RT5668_SAR_BYPASS_DIS (0x0 << 10) +#define RT5668_SAR_SEL_MB1_MASK (0x1 << 9) +#define RT5668_SAR_SEL_MB1_SEL (0x1 << 9) +#define RT5668_SAR_SEL_MB1_NOSEL (0x0 << 9) +#define RT5668_SAR_SEL_MB2_MASK (0x1 << 8) +#define RT5668_SAR_SEL_MB2_SEL (0x1 << 8) +#define RT5668_SAR_SEL_MB2_NOSEL (0x0 << 8) +#define RT5668_SAR_SEL_MODE_MASK (0x1 << 7) +#define RT5668_SAR_SEL_MODE_CMP (0x1 << 7) +#define RT5668_SAR_SEL_MODE_ADC (0x0 << 7) +#define RT5668_SAR_SEL_MB1_MB2_MASK (0x1 << 5) +#define RT5668_SAR_SEL_MB1_MB2_AUTO (0x1 << 5) +#define RT5668_SAR_SEL_MB1_MB2_MANU (0x0 << 5) +#define RT5668_SAR_SEL_SIGNAL_MASK (0x1 << 4) +#define RT5668_SAR_SEL_SIGNAL_AUTO (0x1 << 4) +#define RT5668_SAR_SEL_SIGNAL_MANU (0x0 << 4) + +/* SAR ADC Inline Command Control 13 (0x021c) */ +#define RT5668_SAR_SOUR_MASK (0x3f) +#define RT5668_SAR_SOUR_BTN (0x3f) +#define RT5668_SAR_SOUR_TYPE (0x0) + + +/* System Clock Source */ +enum { + RT5668_SCLK_S_MCLK, + RT5668_SCLK_S_PLL1, + RT5668_SCLK_S_PLL2, + RT5668_SCLK_S_RCCLK, +}; + +/* PLL Source */ +enum { + RT5668_PLL1_S_MCLK, + RT5668_PLL1_S_BCLK1, + RT5668_PLL1_S_RCCLK, +}; + +enum { + RT5668_AIF1, + RT5668_AIF2, + RT5668_AIFS +}; + +/* filter mask */ +enum { + RT5668_DA_STEREO1_FILTER = 0x1, + RT5668_AD_STEREO1_FILTER = (0x1 << 1), +}; + +enum { + RT5668_CLK_SEL_SYS, + RT5668_CLK_SEL_I2S1_ASRC, + RT5668_CLK_SEL_I2S2_ASRC, +}; + +int rt5668_sel_asrc_clk_src(struct snd_soc_component *component, + unsigned int filter_mask, unsigned int clk_src); + +#endif /* __RT5668_H__ */ -- 2.17.0

1 0

[alsa-devel] [PATCH] ASoC: max98357a: Fix speaker pop when starting playback
by Ezequiel Garcia 16 Apr '18

16 Apr '18

From: Mac Chiang <mac.chiang(a)intel.com> Currently, speaker pop/crack noise can happen during PCM playback. This is due to the LRCLK being removed while the BCLK being present, which as per the datasheet can cause large DC output voltage. While on some platforms simply controlling the SD_MODE pin is enough to remove the noise, others have a significant delay between the BCLK and the LRCLK activation. Therefore, this commit adds a delay in the codec driver to guarantee clocks are both active by the time SD_MODE is asserted. Link: https://github.com/raspberrypi/linux/issues/2212 Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> Signed-off-by: Stephen Barber <smbarber(a)chromium.org> Signed-off-by: Ezequiel Garcia <ezequiel(a)collabora.co.uk> --- .../devicetree/bindings/sound/max98357a.txt | 6 ++ sound/soc/codecs/max98357a.c | 69 +++++++++++++++++++--- 2 files changed, 66 insertions(+), 9 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/max98357a.txt b/Documentation/devicetree/bindings/sound/max98357a.txt index 28645a2ff885..de42c9f372ed 100644 --- a/Documentation/devicetree/bindings/sound/max98357a.txt +++ b/Documentation/devicetree/bindings/sound/max98357a.txt @@ -9,10 +9,16 @@ Optional properties: - sdmode-gpios : GPIO specifier for the chip's SD_MODE pin. If this option is not specified then driver does not manage the pin state (e.g. chip is always on). +- sdmode-delay : specify a delay time for SD_MODE pin. According + to the DAC datasheet, if LRCLK is removed while BCLK is present, + the DAC output can cause loud pop/crack noises. This property + specifies a delay for the SD_MODE pin assert, required to + eliminate the noise. Example: max98357a { compatible = "maxim,max98357a"; sdmode-gpios = <&qcom_pinmux 25 0>; + sdmode-delay = <5>; }; diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index 426ed2dae6ca..9666fa3602a5 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -27,27 +27,57 @@ #include <sound/soc-dai.h> #include <sound/soc-dapm.h> +struct max98357a_priv { + struct delayed_work enable_sdmode_work; + struct gpio_desc *sdmode; + unsigned int sdmode_delay; + int sdmode_enabled; + spinlock_t sdmode_lock; +}; + +static void max98357a_enable_sdmode_work(struct work_struct *work) +{ + struct max98357a_priv *max98357a = container_of(work, + struct max98357a_priv, enable_sdmode_work.work); + unsigned long flags; + + spin_lock_irqsave(&max98357a->sdmode_lock, flags); + gpiod_set_value(max98357a->sdmode, max98357a->sdmode_enabled); + spin_unlock_irqrestore(&max98357a->sdmode_lock, flags); +} + static int max98357a_daiops_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - struct gpio_desc *sdmode = snd_soc_dai_get_drvdata(dai); + struct snd_soc_codec *codec = dai->codec; + struct max98357a_priv *max98357a = snd_soc_codec_get_drvdata(codec); + unsigned long flags; - if (!sdmode) + if (!max98357a->sdmode) return 0; switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - gpiod_set_value(sdmode, 1); + max98357a->sdmode_enabled = 1; + if (max98357a->sdmode_delay) { + queue_delayed_work(system_power_efficient_wq, + &max98357a->enable_sdmode_work, + msecs_to_jiffies(max98357a->sdmode_delay)); + return 0; + } break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - gpiod_set_value(sdmode, 0); + max98357a->sdmode_enabled = 0; break; } + spin_lock_irqsave(&max98357a->sdmode_lock, flags); + gpiod_set_value(max98357a->sdmode, max98357a->sdmode_enabled); + spin_unlock_irqrestore(&max98357a->sdmode_lock, flags); return 0; } @@ -61,19 +91,30 @@ static const struct snd_soc_dapm_route max98357a_dapm_routes[] = { static int max98357a_codec_probe(struct snd_soc_codec *codec) { - struct gpio_desc *sdmode; + struct max98357a_priv *max98357a = snd_soc_codec_get_drvdata(codec); - sdmode = devm_gpiod_get_optional(codec->dev, "sdmode", GPIOD_OUT_LOW); - if (IS_ERR(sdmode)) - return PTR_ERR(sdmode); + max98357a->sdmode = devm_gpiod_get_optional(codec->dev, + "sdmode", GPIOD_OUT_LOW); + if (IS_ERR(max98357a->sdmode)) + return PTR_ERR(max98357a->sdmode); - snd_soc_codec_set_drvdata(codec, sdmode); + spin_lock_init(&max98357a->sdmode_lock); + INIT_DELAYED_WORK(&max98357a->enable_sdmode_work, + max98357a_enable_sdmode_work); + return 0; +} + +static int max98357a_codec_remove(struct snd_soc_codec *codec) +{ + struct max98357a_priv *max98357a = snd_soc_codec_get_drvdata(codec); + cancel_delayed_work_sync(&max98357a->enable_sdmode_work); return 0; } static const struct snd_soc_codec_driver max98357a_codec_driver = { .probe = max98357a_codec_probe, + .remove = max98357a_codec_remove, .component_driver = { .dapm_widgets = max98357a_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(max98357a_dapm_widgets), @@ -107,6 +148,16 @@ static struct snd_soc_dai_driver max98357a_dai_driver = { static int max98357a_platform_probe(struct platform_device *pdev) { + struct max98357a_priv *max98357a; + + max98357a = devm_kzalloc(&pdev->dev, sizeof(*max98357a), GFP_KERNEL); + if (!max98357a) + return -ENOMEM; + dev_set_drvdata(&pdev->dev, max98357a); + + device_property_read_u32(&pdev->dev, "sdmode-delay", + &max98357a->sdmode_delay); + return snd_soc_register_codec(&pdev->dev, &max98357a_codec_driver, &max98357a_dai_driver, 1); } -- 2.16.2

4 7

[alsa-devel] [FIX][PATCH] ASoC: topology: Fix logical inversion in set_link_hw_format()
by Xiuli Pan 16 Apr '18

16 Apr '18

From: Pan Xiuli <xiuli.pan(a)linux.intel.com> The master/slave conventions are wrt. the codec. The topology code contains a logical inversion, fix to follow ASoC usage. Signed-off-by: Pan Xiuli <xiuli.pan(a)linux.intel.com> --- sound/soc/soc-topology.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 561f74d..2931248 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -2030,11 +2030,11 @@ static void set_link_hw_format(struct snd_soc_dai_link *link, /* clock masters */ bclk_master = hw_config->bclk_master; fsync_master = hw_config->fsync_master; - if (!bclk_master && !fsync_master) + if (bclk_master && fsync_master) link->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; - else if (bclk_master && !fsync_master) - link->dai_fmt |= SND_SOC_DAIFMT_CBS_CFM; else if (!bclk_master && fsync_master) + link->dai_fmt |= SND_SOC_DAIFMT_CBS_CFM; + else if (bclk_master && !fsync_master) link->dai_fmt |= SND_SOC_DAIFMT_CBM_CFS; else link->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; -- 2.7.4

6 24

[alsa-devel] [PATCH v3 0/2] ASoC: topology: Improve parsing hw_configs
by Kirill Marinushkin 16 Apr '18

16 Apr '18

Hello Jaroslav, Takashi, Mark, This patch series is a resend of [1] and [2], rebased on top of the latest head. It was logical to resend them together. It includes 2 patches for linux + 2 patches for alsa-lib. Please have a look. Best Regards, Kirill [1] https://patchwork.kernel.org/patch/10250485/ [2] https://patchwork.kernel.org/patch/10230611/ Kirill Marinushkin (2): ASoC: topology: Fix bclk and fsync inversion in set_link_hw_format() ASoC: topology: Add missing clock gating parameter when parsing hw_configs include/uapi/sound/asoc.h | 23 ++++++++++++++++++++--- sound/soc/soc-topology.c | 19 ++++++++++++++----- 2 files changed, 34 insertions(+), 8 deletions(-) -- 2.13.6

3 12

[alsa-devel] [PATCH] ASoC: rt5514: Add the missing register in the readable table
by Oder Chiou 16 Apr '18

16 Apr '18

The patch adds the missing register in the readable table. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 2d79571..c33f8da 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -89,6 +89,7 @@ static const struct reg_default rt5514_reg[] = { {RT5514_PLL3_CALIB_CTRL5, 0x40220012}, {RT5514_DELAY_BUF_CTRL1, 0x7fff006a}, {RT5514_DELAY_BUF_CTRL3, 0x00000000}, + {RT5514_ASRC_IN_CTRL1, 0x00000003}, {RT5514_DOWNFILTER0_CTRL1, 0x00020c2f}, {RT5514_DOWNFILTER0_CTRL2, 0x00020c2f}, {RT5514_DOWNFILTER0_CTRL3, 0x10000362}, @@ -181,6 +182,7 @@ static bool rt5514_readable_register(struct device *dev, unsigned int reg) case RT5514_PLL3_CALIB_CTRL5: case RT5514_DELAY_BUF_CTRL1: case RT5514_DELAY_BUF_CTRL3: + case RT5514_ASRC_IN_CTRL1: case RT5514_DOWNFILTER0_CTRL1: case RT5514_DOWNFILTER0_CTRL2: case RT5514_DOWNFILTER0_CTRL3: @@ -238,6 +240,7 @@ static bool rt5514_i2c_readable_register(struct device *dev, case RT5514_DSP_MAPPING | RT5514_PLL3_CALIB_CTRL5: case RT5514_DSP_MAPPING | RT5514_DELAY_BUF_CTRL1: case RT5514_DSP_MAPPING | RT5514_DELAY_BUF_CTRL3: + case RT5514_DSP_MAPPING | RT5514_ASRC_IN_CTRL1: case RT5514_DSP_MAPPING | RT5514_DOWNFILTER0_CTRL1: case RT5514_DSP_MAPPING | RT5514_DOWNFILTER0_CTRL2: case RT5514_DSP_MAPPING | RT5514_DOWNFILTER0_CTRL3: -- 2.7.4

2 1

[alsa-devel] [PATCH 0/2] Minor GFP_ATOMIC removal
by Takashi Iwai 16 Apr '18

16 Apr '18

Hi, this is small fixes for the old PCI drivers that still used GFP_ATOMIC by some reason. It became a nice cleanup in the end. Takashi === Takashi Iwai (2): ALSA: cmipci: Allocate with GFP_KERNEL instead of GFP_ATOMIC ALSA: emu10k1: Reduce GFP_ATOMIC allocation include/sound/emu10k1.h | 4 ++-- sound/pci/cmipci.c | 2 +- sound/pci/emu10k1/emufx.c | 9 +-------- sound/pci/emu10k1/emupcm.c | 2 +- 4 files changed, 5 insertions(+), 12 deletions(-) -- 2.16.3

1 2

[alsa-devel] [PATCH] ASoC: fsl_esai: Add freq check in set_dai_sysclk()
by Nicolin Chen 16 Apr '18

16 Apr '18

The freq parameter indicates the physical frequency of an actual input clock or a desired frequency of an output clock for HCKT/R. It should never be passed 0. This might cause Division-by-zero. So this patch adds a check to fix it. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/fsl/fsl_esai.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index da8fd98..d79e99e 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -226,6 +226,12 @@ static int fsl_esai_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned long clk_rate; int ret; + if (freq == 0) { + dev_err(dai->dev, "%sput freq of HCK%c should not be 0Hz\n", + in ? "in" : "out", tx ? 'T' : 'R'); + return -EINVAL; + } + /* Bypass divider settings if the requirement doesn't change */ if (freq == esai_priv->hck_rate[tx] && dir == esai_priv->hck_dir[tx]) return 0; -- 2.7.4

3 2

[alsa-devel] [PATCH 1/2] ASoC: max9860: switch to using .probe_new
by Peter Rosin 16 Apr '18

16 Apr '18

Use the new probe style for i2c drivers. Signed-off-by: Peter Rosin <peda(a)axentia.se> --- sound/soc/codecs/max9860.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/max9860.c b/sound/soc/codecs/max9860.c index a2dc6a47f466..376bd48e4790 100644 --- a/sound/soc/codecs/max9860.c +++ b/sound/soc/codecs/max9860.c @@ -599,8 +599,7 @@ static const struct dev_pm_ops max9860_pm_ops = { SET_RUNTIME_PM_OPS(max9860_suspend, max9860_resume, NULL) }; -static int max9860_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) +static int max9860_probe(struct i2c_client *i2c) { struct device *dev = &i2c->dev; struct max9860_priv *max9860; @@ -738,7 +737,7 @@ static const struct of_device_id max9860_of_match[] = { MODULE_DEVICE_TABLE(of, max9860_of_match); static struct i2c_driver max9860_i2c_driver = { - .probe = max9860_probe, + .probe_new = max9860_probe, .remove = max9860_remove, .id_table = max9860_i2c_id, .driver = { -- 2.11.0

3 29

[alsa-devel] [v3] ASoC: AMD: Support headset button on Stoney DA7219
by Akshu Agrawal 16 Apr '18

16 Apr '18

Adds headset button support. TEST=Tested Volume UP/Down, Play/Pause functionality Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- v2: Changed KEY_MEDIA to KEY_PLAYPAUSE v3: Using cz_jack.jack instead of jack->jack sound/soc/amd/acp-da7219-max98357a.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index b205c78..d281c22 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -33,6 +33,7 @@ #include <linux/gpio.h> #include <linux/module.h> #include <linux/i2c.h> +#include <linux/input.h> #include <linux/acpi.h> #include "../codecs/da7219.h" @@ -80,6 +81,11 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) return ret; } + snd_jack_set_key(cz_jack.jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(cz_jack.jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(cz_jack.jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(cz_jack.jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + da7219_aad_jack_det(component, &cz_jack); return 0; -- 1.9.1

3 3

[alsa-devel] [PATCH] ASoC: rsnd: Enable IPMMU v2
by Kuninori Morimoto 16 Apr '18

16 Apr '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 4821d914fe747 ("ASoC: rsnd: use dma_sync_single_for_xxx() for IOMMU") (= v1) which have been already reverted had supported IPMMU support on rsnd driver. Because memory allocating timing and DMAEngine access timing were different, it used continuous memory and called dma map function by itself. OTOH, it is using DMA descriptor mode (= DMA cyclic mode), thus, there was timing conflict between DMA sync/unsync and DMA transfer starting, and it maked sound noise. This patch supports IPMMU with coherent memory, and, it uses Audio DMAC dev for allocating memory by snd_pcm_lib_preallocate_pages_for_all() to indicate memory area to IPMMU. One note is that Playback/Capture need each paired Audio DMAC dev. Because of this, we need to keep each paired Audio DMAC dev when probing, and use it when allocating each memory for IPMMU. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 47 ++++++++++++++++++++++++++++++++++++++++++----- sound/soc/sh/rcar/dma.c | 7 +++++++ sound/soc/sh/rcar/rsnd.h | 1 + 3 files changed, 50 insertions(+), 5 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 4995e73..af04d41 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1352,6 +1352,37 @@ int rsnd_kctrl_new(struct rsnd_mod *mod, #define PREALLOC_BUFFER (32 * 1024) #define PREALLOC_BUFFER_MAX (32 * 1024) +static int rsnd_preallocate_pages(struct snd_soc_pcm_runtime *rtd, + struct rsnd_dai_stream *io, + int stream) +{ + struct rsnd_priv *priv = rsnd_io_to_priv(io); + struct device *dev = rsnd_priv_to_dev(priv); + struct snd_pcm_substream *substream; + int err; + + /* + * use Audio-DMAC dev if we can use IPMMU + * see + * rsnd_dmaen_attach() + */ + if (io->dmac_dev) + dev = io->dmac_dev; + + for (substream = rtd->pcm->streams[stream].substream; + substream; + substream = substream->next) { + err = snd_pcm_lib_preallocate_pages(substream, + SNDRV_DMA_TYPE_DEV, + dev, + PREALLOC_BUFFER, PREALLOC_BUFFER_MAX); + if (err < 0) + return err; + } + + return 0; +} + static int rsnd_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *dai = rtd->cpu_dai; @@ -1366,11 +1397,17 @@ static int rsnd_pcm_new(struct snd_soc_pcm_runtime *rtd) if (ret) return ret; - return snd_pcm_lib_preallocate_pages_for_all( - rtd->pcm, - SNDRV_DMA_TYPE_DEV, - rtd->card->snd_card->dev, - PREALLOC_BUFFER, PREALLOC_BUFFER_MAX); + ret = rsnd_preallocate_pages(rtd, &rdai->playback, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + return ret; + + ret = rsnd_preallocate_pages(rtd, &rdai->capture, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + return ret; + + return 0; } static const struct snd_soc_component_driver rsnd_soc_component = { diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 32ac97b..ef82b94 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -253,6 +253,13 @@ static int rsnd_dmaen_attach(struct rsnd_dai_stream *io, return -EAGAIN; } + /* + * use it for IPMMU if needed + * see + * rsnd_preallocate_pages() + */ + io->dmac_dev = chan->device->dev; + dma_release_channel(chan); dmac->dmaen_num++; diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index b1896f1..6d7280d 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -435,6 +435,7 @@ struct rsnd_dai_stream { struct snd_pcm_substream *substream; struct rsnd_mod *mod[RSND_MOD_MAX]; struct rsnd_dai *rdai; + struct device *dmac_dev; /* for IPMMU */ u32 parent_ssi_status; }; #define rsnd_io_to_mod(io, i) ((i) < RSND_MOD_MAX ? (io)->mod[(i)] : NULL) -- 1.9.1

2 2

[alsa-devel] [PATCH] ASoC: rsnd: ssi: wait maximum 5ms for status check
by Kuninori Morimoto 16 Apr '18

16 Apr '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> It is waiting udelay(50) x 1024 (= 50ms) for status check in worst case, but it is overkill. And we shouldn't use udelay() for 50us (linux/Documentation/timers/timers-howto.txt) Waiting maximum udelay(5) x 1024 (= 5ms) is very enough for status check. This patch fixes these issue. Reported-by: Hiromitsu Yamasaki <hiromitsu.yamasaki.ym(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 31ffe3f..9538f76 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -171,7 +171,7 @@ static void rsnd_ssi_status_check(struct rsnd_mod *mod, if (status & bit) return; - udelay(50); + udelay(5); } dev_warn(dev, "%s[%d] status check failed\n", -- 1.9.1

2 2

[alsa-devel] [PATCH v2 1/2] ASoC: adau17x1: Handling of DSP_RUN register during fw setup
by Robert Rosengren 16 Apr '18

16 Apr '18

From: Danny Smith <dannys(a)axis.com> DSP_RUN needs to be disabled during firmware write otherwise we can end up with undefined behavior if writing to a dsp which is already running firmware. Signed-off-by: Danny Smith <dannys(a)axis.com> Signed-off-by: Robert Rosengren <robert.rosengren(a)axis.com> --- sound/soc/codecs/adau17x1.c | 26 ++++++++++++++++++++------ sound/soc/codecs/adau17x1.h | 3 ++- 2 files changed, 22 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 80c2a06285bb..12bf24c26818 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -502,7 +502,7 @@ static int adau17x1_hw_params(struct snd_pcm_substream *substream, } if (adau->sigmadsp) { - ret = adau17x1_setup_firmware(adau, params_rate(params)); + ret = adau17x1_setup_firmware(component, params_rate(params)); if (ret < 0) return ret; } @@ -835,26 +835,40 @@ bool adau17x1_volatile_register(struct device *dev, unsigned int reg) } EXPORT_SYMBOL_GPL(adau17x1_volatile_register); -int adau17x1_setup_firmware(struct adau *adau, unsigned int rate) +int adau17x1_setup_firmware(struct snd_soc_component *component, + unsigned int rate) { int ret; - int dspsr; + int dspsr, dsp_run; + struct adau *adau = snd_soc_component_get_drvdata(component); + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + + snd_soc_dapm_mutex_lock(dapm); ret = regmap_read(adau->regmap, ADAU17X1_DSP_SAMPLING_RATE, &dspsr); if (ret) - return ret; + goto err; + + ret = regmap_read(adau->regmap, ADAU17X1_DSP_RUN, &dsp_run); + if (ret) + goto err; regmap_write(adau->regmap, ADAU17X1_DSP_ENABLE, 1); regmap_write(adau->regmap, ADAU17X1_DSP_SAMPLING_RATE, 0xf); + regmap_write(adau->regmap, ADAU17X1_DSP_RUN, 0); ret = sigmadsp_setup(adau->sigmadsp, rate); if (ret) { regmap_write(adau->regmap, ADAU17X1_DSP_ENABLE, 0); - return ret; + goto err; } regmap_write(adau->regmap, ADAU17X1_DSP_SAMPLING_RATE, dspsr); + regmap_write(adau->regmap, ADAU17X1_DSP_RUN, dsp_run); - return 0; +err: + snd_soc_dapm_mutex_unlock(dapm); + + return ret; } EXPORT_SYMBOL_GPL(adau17x1_setup_firmware); diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h index a7b1cb770814..e6fe87beec07 100644 --- a/sound/soc/codecs/adau17x1.h +++ b/sound/soc/codecs/adau17x1.h @@ -68,7 +68,8 @@ int adau17x1_resume(struct snd_soc_component *component); extern const struct snd_soc_dai_ops adau17x1_dai_ops; -int adau17x1_setup_firmware(struct adau *adau, unsigned int rate); +int adau17x1_setup_firmware(struct snd_soc_component *component, + unsigned int rate); bool adau17x1_has_dsp(struct adau *adau); #define ADAU17X1_CLOCK_CONTROL 0x4000 -- 2.11.0

3 6

[alsa-devel] [PATCH] ALSA: core: Report audio_tstamp in snd_pcm_sync_ptr
by Takashi Iwai 16 Apr '18

16 Apr '18

From: David Henningsson <diwic(a)ubuntu.com> It looks like a simple mistake that this struct member was forgotten. Audio_tstamp isn't used much, and on some archs (such as x86) this ioctl is not used by default, so that might be the reason why this has slipped for so long. Signed-off-by: David Henningsson <diwic(a)ubuntu.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_native.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 35ffccea94c3..632228bfd36b 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2751,6 +2751,7 @@ static int snd_pcm_sync_ptr(struct snd_pcm_substream *substream, sync_ptr.s.status.hw_ptr = status->hw_ptr; sync_ptr.s.status.tstamp = status->tstamp; sync_ptr.s.status.suspended_state = status->suspended_state; + sync_ptr.s.status.audio_tstamp = status->audio_tstamp; snd_pcm_stream_unlock_irq(substream); if (copy_to_user(_sync_ptr, &sync_ptr, sizeof(sync_ptr))) return -EFAULT; -- 2.16.3

1 0

[alsa-devel] [PATCH] ALSA: hda - New VIA controller suppor no-snoop path
by David Wang 16 Apr '18

16 Apr '18

This patch is used to tell kernel that new VIA HDAC controller also support no-snoop path. Signed-off-by: David Wang <davidwang(a)zhaoxin.com> --- sound/pci/hda/hda_intel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 96143df..439e8e9 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1640,7 +1640,7 @@ static void azx_check_snoop_available(struct azx *chip) */ u8 val; pci_read_config_byte(chip->pci, 0x42, &val); - if (!(val & 0x80) && chip->pci->revision == 0x30) + if (!(val & 0x80) && (chip->pci->revision == 0x30 || chip->pci->revision == 0x20)) snoop = false; } -- 1.9.1

2 1

[alsa-devel] [PATCH 0/2] ASoC: rsnd: reborn HDMI sound probing
by Kuninori Morimoto 16 Apr '18

16 Apr '18

Hi Mark I noticed rsnd HDMI sound probing had been broken since v4.15. No board is using it at this point, thus, there is no real issue. Kuninori Morimoto (2): ASoC: rsnd: add RSND_GEN3 for R-Car Gen3 ASoC: rsnd: don't assume node full path name for HDMI probing sound/soc/sh/rcar/core.c | 2 +- sound/soc/sh/rcar/dma.c | 4 ++-- sound/soc/sh/rcar/gen.c | 3 ++- sound/soc/sh/rcar/rsnd.h | 2 ++ sound/soc/sh/rcar/ssi.c | 11 +++++++++-- 5 files changed, 16 insertions(+), 6 deletions(-) -- 1.9.1

2 8

[alsa-devel] possible bug: input and output devices are getting mixed up
by Raymond Jennings 16 Apr '18

16 Apr '18

I used arecord -L to list some things, and I got weird results, saying among other things that my webcam was capable of surround sound output even though it's strictly an input device. I already reported it downstream with my distro but so far no news. I was told it was a bug with alsa. As for the card in question, U0x46d0x825 is my logitech brand webcam connected via USB. $ arecord -L null Discard all samples (playback) or generate zero samples (capture) pulse PulseAudio Sound Server sysdefault:CARD=PCH HDA Intel PCH, ALC887-VD Analog Default Audio Device front:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog Front speakers sysdefault:CARD=U0x46d0x825 USB Device 0x46d:0x825, USB Audio Default Audio Device front:CARD=U0x46d0x825,DEV=0 USB Device 0x46d:0x825, USB Audio Front speakers surround21:CARD=U0x46d0x825,DEV=0 USB Device 0x46d:0x825, USB Audio 2.1 Surround output to Front and Subwoofer speakers surround40:CARD=U0x46d0x825,DEV=0 USB Device 0x46d:0x825, USB Audio 4.0 Surround output to Front and Rear speakers surround41:CARD=U0x46d0x825,DEV=0 USB Device 0x46d:0x825, USB Audio 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=U0x46d0x825,DEV=0 USB Device 0x46d:0x825, USB Audio 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=U0x46d0x825,DEV=0 USB Device 0x46d:0x825, USB Audio 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=U0x46d0x825,DEV=0 USB Device 0x46d:0x825, USB Audio 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=U0x46d0x825,DEV=0 USB Device 0x46d:0x825, USB Audio IEC958 (S/PDIF) Digital Audio Output

1 0

[alsa-devel] Avoiding wordexp prevents environment variables being used
by Mark Hills 13 Apr '18

13 Apr '18

I just came up against the patch below; it prevents useful snippets of alsa-conf like this: @hooks [ { func load files [ "~/.asoundrc-$HOSTNAME" ] errors false } ] as the evalutation of all but "~" has been removed. Seems like removal of a perfectly good feature in the name of security; because wordexp() 1) is not used (and should not be used) on data originating from an untrusted source 2) is already used with WRDE_NOCMD, which the same POSIX spec documents as: "The WRDE_NOCMD flag is provided for applications that, for security or other reasons, want to prevent a user from executing shell commands." 3) on glibc can be seen (with strace) not to execute other commands If one is to treat the POSIX doc as gospel (as cited by the patch) the cause of firefox (circa July 2017) not working would actually be that musl does not honour WRDE_NOCMD to the letter. I agree the spec of wordexp() could be more useful, though. Also, hypothesising the attacks of an already-compromised application would get into a sticky conversation about the thread safety of getenv("HOME") (and associated buffer wrangling) vs. a library function being used for its intended purpose. In practice, Firefox may have moved on here (no ALSA support anymore) so should quirks of its sandbox be driving this? -- Mark >From cb34cee0d8da2fb131986d5782ddf5cec985c532 Mon Sep 17 00:00:00 2001 From: Natanael Copa <ncopa(a)alpinelinux.org> Date: Fri, 14 Jul 2017 18:47:05 +0200 Subject: [PATCH] snd_user_file: avoid use wordexp As suggested in POSIX[1], wordexp might execute the shell. If the libc implementation does so, it will break the firefox sandbox which does not allow exec. This happened on Alpine Linux with musl libc[2]. Since we cannot guarantee that the system wordexp implementation does not execute shell, we cannot really use it, and need to implement the ~/ expansion ourselves. We provide a configure option --with-wordexp for users that still may need it, but we leave this off by default because wordexp is a large attack vector and it is better to avoid it. [1]: http://pubs.opengroup.org/onlinepubs/9699919799/functions/wordexp.html#tag_… [2]: http://bugs.alpinelinux.org/issues/7454#note-2 Signed-off-by: Natanael Copa <ncopa(a)alpinelinux.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- configure.ac | 19 ++++++++++++++++- src/userfile.c | 65 ++++++++++++++++++++++++++++++++++++++++++++++++++-------- 2 files changed, 75 insertions(+), 9 deletions(-) diff --git a/configure.ac b/configure.ac index 26e5d125..fbcfa829 100644 --- a/configure.ac +++ b/configure.ac @@ -303,8 +303,25 @@ fi AC_SUBST(ALSA_DEPLIBS) +dnl Check for use of wordexp... +AC_MSG_CHECKING(for use of wordexp) +AC_ARG_WITH(wordexp, + AS_HELP_STRING([--with-wordexp], + [Use wordexp when expanding configs (default = no)]), + [case "$withval" in + y|yes) wordexp=yes ;; + *) wordexp=no ;; + esac],) +if test "$wordexp" = "yes" ; then + AC_DEFINE(HAVE_WORDEXP, "1", [Enable use of wordexp]) + AC_MSG_RESULT(yes) + AC_CHECK_HEADER([wordexp.h],[], [AC_MSG_ERROR([Couldn't find wordexp.h])]) +else + AC_MSG_RESULT(no) +fi + dnl Check for headers -AC_CHECK_HEADERS([wordexp.h endian.h sys/endian.h sys/shm.h]) +AC_CHECK_HEADERS([endian.h sys/endian.h sys/shm.h]) dnl Check for resmgr support... AC_MSG_CHECKING(for resmgr support) diff --git a/src/userfile.c b/src/userfile.c index 72779da4..f2145470 100644 --- a/src/userfile.c +++ b/src/userfile.c @@ -21,6 +21,7 @@ #include <config.h> #include <string.h> #include <errno.h> +#include <assert.h> /** * \brief Get the full file name @@ -32,14 +33,13 @@ * stores the first matchine one. The returned string is strdup'ed. */ -#ifdef HAVE_WORDEXP_H +#ifdef HAVE_WORDEXP #include <wordexp.h> -#include <assert.h> int snd_user_file(const char *file, char **result) { wordexp_t we; int err; - + assert(file && result); err = wordexp(file, &we, WRDE_NOCMD); switch (err) { @@ -61,13 +61,62 @@ int snd_user_file(const char *file, char **result) return 0; } -#else /* !HAVE_WORDEXP_H */ -/* just copy the string - would be nicer to expand by ourselves, though... */ +#else /* !HAVE_WORDEX */ + +#include <sys/types.h> +#include <unistd.h> +#include <pwd.h> +#include <stdio.h> +#include <stdlib.h> + int snd_user_file(const char *file, char **result) { - *result = strdup(file); - if (! *result) + int err; + size_t len; + char *buf = NULL; + + assert(file && result); + *result = NULL; + + /* expand ~/ if needed */ + if (file[0] == '~' && file[1] == '/') { + const char *home = getenv("HOME"); + if (home == NULL) { + struct passwd pwent, *p = NULL; + uid_t id = getuid(); + size_t bufsize = 1024; + + buf = malloc(bufsize); + if (buf == NULL) + goto out; + + while ((err = getpwuid_r(id, &pwent, buf, bufsize, &p)) == ERANGE) { + char *newbuf; + bufsize += 1024; + if (bufsize < 1024) + break; + newbuf = realloc(buf, bufsize); + if (newbuf == NULL) + goto out; + buf = newbuf; + } + home = err ? "" : pwent.pw_dir; + } + len = strlen(home) + strlen(&file[2]) + 2; + *result = malloc(len); + if (*result) + snprintf(*result, len, "%s/%s", home, &file[2]); + } else { + *result = strdup(file); + } + +out: + if (buf) + free(buf); + + if (*result == NULL) return -ENOMEM; return 0; } -#endif /* HAVE_WORDEXP_H */ + +#endif /* HAVE_WORDEXP */ -- 2.14.1

2 3

[alsa-devel] [PATCH] ASoC: atmel_ssc_dai: fix spelling mistake: "Stoping" -> "Stopping"
by Colin King 13 Apr '18

13 Apr '18

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to spelling mistake in pr_debug message text Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/atmel/atmel_ssc_dai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index a1e2c5682dcd..1c7af0ca98ec 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -820,7 +820,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, if (ret < 0) { printk(KERN_WARNING "atmel_ssc_dai: request_irq failure\n"); - pr_debug("Atmel_ssc_dai: Stoping clock\n"); + pr_debug("Atmel_ssc_dai: Stopping clock\n"); clk_disable(ssc_p->ssc->clk); return ret; } -- 2.15.1

8 15

[alsa-devel] [PATCH][RFC] ASoC: wm8350: remove snd_soc_codec
by Kuninori Morimoto 13 Apr '18

13 Apr '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> codec is replace to component. It seems no-one is using it, Let's remove it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Hi Mark, Liam I think no-one is using snd_soc_codec, but please double-check it. I added [RFC] on this patch include/linux/mfd/wm8350/audio.h | 3 --- 1 file changed, 3 deletions(-) diff --git a/include/linux/mfd/wm8350/audio.h b/include/linux/mfd/wm8350/audio.h index bd581c6..0bc41c4c 100644 --- a/include/linux/mfd/wm8350/audio.h +++ b/include/linux/mfd/wm8350/audio.h @@ -617,11 +617,8 @@ struct wm8350_audio_platform_data { u32 codec_current_charge:2; /* codec current @ vmid charge */ }; -struct snd_soc_codec; - struct wm8350_codec { struct platform_device *pdev; - struct snd_soc_codec *codec; struct wm8350_audio_platform_data *platform_data; }; -- 1.9.1

3 6

[alsa-devel] [PATCH] sound: soc: intel: bxt_da7219_max98357a: Replace GFP_ATOMIC with GFP_KERNEL in broxton_audio_probe
by Jia-Ju Bai 13 Apr '18

13 Apr '18

broxton_audio_probe() is never called in atomic context. This function is only set as ".probe" in "struct platform_driver". Despite never getting called from atomic context, broxton_audio_probe() calls devm_kzalloc() with GFP_ATOMIC, which waits busily for allocation. GFP_ATOMIC is not necessary and can be replaced with GFP_KERNEL, to avoid busy waiting and improve the possibility of sucessful allocation. This is found by a static analysis tool named DCNS written by myself. Signed-off-by: Jia-Ju Bai <baijiaju1990(a)gmail.com> --- sound/soc/intel/boards/bxt_da7219_max98357a.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index ce35ec7..f4361b7 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -585,7 +585,7 @@ static int broxton_audio_probe(struct platform_device *pdev) { struct bxt_card_private *ctx; - ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) return -ENOMEM; -- 1.9.1

5 9

[alsa-devel] [PATCH] sound: soc: intel: bxt_rt298: Replace GFP_ATOMIC with GFP_KERNEL in broxton_audio_probe
by Jia-Ju Bai 13 Apr '18

13 Apr '18

broxton_audio_probe() is never called in atomic context. This function is only set as ".probe" in "struct platform_driver". Despite never getting called from atomic context, broxton_audio_probe() calls devm_kzalloc() with GFP_ATOMIC, which waits busily for allocation. GFP_ATOMIC is not necessary and can be replaced with GFP_KERNEL, to avoid busy waiting and improve the possibility of sucessful allocation. This is found by a static analysis tool named DCNS written by myself. Signed-off-by: Jia-Ju Bai <baijiaju1990(a)gmail.com> --- sound/soc/intel/boards/bxt_rt298.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c index 7843104..8257830 100644 --- a/sound/soc/intel/boards/bxt_rt298.c +++ b/sound/soc/intel/boards/bxt_rt298.c @@ -578,7 +578,7 @@ static int broxton_audio_probe(struct platform_device *pdev) } } - ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) return -ENOMEM; -- 1.9.1

2 2

[alsa-devel] Applied "ASoC: intel: bytcr_rt5640: Replace GFP_ATOMIC with GFP_KERNEL in snd_byt_rt5640_mc_probe" to the asoc tree
by Mark Brown 13 Apr '18

13 Apr '18

The patch ASoC: intel: bytcr_rt5640: Replace GFP_ATOMIC with GFP_KERNEL in snd_byt_rt5640_mc_probe has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From f5c74c2918e51be14bc9c11c76fa118e354f4789 Mon Sep 17 00:00:00 2001 From: Jia-Ju Bai <baijiaju1990(a)gmail.com> Date: Mon, 9 Apr 2018 18:47:37 +0800 Subject: [PATCH] ASoC: intel: bytcr_rt5640: Replace GFP_ATOMIC with GFP_KERNEL in snd_byt_rt5640_mc_probe snd_byt_rt5640_mc_probe() is never called in atomic context. This function is only set as ".probe" in "struct platform_driver". Despite never getting called from atomic context, snd_byt_rt5640_mc_probe() calls devm_kzalloc() with GFP_ATOMIC, which waits busily for allocation. GFP_ATOMIC is not necessary and can be replaced with GFP_KERNEL, to avoid busy waiting and improve the possibility of sucessful allocation. This is found by a static analysis tool named DCNS written by myself. Signed-off-by: Jia-Ju Bai <baijiaju1990(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index a8d8bff788e7..ad5fcd5a1762 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -744,7 +744,7 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) int i; is_bytcr = false; - priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_ATOMIC); + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); if (!priv) return -ENOMEM; -- 2.17.0

1 1

[alsa-devel] [PATCH] sound: soc: intel: cht_bsw_max98090_ti: Replace GFP_ATOMIC with GFP_KERNEL in snd_cht_mc_probe
by Jia-Ju Bai 13 Apr '18

13 Apr '18

snd_cht_mc_probe() is never called in atomic context. This function is only set as ".probe" in "struct platform_driver". Despite never getting called from atomic context, snd_cht_mc_probe() calls devm_kzalloc() with GFP_ATOMIC, which waits busily for allocation. GFP_ATOMIC is not necessary and can be replaced with GFP_KERNEL, to avoid busy waiting and improve the possibility of sucessful allocation. This is found by a static analysis tool named DCNS written by myself. Signed-off-by: Jia-Ju Bai <baijiaju1990(a)gmail.com> --- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index 20755ec..bc2ec36 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -280,7 +280,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) int ret_val = 0; struct cht_mc_private *drv; - drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_ATOMIC); + drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL); if (!drv) return -ENOMEM; -- 1.9.1

2 2

[alsa-devel] Applied "ASoC: intel: cht_bsw_rt5645: Replace GFP_ATOMIC with GFP_KERNEL in snd_cht_mc_probe" to the asoc tree
by Mark Brown 13 Apr '18

13 Apr '18

The patch ASoC: intel: cht_bsw_rt5645: Replace GFP_ATOMIC with GFP_KERNEL in snd_cht_mc_probe has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From ca243b56cc07b63da38bddfd30d48792f442cca8 Mon Sep 17 00:00:00 2001 From: Jia-Ju Bai <baijiaju1990(a)gmail.com> Date: Mon, 9 Apr 2018 18:48:31 +0800 Subject: [PATCH] ASoC: intel: cht_bsw_rt5645: Replace GFP_ATOMIC with GFP_KERNEL in snd_cht_mc_probe snd_cht_mc_probe() is never called in atomic context. This function is only set as ".probe" in "struct platform_driver". Despite never getting called from atomic context, snd_cht_mc_probe() calls devm_kzalloc() with GFP_ATOMIC, which waits busily for allocation. GFP_ATOMIC is not necessary and can be replaced with GFP_KERNEL, to avoid busy waiting and improve the possibility of sucessful allocation. This is found by a static analysis tool named DCNS written by myself. Signed-off-by: Jia-Ju Bai <baijiaju1990(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/cht_bsw_rt5645.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 49ba1a956a06..f5a5ea6a093c 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -539,7 +539,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) int ret_val = 0; int i; - drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_ATOMIC); + drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL); if (!drv) return -ENOMEM; -- 2.17.0

1 1

[alsa-devel] [PATCH] sound: soc: intel: skl_nau88l25_max98357a: Replace GFP_ATOMIC with GFP_KERNEL in skylake_audio_probe
by Jia-Ju Bai 13 Apr '18

13 Apr '18

skylake_audio_probe() is never called in atomic context. This function is only set as ".probe" in "struct platform_driver". Despite never getting called from atomic context, skylake_audio_probe() calls devm_kzalloc() with GFP_ATOMIC, which waits busily for allocation. GFP_ATOMIC is not necessary and can be replaced with GFP_KERNEL, to avoid busy waiting and improve the possibility of sucessful allocation. This is found by a static analysis tool named DCNS written by myself. Signed-off-by: Jia-Ju Bai <baijiaju1990(a)gmail.com> --- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c index 5ed0aa2..8b80d1a 100644 --- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c +++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c @@ -657,7 +657,7 @@ static int skylake_audio_probe(struct platform_device *pdev) struct skl_nau8825_private *ctx; struct skl_machine_pdata *pdata; - ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) return -ENOMEM; -- 1.9.1

2 2

[alsa-devel] [PATCH] sound: soc: intel: skl_nau88l25_ssm4567: Replace GFP_ATOMIC with GFP_KERNEL in skylake_audio_probe
by Jia-Ju Bai 13 Apr '18

13 Apr '18

skylake_audio_probe() is never called in atomic context. This function is only set as ".probe" in "struct platform_driver". Despite never getting called from atomic context, skylake_audio_probe() calls devm_kzalloc() with GFP_ATOMIC, which waits busily for allocation. GFP_ATOMIC is not necessary and can be replaced with GFP_KERNEL, to avoid busy waiting and improve the possibility of sucessful allocation. This is found by a static analysis tool named DCNS written by myself. Signed-off-by: Jia-Ju Bai <baijiaju1990(a)gmail.com> --- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c index 01b8b14..033f9d3 100644 --- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c +++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c @@ -710,7 +710,7 @@ static int skylake_audio_probe(struct platform_device *pdev) struct skl_nau88125_private *ctx; struct skl_machine_pdata *pdata; - ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) return -ENOMEM; -- 1.9.1

2 2

[alsa-devel] [PATCH] sound: soc: intel: skl_rt286: Replace GFP_ATOMIC with GFP_KERNEL in skylake_audio_probe
by Jia-Ju Bai 13 Apr '18

13 Apr '18

skylake_audio_probe() is never called in atomic context. This function is only set as ".probe" in "struct platform_driver". Despite never getting called from atomic context, skylake_audio_probe() calls devm_kzalloc() with GFP_ATOMIC, which waits busily for allocation. GFP_ATOMIC is not necessary and can be replaced with GFP_KERNEL, to avoid busy waiting and improve the possibility of sucessful allocation. This is found by a static analysis tool named DCNS written by myself. Signed-off-by: Jia-Ju Bai <baijiaju1990(a)gmail.com> --- sound/soc/intel/boards/skl_rt286.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c index 2bc4cfc..bb7187b 100644 --- a/sound/soc/intel/boards/skl_rt286.c +++ b/sound/soc/intel/boards/skl_rt286.c @@ -527,7 +527,7 @@ static int skylake_audio_probe(struct platform_device *pdev) { struct skl_rt286_private *ctx; - ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) return -ENOMEM; -- 1.9.1

2 2

[alsa-devel] [PATCH] ASoC: rsnd: makes rsnd_cmd_mod_get() static
by Kuninori Morimoto 13 Apr '18

13 Apr '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> rsnd_cmd_mod_get() is used from cmd.c only. Let's makes it static function Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/cmd.c | 15 +++++++-------- sound/soc/sh/rcar/rsnd.h | 1 - 2 files changed, 7 insertions(+), 9 deletions(-) diff --git a/sound/soc/sh/rcar/cmd.c b/sound/soc/sh/rcar/cmd.c index f1d4fb5..4221937 100644 --- a/sound/soc/sh/rcar/cmd.c +++ b/sound/soc/sh/rcar/cmd.c @@ -125,6 +125,13 @@ static int rsnd_cmd_stop(struct rsnd_mod *mod, .stop = rsnd_cmd_stop, }; +static struct rsnd_mod *rsnd_cmd_mod_get(struct rsnd_priv *priv, int id) +{ + if (WARN_ON(id < 0 || id >= rsnd_cmd_nr(priv))) + id = 0; + + return rsnd_mod_get((struct rsnd_cmd *)(priv->cmd) + id); +} int rsnd_cmd_attach(struct rsnd_dai_stream *io, int id) { struct rsnd_priv *priv = rsnd_io_to_priv(io); @@ -133,14 +140,6 @@ int rsnd_cmd_attach(struct rsnd_dai_stream *io, int id) return rsnd_dai_connect(mod, io, mod->type); } -struct rsnd_mod *rsnd_cmd_mod_get(struct rsnd_priv *priv, int id) -{ - if (WARN_ON(id < 0 || id >= rsnd_cmd_nr(priv))) - id = 0; - - return rsnd_mod_get((struct rsnd_cmd *)(priv->cmd) + id); -} - int rsnd_cmd_probe(struct rsnd_priv *priv) { struct device *dev = rsnd_priv_to_dev(priv); diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 172c8d6..ab4d555 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -775,7 +775,6 @@ unsigned int rsnd_src_get_rate(struct rsnd_priv *priv, int rsnd_cmd_probe(struct rsnd_priv *priv); void rsnd_cmd_remove(struct rsnd_priv *priv); int rsnd_cmd_attach(struct rsnd_dai_stream *io, int id); -struct rsnd_mod *rsnd_cmd_mod_get(struct rsnd_priv *priv, int id); void rsnd_mod_make_sure(struct rsnd_mod *mod, enum rsnd_mod_type type); #ifdef DEBUG -- 1.9.1

2 2

[alsa-devel] [PATCH 1/3] ASoC: TSCS42xx: Shorten lines and other cleanup
by Steven Eckhoff 13 Apr '18

13 Apr '18

Shorten lines greater than 80 chars Add const to struct snd_soc_component_driver Signed-off-by: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> --- sound/soc/codecs/tscs42xx.c | 87 ++++++++++++++++++++++++++++----------------- 1 file changed, 55 insertions(+), 32 deletions(-) diff --git a/sound/soc/codecs/tscs42xx.c b/sound/soc/codecs/tscs42xx.c index cfc71b5411ee..37636d6efe60 100644 --- a/sound/soc/codecs/tscs42xx.c +++ b/sound/soc/codecs/tscs42xx.c @@ -204,7 +204,8 @@ static int power_up_audio_plls(struct snd_soc_component *component) break; default: ret = -EINVAL; - dev_err(component->dev, "Unrecognized PLL output freq (%d)\n", ret); + dev_err(component->dev, + "Unrecognized PLL output freq (%d)\n", ret); return ret; } @@ -261,7 +262,8 @@ static int power_down_audio_plls(struct snd_soc_component *component) static int coeff_ram_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); struct tscs42xx *tscs42xx = snd_soc_component_get_drvdata(component); struct coeff_ram_ctl *ctl = (struct coeff_ram_ctl *)kcontrol->private_value; @@ -280,7 +282,8 @@ static int coeff_ram_get(struct snd_kcontrol *kcontrol, static int coeff_ram_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); struct tscs42xx *tscs42xx = snd_soc_component_get_drvdata(component); struct coeff_ram_ctl *ctl = (struct coeff_ram_ctl *)kcontrol->private_value; @@ -363,7 +366,8 @@ static int dapm_micb_event(struct snd_soc_dapm_widget *w, static int pll_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); int ret; if (SND_SOC_DAPM_EVENT_ON(event)) @@ -377,7 +381,8 @@ static int pll_event(struct snd_soc_dapm_widget *w, static int dac_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); struct tscs42xx *tscs42xx = snd_soc_component_get_drvdata(component); int ret; @@ -819,16 +824,19 @@ static int setup_sample_format(struct snd_soc_component *component, dev_err(component->dev, "Unsupported format width (%d)\n", ret); return ret; } - ret = snd_soc_component_update_bits(component, R_AIC1, RM_AIC1_WL, width); + ret = snd_soc_component_update_bits(component, + R_AIC1, RM_AIC1_WL, width); if (ret < 0) { - dev_err(component->dev, "Failed to set sample width (%d)\n", ret); + dev_err(component->dev, + "Failed to set sample width (%d)\n", ret); return ret; } return 0; } -static int setup_sample_rate(struct snd_soc_component *component, unsigned int rate) +static int setup_sample_rate(struct snd_soc_component *component, + unsigned int rate) { struct tscs42xx *tscs42xx = snd_soc_component_get_drvdata(component); unsigned int br, bm; @@ -881,24 +889,32 @@ static int setup_sample_rate(struct snd_soc_component *component, unsigned int r } /* DAC and ADC share bit and frame clock */ - ret = snd_soc_component_update_bits(component, R_DACSR, RM_DACSR_DBR, br); + ret = snd_soc_component_update_bits(component, + R_DACSR, RM_DACSR_DBR, br); if (ret < 0) { - dev_err(component->dev, "Failed to update register (%d)\n", ret); + dev_err(component->dev, + "Failed to update register (%d)\n", ret); return ret; } - ret = snd_soc_component_update_bits(component, R_DACSR, RM_DACSR_DBM, bm); + ret = snd_soc_component_update_bits(component, + R_DACSR, RM_DACSR_DBM, bm); if (ret < 0) { - dev_err(component->dev, "Failed to update register (%d)\n", ret); + dev_err(component->dev, + "Failed to update register (%d)\n", ret); return ret; } - ret = snd_soc_component_update_bits(component, R_ADCSR, RM_DACSR_DBR, br); + ret = snd_soc_component_update_bits(component, + R_ADCSR, RM_DACSR_DBR, br); if (ret < 0) { - dev_err(component->dev, "Failed to update register (%d)\n", ret); + dev_err(component->dev, + "Failed to update register (%d)\n", ret); return ret; } - ret = snd_soc_component_update_bits(component, R_ADCSR, RM_DACSR_DBM, bm); + ret = snd_soc_component_update_bits(component, + R_ADCSR, RM_DACSR_DBM, bm); if (ret < 0) { - dev_err(component->dev, "Failed to update register (%d)\n", ret); + dev_err(component->dev, + "Failed to update register (%d)\n", ret); return ret; } @@ -1076,7 +1092,8 @@ static int tscs42xx_hw_params(struct snd_pcm_substream *substream, ret = setup_sample_rate(component, params_rate(params)); if (ret < 0) { - dev_err(component->dev, "Failed to setup sample rate (%d)\n", ret); + dev_err(component->dev, + "Failed to setup sample rate (%d)\n", ret); return ret; } @@ -1087,7 +1104,8 @@ static inline int dac_mute(struct snd_soc_component *component) { int ret; - ret = snd_soc_component_update_bits(component, R_CNVRTR1, RM_CNVRTR1_DACMU, + ret = snd_soc_component_update_bits(component, + R_CNVRTR1, RM_CNVRTR1_DACMU, RV_CNVRTR1_DACMU_ENABLE); if (ret < 0) { dev_err(component->dev, "Failed to mute DAC (%d)\n", @@ -1102,7 +1120,8 @@ static inline int dac_unmute(struct snd_soc_component *component) { int ret; - ret = snd_soc_component_update_bits(component, R_CNVRTR1, RM_CNVRTR1_DACMU, + ret = snd_soc_component_update_bits(component, + R_CNVRTR1, RM_CNVRTR1_DACMU, RV_CNVRTR1_DACMU_DISABLE); if (ret < 0) { dev_err(component->dev, "Failed to unmute DAC (%d)\n", @@ -1117,8 +1136,8 @@ static inline int adc_mute(struct snd_soc_component *component) { int ret; - ret = snd_soc_component_update_bits(component, R_CNVRTR0, RM_CNVRTR0_ADCMU, - RV_CNVRTR0_ADCMU_ENABLE); + ret = snd_soc_component_update_bits(component, + R_CNVRTR0, RM_CNVRTR0_ADCMU, RV_CNVRTR0_ADCMU_ENABLE); if (ret < 0) { dev_err(component->dev, "Failed to mute ADC (%d)\n", ret); @@ -1132,8 +1151,8 @@ static inline int adc_unmute(struct snd_soc_component *component) { int ret; - ret = snd_soc_component_update_bits(component, R_CNVRTR0, RM_CNVRTR0_ADCMU, - RV_CNVRTR0_ADCMU_DISABLE); + ret = snd_soc_component_update_bits(component, + R_CNVRTR0, RM_CNVRTR0_ADCMU, RV_CNVRTR0_ADCMU_DISABLE); if (ret < 0) { dev_err(component->dev, "Failed to unmute ADC (%d)\n", ret); @@ -1171,8 +1190,8 @@ static int tscs42xx_set_dai_fmt(struct snd_soc_dai *codec_dai, /* Slave mode not supported since it needs always-on frame clock */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: - ret = snd_soc_component_update_bits(component, R_AIC1, RM_AIC1_MS, - RV_AIC1_MS_MASTER); + ret = snd_soc_component_update_bits(component, + R_AIC1, RM_AIC1_MS, RV_AIC1_MS_MASTER); if (ret < 0) { dev_err(component->dev, "Failed to set codec DAI master (%d)\n", ret); @@ -1211,14 +1230,18 @@ static int tscs42xx_set_dai_bclk_ratio(struct snd_soc_dai *codec_dai, return -EINVAL; } - ret = snd_soc_component_update_bits(component, R_DACSR, RM_DACSR_DBCM, value); + ret = snd_soc_component_update_bits(component, + R_DACSR, RM_DACSR_DBCM, value); if (ret < 0) { - dev_err(component->dev, "Failed to set DAC BCLK ratio (%d)\n", ret); + dev_err(component->dev, + "Failed to set DAC BCLK ratio (%d)\n", ret); return ret; } - ret = snd_soc_component_update_bits(component, R_ADCSR, RM_ADCSR_ABCM, value); + ret = snd_soc_component_update_bits(component, + R_ADCSR, RM_ADCSR_ABCM, value); if (ret < 0) { - dev_err(component->dev, "Failed to set ADC BCLK ratio (%d)\n", ret); + dev_err(component->dev, + "Failed to set ADC BCLK ratio (%d)\n", ret); return ret; } @@ -1309,7 +1332,7 @@ static int part_is_valid(struct tscs42xx *tscs42xx) }; } -static struct snd_soc_component_driver soc_codec_dev_tscs42xx = { +static const struct snd_soc_component_driver soc_codec_dev_tscs42xx = { .dapm_widgets = tscs42xx_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(tscs42xx_dapm_widgets), .dapm_routes = tscs42xx_intercon, @@ -1416,8 +1439,8 @@ static int tscs42xx_i2c_probe(struct i2c_client *i2c, mutex_init(&tscs42xx->coeff_ram_lock); mutex_init(&tscs42xx->pll_lock); - ret = devm_snd_soc_register_component(tscs42xx->dev, &soc_codec_dev_tscs42xx, - &tscs42xx_dai, 1); + ret = devm_snd_soc_register_component(tscs42xx->dev, + &soc_codec_dev_tscs42xx, &tscs42xx_dai, 1); if (ret) { dev_err(tscs42xx->dev, "Failed to register codec (%d)\n", ret); return ret; -- 2.15.1

2 2

[alsa-devel] [PATCH 2/3] ASoC: TSCS42xx: Cleanup private data members
by Steven Eckhoff 13 Apr '18

13 Apr '18

In the TSCS42xx driver there are some unnecesarry private data members Remove blrcm from private data Remove dev from private data Signed-off-by: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> --- sound/soc/codecs/tscs42xx.c | 16 ++++++---------- 1 file changed, 6 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/tscs42xx.c b/sound/soc/codecs/tscs42xx.c index 37636d6efe60..d41852a37ac6 100644 --- a/sound/soc/codecs/tscs42xx.c +++ b/sound/soc/codecs/tscs42xx.c @@ -31,7 +31,6 @@ struct tscs42xx { int bclk_ratio; int samplerate; - unsigned int blrcm; struct mutex audio_params_lock; u8 coeff_ram[COEFF_RAM_SIZE]; @@ -41,8 +40,6 @@ struct tscs42xx { struct mutex pll_lock; struct regmap *regmap; - - struct device *dev; }; struct coeff_ram_ctl { @@ -1404,12 +1401,11 @@ static int tscs42xx_i2c_probe(struct i2c_client *i2c, return ret; } i2c_set_clientdata(i2c, tscs42xx); - tscs42xx->dev = &i2c->dev; tscs42xx->regmap = devm_regmap_init_i2c(i2c, &tscs42xx_regmap); if (IS_ERR(tscs42xx->regmap)) { ret = PTR_ERR(tscs42xx->regmap); - dev_err(tscs42xx->dev, "Failed to allocate regmap (%d)\n", ret); + dev_err(&i2c->dev, "Failed to allocate regmap (%d)\n", ret); return ret; } @@ -1417,21 +1413,21 @@ static int tscs42xx_i2c_probe(struct i2c_client *i2c, ret = part_is_valid(tscs42xx); if (ret <= 0) { - dev_err(tscs42xx->dev, "No valid part (%d)\n", ret); + dev_err(&i2c->dev, "No valid part (%d)\n", ret); ret = -ENODEV; return ret; } ret = regmap_write(tscs42xx->regmap, R_RESET, RV_RESET_ENABLE); if (ret < 0) { - dev_err(tscs42xx->dev, "Failed to reset device (%d)\n", ret); + dev_err(&i2c->dev, "Failed to reset device (%d)\n", ret); return ret; } ret = regmap_register_patch(tscs42xx->regmap, tscs42xx_patch, ARRAY_SIZE(tscs42xx_patch)); if (ret < 0) { - dev_err(tscs42xx->dev, "Failed to apply patch (%d)\n", ret); + dev_err(&i2c->dev, "Failed to apply patch (%d)\n", ret); return ret; } @@ -1439,10 +1435,10 @@ static int tscs42xx_i2c_probe(struct i2c_client *i2c, mutex_init(&tscs42xx->coeff_ram_lock); mutex_init(&tscs42xx->pll_lock); - ret = devm_snd_soc_register_component(tscs42xx->dev, + ret = devm_snd_soc_register_component(&i2c->dev, &soc_codec_dev_tscs42xx, &tscs42xx_dai, 1); if (ret) { - dev_err(tscs42xx->dev, "Failed to register codec (%d)\n", ret); + dev_err(&i2c->dev, "Failed to register codec (%d)\n", ret); return ret; } -- 2.15.1

2 2

[alsa-devel] [PATCH 3/3 v3] ASoC: TSCS42xx: Add CCF support to get sysclk
by Steven Eckhoff 13 Apr '18

13 Apr '18

The TSCS42xx relies on set_sysclk to get a unique clock id and rate, which prevents it from being used with the simple-card. Remove set_sysclk callback Add CCF support to get clock id and rate Add clocks and clock-names to device tree binding Signed-off-by: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> --- .../devicetree/bindings/sound/tscs42xx.txt | 6 ++ sound/soc/codecs/tscs42xx.c | 104 ++++++++++++++------- sound/soc/codecs/tscs42xx.h | 2 +- 3 files changed, 75 insertions(+), 37 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/tscs42xx.txt b/Documentation/devicetree/bindings/sound/tscs42xx.txt index 2ac2f0996697..7eea32e9d078 100644 --- a/Documentation/devicetree/bindings/sound/tscs42xx.txt +++ b/Documentation/devicetree/bindings/sound/tscs42xx.txt @@ -8,9 +8,15 @@ Required Properties: - reg : <0x71> for analog mic <0x69> for digital mic + - clock-names: Must one of the following "mclk1", "xtal", "mclk2" + + - clocks: phandle of the clock that provides the codec sysclk + Example: wookie: codec@69 { compatible = "tempo,tscs42A2"; reg = <0x69>; + clock-names = "xtal"; + clocks = <&audio_xtal>; }; diff --git a/sound/soc/codecs/tscs42xx.c b/sound/soc/codecs/tscs42xx.c index d41852a37ac6..0aaa6a826096 100644 --- a/sound/soc/codecs/tscs42xx.c +++ b/sound/soc/codecs/tscs42xx.c @@ -12,6 +12,7 @@ #include <linux/module.h> #include <linux/delay.h> #include <linux/mutex.h> +#include <linux/clk.h> #include <sound/tlv.h> #include <sound/pcm_params.h> #include <sound/soc.h> @@ -40,6 +41,9 @@ struct tscs42xx { struct mutex pll_lock; struct regmap *regmap; + + struct clk *sysclk; + int sysclk_src_id; }; struct coeff_ram_ctl { @@ -1251,13 +1255,46 @@ static int tscs42xx_set_dai_bclk_ratio(struct snd_soc_dai *codec_dai, return 0; } -static int tscs42xx_set_dai_sysclk(struct snd_soc_dai *codec_dai, - int clk_id, unsigned int freq, int dir) +static const struct snd_soc_dai_ops tscs42xx_dai_ops = { + .hw_params = tscs42xx_hw_params, + .mute_stream = tscs42xx_mute_stream, + .set_fmt = tscs42xx_set_dai_fmt, + .set_bclk_ratio = tscs42xx_set_dai_bclk_ratio, +}; + +static int part_is_valid(struct tscs42xx *tscs42xx) { - struct snd_soc_component *component = codec_dai->component; + int val; int ret; + unsigned int reg; + + ret = regmap_read(tscs42xx->regmap, R_DEVIDH, &reg); + if (ret < 0) + return ret; - switch (clk_id) { + val = reg << 8; + ret = regmap_read(tscs42xx->regmap, R_DEVIDL, &reg); + if (ret < 0) + return ret; + + val |= reg; + + switch (val) { + case 0x4A74: + case 0x4A73: + return true; + default: + return false; + }; +} + +static int set_sysclk(struct snd_soc_component *component) +{ + struct tscs42xx *tscs42xx = snd_soc_component_get_drvdata(component); + unsigned long freq; + int ret; + + switch (tscs42xx->sysclk_src_id) { case TSCS42XX_PLL_SRC_XTAL: case TSCS42XX_PLL_SRC_MCLK1: ret = snd_soc_component_write(component, R_PLLREFSEL, @@ -1285,6 +1322,7 @@ static int tscs42xx_set_dai_sysclk(struct snd_soc_dai *codec_dai, return -EINVAL; } + freq = clk_get_rate(tscs42xx->sysclk); ret = set_pll_ctl_from_input_freq(component, freq); if (ret < 0) { dev_err(component->dev, @@ -1295,41 +1333,13 @@ static int tscs42xx_set_dai_sysclk(struct snd_soc_dai *codec_dai, return 0; } -static const struct snd_soc_dai_ops tscs42xx_dai_ops = { - .hw_params = tscs42xx_hw_params, - .mute_stream = tscs42xx_mute_stream, - .set_fmt = tscs42xx_set_dai_fmt, - .set_bclk_ratio = tscs42xx_set_dai_bclk_ratio, - .set_sysclk = tscs42xx_set_dai_sysclk, -}; - -static int part_is_valid(struct tscs42xx *tscs42xx) +static int tscs42xx_probe(struct snd_soc_component *component) { - int val; - int ret; - unsigned int reg; - - ret = regmap_read(tscs42xx->regmap, R_DEVIDH, &reg); - if (ret < 0) - return ret; - - val = reg << 8; - ret = regmap_read(tscs42xx->regmap, R_DEVIDL, &reg); - if (ret < 0) - return ret; - - val |= reg; - - switch (val) { - case 0x4A74: - case 0x4A73: - return true; - default: - return false; - }; + return set_sysclk(component); } static const struct snd_soc_component_driver soc_codec_dev_tscs42xx = { + .probe = tscs42xx_probe, .dapm_widgets = tscs42xx_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(tscs42xx_dapm_widgets), .dapm_routes = tscs42xx_intercon, @@ -1387,11 +1397,15 @@ static const struct reg_sequence tscs42xx_patch[] = { { R_AIC2, RV_AIC2_BLRCM_DAC_BCLK_LRCLK_SHARED }, }; +static char const * const src_names[TSCS42XX_PLL_SRC_CNT] = { + "xtal", "mclk1", "mclk2"}; + static int tscs42xx_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct tscs42xx *tscs42xx; - int ret = 0; + int src; + int ret; tscs42xx = devm_kzalloc(&i2c->dev, sizeof(*tscs42xx), GFP_KERNEL); if (!tscs42xx) { @@ -1402,6 +1416,24 @@ static int tscs42xx_i2c_probe(struct i2c_client *i2c, } i2c_set_clientdata(i2c, tscs42xx); + for (src = TSCS42XX_PLL_SRC_XTAL; src < TSCS42XX_PLL_SRC_CNT; src++) { + tscs42xx->sysclk = devm_clk_get(&i2c->dev, src_names[src]); + if (!IS_ERR(tscs42xx->sysclk)) { + break; + } else if (PTR_ERR(tscs42xx->sysclk) != -ENOENT) { + ret = PTR_ERR(tscs42xx->sysclk); + dev_err(&i2c->dev, "Failed to get sysclk (%d)\n", ret); + return ret; + } + } + if (src == TSCS42XX_PLL_SRC_CNT) { + ret = -EINVAL; + dev_err(&i2c->dev, "Failed to get a valid clock name (%d)\n", + ret); + return ret; + } + tscs42xx->sysclk_src_id = src; + tscs42xx->regmap = devm_regmap_init_i2c(i2c, &tscs42xx_regmap); if (IS_ERR(tscs42xx->regmap)) { ret = PTR_ERR(tscs42xx->regmap); diff --git a/sound/soc/codecs/tscs42xx.h b/sound/soc/codecs/tscs42xx.h index d4a30bcbf64b..814c8f3c4a68 100644 --- a/sound/soc/codecs/tscs42xx.h +++ b/sound/soc/codecs/tscs42xx.h @@ -7,10 +7,10 @@ #define __WOOKIE_H__ enum { - TSCS42XX_PLL_SRC_NONE, TSCS42XX_PLL_SRC_XTAL, TSCS42XX_PLL_SRC_MCLK1, TSCS42XX_PLL_SRC_MCLK2, + TSCS42XX_PLL_SRC_CNT, }; #define R_HPVOLL 0x0 -- 2.15.1

2 4

[alsa-devel] [PATCH] ALSA: hda - Silence PM ops build warning
by Lukas Wunner 13 Apr '18

13 Apr '18

The system sleep PM ops azx_suspend() and azx_resume() were previously called by vga_switcheroo, but commit 07f4f97d7b4b ("vga_switcheroo: Use device link for HDA controller") removed their invocation. Unfortunately the commit neglected to update the #ifdef surrounding the two functions, so if CONFIG_PM_SLEEP is *not* enabled but all three of CONFIG_PM, CONFIG_VGA_SWITCHEROO and CONFIG_SND_HDA_CODEC_HDMI *are* enabled, the compiler now emits the following warning: sound/pci/hda/hda_intel.c:1024:12: warning: 'azx_resume' defined but not used [-Wunused-function] static int azx_resume(struct device *dev) ^~~~~~~~~~ sound/pci/hda/hda_intel.c:989:12: warning: 'azx_suspend' defined but not used [-Wunused-function] static int azx_suspend(struct device *dev) ^~~~~~~~~~~ Silence by updating the #ifdef. Because the #ifdef block now uses the same condition as the one immediately succeeding it, the two blocks can be collapsed together, shaving off another two lines. Fixes: 07f4f97d7b4b ("vga_switcheroo: Use device link for HDA controller") Cc: Takashi Iwai <tiwai(a)suse.com> Reported-by: Arnd Bergmann <arnd(a)arndb.de> Signed-off-by: Lukas Wunner <lukas(a)wunner.de> --- sound/pci/hda/hda_intel.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index ec4e6b829ee2..d653c56e9141 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -982,7 +982,7 @@ static int param_set_xint(const char *val, const struct kernel_param *kp) #define azx_del_card_list(chip) /* NOP */ #endif /* CONFIG_PM */ -#if defined(CONFIG_PM_SLEEP) || defined(SUPPORT_VGA_SWITCHEROO) +#ifdef CONFIG_PM_SLEEP /* * power management */ @@ -1063,9 +1063,7 @@ static int azx_resume(struct device *dev) trace_azx_resume(chip); return 0; } -#endif /* CONFIG_PM_SLEEP || SUPPORT_VGA_SWITCHEROO */ -#ifdef CONFIG_PM_SLEEP /* put codec down to D3 at hibernation for Intel SKL+; * otherwise BIOS may still access the codec and screw up the driver */ -- 2.16.2

2 3

[alsa-devel] [PATCH] Documentation: sound: Corrected my_chip port element
by Christina Quast 12 Apr '18

12 Apr '18

Hi all! A small correction in the documentation. C.

1 0

[alsa-devel] [PATCH 0/5] for-linux/sndif: add explicit back and front synchronization
by Oleksandr Andrushchenko 12 Apr '18

12 Apr '18

Hello, all! This is the syncup version of the sound protocol changes for Linux Kernel with the only difference from the corresponding Xen version being spaces to tabs conversion. Regradless of this only change I have dropped R-b tags received for Xen version. In order to provide explicit synchronization between backend and frontend the following changes are introduced in the protocol: - bump protocol version to 2 - add new ring buffer for sending asynchronous events from backend to frontend to report number of bytes played by the frontend (XENSND_EVT_CUR_POS) - introduce trigger events for playback control: start/stop/pause/resume - add "req-" prefix to event-channel and ring-ref to unify naming of the Xen event channels for requests and events - add XENSND_OP_HW_PARAM_QUERY request to read/update stream configuration space: request passes desired intervals/formats for the stream parameters and the response returns allowed intervals and formats mask that can be used. - MAJOR: changed req/resp/evt packet sizes from 32 to 64 octets - Reworked XENSND_OP_HW_PARAM_QUERY so it now sends all parameters at once, allowing to check all the configuration space. - Minor documentation cleanup (added missed "reserved" fields) Oleksandr Andrushchenko (5): xen/sndif: Introduce protocol version xen/sndif: Fix missed "reserved" fields in comments xen/sndif: Make requests and responses 64 octets long xen/sndif: Add explicit back and front synchronization xen/sndif: Add explicit back and front parameter negotiation include/xen/interface/io/sndif.h | 322 +++++++++++++++++++++++++++++-- 1 file changed, 306 insertions(+), 16 deletions(-) -- 2.17.0

3 10

Re: [alsa-devel] [PATCH 08/15] ASoC: pxa: remove the dmaengine compat need
by Mark Brown 12 Apr '18

12 Apr '18

On Mon, Apr 02, 2018 at 04:26:49PM +0200, Robert Jarzmik wrote: > As the pxa architecture switched towards the dmaengine slave map, the > old compatibility mechanism to acquire the dma requestor line number and > priority are not needed anymore. Acked-by: Mark Brown <broonie(a)kernel.org> If there's no dependency I'm happy to take this for 4.18.

2 1

[alsa-devel] [PATCH] ASoC: topology: fix some tiny memory leaks
by Dan Carpenter 12 Apr '18

12 Apr '18

These tiny memory leaks don't have a huge real life impact but they cause static checker warnings so let's fix them. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index fa27d0fca6dc..942c6e5eb4b7 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1325,8 +1325,10 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create( ec->hdr.name); kc[i].name = kstrdup(ec->hdr.name, GFP_KERNEL); - if (kc[i].name == NULL) + if (kc[i].name == NULL) { + kfree(se); goto err_se; + } kc[i].private_value = (long)se; kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; kc[i].access = ec->hdr.access; @@ -1442,8 +1444,10 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dbytes_create( be->hdr.name, be->hdr.access); kc[i].name = kstrdup(be->hdr.name, GFP_KERNEL); - if (kc[i].name == NULL) + if (kc[i].name == NULL) { + kfree(sbe); goto err; + } kc[i].private_value = (long)sbe; kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; kc[i].access = be->hdr.access;

2 1

[alsa-devel] Missing alsa-utils 1.1.6 tag
by Jordi Mallach 12 Apr '18

12 Apr '18

Hello Jaroslav, Can you push a tag for alsa-utils 1.1.6? It's missing from git AFAICT. Thank you, Jordi -- Jordi Mallach Pérez -- Debian developer https://www.debian.org/ jordi(a)sindominio.net jordi(a)debian.org https://www.sindominio.net/ GnuPG public key information available at: https://oskuro.net/

2 1

[alsa-devel] snd_seq_query_port_subscribers not updating subs index
by Charles Flèche 12 Apr '18

12 Apr '18

Hi all, I'm iterating through subscribers with snd_seq_query_port_subscribers to disconnect them. It works most of the time, but when a2j (an alsa -> jack MIDI bridge) runs, snd_seq_query_port_subscribers does not update the subs index, and the next call returns 0, leading to an infinite loop. I have to manually increase the index when the disconnection fails (meaning that I hit an a2j connection). while (snd_seq_query_port_subscribers(seq, subs) >= 0) { addr = *snd_seq_query_subscribe_get_addr(subs); if (snd_seq_disconnect_to(seq, port, addr.client, addr.port) < 0) { snd_seq_query_subscribe_set_index(subs, snd_seq_query_subscribe_get_index(subs) + 1); } } 1. Is the non-updated subs index an expected behavior ? 2. Is my fix a correct one or could it leads to other issues ? Many thanks,

3 4

[alsa-devel] [PATCH] ASoC: fsl_esai: Fix divisor calculation failure at lower ratio
by Nicolin Chen 12 Apr '18

12 Apr '18

When the desired ratio is less than 256, the savesub (tolerance) in the calculation would become 0. This will then fail the loop- search immediately without reporting any errors. But if the ratio is smaller enough, there is no need to calculate the tolerance because PM divisor alone is enough to get the ratio. So a simple fix could be just to set PM directly instead of going into the loop-search. Reported-by: Marek Vasut <marex(a)denx.de> Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: Marek Vasut <marex(a)denx.de> --- sound/soc/fsl/fsl_esai.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 40a7004..da8fd98 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -144,6 +144,13 @@ static int fsl_esai_divisor_cal(struct snd_soc_dai *dai, bool tx, u32 ratio, psr = ratio <= 256 * maxfp ? ESAI_xCCR_xPSR_BYPASS : ESAI_xCCR_xPSR_DIV8; + /* Do not loop-search if PM (1 ~ 256) alone can serve the ratio */ + if (ratio <= 256) { + pm = ratio; + fp = 1; + goto out; + } + /* Set the max fluctuation -- 0.1% of the max devisor */ savesub = (psr ? 1 : 8) * 256 * maxfp / 1000; -- 2.7.4

4 4

[alsa-devel] [PATCH] ASoC: fsl_ssi: Fix mode setting when changing channel number
by Nicolin Chen 12 Apr '18

12 Apr '18

This is a partial revert (in a cleaner way) of commit ebf08ae3bc90 ("ASoC: fsl_ssi: Keep ssi->i2s_net updated") to fix a regression at test cases when switching between mono and stereo audio. The problem is that ssi->i2s_net is initialized in set_dai_fmt() only, while this set_dai_fmt() is only called during the dai-link probe(). The original patch assumed set_dai_fmt() would be called during every playback instance, so it failed at the overriding use cases. This patch adds the local variable i2s_net back to let regular use cases still follow the mode settings from the set_dai_fmt(). Meanwhile, the original commit of keeping ssi->i2s_net updated was to make set_tdm_slot() clean by checking the ssi->i2s_net directly instead of reading SCR register. However, the change itself is not necessary (or even harmful) because the set_tdm_slot() might fail to check the slot number for Normal-Mode-None-Net settings while mono audio cases still need 2 slots. So this patch can also fix it. And it adds an extra line of comments to declare ssi->i2s_net does not reflect the register value but merely the initial setting from the set_dai_fmt(). Reported-by: Mika Penttilä <mika.penttila(a)nextfour.com> Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: Mika Penttilä <mika.penttila(a)nextfour.com> --- sound/soc/fsl/fsl_ssi.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 0823b08..89df2d9 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -217,6 +217,7 @@ struct fsl_ssi_soc_data { * @dai_fmt: DAI configuration this device is currently used with * @streams: Mask of current active streams: BIT(TX) and BIT(RX) * @i2s_net: I2S and Network mode configurations of SCR register + * (this is the initial settings based on the DAI format) * @synchronous: Use synchronous mode - both of TX and RX use STCK and SFCK * @use_dma: DMA is used or FIQ with stream filter * @use_dual_fifo: DMA with support for dual FIFO mode @@ -829,16 +830,23 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, } if (!fsl_ssi_is_ac97(ssi)) { + /* + * Keep the ssi->i2s_net intact while having a local variable + * to override settings for special use cases. Otherwise, the + * ssi->i2s_net will lose the settings for regular use cases. + */ + u8 i2s_net = ssi->i2s_net; + /* Normal + Network mode to send 16-bit data in 32-bit frames */ if (fsl_ssi_is_i2s_cbm_cfs(ssi) && sample_size == 16) - ssi->i2s_net = SSI_SCR_I2S_MODE_NORMAL | SSI_SCR_NET; + i2s_net = SSI_SCR_I2S_MODE_NORMAL | SSI_SCR_NET; /* Use Normal mode to send mono data at 1st slot of 2 slots */ if (channels == 1) - ssi->i2s_net = SSI_SCR_I2S_MODE_NORMAL; + i2s_net = SSI_SCR_I2S_MODE_NORMAL; regmap_update_bits(regs, REG_SSI_SCR, - SSI_SCR_I2S_NET_MASK, ssi->i2s_net); + SSI_SCR_I2S_NET_MASK, i2s_net); } /* In synchronous mode, the SSI uses STCCR for capture */ -- 2.7.4

3 2

[alsa-devel] [PATCH] ASoC: rsnd: mark PM functions __maybe_unused
by Arnd Bergmann 12 Apr '18

12 Apr '18

The suspend/resume callbacks are now optional, leading to a warning when they are unused: sound/soc/sh/rcar/core.c:1548:12: error: 'rsnd_resume' defined but not used [-Werror=unused-function] static int rsnd_resume(struct device *dev) ^~~~~~~~~~~ sound/soc/sh/rcar/core.c:1539:12: error: 'rsnd_suspend' defined but not used [-Werror=unused-function] static int rsnd_suspend(struct device *dev) This marks the as __maybe_unused to avoid the warning. Fixes: f8a9a29c4fe9 ("ASoC: rsnd: set pm_ops in hibernate-compatible way") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/sh/rcar/core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 6a76688a8ba9..94f081b93258 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1536,7 +1536,7 @@ static int rsnd_remove(struct platform_device *pdev) return ret; } -static int rsnd_suspend(struct device *dev) +static int __maybe_unused rsnd_suspend(struct device *dev) { struct rsnd_priv *priv = dev_get_drvdata(dev); @@ -1545,7 +1545,7 @@ static int rsnd_suspend(struct device *dev) return 0; } -static int rsnd_resume(struct device *dev) +static int __maybe_unused rsnd_resume(struct device *dev) { struct rsnd_priv *priv = dev_get_drvdata(dev); -- 2.9.0

2 1

[alsa-devel] [PATCH] ASoC: Intel: atom: fix ACPI/PCI Kconfig
by Pierre-Louis Bossart 12 Apr '18

12 Apr '18

The split between ACPI and PCI platforms generated issues with randconfig: with SND_SST_ATOM_HIFI2_PLATFORM_PCI=y and SND_SST_ATOM_HIFI2_PLATFORM=m, we get this module link failure: ERROR: "sst_context_init" [sound/soc/intel/atom/sst/snd-intel-sst-acpi.ko] undefined! ERROR: "sst_context_cleanup" [sound/soc/intel/atom/sst/snd-intel-sst-acpi.ko] undefined! ERROR: "sst_alloc_drv_context" [sound/soc/intel/atom/sst/snd-intel-sst-acpi.ko] undefined! ERROR: "intel_sst_pm" [sound/soc/intel/atom/sst/snd-intel-sst-acpi.ko] undefined! ERROR: "sst_configure_runtime_pm" [sound/soc/intel/atom/sst/snd-intel-sst-acpi.ko] undefined! To keep things simple, let's expose two configs for SND_SST_ATOM_HIFI2_PLATFORM_PCI and SND_SST_ATOM_HIFI2_PLATFORM_ACPI, which select a common SND_SST_ATOM_HIFI2_PLATFORM option. To avoid breaking existing solutions with the semantics change, SND_SST_ATOM_HIFI2_PLATFORM_ACPI uses "default ACPI" so that "make oldnoconfig" and "make olddefconfig" still work as expected. Also remove mentions of Medfield while we are at it since it was removed recently. Reported-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: 4772c16ede52 ("ASoC: Intel: Kconfig: Simplify-clarify ACPI/PCI dependencies") Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/Kconfig | 22 +++++++++++++--------- 1 file changed, 13 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index f2c9e8c5970a..d7ac77b0197b 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -72,24 +72,28 @@ config SND_SOC_INTEL_BAYTRAIL for Baytrail Chromebooks but this option is now deprecated and is not recommended, use SND_SST_ATOM_HIFI2_PLATFORM instead. +config SND_SST_ATOM_HIFI2_PLATFORM + tristate + select SND_SOC_COMPRESS + config SND_SST_ATOM_HIFI2_PLATFORM_PCI - tristate "PCI HiFi2 (Medfield, Merrifield) Platforms" + tristate "PCI HiFi2 (Merrifield) Platforms" depends on X86 && PCI select SND_SST_IPC_PCI - select SND_SOC_COMPRESS + select SND_SST_ATOM_HIFI2_PLATFORM help - If you have a Intel Medfield or Merrifield/Edison platform, then + If you have a Intel Merrifield/Edison platform, then enable this option by saying Y or m. Distros will typically not - enable this option: Medfield devices are not available to - developers and while Merrifield/Edison can run a mainline kernel with - limited functionality it will require a firmware file which - is not in the standard firmware tree + enable this option: while Merrifield/Edison can run a mainline + kernel with limited functionality it will require a firmware file + which is not in the standard firmware tree -config SND_SST_ATOM_HIFI2_PLATFORM +config SND_SST_ATOM_HIFI2_PLATFORM_ACPI tristate "ACPI HiFi2 (Baytrail, Cherrytrail) Platforms" + default ACPI depends on X86 && ACPI select SND_SST_IPC_ACPI - select SND_SOC_COMPRESS + select SND_SST_ATOM_HIFI2_PLATFORM select SND_SOC_ACPI_INTEL_MATCH select IOSF_MBI help -- 2.14.1

4 3

[alsa-devel] [Xen-devel][PATCH v3 0/5] sndif: add explicit back and front synchronization
by Oleksandr Andrushchenko 12 Apr '18

12 Apr '18

From: Oleksandr Andrushchenko <oleksandr_andrushchenko(a)epam.com> Hello, all! In order to provide explicit synchronization between backend and frontend the following changes are introduced in the protocol: - bump protocol version to 2 - add new ring buffer for sending asynchronous events from backend to frontend to report number of bytes played by the frontend (XENSND_EVT_CUR_POS) - introduce trigger events for playback control: start/stop/pause/resume - add "req-" prefix to event-channel and ring-ref to unify naming of the Xen event channels for requests and events - add XENSND_OP_HW_PARAM_QUERY request to read/update stream configuration space: request passes desired intervals/formats for the stream parameters and the response returns allowed intervals and formats mask that can be used. Changes since v2: 1. Konrad's r-b tag for version patch 2. MAJOR: changed req/resp/evt packet sizes from 32 to 64 octets 3. Reworked XENSND_OP_HW_PARAM_QUERY so it now sends all parameters at once, allowing to check all the configuration space. 4. Minor documentation cleanup (added missed "reserved" fields) Changes since v1: 1. Changed protocol version definition from string to integer, so it can easily be used in comparisons. Konrad, I have removed your r-b tag for the reason of this change. 2. In order to provide explicit stream parameter negotiation between backend and frontend the following changes are introduced in the protocol: add XENSND_OP_HW_PARAM_QUERY request to read/update configuration space for the parameter given: request passes desired parameter interval (mask) and the response to this request returns min/max interval (mask) for the parameter to be used. Parameters supported by this request/response: - format mask - sample rate interval - number of channels interval - buffer size, interval, frames - period size, interval, frames Oleksandr Andrushchenko (5): sndif: Introduce protocol version sndif: Fix missed "reserved" fields in comments sndif: Make requests and responses 64 octets long sndif: Add explicit back and front synchronization sndif: Add explicit back and front parameter negotiation xen/include/public/io/sndif.h | 322 +++++++++++++++++++++++++++++++++++++++--- 1 file changed, 306 insertions(+), 16 deletions(-) -- 2.7.4

4 16

[alsa-devel] [PATCH] ASoC: dmic: Fix clock parenting
by Peter Ujfalusi 12 Apr '18

12 Apr '18

From: Tero Kristo <t-kristo(a)ti.com> In 4.16 the clock hierarchy got changed by a5c82a09d876 ARM: dts: omap4: add clkctrl nodes The fck of dmic is no longer a mux clock, it's parent is. Signed-off-by: Tero Kristo <t-kristo(a)ti.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Cc: stable(a)vger.kernel.org # 4.16+ --- Mark, can you apply this for 4.17 as DMIC got broken already in 4.16. Regards, Peter sound/soc/omap/omap-dmic.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c index c7101d851213..51dd7c65096b 100644 --- a/sound/soc/omap/omap-dmic.c +++ b/sound/soc/omap/omap-dmic.c @@ -281,7 +281,7 @@ static int omap_dmic_dai_trigger(struct snd_pcm_substream *substream, static int omap_dmic_select_fclk(struct omap_dmic *dmic, int clk_id, unsigned int freq) { - struct clk *parent_clk; + struct clk *parent_clk, *mux; char *parent_clk_name; int ret = 0; @@ -329,14 +329,21 @@ static int omap_dmic_select_fclk(struct omap_dmic *dmic, int clk_id, return -ENODEV; } + mux = clk_get_parent(dmic->fclk); + if (IS_ERR(mux)) { + dev_err(dmic->dev, "can't get fck mux parent\n"); + clk_put(parent_clk); + return -ENODEV; + } + mutex_lock(&dmic->mutex); if (dmic->active) { /* disable clock while reparenting */ pm_runtime_put_sync(dmic->dev); - ret = clk_set_parent(dmic->fclk, parent_clk); + ret = clk_set_parent(mux, parent_clk); pm_runtime_get_sync(dmic->dev); } else { - ret = clk_set_parent(dmic->fclk, parent_clk); + ret = clk_set_parent(mux, parent_clk); } mutex_unlock(&dmic->mutex); @@ -349,6 +356,7 @@ static int omap_dmic_select_fclk(struct omap_dmic *dmic, int clk_id, dmic->fclk_freq = freq; err_busy: + clk_put(mux); clk_put(parent_clk); return ret; -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

2 1

[alsa-devel] [PATCH] ALSA: line6: Use correct endpoint type for midi output
by Fabián Inostroza 12 Apr '18

12 Apr '18

Hi, Sending MIDI messages to a PODxt through the USB connection shows "usb_submit_urb failed" in dmesg and the message is not received by the POD. The error is caused because in the funcion send_midi_async() in midi.c there is a call to usb_sndbulkpipe() for endpoint 3 OUT, but the PODxt USB descriptor shows that this endpoint it's an interrupt endpoint. Patch tested with PODxt only. Signed-off-by: Fabián Inostroza <fabianinostroza(a)udec.cl> --- sound/usb/line6/midi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/line6/midi.c b/sound/usb/line6/midi.c index 6d7cde56..e2cf55c5 100644 --- a/sound/usb/line6/midi.c +++ b/sound/usb/line6/midi.c @@ -125,7 +125,7 @@ static int send_midi_async(struct usb_line6 *line6, unsigned char *data, } usb_fill_int_urb(urb, line6->usbdev, - usb_sndbulkpipe(line6->usbdev, + usb_sndintpipe(line6->usbdev, line6->properties->ep_ctrl_w), transfer_buffer, length, midi_sent, line6, line6->interval); -- 2.16.3

2 1

[alsa-devel] [PATCH 1/2] ASoC: AMD: Support headset button on Stoney DA7219
by Akshu Agrawal 12 Apr '18

12 Apr '18

Adds headset button support. TEST=Tested Volume UP/Down functionality Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp-da7219-max98357a.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index b205c78..d9491e1 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -33,6 +33,7 @@ #include <linux/gpio.h> #include <linux/module.h> #include <linux/i2c.h> +#include <linux/input.h> #include <linux/acpi.h> #include "../codecs/da7219.h" @@ -51,6 +52,7 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_card *card = rtd->card; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_component *component = codec_dai->component; + struct snd_soc_jack *jack; dev_info(rtd->dev, "codec dai name = %s\n", codec_dai->name); @@ -80,6 +82,12 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) return ret; } + jack = &cz_jack; + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + da7219_aad_jack_det(component, &cz_jack); return 0; -- 1.9.1

4 10

[alsa-devel] BUG: unable to handle kernel paging request in snd_pcm_format_set_silence
by syzbot 11 Apr '18

11 Apr '18

Hello, syzbot hit the following crash on upstream commit f2d285669aae656dfeafa0bf25e86bbbc5d22329 (Tue Apr 3 17:45:39 2018 +0000) Merge tag 'pm-4.17-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/rafael/linux-pm syzbot dashboard link: https://syzkaller.appspot.com/bug?extid=f973e52978d95f818f69 So far this crash happened 2 times on upstream. syzkaller reproducer: https://syzkaller.appspot.com/x/repro.syz?id=5368241429938176 Raw console output: https://syzkaller.appspot.com/x/log.txt?id=6066534392791040 Kernel config: https://syzkaller.appspot.com/x/.config?id=686016073509112605 compiler: gcc (GCC) 7.1.1 20170620 user-space arch: i386 IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+f973e52978d95f818f69(a)syzkaller.appspotmail.com It will help syzbot understand when the bug is fixed. See footer for details. If you forward the report, please keep this part and the footer. IPVS: ftp: loaded support on port[0] = 21 IPVS: ftp: loaded support on port[0] = 21 IPVS: ftp: loaded support on port[0] = 21 IPVS: ftp: loaded support on port[0] = 21 IPVS: ftp: loaded support on port[0] = 21 BUG: unable to handle kernel paging request at ffffc90004cc7000 PGD 1dad42067 P4D 1dad42067 PUD 1dad43067 PMD 1c5cb3067 PTE 0 Oops: 0002 [#1] SMP KASAN Dumping ftrace buffer: (ftrace buffer empty) Modules linked in: CPU: 0 PID: 5516 Comm: syz-executor3 Not tainted 4.16.0+ #288 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 RIP: 0010:memset_erms+0x9/0x10 arch/x86/lib/memset_64.S:65 RSP: 0018:ffff8801d7fc7368 EFLAGS: 00010246 RAX: fffff520009b9600 RBX: ffffc90004cc7000 RCX: 0000000000104500 RDX: 0000000000104500 RSI: 0000000000000000 RDI: ffffc90004cc7000 RBP: ffff8801d7fc7388 R08: 0000000000000000 R09: ffffc90004cc7000 R10: 00000000000208a0 R11: fffff520009b969f R12: 0000000000104500 R13: 0000000000000000 R14: ffffffff88c9185c R15: 0000000000000010 FS: 0000000000000000(0000) GS:ffff8801db000000(0063) knlGS:00000000f7f5fb40 CS: 0010 DS: 002b ES: 002b CR0: 0000000080050033 CR2: ffffc90004cc7000 CR3: 00000001d765c000 CR4: 00000000001406f0 DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000400 Call Trace: memset include/linux/string.h:330 [inline] snd_pcm_format_set_silence+0x13f/0x4b0 sound/core/pcm_misc.c:431 loopback_prepare+0x92b/0xbf0 sound/drivers/aloop.c:347 snd_pcm_do_prepare+0x58/0x90 sound/core/pcm_native.c:1640 snd_pcm_action_single+0x70/0x110 sound/core/pcm_native.c:1084 snd_pcm_action_nonatomic+0x95/0xa0 sound/core/pcm_native.c:1151 snd_pcm_prepare+0x115/0x1a0 sound/core/pcm_native.c:1687 snd_pcm_kernel_ioctl+0x97/0x240 sound/core/pcm_native.c:2998 snd_pcm_oss_prepare+0x49/0x150 sound/core/oss/pcm_oss.c:1104 snd_pcm_oss_make_ready+0xf9/0x130 sound/core/oss/pcm_oss.c:1132 snd_pcm_oss_set_trigger+0x384/0x930 sound/core/oss/pcm_oss.c:1959 snd_pcm_oss_poll+0x7aa/0xd60 sound/core/oss/pcm_oss.c:2727 do_pollfd fs/select.c:832 [inline] do_poll fs/select.c:883 [inline] do_sys_poll+0x719/0x1050 fs/select.c:977 SYSC_poll fs/select.c:1035 [inline] SyS_poll+0x10d/0x450 fs/select.c:1023 do_syscall_32_irqs_on arch/x86/entry/common.c:330 [inline] do_fast_syscall_32+0x3ec/0xf9f arch/x86/entry/common.c:392 entry_SYSENTER_compat+0x70/0x7f arch/x86/entry/entry_64_compat.S:139 RIP: 0023:0xf7f63c99 RSP: 002b:00000000f7f5f0ac EFLAGS: 00000282 ORIG_RAX: 00000000000000a8 RAX: ffffffffffffffda RBX: 0000000020000100 RCX: 0000000000000001 RDX: 0000000000000000 RSI: 0000000000000000 RDI: 0000000000000000 RBP: 0000000000000000 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000000 R12: 0000000000000000 R13: 0000000000000000 R14: 0000000000000000 R15: 0000000000000000 Code: 48 c1 e9 03 40 0f b6 f6 48 b8 01 01 01 01 01 01 01 01 48 0f af c6 f3 48 ab 89 d1 f3 aa 4c 89 c8 c3 90 49 89 f9 40 88 f0 48 89 d1 <f3> aa 4c 89 c8 c3 90 49 89 fa 40 0f b6 ce 48 b8 01 01 01 01 01 RIP: memset_erms+0x9/0x10 arch/x86/lib/memset_64.S:65 RSP: ffff8801d7fc7368 CR2: ffffc90004cc7000 ---[ end trace 95a476f9da224f32 ]--- --- This bug is generated by a dumb bot. It may contain errors. See https://goo.gl/tpsmEJ for details. Direct all questions to syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. If you forgot to add the Reported-by tag, once the fix for this bug is merged into any tree, please reply to this email with: #syz fix: exact-commit-title If you want to test a patch for this bug, please reply with: #syz test: git://repo/address.git branch and provide the patch inline or as an attachment. To mark this as a duplicate of another syzbot report, please reply with: #syz dup: exact-subject-of-another-report If it's a one-off invalid bug report, please reply with: #syz invalid Note: if the crash happens again, it will cause creation of a new bug report. Note: all commands must start from beginning of the line in the email body.

1 0

[alsa-devel] [PATCH v2 0/5] ASoC: Intel: geminilake: Adds da7219+max98537 machine driver support
by Naveen Manohar 11 Apr '18

11 Apr '18

Patch adds required changes in bxt machine & corresponding driver entry in machine to use MAX98357A codec as speaker on SSP1 and DA7219 codec as headset on SSP2 for GLK board. Naveen Manohar (5): ASoC: Intel: broxton: reduce machine name for bxt_da7219_max98357a ASoC: Intel: Headset button support in broxton machine driver ASoC: Intel: Add Geminilake Dialog Maxim machine driver support ASoC: Intel: glk: Add DAI links for Multi-Playback ASoC: Intel: Add Geminilake Dialog+Maxim machine driver entry sound/soc/intel/boards/bxt_da7219_max98357a.c | 152 ++++++++++++++++++++++---- sound/soc/intel/skylake/skl.c | 9 +- 2 files changed, 141 insertions(+), 20 deletions(-) -- v1->v2: Unified dai-link usage between bxt & glk mach, corrected jack CTRL 2.7.4

4 9

Re: [alsa-devel] call/normal switch was Re: omap4-droid4: voice call support was
by Tony Lindgren 11 Apr '18

11 Apr '18

* Dan Williams <dcbw(a)redhat.com> [180402 15:51]: > On Sun, 2018-04-01 at 10:30 -0700, Tony Lindgren wrote: > > * Tony Lindgren <tony(a)atomide.com> [180401 15:38]: > > Found it! Here's what I need to do over n_gsm: > > > > ngsm 1 "AT+CFUN=1" > > ngsm 1 "AT+CFUN?" > > ngsm 2 "AT+EACC=3,0" # enable mic > > ngsm 2 "AT+CLVL=4" # set speaker volume > > ngsm 2 "AT+CMUT=0" # unmute mic > > I tried to look through the QMI dumps we have in libqmi from 2013 > (latest Qualcomm posted) and couldn't find anything to do with mic > control, speaker volume, or anything like that. > > If the modem supports the AT service (which I think it does? Not > looking at the libqmi dumps right now) then it could potentially tunnel > these AT commands through QMI too. > > Perhaps Qualcomm added something to the Voice service after 2013, or > perhaps there are other services that might control speaker/mic that we > don't have public dumps for yet though. OK thanks for checking. So probably only n_gsm channel 1 is for normal Qualcomm at commands, and then channel 2 and others are commands implemented by Motorola on the mdm6600. I guess we'd have to add support for reading and writing to /dev/gsmtty2 at least as it looks like these cannot be accessed via /dev/ttyUSB4. Or at least I have not figured out any other way to access them. Regards, Tony

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[alsa-devel] [PATCH v2, 1/2] ASoC: AMD: Support headset button on Stoney DA7219
by Akshu Agrawal 11 Apr '18

11 Apr '18

Adds headset button support. TEST=Tested Volume UP/Down, Play/Pause functionality Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- v2: Changed KEY_MEDIA to KEY_PLAYPAUSE sound/soc/amd/acp-da7219-max98357a.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index b205c78..1012a80 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -33,6 +33,7 @@ #include <linux/gpio.h> #include <linux/module.h> #include <linux/i2c.h> +#include <linux/input.h> #include <linux/acpi.h> #include "../codecs/da7219.h" @@ -51,6 +52,7 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_card *card = rtd->card; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_component *component = codec_dai->component; + struct snd_soc_jack *jack; dev_info(rtd->dev, "codec dai name = %s\n", codec_dai->name); @@ -80,6 +82,12 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) return ret; } + jack = &cz_jack; + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + da7219_aad_jack_det(component, &cz_jack); return 0; -- 1.9.1

4 5

[alsa-devel] [PATCH 1/2] ASoC: adau17x1: Handling of DSP_RUN register during fw setup
by Robert Rosengren 11 Apr '18

11 Apr '18

From: Danny Smith <dannys(a)axis.com> DSP_RUN needs to be disabled during firmware write otherwise we can end up with undefined behavior if writing to a dsp which is already running firmware. Signed-off-by: Danny Smith <dannys(a)axis.com> --- sound/soc/codecs/adau17x1.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 80c2a06285bb..5636b9522462 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -838,14 +838,19 @@ EXPORT_SYMBOL_GPL(adau17x1_volatile_register); int adau17x1_setup_firmware(struct adau *adau, unsigned int rate) { int ret; - int dspsr; + int dspsr, dsp_run; ret = regmap_read(adau->regmap, ADAU17X1_DSP_SAMPLING_RATE, &dspsr); if (ret) return ret; + ret = regmap_read(adau->regmap, ADAU17X1_DSP_RUN, &dsp_run); + if (ret) + return ret; + regmap_write(adau->regmap, ADAU17X1_DSP_ENABLE, 1); regmap_write(adau->regmap, ADAU17X1_DSP_SAMPLING_RATE, 0xf); + regmap_write(adau->regmap, ADAU17X1_DSP_RUN, 0); ret = sigmadsp_setup(adau->sigmadsp, rate); if (ret) { @@ -853,6 +858,7 @@ int adau17x1_setup_firmware(struct adau *adau, unsigned int rate) return ret; } regmap_write(adau->regmap, ADAU17X1_DSP_SAMPLING_RATE, dspsr); + regmap_write(adau->regmap, ADAU17X1_DSP_RUN, dsp_run); return 0; } -- 2.11.0

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[alsa-devel] [PATCH v2 00/13] soundwire: Add stream support
by Vinod Koul 11 Apr '18

11 Apr '18

This series adds support in SoundWire subsystem for: - Documentation for stream support - stream management - data port management - DAI ops in cadence and Intel drivers - ASoC API to propagate SDW stream Updates in v2: - Make ASoC API inlined - Make stream states as states and not action - Update the direction enum - Fix some typos and comment updates Sanyog Kale (7): soundwire: Add more documentation soundwire: Add support for SoundWire stream management soundwire: Add support for port management soundwire: Add Master and Slave port programming soundwire: Add helpers for ports operations soundwire: Add bank switch routine soundwire: Add stream configuration APIs Shreyas NC (2): ASoC: Add SoundWire stream programming interface soundwire: Remove cdns_master_ops Vinod Koul (4): soundwire: cdns: Add port routines soundwire: cdns: Add stream routines soundwire: intel: Add stream initialization soundwire: intel: Add audio DAI ops .../driver-api/soundwire/error_handling.rst | 65 + Documentation/driver-api/soundwire/index.rst | 3 + Documentation/driver-api/soundwire/locking.rst | 106 ++ Documentation/driver-api/soundwire/stream.rst | 368 +++++ drivers/soundwire/Kconfig | 2 +- drivers/soundwire/Makefile | 2 +- drivers/soundwire/bus.c | 43 + drivers/soundwire/bus.h | 72 + drivers/soundwire/cadence_master.c | 449 +++++- drivers/soundwire/cadence_master.h | 151 ++ drivers/soundwire/intel.c | 528 ++++++- drivers/soundwire/intel.h | 4 + drivers/soundwire/intel_init.c | 3 + drivers/soundwire/stream.c | 1549 ++++++++++++++++++++ include/linux/soundwire/sdw.h | 332 ++++- include/linux/soundwire/sdw_intel.h | 14 + include/sound/soc-dai.h | 21 + 17 files changed, 3698 insertions(+), 14 deletions(-) create mode 100644 Documentation/driver-api/soundwire/error_handling.rst create mode 100644 Documentation/driver-api/soundwire/locking.rst create mode 100644 Documentation/driver-api/soundwire/stream.rst create mode 100644 drivers/soundwire/stream.c -- 2.7.4

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Re: [alsa-devel] No audio on iMac18,1 (A1418) with kernel 4.15.14
by Freaky 10 Apr '18

10 Apr '18

FYI: Tested all the pin outs with hda-analyzer, didn't get sound on the speakers with any of them. On Tue, Apr 10, 2018 at 3:57 PM, Freaky <freaky2000(a)gmail.com> wrote: > Hi there, > > we got an iMac18,1 here on which we have no audio. > > Maybe the codec is just too new, although I see people using it on > non-Apple hardware. Using model parameters imac27 or imac27_122 doesn't > help. > > Have alsa-info attached. lspci (on this distro (but it's dated although it > has a recent kernel)) doesn't recognize it yet, the online PCI-IDs says: > > Sunrise Point-LP HD Audio > > Found some other people mentioning this issue at Arch, thread is here: > https://bbs.archlinux.org/viewtopic.php?id=233449&p=2 - no solution so > far. > > Will see if flipping bits with hda-analyzer will make it go. > > If there's a solution for this already we'd be much obliged if someone can > point us to it. Otherwise very willing to help, but can't code drivers > unfortunately. Can apply patches and build kernels however :). We'll have > to return the machine in about a week, so if anyone desires > output/testing/etc. please let us know asap as we only have it available > for a short while. > > Will post the solution, if we can find one, to the Arch thread as well. > > Thanks in advance. > > Kind regards >

1 0

[alsa-devel] INFO: rcu detected stall in __snd_pcm_lib_xfer (2)
by syzbot 10 Apr '18

10 Apr '18

Hello, syzbot hit the following crash on upstream commit e02d37bf55a9a36f22427fd6dd733fe104d817b6 (Thu Apr 5 17:42:07 2018 +0000) Merge tag 'sound-4.17-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound syzbot dashboard link: https://syzkaller.appspot.com/bug?extid=7e3f31a52646f939c052 So far this crash happened 43 times on upstream. Unfortunately, I don't have any reproducer for this crash yet. Raw console output: https://syzkaller.appspot.com/x/log.txt?id=5148264345108480 Kernel config: https://syzkaller.appspot.com/x/.config?id=-4805825610197092128 compiler: gcc (GCC) 8.0.1 20180301 (experimental) IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+7e3f31a52646f939c052(a)syzkaller.appspotmail.com It will help syzbot understand when the bug is fixed. See footer for details. If you forward the report, please keep this part and the footer. INFO: rcu_sched self-detected stall on CPU 1-...!: (124999 ticks this GP) idle=522/1/4611686018427387906 softirq=65432/65432 fqs=169 (t=125000 jiffies g=35384 c=35383 q=537) rcu_sched kthread starved for 124294 jiffies! g35384 c35383 f0x0 RCU_GP_WAIT_FQS(3) ->state=0x0 ->cpu=0 RCU grace-period kthread stack dump: rcu_sched R running task 23768 9 2 0x80000000 Call Trace: context_switch kernel/sched/core.c:2848 [inline] __schedule+0x807/0x1e40 kernel/sched/core.c:3490 schedule+0xef/0x430 kernel/sched/core.c:3549 schedule_timeout+0x138/0x240 kernel/time/timer.c:1801 rcu_gp_kthread+0x6b5/0x1940 kernel/rcu/tree.c:2231 kthread+0x345/0x410 kernel/kthread.c:238 ret_from_fork+0x3a/0x50 arch/x86/entry/entry_64.S:411 NMI backtrace for cpu 1 CPU: 1 PID: 15299 Comm: syz-executor3 Not tainted 4.16.0+ #2 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: <IRQ> __dump_stack lib/dump_stack.c:17 [inline] dump_stack+0x1b9/0x29f lib/dump_stack.c:53 nmi_cpu_backtrace.cold.4+0x19/0xce lib/nmi_backtrace.c:103 nmi_trigger_cpumask_backtrace+0x151/0x192 lib/nmi_backtrace.c:62 arch_trigger_cpumask_backtrace+0x14/0x20 arch/x86/kernel/apic/hw_nmi.c:38 trigger_single_cpu_backtrace include/linux/nmi.h:156 [inline] rcu_dump_cpu_stacks+0x175/0x1c2 kernel/rcu/tree.c:1376 print_cpu_stall kernel/rcu/tree.c:1525 [inline] check_cpu_stall.isra.61.cold.80+0x36c/0x59a kernel/rcu/tree.c:1593 __rcu_pending kernel/rcu/tree.c:3356 [inline] rcu_pending kernel/rcu/tree.c:3401 [inline] rcu_check_callbacks+0x21b/0xad0 kernel/rcu/tree.c:2763 update_process_times+0x2d/0x70 kernel/time/timer.c:1636 tick_sched_handle+0xa0/0x180 kernel/time/tick-sched.c:171 tick_sched_timer+0x42/0x130 kernel/time/tick-sched.c:1179 __run_hrtimer kernel/time/hrtimer.c:1337 [inline] __hrtimer_run_queues+0x3e3/0x10a0 kernel/time/hrtimer.c:1399 hrtimer_interrupt+0x286/0x650 kernel/time/hrtimer.c:1457 local_apic_timer_interrupt arch/x86/kernel/apic/apic.c:1025 [inline] smp_apic_timer_interrupt+0x15d/0x710 arch/x86/kernel/apic/apic.c:1050 apic_timer_interrupt+0xf/0x20 arch/x86/entry/entry_64.S:862 </IRQ> RIP: 0010:arch_local_irq_enable arch/x86/include/asm/paravirt.h:793 [inline] RIP: 0010:snd_pcm_stream_unlock_irq+0xb7/0xf0 sound/core/pcm_native.c:166 RSP: 0018:ffff8801c3f27560 EFLAGS: 00000246 ORIG_RAX: ffffffffffffff13 RAX: 0000000000040000 RBX: 0000000000000000 RCX: ffffc900038d5000 RDX: 0000000000040000 RSI: ffffffff859ca760 RDI: ffffffff88b172b8 RBP: ffff8801c3f27568 R08: ffff8801abf86af8 R09: 0000000000000006 R10: ffff8801abf86280 R11: 0000000000000000 R12: 0000000000000004 R13: 00000000ffffffe0 R14: ffff8801d7d9e1c0 R15: ffff8801ceac6940 __snd_pcm_lib_xfer+0x739/0x1d10 sound/core/pcm_lib.c:2246 snd_pcm_oss_write3+0xe9/0x220 sound/core/oss/pcm_oss.c:1236 io_playback_transfer+0x274/0x310 sound/core/oss/io.c:47 snd_pcm_plug_write_transfer+0x36c/0x470 sound/core/oss/pcm_plugin.c:619 snd_pcm_oss_write2+0x25c/0x460 sound/core/oss/pcm_oss.c:1365 snd_pcm_oss_write1 sound/core/oss/pcm_oss.c:1415 [inline] snd_pcm_oss_write+0x764/0xa20 sound/core/oss/pcm_oss.c:2774 __vfs_write+0x10b/0x880 fs/read_write.c:485 vfs_write+0x1f8/0x560 fs/read_write.c:549 ksys_write+0xf9/0x250 fs/read_write.c:598 SYSC_write fs/read_write.c:610 [inline] SyS_write+0x24/0x30 fs/read_write.c:607 do_syscall_64+0x29e/0x9d0 arch/x86/entry/common.c:287 entry_SYSCALL_64_after_hwframe+0x42/0xb7 RIP: 0033:0x4552d9 RSP: 002b:00007f1568de6c68 EFLAGS: 00000246 ORIG_RAX: 0000000000000001 RAX: ffffffffffffffda RBX: 00007f1568de76d4 RCX: 00000000004552d9 RDX: 0000000000000060 RSI: 0000000020000000 RDI: 0000000000000013 RBP: 000000000072bea0 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 00000000ffffffff R13: 00000000000006c0 R14: 00000000006fd2a0 R15: 0000000000000000 --- This bug is generated by a dumb bot. It may contain errors. See https://goo.gl/tpsmEJ for details. Direct all questions to syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. If you forgot to add the Reported-by tag, once the fix for this bug is merged into any tree, please reply to this email with: #syz fix: exact-commit-title To mark this as a duplicate of another syzbot report, please reply with: #syz dup: exact-subject-of-another-report If it's a one-off invalid bug report, please reply with: #syz invalid Note: if the crash happens again, it will cause creation of a new bug report. Note: all commands must start from beginning of the line in the email body.

3 3

[alsa-devel] [PATCH] ASoC: fsl_esai: Clear the xPM bit when using xFP
by Marek Vasut 10 Apr '18

10 Apr '18

When setting xFP directly, set the xPM predivider to 1, otherwise it could remain set to previously set incorrect value and interfere with the correct clocking. Signed-off-by: Marek Vasut <marex(a)denx.de> Cc: Fabio Estevam <fabio.estevam(a)nxp.com> Cc: Gustavo A. R. Silva <garsilva(a)embeddedor.com> Cc: Mark Brown <broonie(a)kernel.org> --- sound/soc/fsl/fsl_esai.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 40a700493f4c..9f69823b50d7 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -128,8 +128,11 @@ static int fsl_esai_divisor_cal(struct snd_soc_dai *dai, bool tx, u32 ratio, maxfp = usefp ? 16 : 1; - if (usefp && fp) + if (usefp && fp) { + regmap_update_bits(esai_priv->regmap, REG_ESAI_xCCR(tx), + ESAI_xCCR_xPM_MASK, 0); goto out_fp; + } if (ratio > 2 * 8 * 256 * maxfp || ratio < 2) { dev_err(dai->dev, "the ratio is out of range (2 ~ %d)\n", -- 2.16.2

3 9

[alsa-devel] WARNING in snd_pcm_hw_params
by syzbot 09 Apr '18

09 Apr '18

Hello, syzbot hit the following crash on upstream commit 3c8ba0d61d04ced9f8d9ff93977995a9e4e96e91 (Sat Mar 31 01:52:36 2018 +0000) kernel.h: Retain constant expression output for max()/min() syzbot dashboard link: https://syzkaller.appspot.com/bug?extid=803e0047ac3a3096bb4f So far this crash happened 6 times on upstream. C reproducer: https://syzkaller.appspot.com/x/repro.c?id=5886672973594624 syzkaller reproducer: https://syzkaller.appspot.com/x/repro.syz?id=4913841943085056 Raw console output: https://syzkaller.appspot.com/x/log.txt?id=6314122077536256 Kernel config: https://syzkaller.appspot.com/x/.config?id=-1647968177339044852 compiler: gcc (GCC) 8.0.1 20180301 (experimental) IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+803e0047ac3a3096bb4f(a)syzkaller.appspotmail.com It will help syzbot understand when the bug is fixed. See footer for details. If you forward the report, please keep this part and the footer. RAX: ffffffffffffffda RBX: 0000000000000000 RCX: 0000000000441939 RDX: 0000000000000001 RSI: 0000000020000180 RDI: 0000000000000003 RBP: 00007ffe8c3569b0 R08: 0000000000000002 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 0000000000000005 R13: ffffffffffffffff R14: 0000000000000000 R15: 0000000000000000 WARNING: CPU: 0 PID: 4468 at sound/core/pcm_native.c:620 snd_pcm_hw_params_choose sound/core/pcm_native.c:620 [inline] WARNING: CPU: 0 PID: 4468 at sound/core/pcm_native.c:620 snd_pcm_hw_params+0x65a/0x1e20 sound/core/pcm_native.c:672 Kernel panic - not syncing: panic_on_warn set ... CPU: 0 PID: 4468 Comm: syzkaller371267 Not tainted 4.16.0+ #2 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:17 [inline] dump_stack+0x1b9/0x29f lib/dump_stack.c:53 panic+0x22f/0x4de kernel/panic.c:183 __warn.cold.8+0x163/0x1a3 kernel/panic.c:547 report_bug+0x252/0x2d0 lib/bug.c:186 fixup_bug arch/x86/kernel/traps.c:178 [inline] do_error_trap+0x1bc/0x470 arch/x86/kernel/traps.c:296 do_invalid_op+0x1b/0x20 arch/x86/kernel/traps.c:315 invalid_op+0x1b/0x40 arch/x86/entry/entry_64.S:991 RIP: 0010:snd_pcm_hw_params_choose sound/core/pcm_native.c:620 [inline] RIP: 0010:snd_pcm_hw_params+0x65a/0x1e20 sound/core/pcm_native.c:672 RSP: 0018:ffff8801ac66f498 EFLAGS: 00010293 RAX: ffff8801acb86580 RBX: 00000000fffffff4 RCX: ffffffff859d7220 RDX: 0000000000000000 RSI: ffffffff859d752a RDI: 0000000000000005 RBP: ffff8801ac66f6e8 R08: ffff8801acb86580 R09: ffffed003b6046c2 R10: ffffed003b6046c2 R11: ffff8801db023613 R12: ffffffff8826309c R13: ffff8801ceab6940 R14: ffff8801d9509200 R15: dffffc0000000000 snd_pcm_kernel_ioctl+0x6c/0x240 sound/core/pcm_native.c:2999 snd_pcm_oss_change_params_locked+0x1be8/0x3ce0 sound/core/oss/pcm_oss.c:993 snd_pcm_oss_make_ready_locked+0xbc/0x130 sound/core/oss/pcm_oss.c:1182 snd_pcm_oss_write1 sound/core/oss/pcm_oss.c:1396 [inline] snd_pcm_oss_write+0x516/0xa20 sound/core/oss/pcm_oss.c:2774 __vfs_write+0x10b/0x880 fs/read_write.c:485 vfs_write+0x1f8/0x560 fs/read_write.c:549 ksys_write+0xf9/0x250 fs/read_write.c:598 SYSC_write fs/read_write.c:610 [inline] SyS_write+0x24/0x30 fs/read_write.c:607 do_syscall_64+0x29e/0x9d0 arch/x86/entry/common.c:287 entry_SYSCALL_64_after_hwframe+0x42/0xb7 RIP: 0033:0x441939 RSP: 002b:00007ffe8c3568e8 EFLAGS: 00000246 ORIG_RAX: 0000000000000001 RAX: ffffffffffffffda RBX: 0000000000000000 RCX: 0000000000441939 RDX: 0000000000000001 RSI: 0000000020000180 RDI: 0000000000000003 RBP: 00007ffe8c3569b0 R08: 0000000000000002 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 0000000000000005 R13: ffffffffffffffff R14: 0000000000000000 R15: 0000000000000000 Dumping ftrace buffer: (ftrace buffer empty) Kernel Offset: disabled Rebooting in 86400 seconds.. --- This bug is generated by a dumb bot. It may contain errors. See https://goo.gl/tpsmEJ for details. Direct all questions to syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. If you forgot to add the Reported-by tag, once the fix for this bug is merged into any tree, please reply to this email with: #syz fix: exact-commit-title If you want to test a patch for this bug, please reply with: #syz test: git://repo/address.git branch and provide the patch inline or as an attachment. To mark this as a duplicate of another syzbot report, please reply with: #syz dup: exact-subject-of-another-report If it's a one-off invalid bug report, please reply with: #syz invalid Note: if the crash happens again, it will cause creation of a new bug report. Note: all commands must start from beginning of the line in the email body.

3 4

[alsa-devel] BUG: soft lockup in snd_virmidi_output_trigger
by syzbot 09 Apr '18

09 Apr '18

Hello, syzbot hit the following crash on upstream commit 3fd14cdcc05a682b03743683ce3a726898b20555 (Fri Apr 6 19:15:41 2018 +0000) Merge tag 'mtd/for-4.17' of git://git.infradead.org/linux-mtd syzbot dashboard link: https://syzkaller.appspot.com/bug?extid=619d9f40141d826b097e Unfortunately, I don't have any reproducer for this crash yet. Raw console output: https://syzkaller.appspot.com/x/log.txt?id=4594231414882304 Kernel config: https://syzkaller.appspot.com/x/.config?id=-5813481738265533882 compiler: gcc (GCC) 8.0.1 20180301 (experimental) IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+619d9f40141d826b097e(a)syzkaller.appspotmail.com It will help syzbot understand when the bug is fixed. See footer for details. If you forward the report, please keep this part and the footer. 8021q: adding VLAN 0 to HW filter on device team0 IPv6: ADDRCONF(NETDEV_UP): team0: link is not ready 8021q: adding VLAN 0 to HW filter on device team0 IPv6: ADDRCONF(NETDEV_UP): team0: link is not ready 8021q: adding VLAN 0 to HW filter on device team0 watchdog: BUG: soft lockup - CPU#1 stuck for 134s! [syz-executor2:10431] Modules linked in: irq event stamp: 35856 hardirqs last enabled at (35855): [<ffffffff874a1734>] __raw_spin_unlock_irqrestore include/linux/spinlock_api_smp.h:160 [inline] hardirqs last enabled at (35855): [<ffffffff874a1734>] _raw_spin_unlock_irqrestore+0x74/0xc0 kernel/locking/spinlock.c:184 hardirqs last disabled at (35856): [<ffffffff87600901>] interrupt_entry+0xb1/0xf0 arch/x86/entry/entry_64.S:624 softirqs last enabled at (162): [<ffffffff87800778>] __do_softirq+0x778/0xaf5 kernel/softirq.c:311 softirqs last disabled at (95): [<ffffffff81467a61>] invoke_softirq kernel/softirq.c:365 [inline] softirqs last disabled at (95): [<ffffffff81467a61>] irq_exit+0x1d1/0x200 kernel/softirq.c:405 CPU: 1 PID: 10431 Comm: syz-executor2 Not tainted 4.16.0+ #4 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 RIP: 0010:arch_local_irq_restore arch/x86/include/asm/paravirt.h:783 [inline] RIP: 0010:__raw_spin_unlock_irqrestore include/linux/spinlock_api_smp.h:160 [inline] RIP: 0010:_raw_spin_unlock_irqrestore+0xa1/0xc0 kernel/locking/spinlock.c:184 RSP: 0018:ffff880184db7780 EFLAGS: 00000282 ORIG_RAX: ffffffffffffff13 RAX: dffffc0000000000 RBX: 0000000000000282 RCX: 0000000000000000 RDX: 1ffffffff1162e55 RSI: 0000000000000001 RDI: 0000000000000282 RBP: ffff880184db7790 R08: ffffed0035d21962 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000000 R12: ffff8801ae90cb08 R13: ffff880184db7810 R14: 0000000000000001 R15: ffff8801cb9a5880 FS: 00007fc943ad8700(0000) GS:ffff8801db100000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00007fc943ad7db8 CR3: 00000001b070d000 CR4: 00000000001406e0 DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000400 Call Trace: spin_unlock_irqrestore include/linux/spinlock.h:365 [inline] snd_virmidi_output_trigger+0x522/0x6c0 sound/core/seq/seq_virmidi.c:205 snd_rawmidi_output_trigger sound/core/rawmidi.c:150 [inline] snd_rawmidi_kernel_write1+0x519/0x700 sound/core/rawmidi.c:1288 snd_rawmidi_write+0x2e2/0xdc0 sound/core/rawmidi.c:1338 __vfs_write+0x10b/0x880 fs/read_write.c:485 vfs_write+0x1f8/0x560 fs/read_write.c:549 ksys_write+0xf9/0x250 fs/read_write.c:598 SYSC_write fs/read_write.c:610 [inline] SyS_write+0x24/0x30 fs/read_write.c:607 do_syscall_64+0x29e/0x9d0 arch/x86/entry/common.c:287 entry_SYSCALL_64_after_hwframe+0x42/0xb7 RIP: 0033:0x455259 RSP: 002b:00007fc943ad7c68 EFLAGS: 00000246 ORIG_RAX: 0000000000000001 RAX: ffffffffffffffda RBX: 00007fc943ad86d4 RCX: 0000000000455259 RDX: 00000000e78e624c RSI: 0000000020000040 RDI: 0000000000000014 RBP: 000000000072c010 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 00000000ffffffff R13: 00000000000006ca R14: 00000000006fd390 R15: 0000000000000002 Code: c7 a8 72 b1 88 48 b8 00 00 00 00 00 fc ff df 48 89 fa 48 c1 ea 03 80 3c 02 00 75 21 48 83 3d 4e 5b 67 01 00 74 0e 48 89 df 57 9d <0f> 1f 44 00 00 eb bb 0f 0b 0f 0b e8 6f 29 68 fa eb 97 e8 68 29 --- This bug is generated by a dumb bot. It may contain errors. See https://goo.gl/tpsmEJ for details. Direct all questions to syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. If you forgot to add the Reported-by tag, once the fix for this bug is merged into any tree, please reply to this email with: #syz fix: exact-commit-title To mark this as a duplicate of another syzbot report, please reply with: #syz dup: exact-subject-of-another-report If it's a one-off invalid bug report, please reply with: #syz invalid Note: if the crash happens again, it will cause creation of a new bug report. Note: all commands must start from beginning of the line in the email body.

3 2

[alsa-devel] [PATCH 1/2] sound: ad1889: Replace mdelay with usleep_range in snd_ad1889_init
by Jia-Ju Bai 09 Apr '18

09 Apr '18

snd_ad1889_init() is never called in atomic context. The call chain ending up at snd_ad1889_init() is: [1] snd_ad1889_init() <- snd_ad1889_create() <- snd_ad1889_probe() snd_ad1889_probe() is only set as ".probe" in struct pci_driver. This function is not called in atomic context. Despite never getting called from atomic context, snd_ad1889_init() calls mdelay for busy wait. This is not necessary and can be replaced with usleep_range to avoid busy waiting. This is found by a static analysis tool named DCNS written by myself. Signed-off-by: Jia-Ju Bai <baijiaju1990(a)gmail.com> --- sound/pci/ad1889.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c index 0bf2c04..d4965eb 100644 --- a/sound/pci/ad1889.c +++ b/sound/pci/ad1889.c @@ -872,7 +872,7 @@ struct snd_ad1889 { ad1889_writew(chip, AD_DS_CCS, AD_DS_CCS_CLKEN); /* turn on clock */ ad1889_readw(chip, AD_DS_CCS); /* flush posted write */ - mdelay(10); + usleep_range(10000, 11000); /* enable Master and Target abort interrupts */ ad1889_writel(chip, AD_DMA_DISR, AD_DMA_DISR_PMAE | AD_DMA_DISR_PTAE); -- 1.9.1

2 1

[alsa-devel] [PATCH 2/2] sound: ad1889: Replace mdelay with usleep_range in snd_ad1889_ac97_ready
by Jia-Ju Bai 09 Apr '18

09 Apr '18

snd_ad1889_ac97_ready() is never called in atomic context. The call chain ending up at snd_ad1889_ac97_ready() is: [1] snd_ad1889_ac97_ready() <- snd_ad1889_ac97_xinit() <- snd_ad1889_ac97_init() <- snd_ad1889_probe() snd_ad1889_probe() is only set as ".probe" in struct pci_driver. This function is not called in atomic context. Despite never getting called from atomic context, snd_ad1889_ac97_ready() calls mdelay for busy wait. This is not necessary and can be replaced with usleep_range to avoid busy waiting. This is found by a static analysis tool named DCNS written by myself. Signed-off-by: Jia-Ju Bai <baijiaju1990(a)gmail.com> --- sound/pci/ad1889.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c index 0bf2c04..a2a8e1d 100644 --- a/sound/pci/ad1889.c +++ b/sound/pci/ad1889.c @@ -258,7 +258,7 @@ struct snd_ad1889 { while (!(ad1889_readw(chip, AD_AC97_ACIC) & AD_AC97_ACIC_ACRDY) && --retry) - mdelay(1); + usleep_range(1000, 2000); if (!retry) { dev_err(chip->card->dev, "[%s] Link is not ready.\n", __func__); -- 1.9.1

1 0

[alsa-devel] INFO: rcu detected stall in io_playback_transfer
by syzbot 09 Apr '18

09 Apr '18

Hello, syzbot hit the following crash on upstream commit e02d37bf55a9a36f22427fd6dd733fe104d817b6 (Thu Apr 5 17:42:07 2018 +0000) Merge tag 'sound-4.17-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound syzbot dashboard link: https://syzkaller.appspot.com/bug?extid=4f2016cf5185da7759dc Unfortunately, I don't have any reproducer for this crash yet. Raw console output: https://syzkaller.appspot.com/x/log.txt?id=6609942245015552 Kernel config: https://syzkaller.appspot.com/x/.config?id=-4805825610197092128 compiler: gcc (GCC) 8.0.1 20180301 (experimental) IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+4f2016cf5185da7759dc(a)syzkaller.appspotmail.com It will help syzbot understand when the bug is fixed. See footer for details. If you forward the report, please keep this part and the footer. INFO: rcu_sched self-detected stall on CPU 1-....: (124999 ticks this GP) idle=212/1/4611686018427387906 softirq=57947/57947 fqs=31225 (t=125000 jiffies g=30827 c=30826 q=171) NMI backtrace for cpu 1 CPU: 1 PID: 13330 Comm: syz-executor4 Not tainted 4.16.0+ #2 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: <IRQ> __dump_stack lib/dump_stack.c:17 [inline] dump_stack+0x1b9/0x29f lib/dump_stack.c:53 nmi_cpu_backtrace.cold.4+0x19/0xce lib/nmi_backtrace.c:103 nmi_trigger_cpumask_backtrace+0x151/0x192 lib/nmi_backtrace.c:62 arch_trigger_cpumask_backtrace+0x14/0x20 arch/x86/kernel/apic/hw_nmi.c:38 trigger_single_cpu_backtrace include/linux/nmi.h:156 [inline] rcu_dump_cpu_stacks+0x175/0x1c2 kernel/rcu/tree.c:1376 print_cpu_stall kernel/rcu/tree.c:1525 [inline] check_cpu_stall.isra.61.cold.80+0x36c/0x59a kernel/rcu/tree.c:1593 __rcu_pending kernel/rcu/tree.c:3356 [inline] rcu_pending kernel/rcu/tree.c:3401 [inline] rcu_check_callbacks+0x21b/0xad0 kernel/rcu/tree.c:2763 update_process_times+0x2d/0x70 kernel/time/timer.c:1636 tick_sched_handle+0xa0/0x180 kernel/time/tick-sched.c:171 tick_sched_timer+0x42/0x130 kernel/time/tick-sched.c:1179 __run_hrtimer kernel/time/hrtimer.c:1337 [inline] __hrtimer_run_queues+0x3e3/0x10a0 kernel/time/hrtimer.c:1399 hrtimer_interrupt+0x286/0x650 kernel/time/hrtimer.c:1457 local_apic_timer_interrupt arch/x86/kernel/apic/apic.c:1025 [inline] smp_apic_timer_interrupt+0x15d/0x710 arch/x86/kernel/apic/apic.c:1050 apic_timer_interrupt+0xf/0x20 arch/x86/entry/entry_64.S:862 </IRQ> RIP: 0010:snd_pcm_oss_write3+0xf6/0x220 sound/core/oss/pcm_oss.c:1236 RSP: 0018:ffff88018f57ed00 EFLAGS: 00000282 ORIG_RAX: ffffffffffffff13 RAX: ffffffffffffffe0 RBX: ffffffffffffffe0 RCX: 0000000000000000 RDX: 0000000000000000 RSI: 00000000ffffffe0 RDI: 00000000ffffffe0 RBP: ffff88018f57ed48 R08: ffff8801b29daaf8 R09: 0000000000000006 R10: ffff8801b29da280 R11: 0000000000000000 R12: 0000000000000001 R13: ffff8801ce8c56c0 R14: ffff8801aa45a300 R15: ffffffffffffffe0 io_playback_transfer+0x274/0x310 sound/core/oss/io.c:47 snd_pcm_plug_write_transfer+0x36c/0x470 sound/core/oss/pcm_plugin.c:619 snd_pcm_oss_write2+0x25c/0x460 sound/core/oss/pcm_oss.c:1365 snd_pcm_oss_sync1+0x332/0x5a0 sound/core/oss/pcm_oss.c:1606 snd_pcm_oss_sync.isra.29+0x790/0x980 sound/core/oss/pcm_oss.c:1682 snd_pcm_oss_release+0x214/0x290 sound/core/oss/pcm_oss.c:2559 __fput+0x34d/0x890 fs/file_table.c:209 ____fput+0x15/0x20 fs/file_table.c:243 task_work_run+0x1e4/0x290 kernel/task_work.c:113 exit_task_work include/linux/task_work.h:22 [inline] do_exit+0x1aee/0x2730 kernel/exit.c:865 do_group_exit+0x16f/0x430 kernel/exit.c:968 get_signal+0x886/0x1960 kernel/signal.c:2469 do_signal+0x90/0x2020 arch/x86/kernel/signal.c:810 exit_to_usermode_loop+0x28a/0x310 arch/x86/entry/common.c:162 prepare_exit_to_usermode arch/x86/entry/common.c:196 [inline] syscall_return_slowpath arch/x86/entry/common.c:265 [inline] do_syscall_64+0x792/0x9d0 arch/x86/entry/common.c:292 entry_SYSCALL_64_after_hwframe+0x42/0xb7 RIP: 0033:0x4552d9 RSP: 002b:00007f5c2a0e9c68 EFLAGS: 00000246 ORIG_RAX: 0000000000000023 RAX: fffffffffffffdfc RBX: 00007f5c2a0ea6d4 RCX: 00000000004552d9 RDX: 0000000000000000 RSI: 00000000200001c0 RDI: 0000000020000240 RBP: 000000000072c010 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 00000000ffffffff R13: 0000000000000413 R14: 00000000006f9268 R15: 0000000000000002 --- This bug is generated by a dumb bot. It may contain errors. See https://goo.gl/tpsmEJ for details. Direct all questions to syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. If you forgot to add the Reported-by tag, once the fix for this bug is merged into any tree, please reply to this email with: #syz fix: exact-commit-title To mark this as a duplicate of another syzbot report, please reply with: #syz dup: exact-subject-of-another-report If it's a one-off invalid bug report, please reply with: #syz invalid Note: if the crash happens again, it will cause creation of a new bug report. Note: all commands must start from beginning of the line in the email body.

4 4

[alsa-devel] [PATCH] sound: soc: intel: cht_bsw_rt5645: Replace GFP_ATOMIC with GFP_KERNEL in snd_cht_mc_probe
by Jia-Ju Bai 09 Apr '18

09 Apr '18

snd_cht_mc_probe() is never called in atomic context. This function is only set as ".probe" in "struct platform_driver". Despite never getting called from atomic context, snd_cht_mc_probe() calls devm_kzalloc() with GFP_ATOMIC, which waits busily for allocation. GFP_ATOMIC is not necessary and can be replaced with GFP_KERNEL, to avoid busy waiting and improve the possibility of sucessful allocation. This is found by a static analysis tool named DCNS written by myself. Signed-off-by: Jia-Ju Bai <baijiaju1990(a)gmail.com> --- sound/soc/intel/boards/cht_bsw_rt5645.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 5bcde01..d31d0e4 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -555,7 +555,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) bool found = false; bool is_bytcr = false; - drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_ATOMIC); + drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL); if (!drv) return -ENOMEM; -- 1.9.1

1 0

[alsa-devel] [PATCH] sound: soc: intel: bytcr_rt5640: Replace GFP_ATOMIC with GFP_KERNEL in snd_byt_rt5640_mc_probe
by Jia-Ju Bai 09 Apr '18

09 Apr '18

snd_byt_rt5640_mc_probe() is never called in atomic context. This function is only set as ".probe" in "struct platform_driver". Despite never getting called from atomic context, snd_byt_rt5640_mc_probe() calls devm_kzalloc() with GFP_ATOMIC, which waits busily for allocation. GFP_ATOMIC is not necessary and can be replaced with GFP_KERNEL, to avoid busy waiting and improve the possibility of sucessful allocation. This is found by a static analysis tool named DCNS written by myself. Signed-off-by: Jia-Ju Bai <baijiaju1990(a)gmail.com> --- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 4a76b09..e7705e0 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -766,7 +766,7 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) struct byt_rt5640_private *priv; is_bytcr = false; - priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_ATOMIC); + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); if (!priv) return -ENOMEM; -- 1.9.1

1 0

[alsa-devel] Conversion to int16_t and resolution loss with rate converter plugins
by Flavio Protasio Ribeiro 09 Apr '18

09 Apr '18

Hi Folks, The libsamplerate and speex rate converter plugins only implement the convert_s16 callback from snd_pcm_rate_ops_t. This has ALSA convert the buffer to int16_t before handing it over to the plugin for resampling. On devices with more than 16 effective bits, this implies words gets truncated and bits are lost. Specifically for libsamplerate, the resampling is internally implemented with floats, so truncating to 16 bits does not offer a computational benefit. I get why only convert_s16 is implemented for plugins that aren't part of the main alsa-lib package. The more generic convert callback comes with the burden having the plugin support conversion from/to any of the ALSA data types (or alternatively, having the plugin source include plugin_ops.h and its dependencies). So here's what I propose: replace convert_s16 with a convert_s32 callback which uses int32_t instead of int16_t. This would preserve all bits from the device and keep the plugin interface simple. Of course this means plugins have to be updated to use convert_s32 and rebuilt. Thoughts? -Flavio

3 9

[alsa-devel] [PATCH] ALSA: hda/realtek: Enable audio line out on ASUS D640SA
by Jian-Hong Pan 09 Apr '18

09 Apr '18

This ASUS D640SA desktop whose mother board is D640MB has - two jacks which are a headphone and a mic on the front panel, - three jacks which are a mic, a line out and a line in on the rear panel - one internal speaker. If I plug a headphone to the front headphone jack, there will be sound through the headphone jack, and no sound through the internal speaker. If I unplug the headphone from the the headphone jack, there will be sound through the internal speaker. And always no sound through rear line out, when I plug a headphone or an externel speaker to the rear line out jack. Besides, I had checked and toggled the Auto-Mute Mode in alsamixer, but the rear line out still was not working. Then I checked the sound settings in GUI, and found there was no "Line Out" could be chosen, only the "Headphones" and "HDMI/DisplayPort". However, system does know that there is an "Intel PCH Line Out". [ 10.089082] snd_hda_codec_realtek hdaudioC0D0: autoconfig for ALC887-VD: line_outs=1 (0x14/0x0/0x0/0x0/0x0) type:line [ 10.089083] snd_hda_codec_realtek hdaudioC0D0: speaker_outs=1 (0x1a/0x0/0x0/0x0/0x0) [ 10.089084] snd_hda_codec_realtek hdaudioC0D0: hp_outs=1 (0x1b/0x0/0x0/0x0/0x0) [ 10.089085] snd_hda_codec_realtek hdaudioC0D0: mono: mono_out=0x0 [ 10.089086] snd_hda_codec_realtek hdaudioC0D0: inputs: [ 10.089087] snd_hda_codec_realtek hdaudioC0D0: Rear Mic=0x18 [ 10.089088] snd_hda_codec_realtek hdaudioC0D0: Front Mic=0x19 [ 10.089089] snd_hda_codec_realtek hdaudioC0D0: Line=0x15 [ 10.104387] input: HDA Intel PCH Rear Mic as /devices/pci0000:00/0000:00:1f.3/sound/card0/input9 [ 10.104416] input: HDA Intel PCH Front Mic as /devices/pci0000:00/0000:00:1f.3/sound/card0/input10 [ 10.104441] input: HDA Intel PCH Line as /devices/pci0000:00/0000:00:1f.3/sound/card0/input11 [ 10.104467] input: HDA Intel PCH Line Out as /devices/pci0000:00/0000:00:1f.3/sound/card0/input12 [ 10.104494] input: HDA Intel PCH Front Headphone as /devices/pci0000:00/0000:00:1f.3/sound/card0/input13 Consequently, I checked the pin widgets' default configuration values: - Node 0x14 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Pin Default 0x01014010: [Jack] Line Out at Ext Rear - Node 0x1b [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out Pin Default 0x02214030: [Jack] HP Out at Ext Front Because the headphone jack (Node ID:0x1b) locates on the desktop's front panel, not rear panel, I change the headphone jack's configuration from primary chassis to separate chassis. So, the configuration value of Node ID:0x1b should be 0x22214030. Additionally, I toggle the Auto-Mute Mode of Realtek codecs to “Speaker Only” which makes signal outputs through line out jack when the "Line Out" is chosen in the sound settings. Signed-off-by: Jian-Hong Pan <jian-hong(a)endlessm.com> --- sound/pci/hda/patch_realtek.c | 34 ++++++++++++++++++++++++++++++++++ 1 file changed, 34 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index aef1f52db7d9..a066d84c6478 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1810,6 +1810,8 @@ enum { ALC887_FIXUP_BASS_CHMAP, ALC1220_FIXUP_GB_DUAL_CODECS, ALC1220_FIXUP_CLEVO_P950, + ALC887_FIXUP_ASUS_AUTOMUTE_MODE, + ALC887_FIXUP_ASUS_FRONT_HP, }; static void alc889_fixup_coef(struct hda_codec *codec, @@ -2044,6 +2046,24 @@ static void alc1220_fixup_clevo_p950(struct hda_codec *codec, snd_hda_override_conn_list(codec, 0x1b, 1, conn1); } +/* fix the default Auto-Mute Mode in mixer */ +static void alc887_fixup_asus_automute_mode(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct snd_kcontrol *kctl; + struct snd_ctl_elem_value *uctl; + + kctl = snd_hda_find_mixer_ctl(codec, "Auto-Mute Mode"); + if (!kctl) + return; + uctl = kzalloc(sizeof(*uctl), GFP_KERNEL); + if (!uctl) + return; + uctl->value.enumerated.item[0] = 1; + kctl->put(kctl, uctl); + kfree(uctl); +} + static const struct hda_fixup alc882_fixups[] = { [ALC882_FIXUP_ABIT_AW9D_MAX] = { .type = HDA_FIXUP_PINS, @@ -2288,6 +2308,19 @@ static const struct hda_fixup alc882_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc1220_fixup_clevo_p950, }, + [ALC887_FIXUP_ASUS_AUTOMUTE_MODE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc887_fixup_asus_automute_mode, + }, + [ALC887_FIXUP_ASUS_FRONT_HP] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + {0x1b, 0x22214030}, + {} + }, + .chained = true, + .chain_id = ALC887_FIXUP_ASUS_AUTOMUTE_MODE, + }, }; static const struct snd_pci_quirk alc882_fixup_tbl[] = { @@ -2321,6 +2354,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x13c2, "Asus A7M", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x1043, 0x1873, "ASUS W90V", ALC882_FIXUP_ASUS_W90V), SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_FIXUP_ASUS_W2JC), + SND_PCI_QUIRK(0x1043, 0x2160, "ASUSPRO D640MB", ALC887_FIXUP_ASUS_FRONT_HP), SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_FIXUP_EEE1601), SND_PCI_QUIRK(0x1043, 0x84bc, "ASUS ET2700", ALC887_FIXUP_ASUS_BASS), SND_PCI_QUIRK(0x1043, 0x8691, "ASUS ROG Ranger VIII", ALC882_FIXUP_GPIO3), -- 2.11.0

2 4

[alsa-devel] INFO: rcu detected stall in snd_pcm_stream_unlock_irq
by syzbot 08 Apr '18

08 Apr '18

Hello, syzbot hit the following crash on upstream commit 3fd14cdcc05a682b03743683ce3a726898b20555 (Fri Apr 6 19:15:41 2018 +0000) Merge tag 'mtd/for-4.17' of git://git.infradead.org/linux-mtd syzbot dashboard link: https://syzkaller.appspot.com/bug?extid=584505e6ccdbedcf7629 Unfortunately, I don't have any reproducer for this crash yet. Raw console output: https://syzkaller.appspot.com/x/log.txt?id=5653573555716096 Kernel config: https://syzkaller.appspot.com/x/.config?id=-5813481738265533882 compiler: gcc (GCC) 8.0.1 20180301 (experimental) IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+584505e6ccdbedcf7629(a)syzkaller.appspotmail.com It will help syzbot understand when the bug is fixed. See footer for details. If you forward the report, please keep this part and the footer. random: crng init done INFO: rcu_sched self-detected stall on CPU 1-....: (124996 ticks this GP) idle=c9a/1/4611686018427387906 softirq=17918/17918 fqs=31239 (t=125000 jiffies g=9607 c=9606 q=187) NMI backtrace for cpu 1 CPU: 1 PID: 6608 Comm: syz-executor4 Not tainted 4.16.0+ #4 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: <IRQ> __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0x1b9/0x294 lib/dump_stack.c:113 nmi_cpu_backtrace.cold.4+0x19/0xce lib/nmi_backtrace.c:103 nmi_trigger_cpumask_backtrace+0x151/0x192 lib/nmi_backtrace.c:62 arch_trigger_cpumask_backtrace+0x14/0x20 arch/x86/kernel/apic/hw_nmi.c:38 trigger_single_cpu_backtrace include/linux/nmi.h:156 [inline] rcu_dump_cpu_stacks+0x175/0x1c2 kernel/rcu/tree.c:1376 print_cpu_stall kernel/rcu/tree.c:1525 [inline] check_cpu_stall.isra.61.cold.80+0x36c/0x59a kernel/rcu/tree.c:1593 __rcu_pending kernel/rcu/tree.c:3356 [inline] rcu_pending kernel/rcu/tree.c:3401 [inline] rcu_check_callbacks+0x21b/0xad0 kernel/rcu/tree.c:2763 update_process_times+0x2d/0x70 kernel/time/timer.c:1636 tick_sched_handle+0xa0/0x180 kernel/time/tick-sched.c:171 tick_sched_timer+0x42/0x130 kernel/time/tick-sched.c:1179 __run_hrtimer kernel/time/hrtimer.c:1337 [inline] __hrtimer_run_queues+0x3e3/0x10a0 kernel/time/hrtimer.c:1399 hrtimer_interrupt+0x286/0x650 kernel/time/hrtimer.c:1457 local_apic_timer_interrupt arch/x86/kernel/apic/apic.c:1025 [inline] smp_apic_timer_interrupt+0x15d/0x710 arch/x86/kernel/apic/apic.c:1050 apic_timer_interrupt+0xf/0x20 arch/x86/entry/entry_64.S:862 </IRQ> RIP: 0010:check_kcov_mode kernel/kcov.c:67 [inline] RIP: 0010:__sanitizer_cov_trace_pc+0x20/0x50 kernel/kcov.c:101 RSP: 0018:ffff8801c82feb50 EFLAGS: 00000246 ORIG_RAX: ffffffffffffff13 RAX: ffff8801b5330040 RBX: 0000000000000000 RCX: 1ffff10036a6611b RDX: 0000000000000000 RSI: ffffffff859d514c RDI: ffffffff88b172b8 RBP: ffff8801c82feb50 R08: ffff8801b53308b8 R09: 0000000000000006 R10: ffff8801b5330040 R11: 0000000000000000 R12: 0000000000000004 R13: 00000000ffffffe0 R14: ffff8801d128ab00 R15: ffff8801ce80e040 snd_pcm_stream_unlock_irq+0xbc/0xf0 sound/core/pcm_native.c:167 __snd_pcm_lib_xfer+0x739/0x1d10 sound/core/pcm_lib.c:2246 snd_pcm_oss_write3+0xe9/0x220 sound/core/oss/pcm_oss.c:1236 snd_pcm_oss_write2+0x34c/0x460 sound/core/oss/pcm_oss.c:1373 snd_pcm_oss_sync1+0x332/0x5a0 sound/core/oss/pcm_oss.c:1606 snd_pcm_oss_sync.isra.29+0x790/0x980 sound/core/oss/pcm_oss.c:1682 snd_pcm_oss_release+0x214/0x290 sound/core/oss/pcm_oss.c:2559 __fput+0x34d/0x890 fs/file_table.c:209 ____fput+0x15/0x20 fs/file_table.c:243 task_work_run+0x1e4/0x290 kernel/task_work.c:113 exit_task_work include/linux/task_work.h:22 [inline] do_exit+0x1aee/0x2730 kernel/exit.c:865 do_group_exit+0x16f/0x430 kernel/exit.c:968 get_signal+0x886/0x1960 kernel/signal.c:2469 do_signal+0x98/0x2040 arch/x86/kernel/signal.c:810 exit_to_usermode_loop+0x28a/0x310 arch/x86/entry/common.c:162 prepare_exit_to_usermode arch/x86/entry/common.c:196 [inline] syscall_return_slowpath arch/x86/entry/common.c:265 [inline] do_syscall_64+0x792/0x9d0 arch/x86/entry/common.c:292 entry_SYSCALL_64_after_hwframe+0x42/0xb7 RIP: 0033:0x455259 RSP: 002b:00007f20e223bce8 EFLAGS: 00000246 ORIG_RAX: 00000000000000ca RAX: fffffffffffffe00 RBX: 000000000072c038 RCX: 0000000000455259 RDX: 0000000000000000 RSI: 0000000000000000 RDI: 000000000072c038 RBP: 000000000072c038 R08: 0000000000000000 R09: 000000000072c010 R10: 0000000000000000 R11: 0000000000000246 R12: 0000000000000000 R13: 0000000000a3e81f R14: 00007f20e223c9c0 R15: 0000000000000002 --- This bug is generated by a dumb bot. It may contain errors. See https://goo.gl/tpsmEJ for details. Direct all questions to syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. If you forgot to add the Reported-by tag, once the fix for this bug is merged into any tree, please reply to this email with: #syz fix: exact-commit-title To mark this as a duplicate of another syzbot report, please reply with: #syz dup: exact-subject-of-another-report If it's a one-off invalid bug report, please reply with: #syz invalid Note: if the crash happens again, it will cause creation of a new bug report. Note: all commands must start from beginning of the line in the email body.

1 0

Re: [alsa-devel] [PATCH 14/15] ARM: pxa: change SSP devices allocation
by Robert Jarzmik 08 Apr '18

08 Apr '18

Arnd Bergmann <arnd(a)arndb.de> writes: chop chop ... removed several mail recipients to leave only the ASoC / PXA subset ... > On Mon, Apr 2, 2018 at 4:26 PM, Robert Jarzmik <robert.jarzmik(a)free.fr> wrote: > >> >> +static struct pxa_ssp_info pxa_ssp_infos[] = { >> + { .dma_chan_rx_name = "ssp1_rx", .dma_chan_tx_name = "ssp1_tx", }, >> + { .dma_chan_rx_name = "ssp1_rx", .dma_chan_tx_name = "ssp1_tx", }, >> + { .dma_chan_rx_name = "ssp2_rx", .dma_chan_tx_name = "ssp2_tx", }, >> + { .dma_chan_rx_name = "ssp2_rx", .dma_chan_tx_name = "ssp2_tx", }, >> + { .dma_chan_rx_name = "ssp3_rx", .dma_chan_tx_name = "ssp3_tx", }, >> + { .dma_chan_rx_name = "ssp3_rx", .dma_chan_tx_name = "ssp3_tx", }, >> + { .dma_chan_rx_name = "ssp4_rx", .dma_chan_tx_name = "ssp4_tx", }, >> + { .dma_chan_rx_name = "ssp4_rx", .dma_chan_tx_name = "ssp4_tx", }, >> +}; > > This part looks odd to me, you're adding an extra level of indirection to > do two stages of lookups in some form of platform data. That's unfortunately right. > Why can't you just always use "rx" and "tx" as the names here? Well I couldn't. I'll explain you why, and maybe you'll find a better solution. That all is related to how ASoC and SSP interact. If I remember correctly, here is how it works : - the DMA channel is requested in sound/arm/pxa2xx-pcm-lib.c:128 return snd_dmaengine_pcm_open( substream, dma_request_slave_channel(rtd->platform->dev, The trick is that the device here is _not_ the SSP one, it's if memory serves me well the pxa-pcm-audio one. As a consequence, the device cannot be used to differenciate which SSP exactly is providing the sound samples stream. This information is nevertheless required to choose the correct requestor line, which is a 1-to-1 match to the SSP port. The indirection in the channel name is used to choose the correct requestor line for a given SSP port providing the samples. It also must be underlined that this dma request serves both AC97 and SSP as sample providers. > (also, I don't see why each line is duplicated, but I'm sure there's > an easy answer for that). Ahh that is an unfortunate rebase most probably :) Cheers. -- Robert

2 4

[alsa-devel] [PATCH] slimbus: Fix out-of-bounds access in slim_slicesize()
by Geert Uytterhoeven 08 Apr '18

08 Apr '18

With gcc-4.1.2: slimbus/messaging.c: In function ‘slim_slicesize’: slimbus/messaging.c:186: warning: statement with no effect Indeed, clamp() is a macro not operating in-place, but returning the clamped value. Hence the value is not clamped at all, which may lead to an out-of-bounds access. Fix this by assigning the clamped value. Fixes: afbdcc7c384b0d44 ("slimbus: Add messaging APIs to slimbus framework") Signed-off-by: Geert Uytterhoeven <geert(a)linux-m68k.org> --- drivers/slimbus/messaging.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/slimbus/messaging.c b/drivers/slimbus/messaging.c index 884419c37e8419c8..457ea1f8db309cb9 100644 --- a/drivers/slimbus/messaging.c +++ b/drivers/slimbus/messaging.c @@ -183,7 +183,7 @@ static u16 slim_slicesize(int code) 0, 1, 2, 3, 3, 4, 4, 5, 5, 5, 5, 6, 6, 6, 6, 7 }; - clamp(code, 1, (int)ARRAY_SIZE(sizetocode)); + code = clamp(code, 1, (int)ARRAY_SIZE(sizetocode)); return sizetocode[code - 1]; } -- 2.7.4

1 0

[alsa-devel] [PATCH v2 1/3] dell-led: Change dell-led.h to dell-common.h
by Kai-Heng Feng 07 Apr '18

07 Apr '18

This header will be used for more than just led. Change it to a more generic name. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- v2: Mario suggested to squash the HDA part into the same series. drivers/platform/x86/dell-laptop.c | 2 +- include/linux/{dell-led.h => dell-common.h} | 4 ++-- sound/pci/hda/dell_wmi_helper.c | 2 +- 3 files changed, 4 insertions(+), 4 deletions(-) rename include/linux/{dell-led.h => dell-common.h} (61%) diff --git a/drivers/platform/x86/dell-laptop.c b/drivers/platform/x86/dell-laptop.c index c52c6723374b..8ba820e6c3d0 100644 --- a/drivers/platform/x86/dell-laptop.c +++ b/drivers/platform/x86/dell-laptop.c @@ -29,7 +29,7 @@ #include <linux/mm.h> #include <linux/i8042.h> #include <linux/debugfs.h> -#include <linux/dell-led.h> +#include <linux/dell-common.h> #include <linux/seq_file.h> #include <acpi/video.h> #include "dell-rbtn.h" diff --git a/include/linux/dell-led.h b/include/linux/dell-common.h similarity index 61% rename from include/linux/dell-led.h rename to include/linux/dell-common.h index 92521471517f..37e4b614dd74 100644 --- a/include/linux/dell-led.h +++ b/include/linux/dell-common.h @@ -1,6 +1,6 @@ /* SPDX-License-Identifier: GPL-2.0 */ -#ifndef __DELL_LED_H__ -#define __DELL_LED_H__ +#ifndef __DELL_COMMON_H__ +#define __DELL_COMMON_H__ int dell_micmute_led_set(int on); diff --git a/sound/pci/hda/dell_wmi_helper.c b/sound/pci/hda/dell_wmi_helper.c index 1b48a8c19d28..56050cc3c0ee 100644 --- a/sound/pci/hda/dell_wmi_helper.c +++ b/sound/pci/hda/dell_wmi_helper.c @@ -4,7 +4,7 @@ */ #if IS_ENABLED(CONFIG_DELL_LAPTOP) -#include <linux/dell-led.h> +#include <linux/dell-common.h> enum { MICMUTE_LED_ON, -- 2.15.1

6 24

Re: [alsa-devel] INFO: rcu detected stall in snd_pcm_oss_write3 (2)
by syzbot 07 Apr '18

07 Apr '18

syzbot has found reproducer for the following crash on upstream commit 38c23685b273cfb4ccf31a199feccce3bdcb5d83 (Fri Apr 6 04:29:35 2018 +0000) Merge tag 'armsoc-drivers' of git://git.kernel.org/pub/scm/linux/kernel/git/arm/arm-soc syzbot dashboard link: https://syzkaller.appspot.com/bug?extid=150189c103427d31a053 So far this crash happened 15 times on upstream. C reproducer: https://syzkaller.appspot.com/x/repro.c?id=5405588854931456 syzkaller reproducer: https://syzkaller.appspot.com/x/repro.syz?id=5561439796330496 Raw console output: https://syzkaller.appspot.com/x/log.txt?id=5697900571000832 Kernel config: https://syzkaller.appspot.com/x/.config?id=-5813481738265533882 compiler: gcc (GCC) 8.0.1 20180301 (experimental) IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+150189c103427d31a053(a)syzkaller.appspotmail.com It will help syzbot understand when the bug is fixed. IPVS: ftp: loaded support on port[0] = 21 INFO: rcu_sched self-detected stall on CPU 1-....: (124999 ticks this GP) idle=622/1/4611686018427387906 softirq=10596/10596 fqs=31239 (t=125000 jiffies g=4952 c=4951 q=20) NMI backtrace for cpu 1 CPU: 1 PID: 4474 Comm: syzkaller631460 Not tainted 4.16.0+ #3 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: <IRQ> __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0x1b9/0x294 lib/dump_stack.c:113 nmi_cpu_backtrace.cold.4+0x19/0xce lib/nmi_backtrace.c:103 nmi_trigger_cpumask_backtrace+0x151/0x192 lib/nmi_backtrace.c:62 arch_trigger_cpumask_backtrace+0x14/0x20 arch/x86/kernel/apic/hw_nmi.c:38 trigger_single_cpu_backtrace include/linux/nmi.h:156 [inline] rcu_dump_cpu_stacks+0x175/0x1c2 kernel/rcu/tree.c:1376 print_cpu_stall kernel/rcu/tree.c:1525 [inline] check_cpu_stall.isra.61.cold.80+0x36c/0x59a kernel/rcu/tree.c:1593 __rcu_pending kernel/rcu/tree.c:3356 [inline] rcu_pending kernel/rcu/tree.c:3401 [inline] rcu_check_callbacks+0x21b/0xad0 kernel/rcu/tree.c:2763 update_process_times+0x2d/0x70 kernel/time/timer.c:1636 tick_sched_handle+0xa0/0x180 kernel/time/tick-sched.c:171 tick_sched_timer+0x42/0x130 kernel/time/tick-sched.c:1179 __run_hrtimer kernel/time/hrtimer.c:1337 [inline] __hrtimer_run_queues+0x3e3/0x10a0 kernel/time/hrtimer.c:1399 hrtimer_interrupt+0x286/0x650 kernel/time/hrtimer.c:1457 local_apic_timer_interrupt arch/x86/kernel/apic/apic.c:1025 [inline] smp_apic_timer_interrupt+0x15d/0x710 arch/x86/kernel/apic/apic.c:1050 apic_timer_interrupt+0xf/0x20 arch/x86/entry/entry_64.S:862 </IRQ> RIP: 0010:__snd_pcm_lib_xfer+0x768/0x1d10 sound/core/pcm_lib.c:2111 RSP: 0018:ffff8801b7b76ea8 EFLAGS: 00000286 ORIG_RAX: ffffffffffffff13 RAX: 0000000000000000 RBX: ffffffffffffffe0 RCX: ffffed0036f6edec RDX: 0000000000000000 RSI: ffffffff859ff04e RDI: ffffed0036f6edf0 RBP: ffff8801b7b77148 R08: ffff8801afeae9b8 R09: 0000000000000006 R10: ffff8801afeae140 R11: 0000000000000000 R12: 0000000000000004 R13: 00000000ffffffe0 R14: ffff8801af2165c0 R15: ffff8801ceaec000 snd_pcm_oss_write3+0xe9/0x220 sound/core/oss/pcm_oss.c:1236 io_playback_transfer+0x274/0x310 sound/core/oss/io.c:47 snd_pcm_plug_write_transfer+0x36c/0x470 sound/core/oss/pcm_plugin.c:619 snd_pcm_oss_write2+0x25c/0x460 sound/core/oss/pcm_oss.c:1365 snd_pcm_oss_sync1+0x332/0x5a0 sound/core/oss/pcm_oss.c:1606 snd_pcm_oss_sync.isra.29+0x790/0x980 sound/core/oss/pcm_oss.c:1682 snd_pcm_oss_release+0x214/0x290 sound/core/oss/pcm_oss.c:2559 __fput+0x34d/0x890 fs/file_table.c:209 ____fput+0x15/0x20 fs/file_table.c:243 task_work_run+0x1e4/0x290 kernel/task_work.c:113 exit_task_work include/linux/task_work.h:22 [inline] do_exit+0x1aee/0x2730 kernel/exit.c:865 do_group_exit+0x16f/0x430 kernel/exit.c:968 SYSC_exit_group kernel/exit.c:979 [inline] SyS_exit_group+0x1d/0x20 kernel/exit.c:977 do_syscall_64+0x29e/0x9d0 arch/x86/entry/common.c:287 entry_SYSCALL_64_after_hwframe+0x42/0xb7 RIP: 0033:0x4446e9 RSP: 002b:00007ffe29466408 EFLAGS: 00000202 ORIG_RAX: 00000000000000e7 RAX: ffffffffffffffda RBX: ffffffffffffffff RCX: 00000000004446e9 RDX: 00000000004446e9 RSI: 0000000000000080 RDI: 0000000000000001 RBP: 00000000006cf018 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000202 R12: 0000000000401ff0 R13: 0000000000402080 R14: 0000000000000000 R15: 0000000000000000

3 4

Re: [alsa-devel] INFO: rcu detected stall in snd_pcm_stream_lock_irq
by Dmitry Vyukov 07 Apr '18

07 Apr '18

On Fri, Apr 6, 2018 at 3:24 PM, syzbot <syzbot+74e011b2046f4322d178(a)syzkaller.appspotmail.com> wrote: > Hello, > > syzbot hit the following crash on upstream commit > 38c23685b273cfb4ccf31a199feccce3bdcb5d83 (Fri Apr 6 04:29:35 2018 +0000) > Merge tag 'armsoc-drivers' of > git://git.kernel.org/pub/scm/linux/kernel/git/arm/arm-soc > syzbot dashboard link: > https://syzkaller.appspot.com/bug?extid=74e011b2046f4322d178 > > Unfortunately, I don't have any reproducer for this crash yet. > Raw console output: > https://syzkaller.appspot.com/x/log.txt?id=4582053840420864 > Kernel config: > https://syzkaller.appspot.com/x/.config?id=-5813481738265533882 > compiler: gcc (GCC) 8.0.1 20180301 (experimental) +ALSA maintainers This was misattributed to kcov.c, I will fix it. > IMPORTANT: if you fix the bug, please add the following tag to the commit: > Reported-by: syzbot+74e011b2046f4322d178(a)syzkaller.appspotmail.com > It will help syzbot understand when the bug is fixed. See footer for > details. > If you forward the report, please keep this part and the footer. > > syz-executor1 (6140) used greatest stack depth: 11720 bytes left > INFO: rcu_sched self-detected stall on CPU > 1-....: (124998 ticks this GP) idle=0b2/1/4611686018427387906 > softirq=15730/15730 fqs=31205 > (t=125000 jiffies g=8316 c=8315 q=1529) > NMI backtrace for cpu 1 > CPU: 1 PID: 6087 Comm: syz-executor0 Not tainted 4.16.0+ #3 > Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS > Google 01/01/2011 > Call Trace: > <IRQ> > __dump_stack lib/dump_stack.c:77 [inline] > dump_stack+0x1b9/0x294 lib/dump_stack.c:113 > nmi_cpu_backtrace.cold.4+0x19/0xce lib/nmi_backtrace.c:103 > nmi_trigger_cpumask_backtrace+0x151/0x192 lib/nmi_backtrace.c:62 > arch_trigger_cpumask_backtrace+0x14/0x20 arch/x86/kernel/apic/hw_nmi.c:38 > trigger_single_cpu_backtrace include/linux/nmi.h:156 [inline] > rcu_dump_cpu_stacks+0x175/0x1c2 kernel/rcu/tree.c:1376 > print_cpu_stall kernel/rcu/tree.c:1525 [inline] > check_cpu_stall.isra.61.cold.80+0x36c/0x59a kernel/rcu/tree.c:1593 > __rcu_pending kernel/rcu/tree.c:3356 [inline] > rcu_pending kernel/rcu/tree.c:3401 [inline] > rcu_check_callbacks+0x21b/0xad0 kernel/rcu/tree.c:2763 > update_process_times+0x2d/0x70 kernel/time/timer.c:1636 > tick_sched_handle+0xa0/0x180 kernel/time/tick-sched.c:171 > tick_sched_timer+0x42/0x130 kernel/time/tick-sched.c:1179 > __run_hrtimer kernel/time/hrtimer.c:1337 [inline] > __hrtimer_run_queues+0x3e3/0x10a0 kernel/time/hrtimer.c:1399 > hrtimer_interrupt+0x286/0x650 kernel/time/hrtimer.c:1457 > local_apic_timer_interrupt arch/x86/kernel/apic/apic.c:1025 [inline] > smp_apic_timer_interrupt+0x15d/0x710 arch/x86/kernel/apic/apic.c:1050 > apic_timer_interrupt+0xf/0x20 arch/x86/entry/entry_64.S:862 > </IRQ> > RIP: 0010:__sanitizer_cov_trace_pc+0x3f/0x50 kernel/kcov.c:106 > RSP: 0018:ffff8801920cf4d0 EFLAGS: 00000216 ORIG_RAX: ffffffffffffff13 > RAX: 0000000000040000 RBX: ffff8801ceb61200 RCX: ffffc90001eb0000 > RDX: 0000000000040000 RSI: ffffffff859d760f RDI: ffff8801ceb61200 > RBP: ffff8801920cf4d0 R08: ffff8801b4946540 R09: 0000000000000006 > R10: ffff8801b4946540 R11: 0000000000000000 R12: ffffffff8826fa20 > R13: 0000000000000020 R14: ffff8801d3008c00 R15: ffff8801ceb61200 > snd_pcm_stream_lock_irq+0xf/0xf0 sound/core/pcm_native.c:150 > __snd_pcm_lib_xfer+0x345/0x1d10 sound/core/pcm_lib.c:2162 > snd_pcm_oss_write3+0xe9/0x220 sound/core/oss/pcm_oss.c:1236 > io_playback_transfer+0x274/0x310 sound/core/oss/io.c:47 > snd_pcm_plug_write_transfer+0x36c/0x470 sound/core/oss/pcm_plugin.c:619 > snd_pcm_oss_write2+0x25c/0x460 sound/core/oss/pcm_oss.c:1365 > snd_pcm_oss_write1 sound/core/oss/pcm_oss.c:1431 [inline] > snd_pcm_oss_write+0x55f/0xa20 sound/core/oss/pcm_oss.c:2774 > do_loop_readv_writev fs/read_write.c:703 [inline] > do_iter_write+0x491/0x5f0 fs/read_write.c:961 > vfs_writev+0x1c7/0x330 fs/read_write.c:1004 > do_writev+0x112/0x2f0 fs/read_write.c:1039 > SYSC_writev fs/read_write.c:1112 [inline] > SyS_writev+0x27/0x30 fs/read_write.c:1109 > do_syscall_64+0x29e/0x9d0 arch/x86/entry/common.c:287 > entry_SYSCALL_64_after_hwframe+0x42/0xb7 > RIP: 0033:0x4552d9 > RSP: 002b:00007f82ef52cc68 EFLAGS: 00000246 ORIG_RAX: 0000000000000014 > RAX: ffffffffffffffda RBX: 00007f82ef52d6d4 RCX: 00000000004552d9 > RDX: 0000000000000001 RSI: 0000000020000180 RDI: 0000000000000015 > RBP: 000000000072bea0 R08: 0000000000000000 R09: 0000000000000000 > R10: 0000000000000000 R11: 0000000000000246 R12: 00000000ffffffff > R13: 00000000000006cc R14: 00000000006fd3c0 R15: 0000000000000000 > > > --- > This bug is generated by a dumb bot. It may contain errors. > See https://goo.gl/tpsmEJ for details. > Direct all questions to syzkaller(a)googlegroups.com. > > syzbot will keep track of this bug report. > If you forgot to add the Reported-by tag, once the fix for this bug is > merged > into any tree, please reply to this email with: > #syz fix: exact-commit-title > To mark this as a duplicate of another syzbot report, please reply with: > #syz dup: exact-subject-of-another-report > If it's a one-off invalid bug report, please reply with: > #syz invalid > Note: if the crash happens again, it will cause creation of a new bug > report. > Note: all commands must start from beginning of the line in the email body.

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[alsa-devel] [PATCH 1/2] ALSA: pcm: Yet finer pending signal check in OSS read/write loop
by Takashi Iwai 07 Apr '18

07 Apr '18

PCM OSS emulation has the nested inner loops, and the loops in snd_pcm_oss_read3() and write3() have no pending signal check, so potentially they can hog for too long time for processing. Add the extra check there. While we're at it, a slight code cleanup using the standard macro for the PCM state check is used. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/oss/pcm_oss.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 481ab0e94ffa..5e009a444d9d 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -1221,8 +1221,7 @@ snd_pcm_sframes_t snd_pcm_oss_write3(struct snd_pcm_substream *substream, const struct snd_pcm_runtime *runtime = substream->runtime; int ret; while (1) { - if (runtime->status->state == SNDRV_PCM_STATE_XRUN || - runtime->status->state == SNDRV_PCM_STATE_SUSPENDED) { + if (!snd_pcm_running(substream)) { #ifdef OSS_DEBUG pcm_dbg(substream->pcm, "pcm_oss: write: recovering from %s\n", @@ -1241,6 +1240,8 @@ snd_pcm_sframes_t snd_pcm_oss_write3(struct snd_pcm_substream *substream, const /* has not been started */ if (runtime->status->state == SNDRV_PCM_STATE_PREPARED) return -EAGAIN; + if (signal_pending(current)) + return -ERESTARTSYS; } return ret; } @@ -1278,10 +1279,11 @@ snd_pcm_sframes_t snd_pcm_oss_read3(struct snd_pcm_substream *substream, char *p if (ret < 0) break; } - continue; - } - if (ret != -ESTRPIPE) + } else if (ret != -ESTRPIPE) { break; + } + if (signal_pending(current)) + return -ERESTARTSYS; } return ret; } -- 2.16.3

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[alsa-devel] [PATCH v3 0/3] ALSA: usb-audio: Minor refactoring and UAC3 fix
by Takashi Iwai 07 Apr '18

07 Apr '18

Hi, this is a v3 patchset, addressing the silly mistakes in validator functions in the previous patchset. Takashi === v1->v2: Rename the helper function to be more specific Make validators to check sizes more strictly v2->v3: Fix clock selector validator v2 and v3 as suggested by Ruslan Takashi Iwai (3): ALSA: usb-audio: Refactor clock finder helpers ALSA: usb-audio: More strict sanity checks for clock parsers ALSA: usb-audio: Add sanity checks in UAC3 clock parsers sound/usb/clock.c | 128 +++++++++++++++++++++++------------------------------- 1 file changed, 54 insertions(+), 74 deletions(-) -- 2.16.2

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[alsa-devel] [PATCH] ALSA: pcm: Fix endless loop for XRUN recovery in OSS emulation
by Takashi Iwai 07 Apr '18

07 Apr '18

The commit 02a5d6925cd3 ("ALSA: pcm: Avoid potential races between OSS ioctls and read/write") split the PCM preparation code to a locked version, and it added a sanity check of runtime->oss.prepare flag along with the change. This leaded to an endless loop when the stream gets XRUN: namely, snd_pcm_oss_write3() and co call snd_pcm_oss_prepare() without setting runtime->oss.prepare flag and the loop continues until the PCM state reaches to another one. As the function is supposed to execute the preparation unconditionally, drop the invalid state check there. The bug was triggered by syzkaller. Fixes: 02a5d6925cd3 ("ALSA: pcm: Avoid potential races between OSS ioctls and read/write") Reported-by: syzbot+7e3f31a52646f939c052(a)syzkaller.appspotmail.com Reported-by: syzbot+4f2016cf5185da7759dc(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/oss/pcm_oss.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 481ab0e94ffa..727647755aab 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -1128,13 +1128,12 @@ static int snd_pcm_oss_get_active_substream(struct snd_pcm_oss_file *pcm_oss_fil } /* call with params_lock held */ +/* NOTE: this doesn't care whether runtime->oss.prepare is set or not */ static int snd_pcm_oss_prepare(struct snd_pcm_substream *substream) { int err; struct snd_pcm_runtime *runtime = substream->runtime; - if (!runtime->oss.prepare) - return 0; err = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_PREPARE, NULL); if (err < 0) { pcm_dbg(substream->pcm, -- 2.16.3

1 0

[alsa-devel] Procedure for alsa modules compilation from git
by Pavel Hofman 07 Apr '18

07 Apr '18

Hi, May I ask for description/link describing current working compilation of alsa modules from git? I would like to compile only alsa modules, against my stock ubuntu kernel. Headers/sources in /usr/src, configured to current configuration from /boot/config... What is the oldest kernel to compile against (isch)? Would 4.4 work? I would like to make a small patch for ice1724, test, and send here for review/inclusion. I very much appreciate any help. I think it would be useful to many other occasional devels. Thanks a lot, Pavel.

2 2

[alsa-devel] [PATCH] usb: audio-v2: Correct the comment for struct uac_clock_selector_descriptor
by Takashi Iwai 06 Apr '18

06 Apr '18

The comment in UAC2 clock selector descriptor definition mentions the bAssocTerminal after baCSourceID[], but it doesn't exist in the actual definition. Let's correct it. Fixes: 5dd360ebd832 ("include/linux/usb/audio-v2.h: add more UAC2 details") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/linux/usb/audio-v2.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/linux/usb/audio-v2.h b/include/linux/usb/audio-v2.h index aaafecf073ff..49699255cfd3 100644 --- a/include/linux/usb/audio-v2.h +++ b/include/linux/usb/audio-v2.h @@ -94,7 +94,7 @@ struct uac_clock_selector_descriptor { __u8 bClockID; __u8 bNrInPins; __u8 baCSourceID[]; - /* bmControls, bAssocTerminal and iClockSource omitted */ + /* bmControls and iClockSource omitted */ } __attribute__((packed)); /* 4.7.2.3 Clock Multiplier Descriptor */ -- 2.16.3

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[alsa-devel] [PATCH v2] ASoC: wm8350: remove snd_soc_codec
by Kuninori Morimoto 06 Apr '18

06 Apr '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> codec is replace to component. It seems no-one is using it, Let's remove it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- v1 -> v2 - added Charles's Acked-by include/linux/mfd/wm8350/audio.h | 3 --- 1 file changed, 3 deletions(-) diff --git a/include/linux/mfd/wm8350/audio.h b/include/linux/mfd/wm8350/audio.h index bd581c6..0bc41c4c 100644 --- a/include/linux/mfd/wm8350/audio.h +++ b/include/linux/mfd/wm8350/audio.h @@ -617,11 +617,8 @@ struct wm8350_audio_platform_data { u32 codec_current_charge:2; /* codec current @ vmid charge */ }; -struct snd_soc_codec; - struct wm8350_codec { struct platform_device *pdev; - struct snd_soc_codec *codec; struct wm8350_audio_platform_data *platform_data; }; -- 1.9.1

1 0

[alsa-devel] KASAN: use-after-free Read in snd_pcm_timer_resolution
by syzbot 05 Apr '18

05 Apr '18

Hello, syzbot hit the following crash on upstream commit 0b412605ef5f5c64b31f19e2910b1d5eba9929c3 (Thu Mar 29 01:07:23 2018 +0000) Merge tag 'drm-fixes-for-v4.16-rc8' of git://people.freedesktop.org/~airlied/linux syzbot dashboard link: https://syzkaller.appspot.com/bug?extid=8e62ff4e07aa2ce87826 So far this crash happened 3 times on upstream. C reproducer: https://syzkaller.appspot.com/x/repro.c?id=6219387178582016 syzkaller reproducer: https://syzkaller.appspot.com/x/repro.syz?id=5176598244360192 Raw console output: https://syzkaller.appspot.com/x/log.txt?id=5815694612496384 Kernel config: https://syzkaller.appspot.com/x/.config?id=-8440362230543204781 compiler: gcc (GCC) 7.1.1 20170620 IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+8e62ff4e07aa2ce87826(a)syzkaller.appspotmail.com It will help syzbot understand when the bug is fixed. See footer for details. If you forward the report, please keep this part and the footer. ================================================================== BUG: KASAN: use-after-free in snd_pcm_timer_resolution+0xa3/0xc0 sound/core/pcm_timer.c:72 Read of size 4 at addr ffff8801bdd58490 by task syzkaller702213/13238 CPU: 0 PID: 13238 Comm: syzkaller702213 Not tainted 4.16.0-rc7+ #370 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:17 [inline] dump_stack+0x194/0x24d lib/dump_stack.c:53 print_address_description+0x73/0x250 mm/kasan/report.c:256 kasan_report_error mm/kasan/report.c:354 [inline] kasan_report+0x23c/0x360 mm/kasan/report.c:412 __asan_report_load4_noabort+0x14/0x20 mm/kasan/report.c:432 snd_pcm_timer_resolution+0xa3/0xc0 sound/core/pcm_timer.c:72 snd_timer_resolution sound/core/timer.c:439 [inline] snd_timer_notify1+0x10f/0x400 sound/core/timer.c:462 snd_timer_start1+0x586/0x8f0 sound/core/timer.c:522 snd_timer_start+0x5d/0xa0 sound/core/timer.c:634 snd_timer_user_start.isra.12+0xde/0x130 sound/core/timer.c:1878 snd_timer_user_continue sound/core/timer.c:1902 [inline] __snd_timer_user_ioctl+0x1771/0x2c10 sound/core/timer.c:1977 snd_timer_user_ioctl+0x5f/0x7d sound/core/timer.c:1992 vfs_ioctl fs/ioctl.c:46 [inline] do_vfs_ioctl+0x1b1/0x1520 fs/ioctl.c:686 SYSC_ioctl fs/ioctl.c:701 [inline] SyS_ioctl+0x8f/0xc0 fs/ioctl.c:692 do_syscall_64+0x281/0x940 arch/x86/entry/common.c:287 entry_SYSCALL_64_after_hwframe+0x42/0xb7 RIP: 0033:0x4453c9 RSP: 002b:00007ffd46abffe8 EFLAGS: 00000217 ORIG_RAX: 0000000000000010 RAX: ffffffffffffffda RBX: 0000000000000000 RCX: 00000000004453c9 RDX: 00000000004453c9 RSI: 00000000000054a2 RDI: 0000000000000003 RBP: 0000000000000000 R08: 0000000046ac0168 R09: 0000000046ac0168 R10: 0000000046ac0168 R11: 0000000000000217 R12: 00000000000142ea R13: 0000000000402c90 R14: 00000000006d1440 R15: 0000000000000000 Allocated by task 13234: save_stack+0x43/0xd0 mm/kasan/kasan.c:447 set_track mm/kasan/kasan.c:459 [inline] kasan_kmalloc+0xad/0xe0 mm/kasan/kasan.c:552 kmem_cache_alloc_trace+0x136/0x740 mm/slab.c:3608 kmalloc include/linux/slab.h:512 [inline] kzalloc include/linux/slab.h:701 [inline] snd_pcm_attach_substream+0x3c5/0xb60 sound/core/pcm.c:1006 snd_pcm_open_substream+0x95/0x3b0 sound/core/pcm_native.c:2372 snd_pcm_oss_open_file sound/core/oss/pcm_oss.c:2309 [inline] snd_pcm_oss_open+0x7cf/0x1540 sound/core/oss/pcm_oss.c:2391 soundcore_open+0x321/0x630 sound/sound_core.c:597 chrdev_open+0x257/0x730 fs/char_dev.c:417 do_dentry_open+0x667/0xd40 fs/open.c:752 vfs_open+0x107/0x220 fs/open.c:866 do_last fs/namei.c:3379 [inline] path_openat+0x1151/0x3530 fs/namei.c:3519 do_filp_open+0x25b/0x3b0 fs/namei.c:3554 do_sys_open+0x502/0x6d0 fs/open.c:1059 SYSC_openat fs/open.c:1086 [inline] SyS_openat+0x30/0x40 fs/open.c:1080 do_syscall_64+0x281/0x940 arch/x86/entry/common.c:287 entry_SYSCALL_64_after_hwframe+0x42/0xb7 Freed by task 13234: save_stack+0x43/0xd0 mm/kasan/kasan.c:447 set_track mm/kasan/kasan.c:459 [inline] __kasan_slab_free+0x11a/0x170 mm/kasan/kasan.c:520 kasan_slab_free+0xe/0x10 mm/kasan/kasan.c:527 __cache_free mm/slab.c:3486 [inline] kfree+0xd9/0x260 mm/slab.c:3801 snd_pcm_detach_substream+0x134/0x290 sound/core/pcm.c:1057 snd_pcm_release_substream.part.44+0x23d/0x300 sound/core/pcm_native.c:2361 snd_pcm_release_substream+0x59/0x70 sound/core/pcm_native.c:2362 snd_pcm_oss_release_file.part.18+0x71/0xa0 sound/core/oss/pcm_oss.c:2268 snd_pcm_oss_release_file sound/core/oss/pcm_oss.c:2263 [inline] snd_pcm_oss_release+0x104/0x280 sound/core/oss/pcm_oss.c:2448 __fput+0x327/0x7e0 fs/file_table.c:209 ____fput+0x15/0x20 fs/file_table.c:243 task_work_run+0x199/0x270 kernel/task_work.c:113 exit_task_work include/linux/task_work.h:22 [inline] do_exit+0x9bb/0x1ad0 kernel/exit.c:865 do_group_exit+0x149/0x400 kernel/exit.c:968 SYSC_exit_group kernel/exit.c:979 [inline] SyS_exit_group+0x1d/0x20 kernel/exit.c:977 do_syscall_64+0x281/0x940 arch/x86/entry/common.c:287 entry_SYSCALL_64_after_hwframe+0x42/0xb7 The buggy address belongs to the object at ffff8801bdd58180 which belongs to the cache kmalloc-2048 of size 2048 The buggy address is located 784 bytes inside of 2048-byte region [ffff8801bdd58180, ffff8801bdd58980) The buggy address belongs to the page: page:ffffea0006f75600 count:1 mapcount:0 mapping:ffff8801bdd58180 index:0x0 compound_mapcount: 0 flags: 0x2fffc0000008100(slab|head) raw: 02fffc0000008100 ffff8801bdd58180 0000000000000000 0000000100000003 raw: ffffea0007330d20 ffffea0007268aa0 ffff8801dac00c40 0000000000000000 page dumped because: kasan: bad access detected Memory state around the buggy address: ffff8801bdd58380: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ffff8801bdd58400: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb > ffff8801bdd58480: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ^ ffff8801bdd58500: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ffff8801bdd58580: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ================================================================== --- This bug is generated by a dumb bot. It may contain errors. See https://goo.gl/tpsmEJ for details. Direct all questions to syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. If you forgot to add the Reported-by tag, once the fix for this bug is merged into any tree, please reply to this email with: #syz fix: exact-commit-title If you want to test a patch for this bug, please reply with: #syz test: git://repo/address.git branch and provide the patch inline or as an attachment. To mark this as a duplicate of another syzbot report, please reply with: #syz dup: exact-subject-of-another-report If it's a one-off invalid bug report, please reply with: #syz invalid Note: if the crash happens again, it will cause creation of a new bug report. Note: all commands must start from beginning of the line in the email body.

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[alsa-devel] [PATCH] ALSA: usb: improve delay reporting
by Georg Chini 05 Apr '18

05 Apr '18

Currently, the kernel uses only the number of USB frames to interpolate the delay between sending or receiving URB's. This limits the granuarity of the delay reports to one millisecond because the frame counter is updated in a millisecond interval. This patch additionally uses time stamps to achieve higher precision of the reported delay. If the time difference computed using time stamps does not deviate more than one millisecond from the difference computed using frames, it is assumed that the system time difference delivers the more precise delay estimate. This will significantly increase the precision of the reported delay in the majority of cases where sound card clock and system clock are almost in sync while falling back to the old method for the rare cases where the two times differ too much. As a downside, the possible worst case error increases from 2ms to 3ms. --- sound/usb/card.h | 1 + sound/usb/pcm.c | 47 +++++++++++++++++++++++++++++++---------------- 2 files changed, 32 insertions(+), 16 deletions(-) diff --git a/sound/usb/card.h b/sound/usb/card.h index ed87cc83eb47..8835d9628cac 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -149,6 +149,7 @@ struct snd_usb_substream { int last_frame_number; /* stored frame number */ int last_delay; /* stored delay */ + struct timespec last_update_time; /* time when last_delay was updated */ struct { int marker; diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 3cbfae6604f9..6d40cb0b8777 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -46,9 +46,8 @@ snd_pcm_uframes_t snd_usb_pcm_delay(struct snd_usb_substream *subs, int current_frame_number; int frame_diff; int est_delay; - - if (!subs->last_delay) - return 0; /* short path */ + int time_diff; + struct timespec now; current_frame_number = usb_get_current_frame_number(subs->dev); /* @@ -58,9 +57,21 @@ snd_pcm_uframes_t snd_usb_pcm_delay(struct snd_usb_substream *subs, */ frame_diff = (current_frame_number - subs->last_frame_number) & 0xff; - /* Approximation based on number of samples per USB frame (ms), - some truncation for 44.1 but the estimate is good enough */ - est_delay = frame_diff * rate / 1000; + /* Estimate time since last update of last_delay */ + snd_pcm_gettime(subs->pcm_substream->runtime, &now); + time_diff = (now.tv_sec - subs->last_update_time.tv_sec) * 1000000000; + time_diff += (int)now.tv_nsec - subs->last_update_time.tv_nsec; + time_diff /= 1000; + + if (abs(time_diff - frame_diff * 1000) < 1000) { + /* System time and sound card time do not differ + * too much. Use system time to calculate delay */ + est_delay = time_diff * rate / 1000000; + } else { + /* Approximation based on number of samples per USB frame (ms), + some truncation for 44.1 but the estimate is good enough */ + est_delay = frame_diff * rate / 1000; + } if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) est_delay = subs->last_delay - est_delay; else @@ -853,6 +864,8 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) subs->hwptr_done = 0; subs->transfer_done = 0; subs->last_delay = 0; + subs->last_update_time.tv_sec = 0; + subs->last_update_time.tv_nsec = 0; subs->last_frame_number = 0; runtime->delay = 0; @@ -1334,7 +1347,9 @@ static void retire_capture_urb(struct snd_usb_substream *subs, /* realign last_frame_number */ subs->last_frame_number = current_frame_number; - subs->last_frame_number &= 0xFF; /* keep 8 LSBs */ + + /* Take time stamp for delay accounting */ + snd_pcm_gettime(runtime, &subs->last_update_time); spin_unlock_irqrestore(&subs->lock, flags); /* copy a data chunk */ @@ -1550,7 +1565,9 @@ static void prepare_playback_urb(struct snd_usb_substream *subs, /* realign last_frame_number */ subs->last_frame_number = usb_get_current_frame_number(subs->dev); - subs->last_frame_number &= 0xFF; /* keep 8 LSBs */ + + /* Take time stamp for delay accounting */ + snd_pcm_gettime(runtime, &subs->last_update_time); if (subs->trigger_tstamp_pending_update) { /* this is the first actual URB submitted, @@ -1597,6 +1614,12 @@ static void retire_playback_urb(struct snd_usb_substream *subs, subs->last_delay -= processed; runtime->delay = subs->last_delay; + /* realign last_frame_number */ + subs->last_frame_number = usb_get_current_frame_number(subs->dev); + + /* Take time stamp for delay accounting */ + snd_pcm_gettime(runtime, &subs->last_update_time); + /* * Report when delay estimate is off by more than 2ms. * The error should be lower than 2ms since the estimate relies @@ -1607,14 +1630,6 @@ static void retire_playback_urb(struct snd_usb_substream *subs, "delay: estimated %d, actual %d\n", est_delay, subs->last_delay); - if (!subs->running) { - /* update last_frame_number for delay counting here since - * prepare_playback_urb won't be called during pause - */ - subs->last_frame_number = - usb_get_current_frame_number(subs->dev) & 0xff; - } - out: spin_unlock_irqrestore(&subs->lock, flags); } -- 2.16.2

2 4

[alsa-devel] [PATCH v2 0/3] ALSA: usb-audio: Minor refactoring and UAC3 fix
by Takashi Iwai 05 Apr '18

05 Apr '18

Hi, this is a v2 patchset (now with a shiny cover letter!) for refactoring and hardening UAC2 and UAC3 clock parsers. BTW, it'll be great if anyone can double-check that this rewrite doesn't break things, especially with a UAC3 device. thanks, Takashi === v1->v2: Rename the helper function to be more specific Make validators to check sizes more strictly Takashi Iwai (3): ALSA: usb-audio: Refactor clock finder helpers ALSA: usb-audio: More strict sanity checks for clock parsers ALSA: usb-audio: Add sanity checks in UAC3 clock parsers sound/usb/clock.c | 128 +++++++++++++++++++++++------------------------------- 1 file changed, 54 insertions(+), 74 deletions(-) -- 2.16.2

2 6

[alsa-devel] SOF 1.1 Release
by Liam Girdwood 05 Apr '18

05 Apr '18

Sound Open Firmware 1.1 has now been released and available here :- git://git.alsa-project.org/sound-open-firmware.git git://git.alsa-project.org/sound-open-firmware-tools.git Significant changes since 1.0 :- o Support for Playback/Capture on Intel Apollolake and Cannonlake platforms. o Initial support for Intel Haswell and Broadwell platforms. o Support for DMA based firmware tracing to host userspace. o SIMD optimisations for SRC on xtensa ISA. o Initial support for running firmware components on host. Plans for 1.2 :- o Playback/Capture on Haswell/Broadwell. o SIMD optimisations for volume and mixer. o Improved host based component testing. o Initial runtime module support. SOF Linux driver dependencies for ALSA/ASoC have also started to upstream. Driver patches will continue to flow in anticipation for base SOF driver in 4.18. Thanks Liam

1 0

Re: [alsa-devel] [PATCH 00/15] ARM: pxa: switch to DMA slave maps
by Ulf Hansson 05 Apr '18

05 Apr '18

On 4 April 2018 at 21:56, Boris Brezillon <boris.brezillon(a)bootlin.com> wrote: > On Wed, 04 Apr 2018 21:49:26 +0200 > Robert Jarzmik <robert.jarzmik(a)free.fr> wrote: > >> Ulf Hansson <ulf.hansson(a)linaro.org> writes: >> >> > On 2 April 2018 at 16:26, Robert Jarzmik <robert.jarzmik(a)free.fr> wrote: >> >> Hi, >> >> >> >> This serie is aimed at removing the dmaengine slave compat use, and transfer >> >> knowledge of the DMA requestors into architecture code. >> >> As this looks like a patch bomb, each maintainer expressing for his tree either >> >> an Ack or "I want to take through my tree" will be spared in the next iterations >> >> of this serie. >> > >> > Perhaps an option is to send this hole series as PR for 3.17 rc1, that >> > would removed some churns and make this faster/easier? Well, if you >> > receive the needed acks of course. >> For 3.17-rc1 it looks a bit optimistic with the review time ... If I have all > > Especially since 3.17-rc1 has been released more than 3 years ago :-), > but I guess you meant 4.17-rc1. Yeah, I realize that I was a bit lost in time yesterday. Even more people have been having fun about it (me too). :-) Kind regards Uffe

2 1

[alsa-devel] [PATCH 00/13] soundwire: Add stream support
by Vinod Koul 05 Apr '18

05 Apr '18

This series adds support in SoundWire subsystem for: - Documentation for stream support - stream management - data port management - DAI ops in cadence and Intel drivers - ASoC API to propagate SDW stream Sanyog Kale (5): soundwire: Add more documentation soundwire: Add support for SoundWire stream management soundwire: Add Master and Slave port programming soundwire: Add helpers for ports operations soundwire: Add bank switch routine Shreyas NC (2): ASoC: Add SoundWire stream programming interface soundwire: Remove cdns_master_ops Vinod Koul (6): soundwire: Add support for port management soundwire: Add stream configuration APIs soundwire: cdns: Add port routines soundwire: cdns: Add stream routines soundwire: intel: Add stream initialization soundwire: intel: Add audio DAI ops .../driver-api/soundwire/error_handling.rst | 65 ++ Documentation/driver-api/soundwire/index.rst | 3 + Documentation/driver-api/soundwire/locking.rst | 106 +++ Documentation/driver-api/soundwire/stream.rst | 369 +++++++++ drivers/soundwire/Kconfig | 2 +- drivers/soundwire/Makefile | 3 +- drivers/soundwire/bus.c | 44 + drivers/soundwire/bus.h | 81 ++ drivers/soundwire/cadence_master.c | 449 +++++++++- drivers/soundwire/cadence_master.h | 151 ++++ drivers/soundwire/intel.c | 589 ++++++++++++- drivers/soundwire/intel.h | 4 + drivers/soundwire/intel_init.c | 3 + drivers/soundwire/runtime.c | 907 +++++++++++++++++++++ drivers/soundwire/stream.c | 678 +++++++++++++++ include/linux/soundwire/sdw.h | 337 +++++++- include/linux/soundwire/sdw_intel.h | 14 + include/sound/soc-dai.h | 5 + sound/soc/soc-core.c | 20 + 19 files changed, 3816 insertions(+), 14 deletions(-) create mode 100644 Documentation/driver-api/soundwire/error_handling.rst create mode 100644 Documentation/driver-api/soundwire/locking.rst create mode 100644 Documentation/driver-api/soundwire/stream.rst create mode 100644 drivers/soundwire/runtime.c create mode 100644 drivers/soundwire/stream.c -- 2.7.4

5 24

[alsa-devel] [RESEND PATCH 1/3 v3] ASoC: TSCS42xx: Shorten lines and other cleanup
by Steven Eckhoff 05 Apr '18

05 Apr '18

Shorten lines greater than 80 chars Add const to struct snd_soc_component_driver Signed-off-by: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> --- sound/soc/codecs/tscs42xx.c | 87 ++++++++++++++++++++++++++++----------------- 1 file changed, 55 insertions(+), 32 deletions(-) diff --git a/sound/soc/codecs/tscs42xx.c b/sound/soc/codecs/tscs42xx.c index bbfc73a79b18..5ad68e5538ae 100644 --- a/sound/soc/codecs/tscs42xx.c +++ b/sound/soc/codecs/tscs42xx.c @@ -204,7 +204,8 @@ static int power_up_audio_plls(struct snd_soc_component *component) break; default: ret = -EINVAL; - dev_err(component->dev, "Unrecognized PLL output freq (%d)\n", ret); + dev_err(component->dev, + "Unrecognized PLL output freq (%d)\n", ret); return ret; } @@ -261,7 +262,8 @@ static int power_down_audio_plls(struct snd_soc_component *component) static int coeff_ram_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); struct tscs42xx *tscs42xx = snd_soc_component_get_drvdata(component); struct coeff_ram_ctl *ctl = (struct coeff_ram_ctl *)kcontrol->private_value; @@ -280,7 +282,8 @@ static int coeff_ram_get(struct snd_kcontrol *kcontrol, static int coeff_ram_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); struct tscs42xx *tscs42xx = snd_soc_component_get_drvdata(component); struct coeff_ram_ctl *ctl = (struct coeff_ram_ctl *)kcontrol->private_value; @@ -363,7 +366,8 @@ static int dapm_micb_event(struct snd_soc_dapm_widget *w, static int pll_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); int ret; if (SND_SOC_DAPM_EVENT_ON(event)) @@ -377,7 +381,8 @@ static int pll_event(struct snd_soc_dapm_widget *w, static int dac_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); struct tscs42xx *tscs42xx = snd_soc_component_get_drvdata(component); int ret; @@ -819,16 +824,19 @@ static int setup_sample_format(struct snd_soc_component *component, dev_err(component->dev, "Unsupported format width (%d)\n", ret); return ret; } - ret = snd_soc_component_update_bits(component, R_AIC1, RM_AIC1_WL, width); + ret = snd_soc_component_update_bits(component, + R_AIC1, RM_AIC1_WL, width); if (ret < 0) { - dev_err(component->dev, "Failed to set sample width (%d)\n", ret); + dev_err(component->dev, + "Failed to set sample width (%d)\n", ret); return ret; } return 0; } -static int setup_sample_rate(struct snd_soc_component *component, unsigned int rate) +static int setup_sample_rate(struct snd_soc_component *component, + unsigned int rate) { struct tscs42xx *tscs42xx = snd_soc_component_get_drvdata(component); unsigned int br, bm; @@ -881,24 +889,32 @@ static int setup_sample_rate(struct snd_soc_component *component, unsigned int r } /* DAC and ADC share bit and frame clock */ - ret = snd_soc_component_update_bits(component, R_DACSR, RM_DACSR_DBR, br); + ret = snd_soc_component_update_bits(component, + R_DACSR, RM_DACSR_DBR, br); if (ret < 0) { - dev_err(component->dev, "Failed to update register (%d)\n", ret); + dev_err(component->dev, + "Failed to update register (%d)\n", ret); return ret; } - ret = snd_soc_component_update_bits(component, R_DACSR, RM_DACSR_DBM, bm); + ret = snd_soc_component_update_bits(component, + R_DACSR, RM_DACSR_DBM, bm); if (ret < 0) { - dev_err(component->dev, "Failed to update register (%d)\n", ret); + dev_err(component->dev, + "Failed to update register (%d)\n", ret); return ret; } - ret = snd_soc_component_update_bits(component, R_ADCSR, RM_DACSR_DBR, br); + ret = snd_soc_component_update_bits(component, + R_ADCSR, RM_DACSR_DBR, br); if (ret < 0) { - dev_err(component->dev, "Failed to update register (%d)\n", ret); + dev_err(component->dev, + "Failed to update register (%d)\n", ret); return ret; } - ret = snd_soc_component_update_bits(component, R_ADCSR, RM_DACSR_DBM, bm); + ret = snd_soc_component_update_bits(component, + R_ADCSR, RM_DACSR_DBM, bm); if (ret < 0) { - dev_err(component->dev, "Failed to update register (%d)\n", ret); + dev_err(component->dev, + "Failed to update register (%d)\n", ret); return ret; } @@ -1076,7 +1092,8 @@ static int tscs42xx_hw_params(struct snd_pcm_substream *substream, ret = setup_sample_rate(component, params_rate(params)); if (ret < 0) { - dev_err(component->dev, "Failed to setup sample rate (%d)\n", ret); + dev_err(component->dev, + "Failed to setup sample rate (%d)\n", ret); return ret; } @@ -1087,7 +1104,8 @@ static inline int dac_mute(struct snd_soc_component *component) { int ret; - ret = snd_soc_component_update_bits(component, R_CNVRTR1, RM_CNVRTR1_DACMU, + ret = snd_soc_component_update_bits(component, + R_CNVRTR1, RM_CNVRTR1_DACMU, RV_CNVRTR1_DACMU_ENABLE); if (ret < 0) { dev_err(component->dev, "Failed to mute DAC (%d)\n", @@ -1102,7 +1120,8 @@ static inline int dac_unmute(struct snd_soc_component *component) { int ret; - ret = snd_soc_component_update_bits(component, R_CNVRTR1, RM_CNVRTR1_DACMU, + ret = snd_soc_component_update_bits(component, + R_CNVRTR1, RM_CNVRTR1_DACMU, RV_CNVRTR1_DACMU_DISABLE); if (ret < 0) { dev_err(component->dev, "Failed to unmute DAC (%d)\n", @@ -1117,8 +1136,8 @@ static inline int adc_mute(struct snd_soc_component *component) { int ret; - ret = snd_soc_component_update_bits(component, R_CNVRTR0, RM_CNVRTR0_ADCMU, - RV_CNVRTR0_ADCMU_ENABLE); + ret = snd_soc_component_update_bits(component, + R_CNVRTR0, RM_CNVRTR0_ADCMU, RV_CNVRTR0_ADCMU_ENABLE); if (ret < 0) { dev_err(component->dev, "Failed to mute ADC (%d)\n", ret); @@ -1132,8 +1151,8 @@ static inline int adc_unmute(struct snd_soc_component *component) { int ret; - ret = snd_soc_component_update_bits(component, R_CNVRTR0, RM_CNVRTR0_ADCMU, - RV_CNVRTR0_ADCMU_DISABLE); + ret = snd_soc_component_update_bits(component, + R_CNVRTR0, RM_CNVRTR0_ADCMU, RV_CNVRTR0_ADCMU_DISABLE); if (ret < 0) { dev_err(component->dev, "Failed to unmute ADC (%d)\n", ret); @@ -1171,8 +1190,8 @@ static int tscs42xx_set_dai_fmt(struct snd_soc_dai *codec_dai, /* Slave mode not supported since it needs always-on frame clock */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: - ret = snd_soc_component_update_bits(component, R_AIC1, RM_AIC1_MS, - RV_AIC1_MS_MASTER); + ret = snd_soc_component_update_bits(component, + R_AIC1, RM_AIC1_MS, RV_AIC1_MS_MASTER); if (ret < 0) { dev_err(component->dev, "Failed to set codec DAI master (%d)\n", ret); @@ -1211,14 +1230,18 @@ static int tscs42xx_set_dai_bclk_ratio(struct snd_soc_dai *codec_dai, return -EINVAL; } - ret = snd_soc_component_update_bits(component, R_DACSR, RM_DACSR_DBCM, value); + ret = snd_soc_component_update_bits(component, + R_DACSR, RM_DACSR_DBCM, value); if (ret < 0) { - dev_err(component->dev, "Failed to set DAC BCLK ratio (%d)\n", ret); + dev_err(component->dev, + "Failed to set DAC BCLK ratio (%d)\n", ret); return ret; } - ret = snd_soc_component_update_bits(component, R_ADCSR, RM_ADCSR_ABCM, value); + ret = snd_soc_component_update_bits(component, + R_ADCSR, RM_ADCSR_ABCM, value); if (ret < 0) { - dev_err(component->dev, "Failed to set ADC BCLK ratio (%d)\n", ret); + dev_err(component->dev, + "Failed to set ADC BCLK ratio (%d)\n", ret); return ret; } @@ -1309,7 +1332,7 @@ static int part_is_valid(struct tscs42xx *tscs42xx) }; } -static struct snd_soc_component_driver soc_codec_dev_tscs42xx = { +static const struct snd_soc_component_driver soc_codec_dev_tscs42xx = { .dapm_widgets = tscs42xx_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(tscs42xx_dapm_widgets), .dapm_routes = tscs42xx_intercon, @@ -1416,8 +1439,8 @@ static int tscs42xx_i2c_probe(struct i2c_client *i2c, mutex_init(&tscs42xx->coeff_ram_lock); mutex_init(&tscs42xx->pll_lock); - ret = devm_snd_soc_register_component(tscs42xx->dev, &soc_codec_dev_tscs42xx, - &tscs42xx_dai, 1); + ret = devm_snd_soc_register_component(tscs42xx->dev, + &soc_codec_dev_tscs42xx, &tscs42xx_dai, 1); if (ret) { dev_err(tscs42xx->dev, "Failed to register codec (%d)\n", ret); return ret; -- 2.15.1

1 2

[alsa-devel] CX20724 headset button detect
by David Woodhouse 04 Apr '18

04 Apr '18

I have an HP EliteBook 850 G3 with CX20724: hdaudioC0D0: CX20724: BIOS auto-probing. hdaudioC0D0: action: 0 gpio_led: 0 hdaudioC0D0: autoconfig for CX20724: line_outs=1 (0x16/0x0/0x0/0x0/0x0) type:line hdaudioC0D0: speaker_outs=1 (0x17/0x0/0x0/0x0/0x0) hdaudioC0D0: hp_outs=1 (0x1d/0x0/0x0/0x0/0x0) hdaudioC0D0: mono: mono_out=0x0 hdaudioC0D0: inputs: hdaudioC0D0: Mic=0x19 hdaudioC0D0: Internal Mic=0x1a hdaudioC0D0: Line=0x18 hdaudioC0D0: Enable sync_write for stable communication hdaudioC0D0: action: 1 gpio_led: 0 hdaudioC0D0: action: 2 gpio_led: 0 ... input: HDA Intel PCH Mic as /devices/pci0000:00/0000:00:1f.3/sound/card0/input21 The CX20724 datasheet¹ suggests that it supports headset button events — at least 'Function A' when the mic pin is shorted to ground, if not the other functions. But it isn't working. Only the jack insertion is reported through the corresponding input device, not the button press. I added a printk into snd_hda_jack_unsol_event(), and it isn't even being invoked at all for the button presses; only insertion/removal. How do we make this work? ¹ https://www.tonymacx86.com/attachments/datasheet_cx20724-pdf.203615/

2 4

[alsa-devel] No audio on headphones on HP X2 210 (Baytrail Atom - RT5640)
by Freaky 04 Apr '18

04 Apr '18

Hi there, we're running Fedora 27 on a HP X2 210. Grabbed the UCM config files for it from the GIT repo of plbossart ( https://github.com/plbossart/UCM/tree/master/bytcr-rt5640) blacklisted the snd_hdmi_lpe_audio and sound now works via the onboard speakers. Current kernel is 4.15.13. Audio via the headphones does not work. Been looking at the source code, but most of the abbreviations are mumbo jumbo to me :). Doesn't look like there are any for the headphones in 4.15.13. Did find this patch: https://webcache.googleusercontent.com/search?q=cache:FlvUx7XgbH4J:https://… Which seems to enable a quirk for headphone detection. Doesn't seem to be in 4.15.14 source. Do we require that patch? Not sure what this line does: #define BYT_RT5640_JACK_PIN(quirk) ((quirk) & GENMASK(15, 13)) Would I need to flip bits 15 and 13? What would the parameter be? The file is called bytcr_rt5640.c, but for the 5645 (Cherrytrail atoms) the kernel parameter is called snd_soc_rt5645.quirk, is it called snd_soc_rt5640.quirk here? Doubt it since there are quirks defined in sound/soc/codecs/rt5645.c whereas I don't see any (4.15.14 sources) in sound/soc/codecs/rt5640.c. Any advise would be most welcome :). Thanks in advance. Kind regards,

2 1

[alsa-devel] [PATCH 1/2] ALSA: usb-audio: Refactor clock finder helpers
by Takashi Iwai 04 Apr '18

04 Apr '18

There are lots of open-coded functions to find a clock source, selector and multiplier. Now there are both v2 and v3, so six variants. This patch refactors the code to use a common helper for the main loop, and define each validator function for each target. There is no functional change. Fixes: 9a2fe9b801f5 ("ALSA: usb: initial USB Audio Device Class 3.0 support") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/clock.c | 127 +++++++++++++++++++++++------------------------------- 1 file changed, 53 insertions(+), 74 deletions(-) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index ab39ccb974c6..c5f0cf532c0c 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -35,105 +35,84 @@ #include "clock.h" #include "quirks.h" -static struct uac_clock_source_descriptor * - snd_usb_find_clock_source(struct usb_host_interface *ctrl_iface, - int clock_id) +static void *find_descriptor(struct usb_host_interface *iface, int id, + bool (*validator)(void *, int), u8 type) { - struct uac_clock_source_descriptor *cs = NULL; + void *cs = NULL; - while ((cs = snd_usb_find_csint_desc(ctrl_iface->extra, - ctrl_iface->extralen, - cs, UAC2_CLOCK_SOURCE))) { - if (cs->bLength >= sizeof(*cs) && cs->bClockID == clock_id) + while ((cs = snd_usb_find_csint_desc(iface->extra, iface->extralen, + cs, type))) { + if (validator(cs, id)) return cs; } return NULL; } -static struct uac3_clock_source_descriptor * - snd_usb_find_clock_source_v3(struct usb_host_interface *ctrl_iface, - int clock_id) +static bool validate_clock_source_v2(void *p, int id) { - struct uac3_clock_source_descriptor *cs = NULL; - - while ((cs = snd_usb_find_csint_desc(ctrl_iface->extra, - ctrl_iface->extralen, - cs, UAC3_CLOCK_SOURCE))) { - if (cs->bClockID == clock_id) - return cs; - } - - return NULL; + struct uac_clock_source_descriptor *cs = p; + return cs->bLength >= sizeof(*cs) && cs->bClockID == id; } -static struct uac_clock_selector_descriptor * - snd_usb_find_clock_selector(struct usb_host_interface *ctrl_iface, - int clock_id) +static bool validate_clock_source_v3(void *p, int id) { - struct uac_clock_selector_descriptor *cs = NULL; - - while ((cs = snd_usb_find_csint_desc(ctrl_iface->extra, - ctrl_iface->extralen, - cs, UAC2_CLOCK_SELECTOR))) { - if (cs->bLength >= sizeof(*cs) && cs->bClockID == clock_id) { - if (cs->bLength < 5 + cs->bNrInPins) - return NULL; - return cs; - } - } - - return NULL; + struct uac3_clock_source_descriptor *cs = p; + return cs->bClockID == id; } -static struct uac3_clock_selector_descriptor * - snd_usb_find_clock_selector_v3(struct usb_host_interface *ctrl_iface, - int clock_id) +static bool validate_clock_selector_v2(void *p, int id) { - struct uac3_clock_selector_descriptor *cs = NULL; - - while ((cs = snd_usb_find_csint_desc(ctrl_iface->extra, - ctrl_iface->extralen, - cs, UAC3_CLOCK_SELECTOR))) { - if (cs->bClockID == clock_id) - return cs; - } - - return NULL; + struct uac_clock_selector_descriptor *cs = p; + return cs->bLength >= sizeof(*cs) && cs->bClockID == id && + cs->bLength >= 5 + cs->bNrInPins; } -static struct uac_clock_multiplier_descriptor * - snd_usb_find_clock_multiplier(struct usb_host_interface *ctrl_iface, - int clock_id) +static bool validate_clock_selector_v3(void *p, int id) { - struct uac_clock_multiplier_descriptor *cs = NULL; - - while ((cs = snd_usb_find_csint_desc(ctrl_iface->extra, - ctrl_iface->extralen, - cs, UAC2_CLOCK_MULTIPLIER))) { - if (cs->bLength >= sizeof(*cs) && cs->bClockID == clock_id) - return cs; - } - - return NULL; + struct uac3_clock_selector_descriptor *cs = p; + return cs->bClockID == id; } -static struct uac3_clock_multiplier_descriptor * - snd_usb_find_clock_multiplier_v3(struct usb_host_interface *ctrl_iface, - int clock_id) +static bool validate_clock_multiplier_v2(void *p, int id) { - struct uac3_clock_multiplier_descriptor *cs = NULL; + struct uac_clock_multiplier_descriptor *cs = p; + return cs->bLength >= sizeof(*cs) && cs->bClockID == id; +} - while ((cs = snd_usb_find_csint_desc(ctrl_iface->extra, - ctrl_iface->extralen, - cs, UAC3_CLOCK_MULTIPLIER))) { - if (cs->bClockID == clock_id) - return cs; - } +static bool validate_clock_multiplier_v3(void *p, int id) +{ + struct uac3_clock_multiplier_descriptor *cs = p; + return cs->bClockID == id; +} - return NULL; +#define DEFINE_FIND_HELPER(name, obj, validator, type) \ +static obj *name(struct usb_host_interface *iface, int id) \ +{ \ + return find_descriptor(iface, id, validator, type); \ } +DEFINE_FIND_HELPER(snd_usb_find_clock_source, + struct uac_clock_source_descriptor, + validate_clock_source_v2, UAC2_CLOCK_SOURCE); +DEFINE_FIND_HELPER(snd_usb_find_clock_source_v3, + struct uac3_clock_source_descriptor, + validate_clock_source_v3, UAC3_CLOCK_SOURCE); + +DEFINE_FIND_HELPER(snd_usb_find_clock_selector, + struct uac_clock_selector_descriptor, + validate_clock_selector_v2, UAC2_CLOCK_SELECTOR); +DEFINE_FIND_HELPER(snd_usb_find_clock_selector_v3, + struct uac3_clock_selector_descriptor, + validate_clock_selector_v3, UAC3_CLOCK_SELECTOR); + +DEFINE_FIND_HELPER(snd_usb_find_clock_multiplier, + struct uac_clock_multiplier_descriptor, + validate_clock_multiplier_v2, UAC2_CLOCK_MULTIPLIER); +DEFINE_FIND_HELPER(snd_usb_find_clock_multiplier_v3, + struct uac3_clock_multiplier_descriptor, + validate_clock_multiplier_v3, UAC3_CLOCK_MULTIPLIER); + static int uac_clock_selector_get_val(struct snd_usb_audio *chip, int selector_id) { unsigned char buf; -- 2.16.2

3 5

Re: [alsa-devel] [PATCH 1/2] ALSA: usb-audio: Refactor clock finder helpers
by Takashi Iwai 04 Apr '18

04 Apr '18

On Tue, 03 Apr 2018 19:13:51 +0200, Andrew Chant wrote: > > On Tue, Apr 3, 2018 at 8:48 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > > > > There are lots of open-coded functions to find a clock source, > > selector and multiplier. Now there are both v2 and v3, so six > > variants. > > > > This patch refactors the code to use a common helper for the main > > loop, and define each validator function for each target. > > There is no functional change. > > > > Fixes: 9a2fe9b801f5 ("ALSA: usb: initial USB Audio Device Class 3.0 support") > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > > --- > > sound/usb/clock.c | 127 +++++++++++++++++++++++------------------------------- > > 1 file changed, 53 insertions(+), 74 deletions(-) > > > > diff --git a/sound/usb/clock.c b/sound/usb/clock.c > > index ab39ccb974c6..c5f0cf532c0c 100644 > > --- a/sound/usb/clock.c > > +++ b/sound/usb/clock.c > > @@ -35,105 +35,84 @@ > > #include "clock.h" > > #include "quirks.h" > > > > -static struct uac_clock_source_descriptor * > > - snd_usb_find_clock_source(struct usb_host_interface *ctrl_iface, > > - int clock_id) > > +static void *find_descriptor(struct usb_host_interface *iface, int id, > > + bool (*validator)(void *, int), u8 type) > > this comment isn't very important, and might be wrong: > find_descriptor is a very generic name, it might make grepping a > little confusing. > Could you use find_clock_descriptor? find_uac_clock_descriptor? Makes sense. Now I renamed it with a bit shorter one, find_uac_clock_desc(). The revised patch is below. thanks, Takashi -- 8< -- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH v2 1/2] ALSA: usb-audio: Refactor clock finder helpers There are lots of open-coded functions to find a clock source, selector and multiplier. Now there are both v2 and v3, so six variants. This patch refactors the code to use a common helper for the main loop, and define each validator function for each target. There is no functional change. Fixes: 9a2fe9b801f5 ("ALSA: usb: initial USB Audio Device Class 3.0 support") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- v1->v2: rename find_descriptor() with find_uac_clock_desc() sound/usb/clock.c | 127 +++++++++++++++++++++++------------------------------- 1 file changed, 53 insertions(+), 74 deletions(-) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index ab39ccb974c6..27c2275a2505 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -35,105 +35,84 @@ #include "clock.h" #include "quirks.h" -static struct uac_clock_source_descriptor * - snd_usb_find_clock_source(struct usb_host_interface *ctrl_iface, - int clock_id) +static void *find_uac_clock_desc(struct usb_host_interface *iface, int id, + bool (*validator)(void *, int), u8 type) { - struct uac_clock_source_descriptor *cs = NULL; + void *cs = NULL; - while ((cs = snd_usb_find_csint_desc(ctrl_iface->extra, - ctrl_iface->extralen, - cs, UAC2_CLOCK_SOURCE))) { - if (cs->bLength >= sizeof(*cs) && cs->bClockID == clock_id) + while ((cs = snd_usb_find_csint_desc(iface->extra, iface->extralen, + cs, type))) { + if (validator(cs, id)) return cs; } return NULL; } -static struct uac3_clock_source_descriptor * - snd_usb_find_clock_source_v3(struct usb_host_interface *ctrl_iface, - int clock_id) +static bool validate_clock_source_v2(void *p, int id) { - struct uac3_clock_source_descriptor *cs = NULL; - - while ((cs = snd_usb_find_csint_desc(ctrl_iface->extra, - ctrl_iface->extralen, - cs, UAC3_CLOCK_SOURCE))) { - if (cs->bClockID == clock_id) - return cs; - } - - return NULL; + struct uac_clock_source_descriptor *cs = p; + return cs->bLength >= sizeof(*cs) && cs->bClockID == id; } -static struct uac_clock_selector_descriptor * - snd_usb_find_clock_selector(struct usb_host_interface *ctrl_iface, - int clock_id) +static bool validate_clock_source_v3(void *p, int id) { - struct uac_clock_selector_descriptor *cs = NULL; - - while ((cs = snd_usb_find_csint_desc(ctrl_iface->extra, - ctrl_iface->extralen, - cs, UAC2_CLOCK_SELECTOR))) { - if (cs->bLength >= sizeof(*cs) && cs->bClockID == clock_id) { - if (cs->bLength < 5 + cs->bNrInPins) - return NULL; - return cs; - } - } - - return NULL; + struct uac3_clock_source_descriptor *cs = p; + return cs->bClockID == id; } -static struct uac3_clock_selector_descriptor * - snd_usb_find_clock_selector_v3(struct usb_host_interface *ctrl_iface, - int clock_id) +static bool validate_clock_selector_v2(void *p, int id) { - struct uac3_clock_selector_descriptor *cs = NULL; - - while ((cs = snd_usb_find_csint_desc(ctrl_iface->extra, - ctrl_iface->extralen, - cs, UAC3_CLOCK_SELECTOR))) { - if (cs->bClockID == clock_id) - return cs; - } - - return NULL; + struct uac_clock_selector_descriptor *cs = p; + return cs->bLength >= sizeof(*cs) && cs->bClockID == id && + cs->bLength >= 5 + cs->bNrInPins; } -static struct uac_clock_multiplier_descriptor * - snd_usb_find_clock_multiplier(struct usb_host_interface *ctrl_iface, - int clock_id) +static bool validate_clock_selector_v3(void *p, int id) { - struct uac_clock_multiplier_descriptor *cs = NULL; - - while ((cs = snd_usb_find_csint_desc(ctrl_iface->extra, - ctrl_iface->extralen, - cs, UAC2_CLOCK_MULTIPLIER))) { - if (cs->bLength >= sizeof(*cs) && cs->bClockID == clock_id) - return cs; - } - - return NULL; + struct uac3_clock_selector_descriptor *cs = p; + return cs->bClockID == id; } -static struct uac3_clock_multiplier_descriptor * - snd_usb_find_clock_multiplier_v3(struct usb_host_interface *ctrl_iface, - int clock_id) +static bool validate_clock_multiplier_v2(void *p, int id) { - struct uac3_clock_multiplier_descriptor *cs = NULL; + struct uac_clock_multiplier_descriptor *cs = p; + return cs->bLength >= sizeof(*cs) && cs->bClockID == id; +} - while ((cs = snd_usb_find_csint_desc(ctrl_iface->extra, - ctrl_iface->extralen, - cs, UAC3_CLOCK_MULTIPLIER))) { - if (cs->bClockID == clock_id) - return cs; - } +static bool validate_clock_multiplier_v3(void *p, int id) +{ + struct uac3_clock_multiplier_descriptor *cs = p; + return cs->bClockID == id; +} - return NULL; +#define DEFINE_FIND_HELPER(name, obj, validator, type) \ +static obj *name(struct usb_host_interface *iface, int id) \ +{ \ + return find_uac_clock_desc(iface, id, validator, type); \ } +DEFINE_FIND_HELPER(snd_usb_find_clock_source, + struct uac_clock_source_descriptor, + validate_clock_source_v2, UAC2_CLOCK_SOURCE); +DEFINE_FIND_HELPER(snd_usb_find_clock_source_v3, + struct uac3_clock_source_descriptor, + validate_clock_source_v3, UAC3_CLOCK_SOURCE); + +DEFINE_FIND_HELPER(snd_usb_find_clock_selector, + struct uac_clock_selector_descriptor, + validate_clock_selector_v2, UAC2_CLOCK_SELECTOR); +DEFINE_FIND_HELPER(snd_usb_find_clock_selector_v3, + struct uac3_clock_selector_descriptor, + validate_clock_selector_v3, UAC3_CLOCK_SELECTOR); + +DEFINE_FIND_HELPER(snd_usb_find_clock_multiplier, + struct uac_clock_multiplier_descriptor, + validate_clock_multiplier_v2, UAC2_CLOCK_MULTIPLIER); +DEFINE_FIND_HELPER(snd_usb_find_clock_multiplier_v3, + struct uac3_clock_multiplier_descriptor, + validate_clock_multiplier_v3, UAC3_CLOCK_MULTIPLIER); + static int uac_clock_selector_get_val(struct snd_usb_audio *chip, int selector_id) { unsigned char buf; -- 2.16.2

2 1

[alsa-devel] [PATCHv5 0/5] Motorola Droid 4 Audio Support
by Sebastian Reichel 04 Apr '18

04 Apr '18

Hi, This adds audio support to Motorola Droid 4. I dropped the regulator from the DT binding as requested by Mark. Tested: - Playing via both DACs using Speaker, Earpiece, Headphone - Recording using internal Mic - Volume Controls Known not to work: - 3.5mm detection support (requires closed source firmware, needs further investigation) - Modem / Bluetooth Audio (this needs something more sophisticated than the audio-graph-card driver. I will have a look once the basic support has been merged) Changes since PATCHv4: * Replace verbose GPL text with SPDX header * Use "GPL v2" as MODULE_LICENSE * Drop "default MFD_CPCAP" for the codec driver * Use SOC_SINGLE instead of SOC_ENUM for Phase invert switches * Find DT node by node name instead of by compatible property - Drop Acked-by from Rob Herring * Add new patch for MFD driver to implement the mentioned DT change Changes since PATCHv3: * Drop regulator from DT binding, request VAUDIO from global regulator namespace instead Changes since PATCHv2: * Fix a whitespace issue * Fix const notes Takashi provided * Fix a DAPM route issue I accidently introduced in v2 Changes since PATCHv1: * Add patch from Tony exporting soc_dpcm_runtime_update * Integrate DT patch for vaudio initial mode * Split dt-binding from codec patch and add Ack from Rob * Fix CPCAP position in Kconfig/Makefile * Avoid "err +=" constructs * Simplify reset function * Drop cpcap_audio_read/write helpers * Do not use tertiary operator for mute register value * Update Input Mux logic * Switch from simple-audio-card to audio-graph-card -- Sebastian Sebastian Reichel (5): dt-bindings: mfd: motorola-cpcap: document audio-codec ASoC: codec: cpcap: new codec mfd: motorola-cpcap: Add audio-codec support ARM: dts: motorola-cpcap-mapphone: add audio-codec ARM: dts: omap4-droid4: add soundcard .../devicetree/bindings/mfd/motorola-cpcap.txt | 42 + arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi | 13 + arch/arm/boot/dts/omap4-droid4-xt894.dts | 78 + drivers/mfd/motorola-cpcap.c | 51 +- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cpcap.c | 1568 ++++++++++++++++++++ 7 files changed, 1757 insertions(+), 1 deletion(-) create mode 100644 sound/soc/codecs/cpcap.c -- 2.16.1

9 90

[alsa-devel] [PATCH 1/3 v3] ASoC: TSCS42xx: Shorten lines and other cleanup
by Steven Eckhoff 03 Apr '18

03 Apr '18

Shorten lines greater than 80 chars Add const to struct snd_soc_component_driver Signed-off-by: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> --- sound/soc/codecs/tscs42xx.c | 87 ++++++++++++++++++++++++++++----------------- 1 file changed, 55 insertions(+), 32 deletions(-) diff --git a/sound/soc/codecs/tscs42xx.c b/sound/soc/codecs/tscs42xx.c index cfc71b5411ee..37636d6efe60 100644 --- a/sound/soc/codecs/tscs42xx.c +++ b/sound/soc/codecs/tscs42xx.c @@ -204,7 +204,8 @@ static int power_up_audio_plls(struct snd_soc_component *component) break; default: ret = -EINVAL; - dev_err(component->dev, "Unrecognized PLL output freq (%d)\n", ret); + dev_err(component->dev, + "Unrecognized PLL output freq (%d)\n", ret); return ret; } @@ -261,7 +262,8 @@ static int power_down_audio_plls(struct snd_soc_component *component) static int coeff_ram_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); struct tscs42xx *tscs42xx = snd_soc_component_get_drvdata(component); struct coeff_ram_ctl *ctl = (struct coeff_ram_ctl *)kcontrol->private_value; @@ -280,7 +282,8 @@ static int coeff_ram_get(struct snd_kcontrol *kcontrol, static int coeff_ram_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); struct tscs42xx *tscs42xx = snd_soc_component_get_drvdata(component); struct coeff_ram_ctl *ctl = (struct coeff_ram_ctl *)kcontrol->private_value; @@ -363,7 +366,8 @@ static int dapm_micb_event(struct snd_soc_dapm_widget *w, static int pll_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); int ret; if (SND_SOC_DAPM_EVENT_ON(event)) @@ -377,7 +381,8 @@ static int pll_event(struct snd_soc_dapm_widget *w, static int dac_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); struct tscs42xx *tscs42xx = snd_soc_component_get_drvdata(component); int ret; @@ -819,16 +824,19 @@ static int setup_sample_format(struct snd_soc_component *component, dev_err(component->dev, "Unsupported format width (%d)\n", ret); return ret; } - ret = snd_soc_component_update_bits(component, R_AIC1, RM_AIC1_WL, width); + ret = snd_soc_component_update_bits(component, + R_AIC1, RM_AIC1_WL, width); if (ret < 0) { - dev_err(component->dev, "Failed to set sample width (%d)\n", ret); + dev_err(component->dev, + "Failed to set sample width (%d)\n", ret); return ret; } return 0; } -static int setup_sample_rate(struct snd_soc_component *component, unsigned int rate) +static int setup_sample_rate(struct snd_soc_component *component, + unsigned int rate) { struct tscs42xx *tscs42xx = snd_soc_component_get_drvdata(component); unsigned int br, bm; @@ -881,24 +889,32 @@ static int setup_sample_rate(struct snd_soc_component *component, unsigned int r } /* DAC and ADC share bit and frame clock */ - ret = snd_soc_component_update_bits(component, R_DACSR, RM_DACSR_DBR, br); + ret = snd_soc_component_update_bits(component, + R_DACSR, RM_DACSR_DBR, br); if (ret < 0) { - dev_err(component->dev, "Failed to update register (%d)\n", ret); + dev_err(component->dev, + "Failed to update register (%d)\n", ret); return ret; } - ret = snd_soc_component_update_bits(component, R_DACSR, RM_DACSR_DBM, bm); + ret = snd_soc_component_update_bits(component, + R_DACSR, RM_DACSR_DBM, bm); if (ret < 0) { - dev_err(component->dev, "Failed to update register (%d)\n", ret); + dev_err(component->dev, + "Failed to update register (%d)\n", ret); return ret; } - ret = snd_soc_component_update_bits(component, R_ADCSR, RM_DACSR_DBR, br); + ret = snd_soc_component_update_bits(component, + R_ADCSR, RM_DACSR_DBR, br); if (ret < 0) { - dev_err(component->dev, "Failed to update register (%d)\n", ret); + dev_err(component->dev, + "Failed to update register (%d)\n", ret); return ret; } - ret = snd_soc_component_update_bits(component, R_ADCSR, RM_DACSR_DBM, bm); + ret = snd_soc_component_update_bits(component, + R_ADCSR, RM_DACSR_DBM, bm); if (ret < 0) { - dev_err(component->dev, "Failed to update register (%d)\n", ret); + dev_err(component->dev, + "Failed to update register (%d)\n", ret); return ret; } @@ -1076,7 +1092,8 @@ static int tscs42xx_hw_params(struct snd_pcm_substream *substream, ret = setup_sample_rate(component, params_rate(params)); if (ret < 0) { - dev_err(component->dev, "Failed to setup sample rate (%d)\n", ret); + dev_err(component->dev, + "Failed to setup sample rate (%d)\n", ret); return ret; } @@ -1087,7 +1104,8 @@ static inline int dac_mute(struct snd_soc_component *component) { int ret; - ret = snd_soc_component_update_bits(component, R_CNVRTR1, RM_CNVRTR1_DACMU, + ret = snd_soc_component_update_bits(component, + R_CNVRTR1, RM_CNVRTR1_DACMU, RV_CNVRTR1_DACMU_ENABLE); if (ret < 0) { dev_err(component->dev, "Failed to mute DAC (%d)\n", @@ -1102,7 +1120,8 @@ static inline int dac_unmute(struct snd_soc_component *component) { int ret; - ret = snd_soc_component_update_bits(component, R_CNVRTR1, RM_CNVRTR1_DACMU, + ret = snd_soc_component_update_bits(component, + R_CNVRTR1, RM_CNVRTR1_DACMU, RV_CNVRTR1_DACMU_DISABLE); if (ret < 0) { dev_err(component->dev, "Failed to unmute DAC (%d)\n", @@ -1117,8 +1136,8 @@ static inline int adc_mute(struct snd_soc_component *component) { int ret; - ret = snd_soc_component_update_bits(component, R_CNVRTR0, RM_CNVRTR0_ADCMU, - RV_CNVRTR0_ADCMU_ENABLE); + ret = snd_soc_component_update_bits(component, + R_CNVRTR0, RM_CNVRTR0_ADCMU, RV_CNVRTR0_ADCMU_ENABLE); if (ret < 0) { dev_err(component->dev, "Failed to mute ADC (%d)\n", ret); @@ -1132,8 +1151,8 @@ static inline int adc_unmute(struct snd_soc_component *component) { int ret; - ret = snd_soc_component_update_bits(component, R_CNVRTR0, RM_CNVRTR0_ADCMU, - RV_CNVRTR0_ADCMU_DISABLE); + ret = snd_soc_component_update_bits(component, + R_CNVRTR0, RM_CNVRTR0_ADCMU, RV_CNVRTR0_ADCMU_DISABLE); if (ret < 0) { dev_err(component->dev, "Failed to unmute ADC (%d)\n", ret); @@ -1171,8 +1190,8 @@ static int tscs42xx_set_dai_fmt(struct snd_soc_dai *codec_dai, /* Slave mode not supported since it needs always-on frame clock */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: - ret = snd_soc_component_update_bits(component, R_AIC1, RM_AIC1_MS, - RV_AIC1_MS_MASTER); + ret = snd_soc_component_update_bits(component, + R_AIC1, RM_AIC1_MS, RV_AIC1_MS_MASTER); if (ret < 0) { dev_err(component->dev, "Failed to set codec DAI master (%d)\n", ret); @@ -1211,14 +1230,18 @@ static int tscs42xx_set_dai_bclk_ratio(struct snd_soc_dai *codec_dai, return -EINVAL; } - ret = snd_soc_component_update_bits(component, R_DACSR, RM_DACSR_DBCM, value); + ret = snd_soc_component_update_bits(component, + R_DACSR, RM_DACSR_DBCM, value); if (ret < 0) { - dev_err(component->dev, "Failed to set DAC BCLK ratio (%d)\n", ret); + dev_err(component->dev, + "Failed to set DAC BCLK ratio (%d)\n", ret); return ret; } - ret = snd_soc_component_update_bits(component, R_ADCSR, RM_ADCSR_ABCM, value); + ret = snd_soc_component_update_bits(component, + R_ADCSR, RM_ADCSR_ABCM, value); if (ret < 0) { - dev_err(component->dev, "Failed to set ADC BCLK ratio (%d)\n", ret); + dev_err(component->dev, + "Failed to set ADC BCLK ratio (%d)\n", ret); return ret; } @@ -1309,7 +1332,7 @@ static int part_is_valid(struct tscs42xx *tscs42xx) }; } -static struct snd_soc_component_driver soc_codec_dev_tscs42xx = { +static const struct snd_soc_component_driver soc_codec_dev_tscs42xx = { .dapm_widgets = tscs42xx_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(tscs42xx_dapm_widgets), .dapm_routes = tscs42xx_intercon, @@ -1416,8 +1439,8 @@ static int tscs42xx_i2c_probe(struct i2c_client *i2c, mutex_init(&tscs42xx->coeff_ram_lock); mutex_init(&tscs42xx->pll_lock); - ret = devm_snd_soc_register_component(tscs42xx->dev, &soc_codec_dev_tscs42xx, - &tscs42xx_dai, 1); + ret = devm_snd_soc_register_component(tscs42xx->dev, + &soc_codec_dev_tscs42xx, &tscs42xx_dai, 1); if (ret) { dev_err(tscs42xx->dev, "Failed to register codec (%d)\n", ret); return ret; -- 2.15.1

2 2

[alsa-devel] [GIT PULL] ASoC updates for v4.17
by Mark Brown 03 Apr '18

03 Apr '18

The following changes since commit 3eb2ce825ea1ad89d20f7a3b5780df850e4be274: Linux 4.16-rc7 (2018-03-25 12:44:30 -1000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v4.17 for you to fetch changes up to 445bb423f6a3c50788529a33b82e45148001e855: Merge remote-tracking branch 'asoc/topic/zx_aud96p22' into asoc-next (2018-03-28 10:32:03 +0800) ---------------------------------------------------------------- ASoC: Updates for v4.17 This is a *very* big release for ASoC. Not much change in the core but there s the transition of all the individual drivers over to components which is intended to support further core work. The goal is to make it easier to do further core work by removing the need to special case all the different driver classes in the core, many of the devices end up being used in multiple roles in modern systems. We also have quite a lot of new drivers added this month of all kinds, quite a few for simple devices but also some more advanced ones with more substantial code. - The biggest thing is the huge series from Morimoto-san which converted everything over to components. This is a huge change by code volume but was fairly mechanical - Many fixes for some of the Realtek based Baytrail systems covering both the CODECs and the CPUs, contributed by Hans de Goode. - Lots of cleanups for Samsung based Odroid systems from Sylwester Nawrocki. - The Freescale SSI driver also got a lot of cleanups from Nicolin Chen. - The Blackfin drivers have been removed as part of the removal of the architecture. - New drivers for AKM AK4458 and AK5558, several AMD based machines, several Intel based machines, Maxim MAX9759, Motorola CPCAP, Socionext Uniphier SoCs, and TI PCM1789 and TDA7419 ---------------------------------------------------------------- I might have some more fixes to come, I was at Linaro Connect and then on holiday for a week so I've got a bunch of e-mail to trawl through but I wanted to get this out promptly. Abhijeet Kumar (4): ALSA: hda: Copying sync power state helper to core ALSA: hda: Make use of core codec functions to sync power state ASoC: hdac_hdmi : Ensuring proper setting of output widget power state ASoC: Intel: Skylake: Fix typo Adam Thomson (2): ASoC: da7219: Add common clock usage for providing DAI clks ASoC: da7219: clkdev_drop usage depends on CONFIG_COMMON_CLK Akshu Agrawal (5): ASoC: AMD: Add machine driver for ST DA7219 MAX98357 ASoC: dwc: Extends DW_I2S_QUIRK_COMP_PARAM1 to playback ASoC: ADAU7002: Adding ACPI id ASoC: amd: Use single dai for da7219 playback and capture ASoC: amd: Enable da7219 master clock using common clock framework Arnd Bergmann (1): ASoC: remove blackfin drivers Bard Liao (6): ASoC: rt5665: remove rt5668 support ASoC: rt5659: fix wrong bit define for ADC L2 power. ASoC: rt5659: fix wrong control register for ADC2 power ASoC: rt5659: Separate adc 1/2 clock control ASoC: rt286: don't turn off HV and VREF if headset is detected ASoC: rt298: don't turn off HV and VREF if headset is detected Charles Keepax (2): ASoC: arizona: Fixup some minor formatting issues ASoC: wm_adsp: Support streams which can start/stop with DSP active Colin Ian King (7): ASoC: Intel: Skylake: make function skl_clk_round_rate static ASoC: uniphier: remove redundant check of blk_id ASoC: TSCS42xx: make const array norm_addrs static, reduces object code size ASoC: sirf: remove duplicated bit-wise or of USP_RXFIFO_THD_INT ASoC: pxa: remove duplicated bit-wise or of SNDRV_PCM_FMTBIT_S24_LE ASoC: cygnus: remove redundant assignment to pointer 'res' ASoC: da7219: make structure da7219_dai_clks_ops static Corentin Labbe (2): ASoC: samsung: clean makefile about inexistant files ASoC: sn95031: remove dead makefile about sn95031 Cosmin-Gabriel Samoila (4): ASoC: ak4458: Add support for AK4458 DAC driver ASoC: ak4458: Add bindings for AK4458 DAC ASoC: ak4458: Modify SPDX license format ASoC: ak4458: Sort headers alphabetically Dan Carpenter (1): ASoC: uniphier: fix an error code in uniphier_aio_comprdma_new() Daniel Baluta (3): ASoC: ak5558: Add support for AK5558 ADC driver ASoC: ak5558: Add bindings for AK5558 ADC ASoC: ak5558: Fix style for SPDX identifier Ezequiel Garcia (3): ASoC: rockchip: Fix dai_name for HDMI codec ASoC: rockchip: Fix compatible string in example ASoC: rockchip: rk3288-hdmi-analog: Select needed codecs Fabio Estevam (17): ASoC: fsl-asoc-card: Use 'snd_pcm_format_t' type when appropriate ASoC: fsl_esai: Use 'const _be *' type for iprop ASoC: fsl_utils: Use 'const _be *' type for iprop ASoC: fsl_ssi: Use 'const _be *' type for iprop ASoC: wm9712: Use empty struct initializer ASoC: tlv320aic3x: Use empty struct initializer ASoC: adau17x1: Use empty struct initializer ASoC: wm9713: Use empty struct initializer ASoC: soc-dapm: Use empty struct initializer ASoC: soc-generic-dmaengine-pcm: Add a DMA debugfs_prefix entry ASoC: soc-generic-dmaengine-pcm: Fix error handling ASoC: soc-generic-dmaengine-pcm: Fix sparse warnings ASoC: Revert "ASoC: soc-generic-dmaengine-pcm: Fix error handling" ASoC: imx-wm8962: Remove machine driver ASoC: wm8524: Remove unit address ASoC: sgtl5000: Pass the required '#sound-dai-cells' ASoC: tscs42xx: Remove owner assignment from i2c_driver Geert Uytterhoeven (3): ASoC: rt5514: Fix rt5514_spi_burst_read() buffer passing ASoC: rt5514: Fix uninitialized calibration value ASoC: rsnd: Document R-Car M3-W support Guneshwor Singh (1): ASoC: Intel: Skylake: Refine skl widget type check Hans de Goede (45): ASoC: Intel: sst: Fix error-code check in sst_pause_stream() ASoC: Intel: sst: Remove 2 unused members from stream_info struct ASoC: Intel: sst: Remove unnecessary sst_init_stream() function ASoC: Intel: sst: Remove unused STREAM_DECODE and STREAM_RESET states ASoC: Intel: sst: Add sst_realloc_stream() function ASoC: Intel: sst: Free streams on suspend, re-alloc on resume ASoC: rt5651: Remove unused rt5651_platform_data ASoC: rt5651: Move all jack-detect initialization to rt5651_set_jack_detect ASoC: rt5651: Move 2 functions higher up in rt5651.c ASoC: rt5651: Use standard component set_jack callback ASoC: rt5651: Add rt5651_apply_properties() helper function ASoC: rt5651: Remove is_sys_clk_from_pll() ASoC: rt5651: Fix bias_level confusion ASoC: rt5651: Do not modify the LDO voltage control bits from set_bias_level() ASoC: rt5651: Do not modify jd and PLL power bits from set_bias_level() ASoC: rt5651: Remove programming of PWR regs before force_bias_level() call ASoC: rt5651: Only configure LDO voltage once at boot ASoC: rt5651: Remove "JD Power" dapm supply ASoC: rt5651: Enable LDO and micbias1 supplies for jack-type detection ASoC: rt5651: Only configure OVCD once at set_jack time ASoC: rt5651: Always keep OVCD enabled ASoC: rt5651: Make rt5651_apply_properties() private ASoC: rt5651: Add devicetree-bindings for jack-detect ASoC: rt5651: Configure jack-detect source through a device-property ASoC: rt5651: Allow specifying over-current threshold through a device-property ASoC: rt5651: Allow specifying the OVCD scale-factor through a device-property ASoC: rt5651: Enable sticky mode for OVCD ASoC: rt5651: Enable Platform Clock during jack-type detect ASoC: rt5651: Add rt5651_jack_inserted() helper ASoC: rt5651: Rewrite jack-type detection ASoC: Intel: bytcr_rt5651: Not being able to find the codec ACPI-dev is an error ASoC: Intel: bytcr_rt5651: Pass jack-src info via device-properties ASoC: Intel: bytcr_rt5651: Actually honor the DMIC_EN quirk if specified ASoC: Intel: bytcr_rt5651: Only create jack if we have a jack-detect source ASoC: Intel: bytcr_rt5651: Add quirk micbias OVCD configuration ASoC: Intel: bytcr_rt5651: Configure PLL1 before using it ASoC: Intel: bytcr_rt5651: Drop snd_soc_dai_set_bclk_ratio() call ASoC: Intel: bytcr_rt5651: Rename IN3_MAP to IN1_HS_IN3_MAP ASoC: Intel: bytcr_rt5651: Add new IN2_HS_IN3 input map and a quirk using it ASoC: Intel: bytcr_rt5651: Add support for Bay Trail CR / SSP0 using boards ASoC: Intel: bytcr_rt5651: Add quirk for the VIOS LTH17 laptop ASoC: Intel: bytcr_rt5651: Change defaults to enable jack-detect, analog mics ASoC: Intel: bytcr_rt5651: Select RCCLK on init() ASoC: rt5651: move definitions of dt-binding constants to include/dt-bindings ASoC: rt5651: Fix jack-dectect typo in the dt-bindings documentation James Kelly (1): ASoC: ssm2602: Replace reg_default_raw with reg_default John Hsu (3): ASoC: Intel: add nau8824 sound card ASoC: Intel: fix argument error in nau8824 machine ASoC: nau8824: recover system clock when device changes Katsuhiro Suzuki (15): ASoC: uniphier: add DT bindings documentation for UniPhier AIO ASoC: uniphier: add support for UniPhier AIO common driver ASoC: uniphier: add support for UniPhier AIO DMA driver ASoC: uniphier: add support for UniPhier AIO CPU DAI driver ASoC: uniphier: add support for UniPhier AIO compress audio ASoC: uniphier: add support for UniPhier LD11/LD20 AIO driver ASoC: uniphier: evea: fix typo 'eva' -> 'evea' ASoC: uniphier: evea: fix lisence comment style ASoC: uniphier: fix broken sound if use SRC in replay ASoC: add DT bindings documentation for ROHM BD28623 codec ASoC: support ROHM BD28623 codec ASoC: uniphier: add support for UniPhier PXs2 AIO ASoC: uniphier: evea: add switch for changing source of line-in ASoC: uniphier: add syscon property to binding document ASoC: uniphier: add syscon property for UniPhier sound system Krzysztof Kozlowski (1): ASoC: samsung: Mark unused Odroid compatibles as deprecated Kuninori Morimoto (243): ASoC: ac97: replace codec to component ASoC: wm0010: replace codec to component ASoC: wm8804: replace codec to component ASoC: cs4271: replace codec to component ASoC: wm2000: replace codec to component ASoC: wm2200: replace codec to component ASoC: wm5100: replace codec to component ASoC: wm8350: replace codec to component ASoC: wm8400: replace codec to component ASoC: wm8580: replace codec to component ASoC: wm8524: replace codec to component ASoC: wm8510: replace codec to component ASoC: wm8523: replace codec to component ASoC: wm8711: replace codec to component ASoC: wm8750: replace codec to component ASoC: wm8737: replace codec to component ASoC: wm8776: replace codec to component ASoC: wm8770: replace codec to component ASoC: wm8727: replace codec to component ASoC: wm8731: replace codec to component ASoC: wm8782: replace codec to component ASoC: wm8728: replace codec to component ASoC: wm8741: replace codec to component ASoC: wm8753: replace codec to component ASoC: wm8900: replace codec to component ASoC: wm8903: replace codec to component ASoC: wm8955: replace codec to component ASoC: wm8960: replace codec to component ASoC: wm8985: replace codec to component ASoC: wm8971: replace codec to component ASoC: wm8978: replace codec to component ASoC: wm8974: replace codec to component ASoC: wm8990: replace codec to component ASoC: wm8988: replace codec to component ASoC: wm8983: replace codec to component ASoC: wm8962: replace codec to component ASoC: wm8996: replace codec to component ASoC: wm8991: replace codec to component ASoC: wm8995: replace codec to component ASoC: wm8961: replace codec to component ASoC: wm8940: replace codec to component ASoC: wm8904: replace codec to component ASoC: wm9081: replace codec to component ASoC: wm9090: replace codec to component ASoC: wm9867: replace codec to component ASoC: wm1250-ev1: replace codec to component ASoC: ak4613: replace codec to component ASoC: ak4642: replace codec to component ASoC: ak5386: replace codec to component ASoC: ak4671: replace codec to component ASoC: ak4104: replace codec to component ASoC: ak4535: replace codec to component ASoC: ak4641: replace codec to component ASoC: ak4554: replace codec to component ASoC: rt274: replace codec to component ASoC: rt5616: replace codec to component ASoC: rt5640: replace codec to component ASoC: rt5651: replace codec to component ASoC: rt5514: replace codec to component ASoC: rt5659: replace codec to component ASoC: rt5670: replace codec to component ASoC: rt5660: replace codec to component ASoC: rt5631: replace codec to component ASoC: rt5665: replace codec to component ASoC: cs4270: replace codec to component ASoC: cs4349: replace codec to component ASoC: cs4265: replace codec to component ASoC: cs35l32: replace codec to component ASoC: cs35l33: replace codec to component ASoC: cs35l35: replace codec to component ASoC: cs35l34: replace codec to component ASoC: cs42xx8: replace codec to component ASoC: cs42l73: replace codec to component ASoC: cs42l52: replace codec to component ASoC: cs42l56: replace codec to component ASoC: cs42l51: replace codec to component ASoC: cs42l42: replace codec to component ASoC: cs43130: replace codec to component ASoC: cs53l30: replace codec to component ASoC: da732x: replace codec to component ASoC: da7210: replace codec to component ASoC: da7218: replace codec to component ASoC: da7213: replace codec to component ASoC: da9055: replace codec to component ASoC: max9860: replace codec to component ASoC: max9850: replace codec to component ASoC: max98371: replace codec to component ASoC: max98095: replace codec to component ASoC: max98090: replace codec to component ASoC: max98926: replace codec to component ASoC: max98088: replace codec to component ASoC: max98925: replace codec to component ASoC: max98927: replace codec to component ASoC: max98357a: replace codec to component ASoC: max98373: replace codec to component ASoC: adav80x: replace codec to component ASoC: adau1373: replace codec to component ASoC: adau7002: replace codec to component ASoC: adau1977: replace codec to component ASoC: adau17x1/adau1761/adau1781: replace codec to component ASoC: tlv320aic23: replace codec to component ASoC: tlv320aic26: replace codec to component ASoC: tlv320aic3x: replace codec to component ASoC: tlv320dac33: replace codec to component ASoC: tlv320aic32x4: replace codec to component ASoC: tlv320aic31xx: replace codec to component ASoC: cx20442: replace codec to component ASoC: tscs42xx: replace codec to component ASoC: pcm179x: replace codec to component ASoC: pcm3008: replace codec to component ASoC: pcm1681: replace codec to component ASoC: pcm512x: replace codec to component ASoC: pcm5102a: replace codec to component ASoC: pcm3168a: replace codec to component ASoC: twl6040: replace codec to component ASoC: twl4030: replace codec to component ASoC: msm8916-wcd-analog: replace codec to component ASoC: msm8916-wcd-digital: replace codec to component ASoC: spdif_transmitter: replace codec to component ASoC: spdif_receiver: replace codec to component ASoC: tas5720: replace codec to component ASoC: tas2552: replace codec to component ASoC: sun4i: replace codec to component ASoC: sun8i: replace codec to component ASoC: ads117x: replace codec to component ASoC: ab8500: replace codec to component ASoC: ad193x: replace codec to component ASoC: ad1836: replace codec to component ASoC: ad73311: replace codec to component ASoC: ssm4567: replace codec to component ASoC: ssm2602: replace codec to component ASoC: ssm2518: replace codec to component ASoC: sta350: replace codec to component ASoC: sta32x: replace codec to component ASoC: sta529: replace codec to component ASoC: tas5086: replace codec to component ASoC: tas571x: replace codec to component ASoC: nau8824: replace codec to component ASoC: nau8810: replace codec to component ASoC: nau8540: replace codec to component ASoC: es8316: replace codec to component ASoC: es7134: replace codec to component ASoC: es8328: replace codec to component ASoC: alc5632: replace codec to component ASoC: alc5623: replace codec to component ASoC: hdmi-codec: replace codec to component ASoC: bt-sco: replace codec to component ASoC: vc4_hdmi: replace codec to component ASoC: zx_aud96p22: replace codec to component ASoC: wl1273: replace codec to component ASoC: mc13783: replace codec to component ASoC: sgtl5000: replace codec to component ASoC: sirf-audio: replace codec to component ASoC: inno_rk3036: replace codec to component ASoC: isabelle: replace codec to component ASoC: tfa9879: replace codec to component ASoC: dmic: replace codec to component ASoC: sti-sas: replace codec to component ASoC: gtm601: replace codec to component ASoC: 88pm860x: replace codec to component ASoC: pistachio: replace codec to component ASoC: lm49453: replace codec to component ASoC: cq93vc: replace codec to component ASoC: jz4740: replace codec to component ASoC: uda1380: replace codec to component ASoC: ml26124: replace codec to component ASoC: si476x: replace codec to component ASoC: uda134x: replace codec to component ASoC: ics43432: replace codec to component ASoC: uniphier: evea: replace codec to component ASoC: pcm186x: replace codec to component ASoC: tas6424: replace codec to component ASoC: atmel-pdmic: replace codec to component ASoC: atmel-classd: replace codec to component ASoC: hdac_hdmi/nau8825/rt286/rt298/rt5663/da7219: replace codec to component ASoC: rt5645/rt5677: replace codec to component ASoC: wm8993/wm8994/wm8958: replace codec to component ASoC: remove rtd->platform checck ASoC: soc-utils: replace platform to component ASoC: soc-generic-dmaengine-pcm: replace platform to component ASoC: intel: atom: replace platform to component ASoC: intel: skylake: replace platform to component ASoC: intel: baytrail: replace platform to component ASoC: intel: haswell: replace platform to component ASoC: bcm: cygnus: replace platform to component ASoC: sh: rsnd: replace platform to component ASoC: sh: dma-sh7760: replace platform to component ASoC: sh: fsi: replace platform to component ASoC: sh: siu: replace platform to component ASoC: atmel: replace platform to component ASoC: amd: replace platform to component ASoC: cs47l24: replace platform to component ASoC: rt5514-spi: replace platform to component ASoC: wm5110: replace platform to component ASoC: wm5102: replace platform to component ASoC: davinci-i2s: replace platform to component ASoC: fsl: dma: replace platform to component ASoC: fsl: asrc: replace platform to component ASoC: fsl: imx: replace platform to component ASoC: fsl: mpc5200: replace platform to component ASoC: dwc-pcm: replace platform to component ASoC: kirkwood: replace platform to component ASoC: pxa: mmp: replace platform to component ASoC: pxa: pxa2xx: replace platform to component ASoC: qcom: lpass-platform: replace platform to component ASoC: samsung: idma: replace platform to component ASoC: xtfpga-i2s: replace platform to component ASoC: blackfin: bf5xx-ac97-pcm: replace platform to component ASoC: blackfin: bf5xx-i2s-pcm: replace platform to component ASoC: omap-pcm: replace platform to component ASoC: au1x: dbdma2: replace platform to component ASoC: au1x: dma: replace platform to component ASoC: nuc900-pcm: replace platform to component ASoC: txx9aclc: replace platform to component ASoC: mt2701/mt8173: replace platform to component ASoC: stm: stm32_adfsdm: replace platform to component ASoC: soc-utils: replace codec to component ASoC: adau1701: replace codec to component ASoC: rsnd: indicate IRQ error status for debug ASoC: rsnd: suppress rsnd_dai_call() debug message ASoC: wm9081: fixup wm9081_digital_mute() reg read ASoC: wm9712: replace codec to component ASoC: wm9713: replace codec to component ASoC: ad73311: replace codec to component ASoC: stac9766: replace codec to component ASoC: wm9705: replace codec to component ASoC: ad1980: replace codec to component ASoC: uniphier: aio-dma: replace platform to component ASoC: arizona: replace codec to component ASoC: ak4613: don't have .use_pmdown_time ASoC: ak4642: don't have .use_pmdown_time ASoC: soc-core: soc_probe_dai() code simplification ASoC: intel: kbl_da7219_max98357: replace codec to component ASoC: amd: acp-da7219-max98357: replace codec to component ASoC: ak4458: replace codec to component ASoC: ak5558: replace codec to component ASoC: pxa/mioa701_wm9713: replace codec to component ASoC: dmic: don't use codec anymore soc-core: don't call kfree() for component ASoC: twl6040: remove duplicated remove callback ASoC: amd: don't use codec anymore ASoC: Intel: bytcr_rt5651: don't use codec anymore ASoC: cpcap: replace codec to component Ladislav Michl (6): ASoC: max9867: Show Kconfig entry ASoC: max9867: Improve error logging ASoC: max9867: Fix codec capabilities ASoC: max9867: DSP mode ASoC: atmel: Remove redundant dev_err() call in probe function ASoC: max9867: Drop probe function Liam Girdwood (3): ASoC: pcm: improve debug output for DPCM BE searching. ASoC: topology: Check widget kcontrols before deref. ASoC: topology: Fix kcontrol name string handling Mark Brown (108): Merge remote-tracking branch 'asoc/fix/core' into asoc-linus Merge remote-tracking branch 'asoc/fix/intel' into asoc-linus Merge remote-tracking branch 'asoc/fix/rockchip' into asoc-linus Merge remote-tracking branches 'asoc/fix/compress' and 'asoc/fix/mtk' into asoc-linus Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branch 'asoc/topic/intel' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm186x' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm512x' into asoc-next Merge remote-tracking branch 'asoc/topic/qcom' into asoc-next Merge remote-tracking branch 'asoc/topic/rcar' into asoc-next Merge remote-tracking branch 'asoc/topic/rockchip' into asoc-next Merge remote-tracking branch 'asoc/topic/twl-breakage' into asoc-next Merge remote-tracking branches 'asoc/topic/88pm860x', 'asoc/topic/ak4613', 'asoc/topic/amd', 'asoc/topic/atmel' and 'asoc/topic/au1x' into asoc-next Merge remote-tracking branches 'asoc/topic/bcm2835', 'asoc/topic/codecs', 'asoc/topic/compress' and 'asoc/topic/cq93vc' into asoc-next Merge remote-tracking branches 'asoc/topic/cs35l32', 'asoc/topic/cs35l34', 'asoc/topic/cs42l52', 'asoc/topic/cs42l56' and 'asoc/topic/cs42l73' into asoc-next Merge remote-tracking branches 'asoc/topic/cs47l24', 'asoc/topic/cx20442', 'asoc/topic/da7213', 'asoc/topic/da7218' and 'asoc/topic/dai-drv' into asoc-next Merge remote-tracking branches 'asoc/topic/davinci', 'asoc/topic/debugfs', 'asoc/topic/disconnect', 'asoc/topic/dmic' and 'asoc/topic/ep93xx' into asoc-next Merge remote-tracking branches 'asoc/topic/eukrea-tlv320', 'asoc/topic/fsl', 'asoc/topic/fsl-ssi' and 'asoc/topic/fsl_asrc' into asoc-next Merge remote-tracking branches 'asoc/topic/hdac_hdmi', 'asoc/topic/hisilicon', 'asoc/topic/iio' and 'asoc/topic/max98373' into asoc-next Merge remote-tracking branches 'asoc/topic/max98926', 'asoc/topic/max98927', 'asoc/topic/mc13783', 'asoc/topic/msm8916' and 'asoc/topic/mt8173' into asoc-next Merge remote-tracking branches 'asoc/topic/mtk', 'asoc/topic/mxs', 'asoc/topic/mxs-sgtl5000', 'asoc/topic/nau8540' and 'asoc/topic/nau8824' into asoc-next Merge remote-tracking branches 'asoc/topic/nau8825' and 'asoc/topic/nuc900' into asoc-next Merge remote-tracking branches 'asoc/topic/rl6231' and 'asoc/topic/rt5514' into asoc-next Merge remote-tracking branches 'asoc/topic/rt5645', 'asoc/topic/sam9g20_wm8731', 'asoc/topic/sam9x5_wm8731', 'asoc/topic/samsung' and 'asoc/topic/sgtl5000' into asoc-next Merge remote-tracking branches 'asoc/topic/si476x', 'asoc/topic/simple', 'asoc/topic/spdif', 'asoc/topic/st-dfsdm' and 'asoc/topic/stm32' into asoc-next Merge remote-tracking branches 'asoc/topic/sun4i-i2s', 'asoc/topic/sun8i-codec', 'asoc/topic/sunxi', 'asoc/topic/symmetry' and 'asoc/topic/tas5720' into asoc-next Merge remote-tracking branches 'asoc/topic/tas6424', 'asoc/topic/tfa9879', 'asoc/topic/tlv320aic31xx', 'asoc/topic/tlv320aic32x4' and 'asoc/topic/tlv320aic3x' into asoc-next Merge remote-tracking branches 'asoc/topic/tlv320dac33', 'asoc/topic/ts3a227e' and 'asoc/topic/tscs42xx' into asoc-next Merge remote-tracking branches 'asoc/topic/uda1380', 'asoc/topic/uniphier' and 'asoc/topic/utils' into asoc-next Merge remote-tracking branches 'asoc/topic/ux500', 'asoc/topic/wm0010', 'asoc/topic/wm2000', 'asoc/topic/wm5102' and 'asoc/topic/wm5110' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8350', 'asoc/topic/wm8400', 'asoc/topic/wm8903', 'asoc/topic/wm8994' and 'asoc/topic/wm8997' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8998' into asoc-next Merge branch 'topic/component-platform' of https://git.kernel.org/.../broonie/sound into asoc-samsung Merge branch 'fix/sgtl5000' of https://git.kernel.org/.../broonie/sound into asoc-sgtl5000 Merge branch 'topic/hda-sync-power' of https://git.kernel.org/.../tiwai/sound into asoc-intel ASoC: Intel: Fix build Merge branch 'fix/sgtl5000' of https://git.kernel.org/.../broonie/sound into asoc-sgtl5000 Merge branch 'fix/hdmi-codec' of https://git.kernel.org/.../broonie/sound into asoc-hdmi-codec Merge branch 'topic/component-platform' of https://git.kernel.org/.../broonie/sound into HEAD Merge branch 'topic/generic-dmaengine' of https://git.kernel.org/.../broonie/sound into asoc-stm32 Merge branch 'topic/component-platform' of https://git.kernel.org/.../broonie/sound into asoc-intel Merge branch 'topic/wm9867' of https://git.kernel.org/.../broonie/sound into asoc-max9867 Merge branch 'topic/rt5651' of https://git.kernel.org/.../broonie/sound into asoc-intel ASoC: core: Fix typo roup->group Merge branch 'fix/rt5659' of https://git.kernel.org/.../broonie/sound into asoc-rt5659 Merge branch 'topic/of' of https://git.kernel.org/.../broonie/sound into asoc-samsung Merge branch 'topic/intel' of https://git.kernel.org/.../broonie/sound into asoc-amd Merge remote-tracking branch 'asoc/fix/component' into asoc-linus Merge remote-tracking branches 'asoc/fix/mtk', 'asoc/fix/rt5659', 'asoc/fix/ssm2602' and 'asoc/fix/topology' into asoc-linus Merge tag 'asoc-v4.16-4' into asoc-next Merge remote-tracking branch 'asoc/topic/component-platform' into asoc-next Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/intel' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm1681' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm1789' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm179x' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm186x' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm3008' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm3168a' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm5102a' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm512x' into asoc-next Merge remote-tracking branch 'asoc/topic/rt5665' into asoc-next Merge remote-tracking branch 'asoc/topic/samsung' into asoc-next Merge remote-tracking branches 'asoc/topic/88pm860x', 'asoc/topic/ab8500', 'asoc/topic/ac97', 'asoc/topic/ad1836' and 'asoc/topic/ad193x' into asoc-next Merge remote-tracking branches 'asoc/topic/ad73311', 'asoc/topic/adau1373', 'asoc/topic/adau17x1', 'asoc/topic/adau1977' and 'asoc/topic/adau7002' into asoc-next Merge remote-tracking branches 'asoc/topic/adav80x', 'asoc/topic/ads117x', 'asoc/topic/ak4104', 'asoc/topic/ak4458' and 'asoc/topic/ak4535' into asoc-next Merge remote-tracking branches 'asoc/topic/ak4554', 'asoc/topic/ak4613', 'asoc/topic/ak4641', 'asoc/topic/ak4642' and 'asoc/topic/ak4671' into asoc-next Merge remote-tracking branches 'asoc/topic/ak5386', 'asoc/topic/ak5558', 'asoc/topic/alc5623', 'asoc/topic/alc5632' and 'asoc/topic/amd' into asoc-next Merge remote-tracking branches 'asoc/topic/arizona', 'asoc/topic/atmel', 'asoc/topic/bd28623', 'asoc/topic/blackfin' and 'asoc/topic/bt-sco' into asoc-next Merge remote-tracking branches 'asoc/topic/cpcap', 'asoc/topic/cq93vc' and 'asoc/topic/cs35l32' into asoc-next Merge remote-tracking branches 'asoc/topic/cs35l33', 'asoc/topic/cs35l34', 'asoc/topic/cs35l35', 'asoc/topic/cs4265' and 'asoc/topic/cs4270' into asoc-next Merge remote-tracking branches 'asoc/topic/cs4271', 'asoc/topic/cs42l42', 'asoc/topic/cs42l51', 'asoc/topic/cs42l52' and 'asoc/topic/cs42l56' into asoc-next Merge remote-tracking branches 'asoc/topic/cs42l73', 'asoc/topic/cs42xx8', 'asoc/topic/cs43130', 'asoc/topic/cs4349' and 'asoc/topic/cs53l30' into asoc-next Merge remote-tracking branches 'asoc/topic/cx20442', 'asoc/topic/cygnus', 'asoc/topic/da7210', 'asoc/topic/da7213' and 'asoc/topic/da7218' into asoc-next Merge remote-tracking branches 'asoc/topic/da7219', 'asoc/topic/da732x', 'asoc/topic/da9055' and 'asoc/topic/dmic' into asoc-next Merge remote-tracking branches 'asoc/topic/dwc', 'asoc/topic/es7134', 'asoc/topic/es8316', 'asoc/topic/es8328' and 'asoc/topic/fsl' into asoc-next Merge remote-tracking branches 'asoc/topic/fsl_esai', 'asoc/topic/fsl_ssi', 'asoc/topic/fsl_utils', 'asoc/topic/generic-dmaengine' and 'asoc/topic/gtm601' into asoc-next Merge remote-tracking branches 'asoc/topic/hdmi-codec', 'asoc/topic/ics43432', 'asoc/topic/imx-wm8962' and 'asoc/topic/inno_rk3036' into asoc-next Merge remote-tracking branches 'asoc/topic/isabelle', 'asoc/topic/jz4740', 'asoc/topic/lm49453', 'asoc/topic/max9759' and 'asoc/topic/max98088' into asoc-next Merge remote-tracking branches 'asoc/topic/max98090', 'asoc/topic/max98095', 'asoc/topic/max98357a', 'asoc/topic/max98371' and 'asoc/topic/max98373' into asoc-next Merge remote-tracking branches 'asoc/topic/max9850', 'asoc/topic/max9860', 'asoc/topic/max9867', 'asoc/topic/max98925' and 'asoc/topic/max98926' into asoc-next Merge remote-tracking branches 'asoc/topic/max98927', 'asoc/topic/mc13783', 'asoc/topic/mediatek', 'asoc/topic/ml26124' and 'asoc/topic/msm8916-wcd-analog' into asoc-next Merge remote-tracking branches 'asoc/topic/msm8916-wcd-digital', 'asoc/topic/mtk', 'asoc/topic/nau8540', 'asoc/topic/nau8810' and 'asoc/topic/nau8824' into asoc-next Merge remote-tracking branches 'asoc/topic/pistachio', 'asoc/topic/pxa', 'asoc/topic/rsnd', 'asoc/topic/rt274' and 'asoc/topic/rt286' into asoc-next Merge remote-tracking branches 'asoc/topic/rt298', 'asoc/topic/rt5514', 'asoc/topic/rt5616', 'asoc/topic/rt5631' and 'asoc/topic/rt5640' into asoc-next Merge remote-tracking branches 'asoc/topic/rt5645', 'asoc/topic/rt5651', 'asoc/topic/rt5659' and 'asoc/topic/rt5660' into asoc-next Merge remote-tracking branches 'asoc/topic/rt5670', 'asoc/topic/sgtl5000', 'asoc/topic/si476x' and 'asoc/topic/sirf' into asoc-next Merge remote-tracking branches 'asoc/topic/sirf-audio', 'asoc/topic/sn95031', 'asoc/topic/soc-dapm', 'asoc/topic/spdif_receiver' and 'asoc/topic/spdif_transmitter' into asoc-next Merge remote-tracking branches 'asoc/topic/ssm2518', 'asoc/topic/ssm2602', 'asoc/topic/ssm4567', 'asoc/topic/sta32x' and 'asoc/topic/sta350' into asoc-next Merge remote-tracking branches 'asoc/topic/sta529', 'asoc/topic/sti-sas', 'asoc/topic/stm32', 'asoc/topic/sun4i' and 'asoc/topic/sun8i' into asoc-next Merge remote-tracking branches 'asoc/topic/tas2552', 'asoc/topic/tas5086', 'asoc/topic/tas571x', 'asoc/topic/tas5720' and 'asoc/topic/tas6424' into asoc-next Merge remote-tracking branches 'asoc/topic/tda7419', 'asoc/topic/tfa9879', 'asoc/topic/tlv320aic23', 'asoc/topic/tlv320aic26' and 'asoc/topic/tlv320aic31xx' into asoc-next Merge remote-tracking branches 'asoc/topic/tlv320aic32x4', 'asoc/topic/tlv320aic3x', 'asoc/topic/tlv320dac33', 'asoc/topic/topology' and 'asoc/topic/tscs42xx' into asoc-next Merge remote-tracking branches 'asoc/topic/twl4030', 'asoc/topic/twl6040', 'asoc/topic/uda134x', 'asoc/topic/uda1380' and 'asoc/topic/uniphier' into asoc-next Merge remote-tracking branches 'asoc/topic/vc4_hdmi', 'asoc/topic/wl1273', 'asoc/topic/wm0010', 'asoc/topic/wm1250-ev1' and 'asoc/topic/wm2000' into asoc-next Merge remote-tracking branches 'asoc/topic/wm2200', 'asoc/topic/wm5100', 'asoc/topic/wm8350', 'asoc/topic/wm8400' and 'asoc/topic/wm8510' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8523', 'asoc/topic/wm8524', 'asoc/topic/wm8580', 'asoc/topic/wm8711' and 'asoc/topic/wm8727' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8728', 'asoc/topic/wm8731', 'asoc/topic/wm8737', 'asoc/topic/wm8741' and 'asoc/topic/wm8750' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8753', 'asoc/topic/wm8770', 'asoc/topic/wm8776', 'asoc/topic/wm8782' and 'asoc/topic/wm8804' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8900', 'asoc/topic/wm8903', 'asoc/topic/wm8904', 'asoc/topic/wm8940' and 'asoc/topic/wm8955' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8960', 'asoc/topic/wm8961', 'asoc/topic/wm8962', 'asoc/topic/wm8971' and 'asoc/topic/wm8974' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8978', 'asoc/topic/wm8983', 'asoc/topic/wm8985', 'asoc/topic/wm8988' and 'asoc/topic/wm8990' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8991', 'asoc/topic/wm8994', 'asoc/topic/wm8995', 'asoc/topic/wm8996' and 'asoc/topic/wm9081' into asoc-next Merge remote-tracking branches 'asoc/topic/wm9090', 'asoc/topic/wm9712', 'asoc/topic/wm9713' and 'asoc/topic/wm_adsp' into asoc-next Merge remote-tracking branch 'asoc/topic/zx_aud96p22' into asoc-next Martin Hundebøll (1): ASoC: soc-core: remove error due to probe deferral Masahiro Yamada (1): ASoC: uniphier: remove superfluous <linux/mfd/syscon.h> inclusion Masanari Iida (1): ASoC: mediatek: mt2701: Fix a typo in printk Matt Porter (2): ASoC: add tda7419 audio processor binding ASoC: add tda7419 audio processor driver Matthias Kaehlcke (1): ASoC: dmic: Add optional wakeup delay Michal Oleszczyk (2): ASoC: sgtl5000: add 'Audio Switch' block implementation ASoC: sgtl5000: add 5 band graphic equalizer Mukunda, Vijendar (2): ASoC: amd: renaming pcm substream names and bytescount params ASoC: amd: Coding style changes for acp dma driver Mylène Josserand (2): ASoC: codecs: Add support for PCM1789 ASoC: Add bindings for PCM1789 Naveen Manohar (3): ASoC: Intel: Add Kabylake-y Dialog Maxim machine driver ASoC: Intel: Add Kabylake Dialog+Maxim machine driver entry ASoC: Intel: kbl: Enable mclk and ssp sclk early for rt5663 Neil Armstrong (2): ASoC: Add DT bindings documentation for max9759 amplifier ASoC: max9759: Add Amplifier Driver Nicolin Chen (17): ASoC: fsl_ssi: Redefine RX and TX macros ASoC: fsl_ssi: Keep ssi->i2s_net updated ASoC: fsl_ssi: Clean up set_dai_tdm_slot() ASoC: fsl_ssi: Maintain a mask of active streams ASoC: fsl_ssi: Rename fsl_ssi_disable_val macro ASoC: fsl_ssi: Clear FIFO directly in fsl_ssi_config() ASoC: fsl_ssi: Clean up helper functions of trigger() ASoC: fsl_ssi: Add DAIFMT define for AC97 ASoC: fsl_ssi: Clean up fsl_ssi_setup_regvals() ASoC: fsl_ssi: Set xFEN0 and xFEN1 together ASoC: fsl_ssi: Use snd_soc_init_dma_data instead ASoC: fsl_ssi: Move one-time configurations to probe() ASoC: fsl_ssi: Setup AC97 in fsl_ssi_hw_init() ASoC: fsl_ssi: Clean up _fsl_ssi_set_dai_fmt() ASoC: fsl_ssi: Add bool synchronous to mark synchronous mode ASoC: fsl_ssi: Move DT related code to a separate probe() ASoC: fsl_ssi: Use ssi->streams instead of reading register Nikita Yushchenko (1): ASoC: rsnd: set pm_ops in hibernate-compatible way Olivier Moysan (4): ASoC: dmaengine_pcm: add processing support ASoC: stm32: Add S/PDIF to SAI bindings ASoC: stm32: sai: Add support of S/PDIF playback ASoC: dmaengine_pcm: document process callback Pawse, GuruprasadX (2): ASoC: Intel: Skylake: Add BE DAI for WoV usecase ASoC: Intel: board: Add BE DAI link for WoV and update DAPM machine map. Peng Donglin (1): ASoC: use DEFINE_SHOW_ATTRIBUTE() to decrease code duplication Peter Ujfalusi (6): ASoC: tlv320aic31xx: Fix master mode clock I2S bus clocks ASoC: tlv320aic31xx: Rename AIF_IN from 'DAC IN' to 'AIF IN' ASoC: tlv320aic31xx: Do not force power on the DAC/ADC in clock master mode ASoC: hdmi-codec: Fix module unloading caused kernel crash ASoC: twl6040: Add back missing write callback ASoC: soc-io: Fix snd_soc_component_update_bits_legacy Ranjani Sridharan (1): ASoC: topology: create TLV data for dapm widgets Richard Fitzgerald (2): ASoC: wm_adsp: Don't init cache from DSP memory if control is write-only ASoC: wm_adsp: Fix some signedness errors in register access Ryder Lee (1): ASoC: mediatek: remove "simple-mfd" in the example Sanyog Kale (1): ASoC: Intel: Skylake: Disable clock and power gating during FW/LIB download Sebastian Reichel (1): ASoC: cpcap: new codec Srinivas Kandagatla (1): ASoC: dapm: add support to pinctrl dapm Sriram Periyasamy (1): ASoC: Intel: Skylake: Find module id from UUID for bind params Steven Eckhoff (1): ASoC: TSCS42xx: Add missing headers Sylwester Nawrocki (19): ASoC: samsung: Add the DT binding files entry to MAINTAINERS ASoC: samsung: i2s: Ensure the RCLK rate is properly determined ASoC: samsung: i2s: Update clock-output-names property documentation ASoC: samsung: i2s: Ensure names of supplied clocks are unique ASoC: samsung: i2s: Prevent external abort on exynos5433 I2S1 access ASoC: samsung: i2s: Define the parameters list for SAMSUNG_I2S_OPCLK ASoC: samsung,tm2-audio DT binding documentation update ASoC: samsung: Add support for HDMI audio on TM2 board ASoC: samsung: Add missing #sound-dai-cells property documentation ASoC: Use proper DT compatible string for Hardkernel Odroid boards ASoC: max98090: Add #sound-dai-cells property documentation ASoC: Add snd_soc_of_put_dai_link_codecs() helper function ASoC: samsung: i2s: Fix rclk_srcrate handling ASoC: samsung: Drop uneeded RCLKSRC setting in the Snow driver ASoC: samsung: Snow DT bindings update ASoC: samsung: Add HDMI audio support for Snow ASoC: samsung: odroid: Fix 32000 sample rate handling ASoC: samsung: odroid: Drop sample rates that cannot be supported from hw_params() ASoC: samsung: Use snd_soc_of_put_dai_link_codecs() in odroid.c Vijendar Mukunda (1): ASoC: amd: modifications in dma stop sequence Wei Yongjun (1): ASoC: mediatek: mt2701: drop unnessary snd_soc_unregister_component() Wu Fengguang (2): ASoC: uniphier: aiodma_rb_get_rp() can be static ASoC: rt5659: rt5659_intel_hd_header_probe_setup() can be static kbuild test robot (1): ASoC: ak4458: ak4458_regmap can be static oder_chiou(a)realtek.com (2): ASoC: rt5659: Add the support of Intel HDA Header ASoC: rt5659: Remove the routing path of the power widget "LDO2" olivier moysan (2): ASoC: stm32: spdifrx: fix typo in function name. ASoC: stm32: spdifrx: Use default dai name weiyongjun (A) (2): ASoC: wm8400: Use devm_snd_soc_register_component() ASoC: da7210: Use devm_snd_soc_register_component() Documentation/devicetree/bindings/sound/ak4458.txt | 23 + Documentation/devicetree/bindings/sound/ak5558.txt | 22 + Documentation/devicetree/bindings/sound/da7219.txt | 6 + Documentation/devicetree/bindings/sound/dmic.txt | 2 + .../devicetree/bindings/sound/fsl-asoc-card.txt | 1 - .../devicetree/bindings/sound/imx-audio-wm8962.txt | 53 - .../devicetree/bindings/sound/max98090.txt | 2 + .../devicetree/bindings/sound/maxim,max9759.txt | 18 + .../devicetree/bindings/sound/mt2701-afe-pcm.txt | 2 +- .../devicetree/bindings/sound/pcm1789.txt | 22 + .../devicetree/bindings/sound/renesas,rsnd.txt | 1 + .../bindings/sound/rockchip,rk3288-hdmi-analog.txt | 2 +- .../devicetree/bindings/sound/rohm,bd28623.txt | 29 + Documentation/devicetree/bindings/sound/rt5651.txt | 17 + Documentation/devicetree/bindings/sound/rt5665.txt | 4 +- .../devicetree/bindings/sound/samsung,odroid.txt | 8 +- .../bindings/sound/samsung,tm2-audio.txt | 14 +- .../devicetree/bindings/sound/samsung-i2s.txt | 22 +- .../devicetree/bindings/sound/sgtl5000.txt | 3 + Documentation/devicetree/bindings/sound/snow.txt | 13 +- .../devicetree/bindings/sound/st,stm32-sai.txt | 6 + .../devicetree/bindings/sound/tda7419.txt | 38 + .../devicetree/bindings/sound/uniphier,aio.txt | 45 + Documentation/devicetree/bindings/sound/wm8524.txt | 2 +- MAINTAINERS | 7 - arch/x86/include/asm/platform_sst_audio.h | 1 + drivers/gpu/drm/vc4/vc4_hdmi.c | 47 +- include/dt-bindings/sound/rt5651.h | 15 + include/sound/da7219.h | 2 + include/sound/dmaengine_pcm.h | 7 + include/sound/hdaudio.h | 2 + include/sound/rt5651.h | 29 - include/sound/rt5659.h | 1 + include/sound/soc-dapm.h | 16 + include/sound/soc.h | 14 +- include/uapi/sound/asound.h | 1 + sound/hda/hdac_device.c | 35 + sound/pci/hda/hda_codec.c | 28 +- sound/pci/hda/hda_local.h | 6 +- sound/soc/Kconfig | 1 - sound/soc/Makefile | 1 - sound/soc/amd/Kconfig | 9 + sound/soc/amd/Makefile | 2 + sound/soc/amd/acp-da7219-max98357a.c | 276 ++++ sound/soc/amd/acp-pcm-dma.c | 206 +-- sound/soc/amd/acp-rt5645.c | 4 +- sound/soc/amd/acp.h | 8 +- sound/soc/atmel/atmel-classd.c | 65 +- sound/soc/atmel/atmel-pcm-pdc.c | 6 +- sound/soc/atmel/atmel-pdmic.c | 67 +- sound/soc/au1x/dbdma2.c | 13 +- sound/soc/au1x/dma.c | 12 +- sound/soc/bcm/cygnus-pcm.c | 7 +- sound/soc/bcm/cygnus-ssp.c | 2 +- sound/soc/blackfin/Kconfig | 205 --- sound/soc/blackfin/Makefile | 40 - sound/soc/blackfin/bf5xx-ac97-pcm.c | 480 ------ sound/soc/blackfin/bf5xx-ac97.c | 388 ----- sound/soc/blackfin/bf5xx-ac97.h | 57 - sound/soc/blackfin/bf5xx-ad1836.c | 109 -- sound/soc/blackfin/bf5xx-ad193x.c | 131 -- sound/soc/blackfin/bf5xx-ad1980.c | 109 -- sound/soc/blackfin/bf5xx-ad73311.c | 212 --- sound/soc/blackfin/bf5xx-i2s-pcm.c | 373 ----- sound/soc/blackfin/bf5xx-i2s-pcm.h | 17 - sound/soc/blackfin/bf5xx-i2s.c | 391 ----- sound/soc/blackfin/bf5xx-sport.c | 1102 -------------- sound/soc/blackfin/bf5xx-sport.h | 174 --- sound/soc/blackfin/bf5xx-ssm2602.c | 126 -- sound/soc/blackfin/bf6xx-i2s.c | 239 --- sound/soc/blackfin/bf6xx-sport.c | 425 ------ sound/soc/blackfin/bf6xx-sport.h | 82 - sound/soc/blackfin/bfin-eval-adau1373.c | 173 --- sound/soc/blackfin/bfin-eval-adau1701.c | 113 -- sound/soc/blackfin/bfin-eval-adau1x61.c | 142 -- sound/soc/blackfin/bfin-eval-adau1x81.c | 129 -- sound/soc/blackfin/bfin-eval-adav80x.c | 145 -- sound/soc/codecs/88pm860x-codec.c | 166 +-- sound/soc/codecs/88pm860x-codec.h | 4 +- sound/soc/codecs/Kconfig | 50 +- sound/soc/codecs/Makefile | 17 +- sound/soc/codecs/ab8500-codec.c | 295 ++-- sound/soc/codecs/ac97.c | 46 +- sound/soc/codecs/ad1836.c | 68 +- sound/soc/codecs/ad193x-i2c.c | 7 - sound/soc/codecs/ad193x-spi.c | 7 - sound/soc/codecs/ad193x.c | 46 +- sound/soc/codecs/ad1980.c | 87 +- sound/soc/codecs/ad73311.c | 27 +- sound/soc/codecs/adau1373.c | 84 +- sound/soc/codecs/adau1701.c | 101 +- sound/soc/codecs/adau1761.c | 88 +- sound/soc/codecs/adau1781.c | 51 +- sound/soc/codecs/adau17x1.c | 69 +- sound/soc/codecs/adau17x1.h | 8 +- sound/soc/codecs/adau1977-i2c.c | 7 - sound/soc/codecs/adau1977-spi.c | 7 - sound/soc/codecs/adau1977.c | 59 +- sound/soc/codecs/adau7002.c | 30 +- sound/soc/codecs/adav801.c | 7 - sound/soc/codecs/adav803.c | 7 - sound/soc/codecs/adav80x.c | 130 +- sound/soc/codecs/ads117x.c | 27 +- sound/soc/codecs/ak4104.c | 68 +- sound/soc/codecs/ak4458.c | 657 ++++++++ sound/soc/codecs/ak4458.h | 86 ++ sound/soc/codecs/ak4535.c | 78 +- sound/soc/codecs/ak4554.c | 27 +- sound/soc/codecs/ak4613.c | 72 +- sound/soc/codecs/ak4641.c | 91 +- sound/soc/codecs/ak4642.c | 104 +- sound/soc/codecs/ak4671.c | 67 +- sound/soc/codecs/ak5386.c | 62 +- sound/soc/codecs/ak5558.c | 415 ++++++ sound/soc/codecs/ak5558.h | 52 + sound/soc/codecs/alc5623.c | 147 +- sound/soc/codecs/alc5632.c | 138 +- sound/soc/codecs/arizona.c | 249 ++-- sound/soc/codecs/arizona.h | 24 +- sound/soc/codecs/bd28623.c | 242 +++ sound/soc/codecs/bt-sco.c | 21 +- sound/soc/codecs/cpcap.c | 1562 ++++++++++++++++++++ sound/soc/codecs/cq93vc.c | 33 +- sound/soc/codecs/cs35l32.c | 45 +- sound/soc/codecs/cs35l33.c | 186 ++- sound/soc/codecs/cs35l34.c | 127 +- sound/soc/codecs/cs35l35.c | 104 +- sound/soc/codecs/cs4265.c | 98 +- sound/soc/codecs/cs4270.c | 124 +- sound/soc/codecs/cs4271-i2c.c | 7 - sound/soc/codecs/cs4271-spi.c | 7 - sound/soc/codecs/cs4271.c | 105 +- sound/soc/codecs/cs42l42.c | 154 +- sound/soc/codecs/cs42l42.h | 2 +- sound/soc/codecs/cs42l51-i2c.c | 8 - sound/soc/codecs/cs42l51.c | 81 +- sound/soc/codecs/cs42l52.c | 146 +- sound/soc/codecs/cs42l56.c | 158 +- sound/soc/codecs/cs42l73.c | 122 +- sound/soc/codecs/cs42xx8-i2c.c | 1 - sound/soc/codecs/cs42xx8.c | 57 +- sound/soc/codecs/cs43130.c | 243 +-- sound/soc/codecs/cs43130.h | 2 +- sound/soc/codecs/cs4349.c | 39 +- sound/soc/codecs/cs47l24.c | 136 +- sound/soc/codecs/cs53l30.c | 57 +- sound/soc/codecs/cx20442.c | 89 +- sound/soc/codecs/da7210.c | 181 ++- sound/soc/codecs/da7213.c | 328 ++-- sound/soc/codecs/da7218.c | 498 +++---- sound/soc/codecs/da7218.h | 2 +- sound/soc/codecs/da7219-aad.c | 236 +-- sound/soc/codecs/da7219-aad.h | 12 +- sound/soc/codecs/da7219.c | 434 ++++-- sound/soc/codecs/da7219.h | 11 +- sound/soc/codecs/da732x.c | 241 +-- sound/soc/codecs/da9055.c | 186 ++- sound/soc/codecs/dmic.c | 92 +- sound/soc/codecs/es7134.c | 27 +- sound/soc/codecs/es8316.c | 80 +- sound/soc/codecs/es8328-i2c.c | 7 - sound/soc/codecs/es8328-spi.c | 7 - sound/soc/codecs/es8328.c | 147 +- sound/soc/codecs/gtm601.c | 28 +- sound/soc/codecs/hdac_hdmi.c | 65 +- sound/soc/codecs/hdac_hdmi.h | 2 +- sound/soc/codecs/hdmi-codec.c | 30 +- sound/soc/codecs/ics43432.c | 16 +- sound/soc/codecs/inno_rk3036.c | 70 +- sound/soc/codecs/isabelle.c | 57 +- sound/soc/codecs/jz4740.c | 50 +- sound/soc/codecs/lm49453.c | 65 +- sound/soc/codecs/max9759.c | 207 +++ sound/soc/codecs/max98088.c | 314 ++-- sound/soc/codecs/max98090.c | 319 ++-- sound/soc/codecs/max98090.h | 4 +- sound/soc/codecs/max98095.c | 448 +++--- sound/soc/codecs/max98095.h | 2 +- sound/soc/codecs/max98357a.c | 29 +- sound/soc/codecs/max98371.c | 43 +- sound/soc/codecs/max98371.h | 1 - sound/soc/codecs/max98373.c | 75 +- sound/soc/codecs/max9850.c | 80 +- sound/soc/codecs/max9860.c | 70 +- sound/soc/codecs/max9867.c | 91 +- sound/soc/codecs/max9867.h | 1 - sound/soc/codecs/max98925.c | 72 +- sound/soc/codecs/max98925.h | 2 +- sound/soc/codecs/max98926.c | 56 +- sound/soc/codecs/max98926.h | 2 +- sound/soc/codecs/max98927.c | 82 +- sound/soc/codecs/max98927.h | 2 +- sound/soc/codecs/mc13783.c | 68 +- sound/soc/codecs/ml26124.c | 103 +- sound/soc/codecs/msm8916-wcd-analog.c | 207 +-- sound/soc/codecs/msm8916-wcd-digital.c | 115 +- sound/soc/codecs/nau8540.c | 79 +- sound/soc/codecs/nau8810.c | 73 +- sound/soc/codecs/nau8824.c | 153 +- sound/soc/codecs/nau8824.h | 2 +- sound/soc/codecs/nau8825.c | 107 +- sound/soc/codecs/nau8825.h | 2 +- sound/soc/codecs/pcm1681.c | 62 +- sound/soc/codecs/pcm1789-i2c.c | 62 + sound/soc/codecs/pcm1789.c | 274 ++++ sound/soc/codecs/pcm1789.h | 17 + sound/soc/codecs/pcm179x-i2c.c | 6 - sound/soc/codecs/pcm179x-spi.c | 6 - sound/soc/codecs/pcm179x.c | 45 +- sound/soc/codecs/pcm179x.h | 1 - sound/soc/codecs/pcm186x-i2c.c | 8 - sound/soc/codecs/pcm186x-spi.c | 8 - sound/soc/codecs/pcm186x.c | 143 +- sound/soc/codecs/pcm186x.h | 1 - sound/soc/codecs/pcm3008.c | 36 +- sound/soc/codecs/pcm3168a.c | 51 +- sound/soc/codecs/pcm5102a.c | 16 +- sound/soc/codecs/pcm512x.c | 162 +- sound/soc/codecs/rt274.c | 206 ++- sound/soc/codecs/rt286.c | 205 ++- sound/soc/codecs/rt286.h | 2 +- sound/soc/codecs/rt298.c | 217 ++- sound/soc/codecs/rt298.h | 2 +- sound/soc/codecs/rt5514-spi.c | 36 +- sound/soc/codecs/rt5514.c | 128 +- sound/soc/codecs/rt5514.h | 2 +- sound/soc/codecs/rt5616.c | 219 ++- sound/soc/codecs/rt5631.c | 324 ++-- sound/soc/codecs/rt5640.c | 260 ++-- sound/soc/codecs/rt5640.h | 6 +- sound/soc/codecs/rt5645.c | 370 +++-- sound/soc/codecs/rt5645.h | 4 +- sound/soc/codecs/rt5651.c | 752 +++++----- sound/soc/codecs/rt5651.h | 23 +- sound/soc/codecs/rt5659.c | 472 +++--- sound/soc/codecs/rt5659.h | 21 +- sound/soc/codecs/rt5660.c | 145 +- sound/soc/codecs/rt5660.h | 2 +- sound/soc/codecs/rt5663.c | 600 ++++---- sound/soc/codecs/rt5663.h | 4 +- sound/soc/codecs/rt5665.c | 350 +++-- sound/soc/codecs/rt5665.h | 3 +- sound/soc/codecs/rt5670.c | 295 ++-- sound/soc/codecs/rt5670.h | 10 +- sound/soc/codecs/rt5677.c | 205 ++- sound/soc/codecs/rt5677.h | 4 +- sound/soc/codecs/sgtl5000.c | 275 ++-- sound/soc/codecs/sgtl5000.h | 7 + sound/soc/codecs/si476x.c | 42 +- sound/soc/codecs/sirf-audio-codec.c | 58 +- sound/soc/codecs/spdif_receiver.c | 26 +- sound/soc/codecs/spdif_transmitter.c | 26 +- sound/soc/codecs/ssm2518.c | 57 +- sound/soc/codecs/ssm2602-i2c.c | 7 - sound/soc/codecs/ssm2602-spi.c | 7 - sound/soc/codecs/ssm2602.c | 96 +- sound/soc/codecs/ssm4567.c | 44 +- sound/soc/codecs/sta32x.c | 134 +- sound/soc/codecs/sta350.c | 117 +- sound/soc/codecs/sta529.c | 69 +- sound/soc/codecs/stac9766.c | 77 +- sound/soc/codecs/sti-sas.c | 84 +- sound/soc/codecs/tas2552.c | 178 ++- sound/soc/codecs/tas5086.c | 94 +- sound/soc/codecs/tas571x.c | 55 +- sound/soc/codecs/tas5720.c | 151 +- sound/soc/codecs/tas6424.c | 99 +- sound/soc/codecs/tda7419.c | 654 ++++++++ sound/soc/codecs/tfa9879.c | 50 +- sound/soc/codecs/tlv320aic23-i2c.c | 7 - sound/soc/codecs/tlv320aic23-spi.c | 7 - sound/soc/codecs/tlv320aic23.c | 120 +- sound/soc/codecs/tlv320aic26.c | 83 +- sound/soc/codecs/tlv320aic31xx.c | 339 +++-- sound/soc/codecs/tlv320aic31xx.h | 1 + sound/soc/codecs/tlv320aic32x4.c | 207 ++- sound/soc/codecs/tlv320aic3x.c | 295 ++-- sound/soc/codecs/tlv320dac33.c | 402 +++-- sound/soc/codecs/tscs42xx.c | 244 +-- sound/soc/codecs/twl4030.c | 383 +++-- sound/soc/codecs/twl6040.c | 269 ++-- sound/soc/codecs/twl6040.h | 10 +- sound/soc/codecs/uda134x.c | 78 +- sound/soc/codecs/uda1380.c | 160 +- sound/soc/codecs/wl1273.c | 93 +- sound/soc/codecs/wl1273.h | 2 +- sound/soc/codecs/wm0010.c | 181 ++- sound/soc/codecs/wm1250-ev1.c | 27 +- sound/soc/codecs/wm2000.c | 78 +- sound/soc/codecs/wm2200.c | 157 +- sound/soc/codecs/wm5100.c | 260 ++-- sound/soc/codecs/wm5100.h | 2 +- sound/soc/codecs/wm5102.c | 154 +- sound/soc/codecs/wm5110.c | 222 ++- sound/soc/codecs/wm8350.c | 183 ++- sound/soc/codecs/wm8350.h | 4 +- sound/soc/codecs/wm8400.c | 211 ++- sound/soc/codecs/wm8510.c | 149 +- sound/soc/codecs/wm8523.c | 98 +- sound/soc/codecs/wm8524.c | 56 +- sound/soc/codecs/wm8580.c | 150 +- sound/soc/codecs/wm8711.c | 116 +- sound/soc/codecs/wm8727.c | 29 +- sound/soc/codecs/wm8728.c | 82 +- sound/soc/codecs/wm8731.c | 101 +- sound/soc/codecs/wm8737.c | 109 +- sound/soc/codecs/wm8741.c | 133 +- sound/soc/codecs/wm8750.c | 115 +- sound/soc/codecs/wm8753.c | 289 ++-- sound/soc/codecs/wm8770.c | 131 +- sound/soc/codecs/wm8776.c | 105 +- sound/soc/codecs/wm8782.c | 27 +- sound/soc/codecs/wm8804.c | 100 +- sound/soc/codecs/wm8900.c | 251 ++-- sound/soc/codecs/wm8903.c | 211 ++- sound/soc/codecs/wm8903.h | 2 +- sound/soc/codecs/wm8904.c | 355 +++-- sound/soc/codecs/wm8940.c | 159 +- sound/soc/codecs/wm8955.c | 189 ++- sound/soc/codecs/wm8958-dsp2.c | 278 ++-- sound/soc/codecs/wm8960.c | 249 ++-- sound/soc/codecs/wm8961.c | 241 ++- sound/soc/codecs/wm8962.c | 397 +++-- sound/soc/codecs/wm8962.h | 2 +- sound/soc/codecs/wm8971.c | 108 +- sound/soc/codecs/wm8974.c | 143 +- sound/soc/codecs/wm8978.c | 185 ++- sound/soc/codecs/wm8983.c | 170 +-- sound/soc/codecs/wm8985.c | 201 ++- sound/soc/codecs/wm8988.c | 127 +- sound/soc/codecs/wm8990.c | 179 ++- sound/soc/codecs/wm8991.c | 167 +-- sound/soc/codecs/wm8993.c | 237 +-- sound/soc/codecs/wm8994.c | 810 +++++----- sound/soc/codecs/wm8994.h | 8 +- sound/soc/codecs/wm8995.c | 286 ++-- sound/soc/codecs/wm8996.c | 401 +++-- sound/soc/codecs/wm8996.h | 4 +- sound/soc/codecs/wm8997.c | 72 +- sound/soc/codecs/wm8998.c | 115 +- sound/soc/codecs/wm9081.c | 228 ++- sound/soc/codecs/wm9090.c | 109 +- sound/soc/codecs/wm9705.c | 83 +- sound/soc/codecs/wm9712.c | 111 +- sound/soc/codecs/wm9713.c | 203 ++- sound/soc/codecs/wm_adsp.c | 99 +- sound/soc/codecs/wm_adsp.h | 14 +- sound/soc/codecs/wm_hubs.c | 254 ++-- sound/soc/codecs/wm_hubs.h | 18 +- sound/soc/codecs/zx_aud96p22.c | 35 +- sound/soc/davinci/davinci-i2s.c | 13 +- sound/soc/dwc/dwc-i2s.c | 4 + sound/soc/dwc/dwc-pcm.c | 5 +- sound/soc/fsl/Kconfig | 11 - sound/soc/fsl/Makefile | 2 - sound/soc/fsl/fsl-asoc-card.c | 6 +- sound/soc/fsl/fsl_asrc.c | 10 - sound/soc/fsl/fsl_asrc.h | 3 +- sound/soc/fsl/fsl_asrc_dma.c | 14 +- sound/soc/fsl/fsl_dma.c | 27 +- sound/soc/fsl/fsl_esai.c | 2 +- sound/soc/fsl/fsl_ssi.c | 758 +++++----- sound/soc/fsl/fsl_ssi.h | 3 - sound/soc/fsl/fsl_utils.c | 2 +- sound/soc/fsl/imx-pcm-fiq.c | 6 +- sound/soc/fsl/imx-wm8962.c | 312 ---- sound/soc/fsl/mpc5200_dma.c | 16 +- sound/soc/fsl/wm1133-ev1.c | 6 +- sound/soc/img/pistachio-internal-dac.c | 34 +- sound/soc/intel/atom/sst-atom-controls.c | 36 +- sound/soc/intel/atom/sst-mfld-platform-compress.c | 4 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 32 +- sound/soc/intel/atom/sst-mfld-platform.h | 4 +- sound/soc/intel/atom/sst/sst.c | 24 +- sound/soc/intel/atom/sst/sst.h | 20 +- sound/soc/intel/atom/sst/sst_acpi.c | 3 +- sound/soc/intel/atom/sst/sst_drv_interface.c | 19 - sound/soc/intel/atom/sst/sst_pvt.c | 8 - sound/soc/intel/atom/sst/sst_stream.c | 117 +- sound/soc/intel/baytrail/sst-baytrail-pcm.c | 66 +- sound/soc/intel/boards/Kconfig | 25 + sound/soc/intel/boards/Makefile | 4 + sound/soc/intel/boards/bdw-rt5677.c | 41 +- sound/soc/intel/boards/broadwell.c | 17 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 12 +- sound/soc/intel/boards/bxt_rt298.c | 27 +- sound/soc/intel/boards/byt-rt5640.c | 4 +- sound/soc/intel/boards/bytcr_rt5640.c | 8 +- sound/soc/intel/boards/bytcr_rt5651.c | 466 +++++- sound/soc/intel/boards/cht_bsw_nau8824.c | 282 ++++ sound/soc/intel/boards/cht_bsw_rt5645.c | 8 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 18 +- sound/soc/intel/boards/haswell.c | 3 +- sound/soc/intel/boards/kbl_da7219_max98357a.c | 613 ++++++++ sound/soc/intel/boards/kbl_rt5663_max98927.c | 19 +- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 14 +- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 12 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 12 +- sound/soc/intel/boards/skl_rt286.c | 12 +- sound/soc/intel/common/soc-acpi-intel-cht-match.c | 9 + sound/soc/intel/haswell/sst-haswell-ipc.h | 2 + sound/soc/intel/haswell/sst-haswell-pcm.c | 118 +- sound/soc/intel/skylake/skl-debug.c | 2 +- sound/soc/intel/skylake/skl-messages.c | 9 +- sound/soc/intel/skylake/skl-pcm.c | 72 +- sound/soc/intel/skylake/skl-ssp-clk.c | 4 +- sound/soc/intel/skylake/skl-sst-ipc.h | 3 + sound/soc/intel/skylake/skl-topology.c | 133 +- sound/soc/intel/skylake/skl-topology.h | 15 +- sound/soc/intel/skylake/skl.c | 40 + sound/soc/intel/skylake/skl.h | 6 +- sound/soc/kirkwood/kirkwood-dma.c | 3 +- sound/soc/kirkwood/kirkwood-i2s.c | 20 +- sound/soc/kirkwood/kirkwood.h | 4 +- sound/soc/mediatek/common/mtk-afe-fe-dai.c | 16 +- .../soc/mediatek/common/mtk-afe-platform-driver.c | 9 +- .../soc/mediatek/common/mtk-afe-platform-driver.h | 3 +- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 54 +- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 42 +- sound/soc/mediatek/mt8173/mt8173-max98090.c | 4 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c | 6 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 12 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 12 +- sound/soc/nuc900/nuc900-pcm.c | 5 +- sound/soc/omap/ams-delta.c | 20 +- sound/soc/omap/omap-abe-twl6040.c | 8 +- sound/soc/omap/omap-pcm.c | 5 +- sound/soc/pxa/mioa701_wm9713.c | 30 +- sound/soc/pxa/mmp-pcm.c | 11 +- sound/soc/pxa/mmp-sspa.c | 1 - sound/soc/pxa/pxa2xx-pcm.c | 5 +- sound/soc/pxa/ttc-dkb.c | 6 +- sound/soc/qcom/lpass-platform.c | 42 +- sound/soc/rockchip/Kconfig | 3 + sound/soc/rockchip/rk3288_hdmi_analog.c | 2 +- sound/soc/rockchip/rk3399_gru_sound.c | 4 +- sound/soc/rockchip/rockchip_rt5645.c | 8 +- sound/soc/samsung/Makefile | 3 - sound/soc/samsung/i2s-regs.h | 11 +- sound/soc/samsung/i2s.c | 52 +- sound/soc/samsung/i2s.h | 11 +- sound/soc/samsung/idma.c | 5 +- sound/soc/samsung/littlemill.c | 8 +- sound/soc/samsung/lowland.c | 12 +- sound/soc/samsung/odroid.c | 33 +- sound/soc/samsung/snow.c | 229 ++- sound/soc/samsung/speyside.c | 12 +- sound/soc/samsung/tm2_wm5110.c | 206 ++- sound/soc/samsung/tobermory.c | 6 +- sound/soc/sh/dma-sh7760.c | 5 +- sound/soc/sh/fsi.c | 24 +- sound/soc/sh/rcar/core.c | 36 +- sound/soc/sh/rcar/rsnd.h | 20 + sound/soc/sh/rcar/src.c | 22 +- sound/soc/sh/rcar/ssi.c | 16 +- sound/soc/sh/siu.h | 2 +- sound/soc/sh/siu_dai.c | 10 +- sound/soc/sh/siu_pcm.c | 9 +- sound/soc/sirf/sirf-usp.h | 2 +- sound/soc/soc-ac97.c | 84 +- sound/soc/soc-core.c | 117 +- sound/soc/soc-dapm.c | 46 +- sound/soc/soc-generic-dmaengine-pcm.c | 121 +- sound/soc/soc-io.c | 7 +- sound/soc/soc-pcm.c | 11 + sound/soc/soc-topology.c | 35 +- sound/soc/soc-utils.c | 28 +- sound/soc/stm/stm32_adfsdm.c | 7 +- sound/soc/stm/stm32_sai.c | 2 + sound/soc/stm/stm32_sai.h | 2 + sound/soc/stm/stm32_sai_sub.c | 153 +- sound/soc/stm/stm32_spdifrx.c | 7 +- sound/soc/sunxi/sun4i-codec.c | 71 +- sound/soc/sunxi/sun8i-codec.c | 23 +- sound/soc/tegra/tegra_wm8903.c | 8 +- sound/soc/txx9/txx9aclc.c | 26 +- sound/soc/uniphier/Kconfig | 33 + sound/soc/uniphier/Makefile | 8 + sound/soc/uniphier/aio-compress.c | 438 ++++++ sound/soc/uniphier/aio-core.c | 1138 ++++++++++++++ sound/soc/uniphier/aio-cpu.c | 578 ++++++++ sound/soc/uniphier/aio-dma.c | 319 ++++ sound/soc/uniphier/aio-ld11.c | 431 ++++++ sound/soc/uniphier/aio-pxs2.c | 320 ++++ sound/soc/uniphier/aio-reg.h | 465 ++++++ sound/soc/uniphier/aio.h | 359 +++++ sound/soc/uniphier/evea.c | 102 +- sound/soc/xtensa/xtfpga-i2s.c | 17 +- 488 files changed, 27614 insertions(+), 22906 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/ak4458.txt create mode 100644 Documentation/devicetree/bindings/sound/ak5558.txt delete mode 100644 Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt create mode 100644 Documentation/devicetree/bindings/sound/maxim,max9759.txt create mode 100644 Documentation/devicetree/bindings/sound/pcm1789.txt create mode 100644 Documentation/devicetree/bindings/sound/rohm,bd28623.txt create mode 100644 Documentation/devicetree/bindings/sound/tda7419.txt create mode 100644 Documentation/devicetree/bindings/sound/uniphier,aio.txt create mode 100644 include/dt-bindings/sound/rt5651.h delete mode 100644 include/sound/rt5651.h create mode 100644 sound/soc/amd/acp-da7219-max98357a.c delete mode 100644 sound/soc/blackfin/Kconfig delete mode 100644 sound/soc/blackfin/Makefile delete mode 100644 sound/soc/blackfin/bf5xx-ac97-pcm.c delete mode 100644 sound/soc/blackfin/bf5xx-ac97.c delete mode 100644 sound/soc/blackfin/bf5xx-ac97.h delete mode 100644 sound/soc/blackfin/bf5xx-ad1836.c delete mode 100644 sound/soc/blackfin/bf5xx-ad193x.c delete mode 100644 sound/soc/blackfin/bf5xx-ad1980.c delete mode 100644 sound/soc/blackfin/bf5xx-ad73311.c delete mode 100644 sound/soc/blackfin/bf5xx-i2s-pcm.c delete mode 100644 sound/soc/blackfin/bf5xx-i2s-pcm.h delete mode 100644 sound/soc/blackfin/bf5xx-i2s.c delete mode 100644 sound/soc/blackfin/bf5xx-sport.c delete mode 100644 sound/soc/blackfin/bf5xx-sport.h delete mode 100644 sound/soc/blackfin/bf5xx-ssm2602.c delete mode 100644 sound/soc/blackfin/bf6xx-i2s.c delete mode 100644 sound/soc/blackfin/bf6xx-sport.c delete mode 100644 sound/soc/blackfin/bf6xx-sport.h delete mode 100644 sound/soc/blackfin/bfin-eval-adau1373.c delete mode 100644 sound/soc/blackfin/bfin-eval-adau1701.c delete mode 100644 sound/soc/blackfin/bfin-eval-adau1x61.c delete mode 100644 sound/soc/blackfin/bfin-eval-adau1x81.c delete mode 100644 sound/soc/blackfin/bfin-eval-adav80x.c create mode 100644 sound/soc/codecs/ak4458.c create mode 100644 sound/soc/codecs/ak4458.h create mode 100644 sound/soc/codecs/ak5558.c create mode 100644 sound/soc/codecs/ak5558.h create mode 100644 sound/soc/codecs/bd28623.c create mode 100644 sound/soc/codecs/cpcap.c create mode 100644 sound/soc/codecs/max9759.c create mode 100644 sound/soc/codecs/pcm1789-i2c.c create mode 100644 sound/soc/codecs/pcm1789.c create mode 100644 sound/soc/codecs/pcm1789.h create mode 100644 sound/soc/codecs/tda7419.c delete mode 100644 sound/soc/fsl/imx-wm8962.c create mode 100644 sound/soc/intel/boards/cht_bsw_nau8824.c create mode 100644 sound/soc/intel/boards/kbl_da7219_max98357a.c create mode 100644 sound/soc/uniphier/aio-compress.c create mode 100644 sound/soc/uniphier/aio-core.c create mode 100644 sound/soc/uniphier/aio-cpu.c create mode 100644 sound/soc/uniphier/aio-dma.c create mode 100644 sound/soc/uniphier/aio-ld11.c create mode 100644 sound/soc/uniphier/aio-pxs2.c create mode 100644 sound/soc/uniphier/aio-reg.h create mode 100644 sound/soc/uniphier/aio.h

2 1

[alsa-devel] [PATCH 1/1] USB-Audio: Add second S/PDIF device on Phiree U2SX
by bruno.n.pagani＠gmail.com 02 Apr '18

02 Apr '18

From: Bruno Pagani <bruno.n.pagani(a)gmail.com> Phiree U2SX is the successor of Phiree U2 and has the same unusual configuration. See ea865bba4615d906144ae5d4f72a4aad2baffe1f for reference. Signed-off-by: Bruno Pagani <bruno.n.pagani(a)gmail.com> diff --git a/src/conf/cards/USB-Audio.conf b/src/conf/cards/USB-Audio.conf index cc8c718c..2b8f9a29 100644 --- a/src/conf/cards/USB-Audio.conf +++ b/src/conf/cards/USB-Audio.conf @@ -64,6 +64,7 @@ USB-Audio.pcm.iec958_device { # Second iec958 device number, if any. USB-Audio.pcm.iec958_2_device { "PHIREE U2" 1 # 0 = PCM S/PDIF, 1 = non-PCM S/PDIF + "PHIREE U2SX" 1 # 0 = PCM S/PDIF, 1 = non-PCM S/PDIF } -- 2.16.3

2 1

[alsa-devel] [PATCH 0/8] Enablement of BT I2S controller instance for AMD APUs
by Vijendar Mukunda 02 Apr '18

02 Apr '18

This patch set updates Audio CoProcessor (ACP) audio drivers to enable BT I2S controller instance. In Audio Coprocessor, There are three I2S controllers can be configured/enabled.(I2S SP, I2S MICSP, BT I2S) Default enabled I2S controller instance is I2S SP instance. There is a requirement to enable BT I2S controller Instance along with I2S SP controller instance. This patch set bring up functionality to render & capture on BT I2S controller instance. The current patch set also includes a new dependent patch. The current code is based on asoc-next, but I'm happy to rebase on whatever tree this ends up going through if there are any problems applying. Vijendar Mukunda (8): ASoC: dwc: I2S Controller instance param added ASoC: amd: dma driver changes for BT I2S controller instance ASoC: amd: dma descriptor changes for BT I2S Instance ASoC: amd: Interrupt handler changes for BT I2S instance ASoC: amd: prepare callback modifications for bt i2s instance ASoC: amd: pointer and trigger callback modifications for bt i2s ASoC: amd: 16bit resolution support for bt i2s instance ASoC: amd: enabling bt i2s config after acp reset include/sound/designware_i2s.h | 6 + sound/soc/amd/acp-pcm-dma.c | 531 +++++++++++++++++++++++++++++++++-------- sound/soc/amd/acp.h | 50 +++- sound/soc/dwc/dwc-i2s.c | 1 + sound/soc/dwc/local.h | 1 + 5 files changed, 481 insertions(+), 108 deletions(-) -- 2.7.4

3 11

[alsa-devel] [PATCH v4] ASoC: AMD: Enable/Disable auxiliary clock via common clock framework
by Akshu Agrawal 30 Mar '18

30 Mar '18

This enables/disables and sets auxiliary clock at 25Mhz. It uses common clock framework for proper ref counting. This approach will save power in comparison to keeping it always On in firmware. V2: Correcting the pin to OSCOUT1 from OSCOUT2 V3: Fix error/warnings from kbuild test V4: Fix build errors for platform which do not support CONFIG_COMMON_CLK TEST= aplay -vv <file> check register to see clock enabled kill aplay check register to see clock disabled Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> Acked-by: Alex Deucher <alexander.deucher(a)amd.com> --- sound/soc/amd/acp-da7219-max98357a.c | 144 ++++++++++++++++++++++++++++++++++- 1 file changed, 142 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 99c6b5c..3c77803 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -30,10 +30,13 @@ #include <sound/soc-dapm.h> #include <sound/jack.h> #include <linux/clk.h> +#include <linux/clkdev.h> +#include <linux/clk-provider.h> #include <linux/gpio.h> #include <linux/module.h> #include <linux/i2c.h> #include <linux/acpi.h> +#include <linux/types.h> #include "../codecs/da7219.h" #include "../codecs/da7219-aad.h" @@ -41,6 +44,20 @@ #define CZ_PLAT_CLK 24000000 #define MCLK_RATE 24576000 #define DUAL_CHANNEL 2 +#define CLKDRVSTR2 0x28 +#define MISCCLKCNTL1 0x40 +#define OSCCLKENB 2 +#define OSCOUT1CLK25MHZ 16 + +struct cz_clock { + const char* acp_clks_name; +#ifdef CONFIG_COMMON_CLK + struct clk_hw acp_clks_hw; +#endif + struct clk_lookup *acp_clks_lookup; + struct clk *acp_clks; + void __iomem *res_base; +}; static struct snd_soc_jack cz_jack; struct clk *da7219_dai_clk; @@ -91,6 +108,8 @@ static int cz_da7219_hw_params(struct snd_pcm_substream *substream, { int ret = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct clk *acpd7219_clk; ret = clk_prepare_enable(da7219_dai_clk); if (ret < 0) { @@ -98,13 +117,27 @@ static int cz_da7219_hw_params(struct snd_pcm_substream *substream, return ret; } + acpd7219_clk = clk_get(card->dev, "acpd7219-clks"); + ret = clk_prepare_enable(acpd7219_clk); + if (ret < 0) { + dev_err(rtd->dev, "can't enable oscillator clock %d\n", ret); + return ret; + } + return ret; } static int cz_da7219_hw_free(struct snd_pcm_substream *substream) { + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct clk *acpd7219_clk; + clk_disable_unprepare(da7219_dai_clk); + acpd7219_clk = clk_get(card->dev, "acpd7219-clks"); + clk_disable_unprepare(acpd7219_clk); + return 0; } @@ -237,9 +270,113 @@ static int cz_fe_startup(struct snd_pcm_substream *substream) .num_controls = ARRAY_SIZE(cz_mc_controls), }; +#ifdef CONFIG_COMMON_CLK +static int acpd7219_clks_prepare(struct clk_hw *hw) +{ + u32 reg_val; + struct cz_clock *cz_clock_obj = + container_of(hw, struct cz_clock, acp_clks_hw); + void __iomem *base = cz_clock_obj->res_base; + + reg_val = readl(base + MISCCLKCNTL1); + reg_val &= ~(0x1 << OSCCLKENB); + writel(reg_val, base + MISCCLKCNTL1); + reg_val = readl(base + CLKDRVSTR2); + reg_val |= (0x1 << OSCOUT1CLK25MHZ); + writel(reg_val, base + CLKDRVSTR2); + + return 0; +} + +static void acpd7219_clks_unprepare(struct clk_hw *hw) +{ + u32 reg_val; + struct cz_clock *cz_clock_obj = + container_of(hw, struct cz_clock, acp_clks_hw); + void __iomem *base = cz_clock_obj->res_base; + + reg_val = readl(base + MISCCLKCNTL1); + reg_val |= (0x1 << OSCCLKENB); + writel(reg_val, base + MISCCLKCNTL1); +} + +static int acpd7219_clks_is_enabled(struct clk_hw *hw) +{ + u32 reg_val; + struct cz_clock *cz_clock_obj = + container_of(hw, struct cz_clock, acp_clks_hw); + void __iomem *base = cz_clock_obj->res_base; + + reg_val = readl(base + MISCCLKCNTL1); + + return !(reg_val & OSCCLKENB); +} + +const struct clk_ops acpd7219_clks_ops = { + .prepare = acpd7219_clks_prepare, + .unprepare = acpd7219_clks_unprepare, + .is_enabled = acpd7219_clks_is_enabled, +}; + +static int register_acpd7219_clocks(struct platform_device *pdev) +{ + struct clk_init_data init = {}; + struct clk *acp_clks; + struct clk_lookup *acp_clks_lookup; + struct cz_clock *cz_clock_obj; + struct resource *res; + struct device dev = pdev->dev; + + cz_clock_obj = kzalloc(sizeof(struct cz_clock), GFP_KERNEL); + if (!cz_clock_obj) + return -ENOMEM; + + cz_clock_obj->acp_clks_name = "acpd7219-clks"; + + init.parent_names = NULL; + init.num_parents = 0; + init.name = cz_clock_obj->acp_clks_name; + init.ops = &acpd7219_clks_ops; + cz_clock_obj->acp_clks_hw.init = &init; + + acp_clks = devm_clk_register(&dev, &cz_clock_obj->acp_clks_hw); + if (IS_ERR(acp_clks)) + { + dev_err(&dev, "Failed to register DAI clocks: %ld\n", + PTR_ERR(acp_clks)); + return -EINVAL; + } + cz_clock_obj->acp_clks = acp_clks; + + acp_clks_lookup = clkdev_create(acp_clks, cz_clock_obj->acp_clks_name, + "%s", dev_name(&dev)); + if (!acp_clks_lookup) + dev_warn(&dev, "Failed to create DAI clkdev"); + else + cz_clock_obj->acp_clks_lookup = acp_clks_lookup; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(&pdev->dev, "Failed to get misc io resource.\n"); + return -EINVAL; + } + cz_clock_obj->res_base = devm_ioremap_nocache(&pdev->dev, res->start, + resource_size(res)); + if (!cz_clock_obj->res_base) + return -ENOMEM; + + return 0; +} +#else +static int register_acpd7219_clocks(struct platform_device *pdev) +{ + return 0; +} +#endif /* CONFIG_COMMON_CLK */ + static int cz_probe(struct platform_device *pdev) { - int ret; + int ret = 0; struct snd_soc_card *card; card = &cz_card; @@ -252,7 +389,10 @@ static int cz_probe(struct platform_device *pdev) cz_card.name, ret); return ret; } - return 0; + + ret = register_acpd7219_clocks(pdev); + + return ret; } static const struct acpi_device_id cz_audio_acpi_match[] = { -- 1.9.1

4 4

[alsa-devel] Regression? No sound with HDMI audio (snd-hda-intel) with kernel 4.15(.*)
by VDR User 30 Mar '18

30 Mar '18

Hi, I upgraded my Debian Sid box from kernel 4.14.31 to 4.15.14 and I no longer get sound over HDMI with my Nividia GT430. Nvidia driver version is 390.48. Neither the Nvidia installer nor dkms makes any difference. My alsa-utils is 1.1.3-1 and my ~/.asoundrc is: ~$ cat .asoundrc pcm.!default "hdmi:NVidia,1" ~$ aplay -l **** List of PLAYBACK Hardware Devices **** card 1: NVidia [HDA NVidia], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 7: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 8: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 9: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 I'm using 'speaker-test -c 2 -t wav -Dhw:CARD=1,7' for testing. After kernel 4.15.14 didn't work, I compiled kernel 4.15 and still got no audio. I then compiled kernel 4.15-rc9 and audio worked. From there I tried to bisect the exact patch that broke the audio but got the following: ~$ cd /usr/src /usr/src$ git clone https://github.com/torvalds/linux.git /usr/src$ cd linux /usr/src/linux$ git bisect start -- sound/hda sound/pci/hda /usr/src/linux$ git bisect good v4.15-rc9 /usr/src/linux$ git bisect bad v4.15 No testable commit found. Maybe you started with bad path parameters? /usr/src/linux$ I'm not sure what went wrong there, hopefully one of you reading this may have an idea. It's worth noting I upgraded another near-identical system and HDMI audio is working fine with 4.15.* on that box. That box has an Nvidia GT520 but other than that the package versions, driver version, etc. are all the same. Also, it's connected to a surround receiver whereas the non-working box is connected directly to a tv. Please let me know if any other info is needed or what I can do/try to debug this further. Any help is greatly appreciated! -Derek

1 0

Re: [alsa-devel] [PATCH] pcm_native: Remove VLA usage
by kbuild test robot 30 Mar '18

30 Mar '18

Hi Kyle, Thank you for the patch! Perhaps something to improve: [auto build test WARNING on v4.16-rc7] [cannot apply to sound/for-next next-20180329] [if your patch is applied to the wrong git tree, please drop us a note to help improve the system] url: https://github.com/0day-ci/linux/commits/Kyle-Spiers/pcm_native-Remove-VLA-… config: x86_64-randconfig-x019-201812 (attached as .config) compiler: gcc-7 (Debian 7.3.0-1) 7.3.0 reproduce: # save the attached .config to linux build tree make ARCH=x86_64 All warnings (new ones prefixed by >>): In file included from include/asm-generic/bug.h:5:0, from arch/x86/include/asm/bug.h:83, from include/linux/bug.h:5, from include/linux/mmdebug.h:5, from include/linux/mm.h:9, from sound/core/pcm_native.c:22: sound/core/pcm_native.c: In function 'constrain_params_by_rules': include/linux/compiler.h:58:2: warning: this 'if' clause does not guard... [-Wmisleading-indentation] if (__builtin_constant_p(!!(cond)) ? !!(cond) : \ ^ include/linux/compiler.h:56:23: note: in expansion of macro '__trace_if' #define if(cond, ...) __trace_if( (cond , ## __VA_ARGS__) ) ^~~~~~~~~~ >> sound/core/pcm_native.c:401:3: note: in expansion of macro 'if' if (changed < 0) ^~ sound/core/pcm_native.c:403:4: note: ...this statement, but the latter is misleadingly indented as if it were guarded by the 'if' return changed; ^~~~~~ vim +/if +401 sound/core/pcm_native.c ^1da177e4 Linus Torvalds 2005-04-16 319 9cc07f55d Takashi Sakamoto 2017-06-09 320 static int constrain_params_by_rules(struct snd_pcm_substream *substream, 877211f5e Takashi Iwai 2005-11-17 321 struct snd_pcm_hw_params *params) ^1da177e4 Linus Torvalds 2005-04-16 322 { 9cc07f55d Takashi Sakamoto 2017-06-09 323 struct snd_pcm_hw_constraints *constrs = 9cc07f55d Takashi Sakamoto 2017-06-09 324 &substream->runtime->hw_constraints; ^1da177e4 Linus Torvalds 2005-04-16 325 unsigned int k; 16c438ade Kyle Spiers 2018-03-28 326 unsigned int *rstamps; ^1da177e4 Linus Torvalds 2005-04-16 327 unsigned int vstamps[SNDRV_PCM_HW_PARAM_LAST_INTERVAL + 1]; d81052f92 Takashi Sakamoto 2017-06-09 328 unsigned int stamp; a1c06e39a Takashi Sakamoto 2017-06-09 329 struct snd_pcm_hw_rule *r; a1c06e39a Takashi Sakamoto 2017-06-09 330 unsigned int d; 9cc07f55d Takashi Sakamoto 2017-06-09 331 struct snd_mask old_mask; 9cc07f55d Takashi Sakamoto 2017-06-09 332 struct snd_interval old_interval; a1c06e39a Takashi Sakamoto 2017-06-09 333 bool again; 9cc07f55d Takashi Sakamoto 2017-06-09 334 int changed; ^1da177e4 Linus Torvalds 2005-04-16 335 d81052f92 Takashi Sakamoto 2017-06-09 336 /* d81052f92 Takashi Sakamoto 2017-06-09 337 * Each application of rule has own sequence number. d81052f92 Takashi Sakamoto 2017-06-09 338 * d81052f92 Takashi Sakamoto 2017-06-09 339 * Each member of 'rstamps' array represents the sequence number of d81052f92 Takashi Sakamoto 2017-06-09 340 * recent application of corresponding rule. d81052f92 Takashi Sakamoto 2017-06-09 341 */ 16c438ade Kyle Spiers 2018-03-28 342 16c438ade Kyle Spiers 2018-03-28 343 rstamps = kcalloc(constrs->rules_num, sizeof(*rstamps), GFP_KERNEL); d81052f92 Takashi Sakamoto 2017-06-09 344 d81052f92 Takashi Sakamoto 2017-06-09 345 /* d81052f92 Takashi Sakamoto 2017-06-09 346 * Each member of 'vstamps' array represents the sequence number of d81052f92 Takashi Sakamoto 2017-06-09 347 * recent application of rule in which corresponding parameters were d81052f92 Takashi Sakamoto 2017-06-09 348 * changed. d81052f92 Takashi Sakamoto 2017-06-09 349 * d81052f92 Takashi Sakamoto 2017-06-09 350 * In initial state, elements corresponding to parameters requested by d81052f92 Takashi Sakamoto 2017-06-09 351 * a caller is 1. For unrequested parameters, corresponding members d81052f92 Takashi Sakamoto 2017-06-09 352 * have 0 so that the parameters are never changed anymore. d81052f92 Takashi Sakamoto 2017-06-09 353 */ ^1da177e4 Linus Torvalds 2005-04-16 354 for (k = 0; k <= SNDRV_PCM_HW_PARAM_LAST_INTERVAL; k++) ^1da177e4 Linus Torvalds 2005-04-16 355 vstamps[k] = (params->rmask & (1 << k)) ? 1 : 0; d81052f92 Takashi Sakamoto 2017-06-09 356 d81052f92 Takashi Sakamoto 2017-06-09 357 /* Due to the above design, actual sequence number starts at 2. */ d81052f92 Takashi Sakamoto 2017-06-09 358 stamp = 2; 0d4e39996 Takashi Sakamoto 2017-06-09 359 retry: d81052f92 Takashi Sakamoto 2017-06-09 360 /* Apply all rules in order. */ a1c06e39a Takashi Sakamoto 2017-06-09 361 again = false; ^1da177e4 Linus Torvalds 2005-04-16 362 for (k = 0; k < constrs->rules_num; k++) { a1c06e39a Takashi Sakamoto 2017-06-09 363 r = &constrs->rules[k]; d81052f92 Takashi Sakamoto 2017-06-09 364 d81052f92 Takashi Sakamoto 2017-06-09 365 /* d81052f92 Takashi Sakamoto 2017-06-09 366 * Check condition bits of this rule. When the rule has d81052f92 Takashi Sakamoto 2017-06-09 367 * some condition bits, parameter without the bits is d81052f92 Takashi Sakamoto 2017-06-09 368 * never processed. SNDRV_PCM_HW_PARAMS_NO_PERIOD_WAKEUP d81052f92 Takashi Sakamoto 2017-06-09 369 * is an example of the condition bits. d81052f92 Takashi Sakamoto 2017-06-09 370 */ ^1da177e4 Linus Torvalds 2005-04-16 371 if (r->cond && !(r->cond & params->flags)) ^1da177e4 Linus Torvalds 2005-04-16 372 continue; d81052f92 Takashi Sakamoto 2017-06-09 373 d81052f92 Takashi Sakamoto 2017-06-09 374 /* d81052f92 Takashi Sakamoto 2017-06-09 375 * The 'deps' array includes maximum three dependencies d81052f92 Takashi Sakamoto 2017-06-09 376 * to SNDRV_PCM_HW_PARAM_XXXs for this rule. The fourth d81052f92 Takashi Sakamoto 2017-06-09 377 * member of this array is a sentinel and should be d81052f92 Takashi Sakamoto 2017-06-09 378 * negative value. d81052f92 Takashi Sakamoto 2017-06-09 379 * d81052f92 Takashi Sakamoto 2017-06-09 380 * This rule should be processed in this time when dependent d81052f92 Takashi Sakamoto 2017-06-09 381 * parameters were changed at former applications of the other d81052f92 Takashi Sakamoto 2017-06-09 382 * rules. d81052f92 Takashi Sakamoto 2017-06-09 383 */ ^1da177e4 Linus Torvalds 2005-04-16 384 for (d = 0; r->deps[d] >= 0; d++) { d656b4a65 Takashi Sakamoto 2017-06-09 385 if (vstamps[r->deps[d]] > rstamps[k]) ^1da177e4 Linus Torvalds 2005-04-16 386 break; ^1da177e4 Linus Torvalds 2005-04-16 387 } d656b4a65 Takashi Sakamoto 2017-06-09 388 if (r->deps[d] < 0) ^1da177e4 Linus Torvalds 2005-04-16 389 continue; be4e31dab Takashi Sakamoto 2017-06-07 390 be4e31dab Takashi Sakamoto 2017-06-07 391 if (trace_hw_mask_param_enabled()) { be4e31dab Takashi Sakamoto 2017-06-07 392 if (hw_is_mask(r->var)) be4e31dab Takashi Sakamoto 2017-06-07 393 old_mask = *hw_param_mask(params, r->var); ^1da177e4 Linus Torvalds 2005-04-16 394 } be4e31dab Takashi Sakamoto 2017-06-07 395 if (trace_hw_interval_param_enabled()) { be4e31dab Takashi Sakamoto 2017-06-07 396 if (hw_is_interval(r->var)) be4e31dab Takashi Sakamoto 2017-06-07 397 old_interval = *hw_param_interval(params, r->var); ^1da177e4 Linus Torvalds 2005-04-16 398 } c6706de0c Takashi Sakamoto 2017-06-07 399 ^1da177e4 Linus Torvalds 2005-04-16 400 changed = r->func(params, r); f74ae15fe Takashi Sakamoto 2017-06-11 @401 if (changed < 0) 16c438ade Kyle Spiers 2018-03-28 402 kfree(rstamps); f74ae15fe Takashi Sakamoto 2017-06-11 403 return changed; c6706de0c Takashi Sakamoto 2017-06-07 404 d81052f92 Takashi Sakamoto 2017-06-09 405 /* 82e7d5012 Takashi Sakamoto 2017-06-11 406 * When the parameter is changed, notify it to the caller d81052f92 Takashi Sakamoto 2017-06-09 407 * by corresponding returned bit, then preparing for next d81052f92 Takashi Sakamoto 2017-06-09 408 * iteration. d81052f92 Takashi Sakamoto 2017-06-09 409 */ ^1da177e4 Linus Torvalds 2005-04-16 410 if (changed && r->var >= 0) { 82e7d5012 Takashi Sakamoto 2017-06-11 411 if (hw_is_mask(r->var)) { 82e7d5012 Takashi Sakamoto 2017-06-11 412 trace_hw_mask_param(substream, r->var, 82e7d5012 Takashi Sakamoto 2017-06-11 413 k + 1, &old_mask, 82e7d5012 Takashi Sakamoto 2017-06-11 414 hw_param_mask(params, r->var)); ^1da177e4 Linus Torvalds 2005-04-16 415 } 82e7d5012 Takashi Sakamoto 2017-06-11 416 if (hw_is_interval(r->var)) { 82e7d5012 Takashi Sakamoto 2017-06-11 417 trace_hw_interval_param(substream, r->var, 82e7d5012 Takashi Sakamoto 2017-06-11 418 k + 1, &old_interval, 82e7d5012 Takashi Sakamoto 2017-06-11 419 hw_param_interval(params, r->var)); ^1da177e4 Linus Torvalds 2005-04-16 420 } 82e7d5012 Takashi Sakamoto 2017-06-11 421 ^1da177e4 Linus Torvalds 2005-04-16 422 params->cmask |= (1 << r->var); ^1da177e4 Linus Torvalds 2005-04-16 423 vstamps[r->var] = stamp; a1c06e39a Takashi Sakamoto 2017-06-09 424 again = true; ^1da177e4 Linus Torvalds 2005-04-16 425 } f74ae15fe Takashi Sakamoto 2017-06-11 426 82e7d5012 Takashi Sakamoto 2017-06-11 427 rstamps[k] = stamp++; ^1da177e4 Linus Torvalds 2005-04-16 428 } 0d4e39996 Takashi Sakamoto 2017-06-09 429 d81052f92 Takashi Sakamoto 2017-06-09 430 /* Iterate to evaluate all rules till no parameters are changed. */ 0d4e39996 Takashi Sakamoto 2017-06-09 431 if (again) 0d4e39996 Takashi Sakamoto 2017-06-09 432 goto retry; 9cc07f55d Takashi Sakamoto 2017-06-09 433 16c438ade Kyle Spiers 2018-03-28 434 kfree(rstamps); 9cc07f55d Takashi Sakamoto 2017-06-09 435 return 0; ^1da177e4 Linus Torvalds 2005-04-16 436 } 9cc07f55d Takashi Sakamoto 2017-06-09 437 :::::: The code at line 401 was first introduced by commit :::::: f74ae15fe3da7905b78e986ad906a333587cf160 ALSA: pcm: return error immediately for parameters handling :::::: TO: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> :::::: CC: Takashi Iwai <tiwai(a)suse.de> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [GIT PULL] Audio firmware update
by Sanyog Kale 30 Mar '18

30 Mar '18

Hi, Please PULL to receive the update for Skylake, Kabylake, Broxton and Geminilake audio DSP firmware. The following changes since commit 44476f2465dac9c22bce90da66e86b2b56ba34f0: BCM-0bb4-0306: Update to Cypress license in WHENCE (2018-03-20 12:20:21 -0400) are available in the git repository at: https://github.com/sanyogrk/firmware kbl for you to fetch changes up to 9c53767c99f1d393e9c0f566ec17e4699788e95d: linux-firmware: intel: Update Geminilake audio firmware (2018-03-23 09:55:51 +0530) ---------------------------------------------------------------- Sanyog Kale (4): linux-firmware: intel: Update Skylake audio firmware linux-firmware: intel: Update Broxton audio firmware linux-firmware: intel: Update Kabylake audio firmware linux-firmware: intel: Update Geminilake audio firmware WHENCE | 16 ++++++++++++---- intel/dsp_fw_bxtn.bin | 2 +- intel/dsp_fw_bxtn_v3366.bin | Bin 0 -> 505608 bytes intel/dsp_fw_glk.bin | 2 +- intel/dsp_fw_glk_v3366.bin | Bin 0 -> 505608 bytes intel/dsp_fw_kbl.bin | 2 +- intel/dsp_fw_kbl_v3402.bin | Bin 0 -> 247112 bytes intel/dsp_fw_release.bin | 2 +- intel/dsp_fw_release_v3402.bin | Bin 0 -> 247112 bytes 9 files changed, 16 insertions(+), 8 deletions(-) create mode 100644 intel/dsp_fw_bxtn_v3366.bin create mode 100644 intel/dsp_fw_glk_v3366.bin create mode 100644 intel/dsp_fw_kbl_v3402.bin create mode 100644 intel/dsp_fw_release_v3402.bin Thanks, Sanyog --

2 1

Re: [alsa-devel] [PATCH] pcm_native: Remove VLA usage
by kbuild test robot 30 Mar '18

30 Mar '18

Hi Kyle, Thank you for the patch! Perhaps something to improve: [auto build test WARNING on v4.16-rc7] [cannot apply to sound/for-next next-20180329] [if your patch is applied to the wrong git tree, please drop us a note to help improve the system] url: https://github.com/0day-ci/linux/commits/Kyle-Spiers/pcm_native-Remove-VLA-… config: i386-randconfig-x075-201812 (attached as .config) compiler: gcc-7 (Debian 7.3.0-1) 7.3.0 reproduce: # save the attached .config to linux build tree make ARCH=i386 All warnings (new ones prefixed by >>): sound/core/pcm_native.c: In function 'constrain_params_by_rules': >> sound/core/pcm_native.c:401:3: warning: this 'if' clause does not guard... [-Wmisleading-indentation] if (changed < 0) ^~ sound/core/pcm_native.c:403:4: note: ...this statement, but the latter is misleadingly indented as if it were guarded by the 'if' return changed; ^~~~~~ vim +/if +401 sound/core/pcm_native.c ^1da177e4 Linus Torvalds 2005-04-16 319 9cc07f55d Takashi Sakamoto 2017-06-09 320 static int constrain_params_by_rules(struct snd_pcm_substream *substream, 877211f5e Takashi Iwai 2005-11-17 321 struct snd_pcm_hw_params *params) ^1da177e4 Linus Torvalds 2005-04-16 322 { 9cc07f55d Takashi Sakamoto 2017-06-09 323 struct snd_pcm_hw_constraints *constrs = 9cc07f55d Takashi Sakamoto 2017-06-09 324 &substream->runtime->hw_constraints; ^1da177e4 Linus Torvalds 2005-04-16 325 unsigned int k; 16c438ade Kyle Spiers 2018-03-28 326 unsigned int *rstamps; ^1da177e4 Linus Torvalds 2005-04-16 327 unsigned int vstamps[SNDRV_PCM_HW_PARAM_LAST_INTERVAL + 1]; d81052f92 Takashi Sakamoto 2017-06-09 328 unsigned int stamp; a1c06e39a Takashi Sakamoto 2017-06-09 329 struct snd_pcm_hw_rule *r; a1c06e39a Takashi Sakamoto 2017-06-09 330 unsigned int d; 9cc07f55d Takashi Sakamoto 2017-06-09 331 struct snd_mask old_mask; 9cc07f55d Takashi Sakamoto 2017-06-09 332 struct snd_interval old_interval; a1c06e39a Takashi Sakamoto 2017-06-09 333 bool again; 9cc07f55d Takashi Sakamoto 2017-06-09 334 int changed; ^1da177e4 Linus Torvalds 2005-04-16 335 d81052f92 Takashi Sakamoto 2017-06-09 336 /* d81052f92 Takashi Sakamoto 2017-06-09 337 * Each application of rule has own sequence number. d81052f92 Takashi Sakamoto 2017-06-09 338 * d81052f92 Takashi Sakamoto 2017-06-09 339 * Each member of 'rstamps' array represents the sequence number of d81052f92 Takashi Sakamoto 2017-06-09 340 * recent application of corresponding rule. d81052f92 Takashi Sakamoto 2017-06-09 341 */ 16c438ade Kyle Spiers 2018-03-28 342 16c438ade Kyle Spiers 2018-03-28 343 rstamps = kcalloc(constrs->rules_num, sizeof(*rstamps), GFP_KERNEL); d81052f92 Takashi Sakamoto 2017-06-09 344 d81052f92 Takashi Sakamoto 2017-06-09 345 /* d81052f92 Takashi Sakamoto 2017-06-09 346 * Each member of 'vstamps' array represents the sequence number of d81052f92 Takashi Sakamoto 2017-06-09 347 * recent application of rule in which corresponding parameters were d81052f92 Takashi Sakamoto 2017-06-09 348 * changed. d81052f92 Takashi Sakamoto 2017-06-09 349 * d81052f92 Takashi Sakamoto 2017-06-09 350 * In initial state, elements corresponding to parameters requested by d81052f92 Takashi Sakamoto 2017-06-09 351 * a caller is 1. For unrequested parameters, corresponding members d81052f92 Takashi Sakamoto 2017-06-09 352 * have 0 so that the parameters are never changed anymore. d81052f92 Takashi Sakamoto 2017-06-09 353 */ ^1da177e4 Linus Torvalds 2005-04-16 354 for (k = 0; k <= SNDRV_PCM_HW_PARAM_LAST_INTERVAL; k++) ^1da177e4 Linus Torvalds 2005-04-16 355 vstamps[k] = (params->rmask & (1 << k)) ? 1 : 0; d81052f92 Takashi Sakamoto 2017-06-09 356 d81052f92 Takashi Sakamoto 2017-06-09 357 /* Due to the above design, actual sequence number starts at 2. */ d81052f92 Takashi Sakamoto 2017-06-09 358 stamp = 2; 0d4e39996 Takashi Sakamoto 2017-06-09 359 retry: d81052f92 Takashi Sakamoto 2017-06-09 360 /* Apply all rules in order. */ a1c06e39a Takashi Sakamoto 2017-06-09 361 again = false; ^1da177e4 Linus Torvalds 2005-04-16 362 for (k = 0; k < constrs->rules_num; k++) { a1c06e39a Takashi Sakamoto 2017-06-09 363 r = &constrs->rules[k]; d81052f92 Takashi Sakamoto 2017-06-09 364 d81052f92 Takashi Sakamoto 2017-06-09 365 /* d81052f92 Takashi Sakamoto 2017-06-09 366 * Check condition bits of this rule. When the rule has d81052f92 Takashi Sakamoto 2017-06-09 367 * some condition bits, parameter without the bits is d81052f92 Takashi Sakamoto 2017-06-09 368 * never processed. SNDRV_PCM_HW_PARAMS_NO_PERIOD_WAKEUP d81052f92 Takashi Sakamoto 2017-06-09 369 * is an example of the condition bits. d81052f92 Takashi Sakamoto 2017-06-09 370 */ ^1da177e4 Linus Torvalds 2005-04-16 371 if (r->cond && !(r->cond & params->flags)) ^1da177e4 Linus Torvalds 2005-04-16 372 continue; d81052f92 Takashi Sakamoto 2017-06-09 373 d81052f92 Takashi Sakamoto 2017-06-09 374 /* d81052f92 Takashi Sakamoto 2017-06-09 375 * The 'deps' array includes maximum three dependencies d81052f92 Takashi Sakamoto 2017-06-09 376 * to SNDRV_PCM_HW_PARAM_XXXs for this rule. The fourth d81052f92 Takashi Sakamoto 2017-06-09 377 * member of this array is a sentinel and should be d81052f92 Takashi Sakamoto 2017-06-09 378 * negative value. d81052f92 Takashi Sakamoto 2017-06-09 379 * d81052f92 Takashi Sakamoto 2017-06-09 380 * This rule should be processed in this time when dependent d81052f92 Takashi Sakamoto 2017-06-09 381 * parameters were changed at former applications of the other d81052f92 Takashi Sakamoto 2017-06-09 382 * rules. d81052f92 Takashi Sakamoto 2017-06-09 383 */ ^1da177e4 Linus Torvalds 2005-04-16 384 for (d = 0; r->deps[d] >= 0; d++) { d656b4a65 Takashi Sakamoto 2017-06-09 385 if (vstamps[r->deps[d]] > rstamps[k]) ^1da177e4 Linus Torvalds 2005-04-16 386 break; ^1da177e4 Linus Torvalds 2005-04-16 387 } d656b4a65 Takashi Sakamoto 2017-06-09 388 if (r->deps[d] < 0) ^1da177e4 Linus Torvalds 2005-04-16 389 continue; be4e31dab Takashi Sakamoto 2017-06-07 390 be4e31dab Takashi Sakamoto 2017-06-07 391 if (trace_hw_mask_param_enabled()) { be4e31dab Takashi Sakamoto 2017-06-07 392 if (hw_is_mask(r->var)) be4e31dab Takashi Sakamoto 2017-06-07 393 old_mask = *hw_param_mask(params, r->var); ^1da177e4 Linus Torvalds 2005-04-16 394 } be4e31dab Takashi Sakamoto 2017-06-07 395 if (trace_hw_interval_param_enabled()) { be4e31dab Takashi Sakamoto 2017-06-07 396 if (hw_is_interval(r->var)) be4e31dab Takashi Sakamoto 2017-06-07 397 old_interval = *hw_param_interval(params, r->var); ^1da177e4 Linus Torvalds 2005-04-16 398 } c6706de0c Takashi Sakamoto 2017-06-07 399 ^1da177e4 Linus Torvalds 2005-04-16 400 changed = r->func(params, r); f74ae15fe Takashi Sakamoto 2017-06-11 @401 if (changed < 0) 16c438ade Kyle Spiers 2018-03-28 402 kfree(rstamps); f74ae15fe Takashi Sakamoto 2017-06-11 403 return changed; c6706de0c Takashi Sakamoto 2017-06-07 404 d81052f92 Takashi Sakamoto 2017-06-09 405 /* 82e7d5012 Takashi Sakamoto 2017-06-11 406 * When the parameter is changed, notify it to the caller d81052f92 Takashi Sakamoto 2017-06-09 407 * by corresponding returned bit, then preparing for next d81052f92 Takashi Sakamoto 2017-06-09 408 * iteration. d81052f92 Takashi Sakamoto 2017-06-09 409 */ ^1da177e4 Linus Torvalds 2005-04-16 410 if (changed && r->var >= 0) { 82e7d5012 Takashi Sakamoto 2017-06-11 411 if (hw_is_mask(r->var)) { 82e7d5012 Takashi Sakamoto 2017-06-11 412 trace_hw_mask_param(substream, r->var, 82e7d5012 Takashi Sakamoto 2017-06-11 413 k + 1, &old_mask, 82e7d5012 Takashi Sakamoto 2017-06-11 414 hw_param_mask(params, r->var)); ^1da177e4 Linus Torvalds 2005-04-16 415 } 82e7d5012 Takashi Sakamoto 2017-06-11 416 if (hw_is_interval(r->var)) { 82e7d5012 Takashi Sakamoto 2017-06-11 417 trace_hw_interval_param(substream, r->var, 82e7d5012 Takashi Sakamoto 2017-06-11 418 k + 1, &old_interval, 82e7d5012 Takashi Sakamoto 2017-06-11 419 hw_param_interval(params, r->var)); ^1da177e4 Linus Torvalds 2005-04-16 420 } 82e7d5012 Takashi Sakamoto 2017-06-11 421 ^1da177e4 Linus Torvalds 2005-04-16 422 params->cmask |= (1 << r->var); ^1da177e4 Linus Torvalds 2005-04-16 423 vstamps[r->var] = stamp; a1c06e39a Takashi Sakamoto 2017-06-09 424 again = true; ^1da177e4 Linus Torvalds 2005-04-16 425 } f74ae15fe Takashi Sakamoto 2017-06-11 426 82e7d5012 Takashi Sakamoto 2017-06-11 427 rstamps[k] = stamp++; ^1da177e4 Linus Torvalds 2005-04-16 428 } 0d4e39996 Takashi Sakamoto 2017-06-09 429 d81052f92 Takashi Sakamoto 2017-06-09 430 /* Iterate to evaluate all rules till no parameters are changed. */ 0d4e39996 Takashi Sakamoto 2017-06-09 431 if (again) 0d4e39996 Takashi Sakamoto 2017-06-09 432 goto retry; 9cc07f55d Takashi Sakamoto 2017-06-09 433 16c438ade Kyle Spiers 2018-03-28 434 kfree(rstamps); 9cc07f55d Takashi Sakamoto 2017-06-09 435 return 0; ^1da177e4 Linus Torvalds 2005-04-16 436 } 9cc07f55d Takashi Sakamoto 2017-06-09 437 :::::: The code at line 401 was first introduced by commit :::::: f74ae15fe3da7905b78e986ad906a333587cf160 ALSA: pcm: return error immediately for parameters handling :::::: TO: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> :::::: CC: Takashi Iwai <tiwai(a)suse.de> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [RESEND][PATCH v4 0/3] Add SPIB Support for Intel Skylake platforms
by Sriram Periyasamy 30 Mar '18

30 Mar '18

Skylake audio controller supports SPIB (Software Position in buffer) capability, which can be used to inform position of application pointer to host DMA controller. When SPIB mode is enabled, driver could write the application pointer position in SPIB register. Host DMA will make sure it won't read/write beyond bytes specified in SPIB register. SPIB mode will be useful in low power use cases, where DSP could pre-fetch large buffers to avoid frequent wakes caused due to interrupts. To support SPIB in the driver, save the spib values in stream context which can be restored during resume from S3. Add new hw_params flag to explicitly tell driver that rewinds will never be used. Please find the references for the previous discussions at [1][2] [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-January/131329. html [2] https://patchwork.kernel.org/patch/9795233/ and the patches of previous versions at [3][4][5] [3] http://mailman.alsa-project.org/pipermail/alsa-devel/2017-June/121967.html [4] http://mailman.alsa-project.org/pipermail/alsa-devel/2017-June/121683.html [5] http://mailman.alsa-project.org/pipermail/alsa-devel/2017-May/120676.html Pierre-Louis Bossart (1): ALSA: core: let low-level driver or userspace disable rewinds Ramesh Babu (2): ALSA: hda: ext: add spib to stream context ASoC: Intel: Skylake: Add support for spib mode include/sound/hdaudio_ext.h | 1 + include/sound/pcm.h | 1 + include/uapi/sound/asound.h | 1 + sound/core/pcm_native.c | 8 ++++++++ sound/hda/ext/hdac_ext_stream.c | 2 ++ sound/soc/intel/skylake/skl-pcm.c | 43 ++++++++++++++++++++++++++++++++++++++- 6 files changed, 55 insertions(+), 1 deletion(-) -- 2.7.4

4 18

[alsa-devel] [PATCH] ALSA: hda_intel: mark PM functions as __maybe_unused
by Arnd Bergmann 29 Mar '18

29 Mar '18

Two callsites of azx_suspend/azx_resume were removed, leaving these functions only called from the optional SET_SYSTEM_SLEEP_PM_OPS() and causing a warning without CONFIG_PM_SLEEP: sound/pci/hda/hda_intel.c:1029:12: error: 'azx_resume' defined but not used [-Werror=unused-function] static int azx_resume(struct device *dev) ^~~~~~~~~~ sound/pci/hda/hda_intel.c:994:12: error: 'azx_suspend' defined but not used [-Werror=unused-function] static int azx_suspend(struct device *dev) ^~~~~~~~~~~ Keeping track of the correct #ifdef checks is hard, so this removes all the #ifdefs for power management in this file and instead uses __maybe_unused annotations that let the compiler do the job right by itself. Fixes: 07f4f97d7b4b ("vga_switcheroo: Use device link for HDA controller") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/pci/hda/hda_intel.c | 52 +++++++++++++++++------------------------ sound/pci/hda/hda_intel_trace.h | 2 -- 2 files changed, 21 insertions(+), 33 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 087c4d861c6e..01a96972d191 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -198,7 +198,9 @@ static bool power_save_controller = 1; module_param(power_save_controller, bool, 0644); MODULE_PARM_DESC(power_save_controller, "Reset controller in power save mode."); #else -#define power_save 0 +#define power_save 0 +#define power_save_controller 0 +#define pm_blacklist 0 #endif /* CONFIG_PM */ static int align_buffer_size = -1; @@ -941,13 +943,16 @@ static unsigned int azx_get_pos_skl(struct azx *chip, struct azx_dev *azx_dev) return azx_get_pos_posbuf(chip, azx_dev); } -#ifdef CONFIG_PM static DEFINE_MUTEX(card_list_lock); static LIST_HEAD(card_list); static void azx_add_card_list(struct azx *chip) { struct hda_intel *hda = container_of(chip, struct hda_intel, chip); + + if (!IS_ENABLED(CONFIG_PM)) + return; + mutex_lock(&card_list_lock); list_add(&hda->list, &card_list); mutex_unlock(&card_list_lock); @@ -956,13 +961,17 @@ static void azx_add_card_list(struct azx *chip) static void azx_del_card_list(struct azx *chip) { struct hda_intel *hda = container_of(chip, struct hda_intel, chip); + + if (!IS_ENABLED(CONFIG_PM)) + return; + mutex_lock(&card_list_lock); list_del_init(&hda->list); mutex_unlock(&card_list_lock); } /* trigger power-save check at writing parameter */ -static int param_set_xint(const char *val, const struct kernel_param *kp) +static int __maybe_unused param_set_xint(const char *val, const struct kernel_param *kp) { struct hda_intel *hda; struct azx *chip; @@ -982,16 +991,11 @@ static int param_set_xint(const char *val, const struct kernel_param *kp) mutex_unlock(&card_list_lock); return 0; } -#else -#define azx_add_card_list(chip) /* NOP */ -#define azx_del_card_list(chip) /* NOP */ -#endif /* CONFIG_PM */ -#if defined(CONFIG_PM_SLEEP) || defined(SUPPORT_VGA_SWITCHEROO) /* * power management */ -static int azx_suspend(struct device *dev) +static int __maybe_unused azx_suspend(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); struct azx *chip; @@ -1026,7 +1030,7 @@ static int azx_suspend(struct device *dev) return 0; } -static int azx_resume(struct device *dev) +static int __maybe_unused azx_resume(struct device *dev) { struct pci_dev *pci = to_pci_dev(dev); struct snd_card *card = dev_get_drvdata(dev); @@ -1068,13 +1072,11 @@ static int azx_resume(struct device *dev) trace_azx_resume(chip); return 0; } -#endif /* CONFIG_PM_SLEEP || SUPPORT_VGA_SWITCHEROO */ -#ifdef CONFIG_PM_SLEEP /* put codec down to D3 at hibernation for Intel SKL+; * otherwise BIOS may still access the codec and screw up the driver */ -static int azx_freeze_noirq(struct device *dev) +static int __maybe_unused azx_freeze_noirq(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); struct azx *chip = card->private_data; @@ -1086,7 +1088,7 @@ static int azx_freeze_noirq(struct device *dev) return 0; } -static int azx_thaw_noirq(struct device *dev) +static int __maybe_unused azx_thaw_noirq(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); struct azx *chip = card->private_data; @@ -1097,10 +1099,8 @@ static int azx_thaw_noirq(struct device *dev) return 0; } -#endif /* CONFIG_PM_SLEEP */ -#ifdef CONFIG_PM -static int azx_runtime_suspend(struct device *dev) +static int __maybe_unused azx_runtime_suspend(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); struct azx *chip; @@ -1132,7 +1132,7 @@ static int azx_runtime_suspend(struct device *dev) return 0; } -static int azx_runtime_resume(struct device *dev) +static int __maybe_unused azx_runtime_resume(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); struct azx *chip; @@ -1185,7 +1185,7 @@ static int azx_runtime_resume(struct device *dev) return 0; } -static int azx_runtime_idle(struct device *dev) +static int __maybe_unused azx_runtime_idle(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); struct azx *chip; @@ -1215,12 +1215,6 @@ static const struct dev_pm_ops azx_pm = { SET_RUNTIME_PM_OPS(azx_runtime_suspend, azx_runtime_resume, azx_runtime_idle) }; -#define AZX_PM_OPS &azx_pm -#else -#define AZX_PM_OPS NULL -#endif /* CONFIG_PM */ - - static int azx_probe_continue(struct azx *chip); #ifdef SUPPORT_VGA_SWITCHEROO @@ -2199,7 +2193,6 @@ static int azx_probe(struct pci_dev *pci, return err; } -#ifdef CONFIG_PM /* On some boards setting power_save to a non 0 value leads to clicking / * popping sounds when ever we enter/leave powersaving mode. Ideally we would * figure out how to avoid these sounds, but that is not always feasible. @@ -2215,7 +2208,6 @@ static struct snd_pci_quirk power_save_blacklist[] = { SND_PCI_QUIRK(0x17aa, 0x2227, "Lenovo X1 Carbon 3rd Gen", 0), {} }; -#endif /* CONFIG_PM */ /* number of codec slots for each chipset: 0 = default slots (i.e. 4) */ static unsigned int azx_max_codecs[AZX_NUM_DRIVERS] = { @@ -2313,8 +2305,7 @@ static int azx_probe_continue(struct azx *chip) azx_add_card_list(chip); val = power_save; -#ifdef CONFIG_PM - if (pm_blacklist) { + if (IS_ENABLED(CONFIG_PM) && pm_blacklist) { const struct snd_pci_quirk *q; q = snd_pci_quirk_lookup(chip->pci, power_save_blacklist); @@ -2324,7 +2315,6 @@ static int azx_probe_continue(struct azx *chip) val = 0; } } -#endif /* CONFIG_PM */ /* * The discrete GPU cannot power down unless the HDA controller runtime @@ -2671,7 +2661,7 @@ static struct pci_driver azx_driver = { .remove = azx_remove, .shutdown = azx_shutdown, .driver = { - .pm = AZX_PM_OPS, + .pm = &azx_pm, }, }; diff --git a/sound/pci/hda/hda_intel_trace.h b/sound/pci/hda/hda_intel_trace.h index 73a7adfa192d..2775fa81a500 100644 --- a/sound/pci/hda/hda_intel_trace.h +++ b/sound/pci/hda/hda_intel_trace.h @@ -34,7 +34,6 @@ DEFINE_EVENT(hda_pm, azx_resume, TP_ARGS(chip) ); -#ifdef CONFIG_PM DEFINE_EVENT(hda_pm, azx_runtime_suspend, TP_PROTO(struct azx *chip), TP_ARGS(chip) @@ -44,7 +43,6 @@ DEFINE_EVENT(hda_pm, azx_runtime_resume, TP_PROTO(struct azx *chip), TP_ARGS(chip) ); -#endif #endif /* _TRACE_HDA_INTEL_H */ -- 2.9.0

3 4

[alsa-devel] [PATCH] ALSA: usb-audio: silence a static checker warning
by Dan Carpenter 29 Mar '18

29 Mar '18

We recently made "format" a u64 variable so now static checkers complain that this shift will wrap around if format is more than 31. I don't think it makes a difference for runtime, but it's simple to silence the warning. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/usb/format.c b/sound/usb/format.c index edbe67eeddfa..49e7ec6d2399 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -55,7 +55,7 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip, struct uac_format_type_i_discrete_descriptor *fmt = _fmt; sample_width = fmt->bBitResolution; sample_bytes = fmt->bSubframeSize; - format = 1 << format; + format = 1ULL << format; break; }

2 1

[alsa-devel] [PATCH] pcm: ioplug: Implement proper drain behavior
by Takashi Iwai 29 Mar '18

29 Mar '18

This patch fixes the draining behavior of ioplug in the following ways: - When no draining ioplug callback is defined, implement the draining loop using snd_pcm_wait*() and sync with the drain finishes. This is equivalent with the implementation in the kernel write(). Similarly as in kernel code, for non-blocking mode, it returns immediately after setting DRAINING state. - The hw_ptr update function checks the PCM state and stops the stream if the draining finishes. - When draining ioplug callback is defined, leave the whole draining operation to it. The callback is supposed to return -EAGAIN for non-blocking case, too. - When an unexpected error happens during draining, it drops the stream, for a safety reason. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- The patch was slightly changed from the draft version. e.g. hw_ptr_update() is called at the beginning of the loop, so that it's updated for non-blocking mode, too. And, the stream is dropped in case of unexpected errors. Also, this patch requires the previous fix ("pcm: Skip avail_min check during draining"). src/pcm/pcm_ioplug.c | 76 +++++++++++++++++++++++++++++++++++++++++++++------- 1 file changed, 67 insertions(+), 9 deletions(-) diff --git a/src/pcm/pcm_ioplug.c b/src/pcm/pcm_ioplug.c index 8c0ed4836365..db64853b3136 100644 --- a/src/pcm/pcm_ioplug.c +++ b/src/pcm/pcm_ioplug.c @@ -47,6 +47,11 @@ typedef struct snd_pcm_ioplug_priv { snd_htimestamp_t trigger_tstamp; } ioplug_priv_t; +static int snd_pcm_ioplug_drop(snd_pcm_t *pcm); +static int snd_pcm_ioplug_poll_descriptors_count(snd_pcm_t *pcm); +static int snd_pcm_ioplug_poll_descriptors(snd_pcm_t *pcm, struct pollfd *pfds, unsigned int space); +static int snd_pcm_ioplug_poll_revents(snd_pcm_t *pcm, struct pollfd *pfds, unsigned int nfds, unsigned short *revents); + /* update the hw pointer */ /* called in lock */ static void snd_pcm_ioplug_hw_ptr_update(snd_pcm_t *pcm) @@ -57,6 +62,7 @@ static void snd_pcm_ioplug_hw_ptr_update(snd_pcm_t *pcm) hw = io->data->callback->pointer(io->data); if (hw >= 0) { snd_pcm_uframes_t delta; + snd_pcm_uframes_t avail; if ((snd_pcm_uframes_t)hw >= io->last_hw) delta = hw - io->last_hw; @@ -67,9 +73,19 @@ static void snd_pcm_ioplug_hw_ptr_update(snd_pcm_t *pcm) delta = wrap_point + hw - io->last_hw; } snd_pcm_mmap_hw_forward(io->data->pcm, delta); + /* stop the stream if all samples are drained */ + if (io->data->state == SND_PCM_STATE_DRAINING) { + avail = snd_pcm_mmap_avail(pcm); + if (avail >= pcm->buffer_size) + snd_pcm_ioplug_drop(pcm); + } io->last_hw = (snd_pcm_uframes_t)hw; - } else - io->data->state = SNDRV_PCM_STATE_XRUN; + } else { + if (io->data->state == SND_PCM_STATE_DRAINING) + snd_pcm_ioplug_drop(pcm); + else + io->data->state = SNDRV_PCM_STATE_XRUN; + } } static int snd_pcm_ioplug_info(snd_pcm_t *pcm, snd_pcm_info_t *info) @@ -488,20 +504,62 @@ static int snd_pcm_ioplug_drop(snd_pcm_t *pcm) return 0; } +static int ioplug_drain_via_poll(snd_pcm_t *pcm) +{ + ioplug_priv_t *io = pcm->private_data; + + while (io->data->state == SND_PCM_STATE_DRAINING) { + snd_pcm_ioplug_hw_ptr_update(pcm); + if (io->data->state != SND_PCM_STATE_DRAINING) + break; + /* in non-blocking mode, let application to poll() by itself */ + if (io->data->nonblock) + return -EAGAIN; + if (snd_pcm_wait_nocheck(pcm, -1) < 0) + break; + } + + return 0; /* force to drop at error */ +} + /* need own locking */ static int snd_pcm_ioplug_drain(snd_pcm_t *pcm) { ioplug_priv_t *io = pcm->private_data; - int err; + int err = 0; - if (io->data->state == SND_PCM_STATE_OPEN) + snd_pcm_lock(pcm); + switch (io->data->state) { + case SND_PCM_STATE_OPEN: + case SND_PCM_STATE_DISCONNECTED: + case SND_PCM_STATE_SUSPENDED: return -EBADFD; + case SND_PCM_STATE_PREPARED: + if (pcm->stream == SND_PCM_STREAM_PLAYBACK) { + err = snd_pcm_ioplug_start(pcm); + if (err < 0) + goto unlock; + io->data->state = SND_PCM_STATE_DRAINING; + } + break; + case SND_PCM_STATE_RUNNING: + io->data->state = SND_PCM_STATE_DRAINING; + break; + } - io->data->state = SND_PCM_STATE_DRAINING; - if (io->data->callback->drain) - io->data->callback->drain(io->data); - snd_pcm_lock(pcm); - err = snd_pcm_ioplug_drop(pcm); + if (io->data->state == SND_PCM_STATE_DRAINING) { + if (io->data->callback->drain) { + snd_pcm_unlock(pcm); /* let plugin own locking */ + err = io->data->callback->drain(io->data); + snd_pcm_lock(pcm); + } else { + err = ioplug_drain_via_poll(pcm); + } + } + + unlock: + if (!err && io->data->state != SND_PCM_STATE_SETUP) + snd_pcm_ioplug_drop(pcm); snd_pcm_unlock(pcm); return err; } -- 2.16.2

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[alsa-devel] [PATCH - IOPLUG DRAIN 0/2]
by twischer＠de.adit-jv.com 29 Mar '18

29 Mar '18

Hi Takashi, > Why not poll()? > IOW, why ioplug must be handled specially regarding the non-blocking >operation? The normal kernel driver behaves like that (returning > -EAGAIN, and let apps to sync with poll()). What do you think about the following solution? (I thought the whole time that you have to use snd_pcm_wait() to wait for drain in nonblocking mode but you have to use the poll_descriptors directly.) Know I am expecting that the user is calling poll() if snd_pcm_drain() returns -EAGAIN and the user has to call snd_pcm_drain() again after poll returns to check if drain is done. Thanks for your help so far. Best regards Timo

3 19

[alsa-devel] [PATCH v2] media: i2c: wm9090: replace codec to component
by Kuninori Morimoto 29 Mar '18

29 Mar '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Now we can replace Codec to Component. Let's do it. Note: xxx_codec_xxx() -> xxx_component_xxx() .idle_bias_off = 0 -> .idle_bias_on = 1 .ignore_pmdown_time = 0 -> .use_pmdown_time = 1 - -> .endianness = 1 - -> .non_legacy_dai_naming = 1 Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - fixup .remove return drivers/media/i2c/tda1997x.c | 21 +++++++++++---------- 1 file changed, 11 insertions(+), 10 deletions(-) diff --git a/drivers/media/i2c/tda1997x.c b/drivers/media/i2c/tda1997x.c index 3021913..2dc4df5 100644 --- a/drivers/media/i2c/tda1997x.c +++ b/drivers/media/i2c/tda1997x.c @@ -2444,7 +2444,7 @@ static int tda1997x_pcm_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct tda1997x_state *state = snd_soc_dai_get_drvdata(dai); - struct snd_soc_codec *codec = dai->codec; + struct snd_soc_component *component = dai->component; struct snd_pcm_runtime *rtd = substream->runtime; int rate, err; @@ -2452,11 +2452,11 @@ static int tda1997x_pcm_startup(struct snd_pcm_substream *substream, err = snd_pcm_hw_constraint_minmax(rtd, SNDRV_PCM_HW_PARAM_RATE, rate, rate); if (err < 0) { - dev_err(codec->dev, "failed to constrain samplerate to %dHz\n", + dev_err(component->dev, "failed to constrain samplerate to %dHz\n", rate); return err; } - dev_info(codec->dev, "set samplerate constraint to %dHz\n", rate); + dev_info(component->dev, "set samplerate constraint to %dHz\n", rate); return 0; } @@ -2479,20 +2479,22 @@ static int tda1997x_pcm_startup(struct snd_pcm_substream *substream, .ops = &tda1997x_dai_ops, }; -static int tda1997x_codec_probe(struct snd_soc_codec *codec) +static int tda1997x_codec_probe(struct snd_soc_component *component) { return 0; } -static int tda1997x_codec_remove(struct snd_soc_codec *codec) +static void tda1997x_codec_remove(struct snd_soc_component *component) { - return 0; } -static struct snd_soc_codec_driver tda1997x_codec_driver = { +static struct snd_soc_component_driver tda1997x_codec_driver = { .probe = tda1997x_codec_probe, .remove = tda1997x_codec_remove, - .reg_word_size = sizeof(u16), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, }; static int tda1997x_probe(struct i2c_client *client, @@ -2737,7 +2739,7 @@ static int tda1997x_probe(struct i2c_client *client, else formats = SNDRV_PCM_FMTBIT_S16_LE; tda1997x_audio_dai.capture.formats = formats; - ret = snd_soc_register_codec(&state->client->dev, + ret = devm_snd_soc_register_component(&state->client->dev, &tda1997x_codec_driver, &tda1997x_audio_dai, 1); if (ret) { @@ -2782,7 +2784,6 @@ static int tda1997x_remove(struct i2c_client *client) struct tda1997x_platform_data *pdata = &state->pdata; if (pdata->audout_format) { - snd_soc_unregister_codec(&client->dev); mutex_destroy(&state->audio_lock); } -- 1.9.1

1 0

Re: [alsa-devel] [PATCH v2] pcm_native: Remove VLA usage
by Takashi Iwai 29 Mar '18

29 Mar '18

On Thu, 29 Mar 2018 00:24:00 +0200, Kyle Spiers wrote: > > As part of the effort to remove VLAs from the kernel[1], this changes > the allocation of the rstamps array from being on the stack to being > kcalloc()ed. This also allows for the removal of the explicit zeroing > loop. > > Signed-off-by: Kyle Spiers <kyle(a)spiers.me> It's been already fixed weeks ago... thanks, Takashi

1 0

[alsa-devel] [PATCH] media: i2c: wm9090: replace codec to component
by Kuninori Morimoto 29 Mar '18

29 Mar '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Now we can replace Codec to Component. Let's do it. Note: xxx_codec_xxx() -> xxx_component_xxx() .idle_bias_off = 0 -> .idle_bias_on = 1 .ignore_pmdown_time = 0 -> .use_pmdown_time = 1 - -> .endianness = 1 - -> .non_legacy_dai_naming = 1 Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Hi Mark, Tim, Mauro ALSA SoC had replaced codec to component, thus, we can't use it anymore. This patch fixes it. But, I don't know who/how can handle it. drivers/media/i2c/tda1997x.c | 20 +++++++++++--------- 1 file changed, 11 insertions(+), 9 deletions(-) diff --git a/drivers/media/i2c/tda1997x.c b/drivers/media/i2c/tda1997x.c index 3021913..5c5de3e 100644 --- a/drivers/media/i2c/tda1997x.c +++ b/drivers/media/i2c/tda1997x.c @@ -2444,7 +2444,7 @@ static int tda1997x_pcm_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct tda1997x_state *state = snd_soc_dai_get_drvdata(dai); - struct snd_soc_codec *codec = dai->codec; + struct snd_soc_component *component = dai->component; struct snd_pcm_runtime *rtd = substream->runtime; int rate, err; @@ -2452,11 +2452,11 @@ static int tda1997x_pcm_startup(struct snd_pcm_substream *substream, err = snd_pcm_hw_constraint_minmax(rtd, SNDRV_PCM_HW_PARAM_RATE, rate, rate); if (err < 0) { - dev_err(codec->dev, "failed to constrain samplerate to %dHz\n", + dev_err(component->dev, "failed to constrain samplerate to %dHz\n", rate); return err; } - dev_info(codec->dev, "set samplerate constraint to %dHz\n", rate); + dev_info(component->dev, "set samplerate constraint to %dHz\n", rate); return 0; } @@ -2479,20 +2479,23 @@ static int tda1997x_pcm_startup(struct snd_pcm_substream *substream, .ops = &tda1997x_dai_ops, }; -static int tda1997x_codec_probe(struct snd_soc_codec *codec) +static int tda1997x_codec_probe(struct snd_soc_component *component) { return 0; } -static int tda1997x_codec_remove(struct snd_soc_codec *codec) +static int tda1997x_codec_remove(struct snd_soc_component *component) { return 0; } -static struct snd_soc_codec_driver tda1997x_codec_driver = { +static struct snd_soc_component_driver tda1997x_codec_driver = { .probe = tda1997x_codec_probe, .remove = tda1997x_codec_remove, - .reg_word_size = sizeof(u16), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, }; static int tda1997x_probe(struct i2c_client *client, @@ -2737,7 +2740,7 @@ static int tda1997x_probe(struct i2c_client *client, else formats = SNDRV_PCM_FMTBIT_S16_LE; tda1997x_audio_dai.capture.formats = formats; - ret = snd_soc_register_codec(&state->client->dev, + ret = devm_snd_soc_register_component(&state->client->dev, &tda1997x_codec_driver, &tda1997x_audio_dai, 1); if (ret) { @@ -2782,7 +2785,6 @@ static int tda1997x_remove(struct i2c_client *client) struct tda1997x_platform_data *pdata = &state->pdata; if (pdata->audout_format) { - snd_soc_unregister_codec(&client->dev); mutex_destroy(&state->audio_lock); } -- 1.9.1

2 2

[alsa-devel] [PATCH v2] ALSA: core api: define offsets for TLV items
by Ranjani Sridharan 29 Mar '18

29 Mar '18

This patch defines accessor offsets for the type, length, min, mute and step items in TLV data. These will be used by drivers to extract the TLV data while loading topology. Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- V2: removed redundant redefinitions and updated prefix --- include/uapi/sound/tlv.h | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/include/uapi/sound/tlv.h b/include/uapi/sound/tlv.h index be5371f09a62..1fd786a5e57b 100644 --- a/include/uapi/sound/tlv.h +++ b/include/uapi/sound/tlv.h @@ -98,4 +98,12 @@ #define SNDRV_CTL_TLVD_DB_GAIN_MUTE -9999999 +/* Accessor offsets for TLV data items */ +#define SNDRV_CTL_TLV_OFFSET_TYPE 0 +#define SNDRV_CTL_TLV_OFFSET_LEN 1 + +/* Accessor offsets for min, mute and step items in dB scale type TLV */ +#define SNDRV_CTL_TLV_OFFSET_DB_MIN 2 +#define SNDRV_CTL_TLV_OFFSET_DB_MUTE_AND_STEP 3 + #endif -- 2.14.1

2 3

[alsa-devel] [PATCH -next] ASoC: amd: acp-da7219-max98357: Make symbol da7219_dai_clk static
by Wei Yongjun 29 Mar '18

29 Mar '18

Fixes the following sparse warning: sound/soc/amd/acp-da7219-max98357a.c:46:12: warning: symbol 'da7219_dai_clk' was not declared. Should it be static? Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/amd/acp-da7219-max98357a.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index b205c78..f41560e 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -43,7 +43,7 @@ #define DUAL_CHANNEL 2 static struct snd_soc_jack cz_jack; -struct clk *da7219_dai_clk; +static struct clk *da7219_dai_clk; static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) {

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Re: [alsa-devel] [PATCH v2] pcm_native: Remove VLA usage
by Takashi Sakamoto 29 Mar '18

29 Mar '18

Hi, On Mar 29 2018 07:24, Kyle Spiers wrote: > As part of the effort to remove VLAs from the kernel[1], this changes > the allocation of the rstamps array from being on the stack to being > kcalloc()ed. This also allows for the removal of the explicit zeroing > loop. > > Signed-off-by: Kyle Spiers <kyle(a)spiers.me> > --- > sound/core/pcm_native.c | 11 +++++++---- > 1 file changed, 7 insertions(+), 4 deletions(-) This bug was already fixed at a commit 5730f9f744cf (' ALSA: pcm: Remove VLA usage')[1][2]. [1] https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/commit/soun… [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-March/133268.html Thanks Takashi Sakamoto

1 0