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[PATCH] ASoC: sh: rz-ssi: Fix dereference of noderef expression warning
by Biju Das 16 Aug '21

16 Aug '21

Fix following sparse warning: sound/soc/sh/rz-ssi.c:156:15: sparse: warning: dereference of noderef expression Signed-off-by: Biju Das <biju.das.jz(a)bp.renesas.com> Reported-by: kernel test robot <lkp(a)intel.com> --- sound/soc/sh/rz-ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rz-ssi.c b/sound/soc/sh/rz-ssi.c index 2f9345415a15..7fa3737b90bf 100644 --- a/sound/soc/sh/rz-ssi.c +++ b/sound/soc/sh/rz-ssi.c @@ -148,7 +148,7 @@ static u32 rz_ssi_reg_readl(struct rz_ssi_priv *priv, uint reg) return readl(priv->base + reg); } -static void rz_ssi_reg_mask_setl(struct rz_ssi_priv __iomem *priv, uint reg, +static void rz_ssi_reg_mask_setl(struct rz_ssi_priv *priv, uint reg, u32 bclr, u32 bset) { u32 val; -- 2.17.1

2 1

[PATCH v3 0/8] ASoC: Intel: boards: use software node API
by Pierre-Louis Bossart 16 Aug '21

16 Aug '21

This is an update on an earlier contribution from Heikki Krogerus The function device_add_properties() is going to be removed. Replacing it with software node API equivalents. Thanks for Hans de Goede and Andy Shevchenko for their comments, suggestions and Reviewed-by tags on GitHub. The review thread can be found at https://github.com/thesofproject/linux/pull/3041) v3 changes: Fixed nit-picks from Andy: label, return value, missing commas/periods. Added Andy's Reviewed-by tag v2 changes: feedback from Andy and Hans Better error handling Codec reference is kept until the .remove callback Remove bus search to find device v1 changes from Heikki's patches: Avoid the use of devm_ routines for Baytrail machine drivers. Heikki Krogerus (1): ASoC: Intel: boards: use software node API in Atom boards Pierre-Louis Bossart (7): ASoC: Intel: boards: harden codec property handling ASoC: Intel: boards: handle errors with acpi_dev_get_first_match_dev() ASoC: Intel: boards: get codec device with ACPI instead of bus search ASoC: Intel: sof_sdw: pass card information to init/exit functions ASoC: Intel: sof_sdw_rt711*: keep codec device reference until remove ASoC: Intel: use software node API in SoundWire machines ASoC: Intel: remove device_properties for Atom boards sound/soc/intel/boards/bytcht_es8316.c | 31 ++++++++-- sound/soc/intel/boards/bytcr_rt5640.c | 60 +++++++++++++++----- sound/soc/intel/boards/bytcr_rt5651.c | 63 ++++++++++++++------- sound/soc/intel/boards/sof_sdw.c | 20 ++++--- sound/soc/intel/boards/sof_sdw_common.h | 37 +++++++----- sound/soc/intel/boards/sof_sdw_max98373.c | 3 +- sound/soc/intel/boards/sof_sdw_rt1308.c | 3 +- sound/soc/intel/boards/sof_sdw_rt1316.c | 3 +- sound/soc/intel/boards/sof_sdw_rt5682.c | 3 +- sound/soc/intel/boards/sof_sdw_rt700.c | 3 +- sound/soc/intel/boards/sof_sdw_rt711.c | 51 +++++++++-------- sound/soc/intel/boards/sof_sdw_rt711_sdca.c | 52 +++++++++-------- sound/soc/intel/boards/sof_sdw_rt715.c | 3 +- sound/soc/intel/boards/sof_sdw_rt715_sdca.c | 3 +- 14 files changed, 223 insertions(+), 112 deletions(-) -- 2.25.1

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[PATCH v2 00/14] ASoC: add Audio Graph Card2 driver
by Kuninori Morimoto 16 Aug '21

16 Aug '21

Hi Mark We already have Audio-Graph-Card which is Of-Graph base general sound card driver. Basically it supports basic CPU-Codec connection, and is also supporting DPCM connection. Because it was forcibly expanded to DPCM, DT parsing is very limited and very difficult to add new features on it, for example Multi CPU/Codec support, Codec2Codec support, etc. And some users want to use Audio-Graph-Card as basic connection and expand/customize its own settings on it. This patch adds more flexible new Audio-Graph-Card2 driver for it. Audio-Graph-Card and Audio-Graph-Card2 are similar, but don't have full compatibility. The reason why I need Audio-Graph-Card2 instead of update Audio-Graph-Card is that it is very difficult to keep compatibilty. Audio-Graph-Card2 supports Normal Connection, DPCM Connection, Multi CPU/Codec Connection, Codec2Codec Connection, and possible to Customizing. This patch-set adds Audio-Graph-Card2 driver and customized driver sample, and DT settings sample which can be used for testing. To enable testing/debuging, this patch-set also adds Test-Component driver. We already have Dummy Component and/or Dummy DAI on soc-utils, but 1) we can't use it from DT, 2) it do nothing. Added new Test-Component can be used from DT, and it can indicate called function name. We can use it to trace callback order / understanding ALSA SoC behavior, etc. Sample DT settings which can be used for testing is using Test-Component. This patch set includes sample DT for custome driver and/or audio-graph-card2. You can easily try to use/test it if you added below line to your DT file. It needs below CONFIGs. It will probe sample Sound Card which has Normal/DPCM/Multi/Codec2Codec connections. #include "../../../../../sound/soc/generic/audio-graph-card2-sample.dtsi" CONFIG_SND_AUDIO_GRAPH_CARD2 CONFIG_SND_SAMPLE_CUSTOM_CARD CONFIG_SND_TEST_COMPONENT There are some notes. Because Audio Graph Card2 is still under experimental stage, it will indicate such warning when probing, and the DT might be updated/exchanged. It can use Codec2Codec, but it will start automatically when probed, and can't stop it so far. It should be updated. 1 - 2 : add Test-Component driver 3 - 9 : add Audio-Graph-Card2 driver 10 : add Audio-Graph-Card2 base custome driver sample 11 - 14 : Audio-Graph-Card2 driver / custome driver DT sample Link: https://lore.kernel.org/r/87k0xszlep.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/871r8u4s6q.wl-kuninori.morimoto.gx@renesas.com v1 -> v2 - don't use "port" base for_each loop Kuninori Morimoto (14): 1 ASoC: test-component: add Test Component YAML bindings 2 ASoC: test-component: add Test Component for Sound debug/test 3 ASoC: simple-card-utils: add asoc_graph_is_ports0() 4 ASoC: simple-card-utils: add codec2codec support 5 ASoC: audio-graph-card2: add Audio Graph Card2 driver 6 ASoC: audio-graph-card2: add DPCM support 7 ASoC: audio-graph-card2: add Multi CPU/Codec support 8 ASoC: audio-graph-card2: add Codec2Codec support 9 ASoC: audio-graph-card2: add Yaml Document 10 ASoC: sample-custom-card: add Audio Graph Card2 custome sample 11 ASoC: audio-graph-card2-sample.dtsi: add Sample DT for Audio Graph Card2 12 ASoC: audio-graph-card2-sample.dtsi: add DPCM sample 13 ASoC: audio-graph-card2-sample.dtsi: add Multi CPU/Codec sample 14 ASoC: audio-graph-card2-sample.dtsi: add Codec2Codec sample. .../sound/audio-graph-card2-items.yaml | 80 ++ .../bindings/sound/audio-graph-card2.yaml | 51 + .../bindings/sound/test-component.yaml | 33 + include/sound/graph_card.h | 24 + include/sound/simple_card_utils.h | 4 + sound/soc/generic/Kconfig | 20 + sound/soc/generic/Makefile | 6 + .../soc/generic/audio-graph-card2-sample.dtsi | 140 ++ sound/soc/generic/audio-graph-card2.c | 1252 +++++++++++++++++ sound/soc/generic/sample-custom-card.c | 160 +++ sound/soc/generic/simple-card-utils.c | 46 +- sound/soc/generic/test-component.c | 659 +++++++++ 12 files changed, 2474 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card2-items.yaml create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card2.yaml create mode 100644 Documentation/devicetree/bindings/sound/test-component.yaml create mode 100644 sound/soc/generic/audio-graph-card2-sample.dtsi create mode 100644 sound/soc/generic/audio-graph-card2.c create mode 100644 sound/soc/generic/sample-custom-card.c create mode 100644 sound/soc/generic/test-component.c -- 2.25.1

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[PATCH] ALSA: hda/realtek: Enable 4-speaker output for Dell XPS 15 9510 laptop
by Kristin Paget 15 Aug '21

15 Aug '21

The 2021-model XPS 15 appears to use the same 4-speakers-on-ALC289 audio setup as the Precision models, so requires the same quirk to enable woofer output. Tested on my own 9510. Signed-off-by: Kristin Paget <kristin(a)tombom.co.uk> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 218d1aa0fd14..8b72064f8321 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8319,6 +8319,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0a2e, "Dell", ALC236_FIXUP_DELL_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1028, 0x0a30, "Dell", ALC236_FIXUP_DELL_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1028, 0x0a58, "Dell", ALC255_FIXUP_DELL_HEADSET_MIC), + SND_PCI_QUIRK(0x1028, 0x0a61, "Dell XPS 15 9510", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2),

2 1

[PATCH] ASoC: sh: rz-ssi: Improve error handling in rz_ssi_dma_request function
by Biju Das 15 Aug '21

15 Aug '21

The rz_ssi_dma_request function only checks the NULL condition for the value returned by the dma_request_chan function, but this function can also return an error. If it happens, the subsequent function call to rz_ssi_dma_slave_config can lead to a kernel crash. This patch fixes the issue by checking both error and NULL condition returned by dma_request_chan. Signed-off-by: Biju Das <biju.das.jz(a)bp.renesas.com> --- sound/soc/sh/rz-ssi.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rz-ssi.c b/sound/soc/sh/rz-ssi.c index ea8d33ede5d2..3867e2efd3e0 100644 --- a/sound/soc/sh/rz-ssi.c +++ b/sound/soc/sh/rz-ssi.c @@ -676,11 +676,19 @@ static void rz_ssi_release_dma_channels(struct rz_ssi_priv *ssi) static int rz_ssi_dma_request(struct rz_ssi_priv *ssi, struct device *dev) { ssi->playback.dma_ch = dma_request_chan(dev, "tx"); + if (IS_ERR_OR_NULL(ssi->playback.dma_ch)) + ssi->playback.dma_ch = NULL; + ssi->capture.dma_ch = dma_request_chan(dev, "rx"); + if (IS_ERR_OR_NULL(ssi->capture.dma_ch)) + ssi->capture.dma_ch = NULL; + if (!ssi->playback.dma_ch && !ssi->capture.dma_ch) { ssi->playback.dma_ch = dma_request_chan(dev, "rt"); - if (!ssi->playback.dma_ch) + if (IS_ERR_OR_NULL(ssi->playback.dma_ch)) { + ssi->playback.dma_ch = NULL; goto no_dma; + } ssi->dma_rt = true; } -- 2.17.1

1 1

[PATCH v2] ASoC: sh: rz-ssi: Improve error handling in rz_ssi_dma_request function
by Biju Das 14 Aug '21

14 Aug '21

The pointer returned by the dma_request_chan function is an error pointer for the error cases. But on the rz_ssi_dma_request function, it checks for NULL pointer instead. This patch fixes the issue by checking the error pointer instead of NULL pointer. Signed-off-by: Biju Das <biju.das.jz(a)bp.renesas.com> --- v1->v2: * Replaced IS_ERR_OR_NULL with IS_ERR for error pointer check, as the dma_request_chan function never returns NULL pointer for error case. --- sound/soc/sh/rz-ssi.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rz-ssi.c b/sound/soc/sh/rz-ssi.c index ea8d33ede5d2..2f9345415a15 100644 --- a/sound/soc/sh/rz-ssi.c +++ b/sound/soc/sh/rz-ssi.c @@ -676,11 +676,19 @@ static void rz_ssi_release_dma_channels(struct rz_ssi_priv *ssi) static int rz_ssi_dma_request(struct rz_ssi_priv *ssi, struct device *dev) { ssi->playback.dma_ch = dma_request_chan(dev, "tx"); + if (IS_ERR(ssi->playback.dma_ch)) + ssi->playback.dma_ch = NULL; + ssi->capture.dma_ch = dma_request_chan(dev, "rx"); + if (IS_ERR(ssi->capture.dma_ch)) + ssi->capture.dma_ch = NULL; + if (!ssi->playback.dma_ch && !ssi->capture.dma_ch) { ssi->playback.dma_ch = dma_request_chan(dev, "rt"); - if (!ssi->playback.dma_ch) + if (IS_ERR(ssi->playback.dma_ch)) { + ssi->playback.dma_ch = NULL; goto no_dma; + } ssi->dma_rt = true; } -- 2.17.1

1 0

unable to play audio in a application running as root.
by GitHub issues - opened 14 Aug '21

14 Aug '21

alsa-project/alsa-lib issue #170 was opened from PeaceNIRA: ALSA lib pcm_dmix.c:1075:(snd_pcm_dmix_open) unable to open slave what ive tried: unistall and reinstalling the app adding root to the audio group application: war thunder Issue URL : https://github.com/alsa-project/alsa-lib/issues/170 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v4 0/3] Add RZ/G2L Sound support
by Biju Das 14 Aug '21

14 Aug '21

This patch series aims to add ASoC support on RZ/G2L SoC's. It is based on the work done by Chris Brandt for RZ/A ASoC driver. v3->v4: * Updated the subject line as per style for the subsystem. * Removed select SND_SIMPLE_CARD from Kconfig * Added C++ comments for copyright and driver description. * Moved validation of channels in hw_params * removed asm issue reported by bot as well as Mark * replaced master/slave macros with provider/consumer macros * Improved locking and added more null pointer checks. v2->v3: * Fixed the dependency on KCONFIG * Merged the binding patch with dma feature added * Updated dt binding example with encoded #dma-cells value. * Improved Error handling in probe function * Removed the passing legacy channel configuration parameters from dmaengine_slave_config function * started using dma_request_chan instead of deprecated dma_request_slave_channel * Removed SoC dtsi and config patches from this series. Will send it later. v1->v2: * Rebased to latest rc kernel Biju Das (3): ASoC: sh: Add RZ/G2L SSIF-2 driver ASoC: dt-bindings: sound: renesas,rz-ssi: Update slave dma channel configuration parameters ASoC: sh: rz-ssi: Add SSI DMAC support .../bindings/sound/renesas,rz-ssi.yaml | 4 +- sound/soc/sh/Kconfig | 6 + sound/soc/sh/Makefile | 4 + sound/soc/sh/rz-ssi.c | 1069 +++++++++++++++++ 4 files changed, 1081 insertions(+), 2 deletions(-) create mode 100644 sound/soc/sh/rz-ssi.c -- 2.17.1

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[PATCH] ALSA: pcm: Add SNDRV_PCM_INFO_EXPLICIT_SYNC flag
by Takashi Iwai 14 Aug '21

14 Aug '21

ALSA PCM core has an optimized way to communicate with user-space for its control and status data via mmap on the supported architectures like x86. Depending on the situation, however, we'd rather want to enforce user-space notifying the applptr or hwptr change explicitly via ioctl. For example, the upcoming non-contig and non-coherent buffer handling would need an explicit sync, and this needs to catch the applptr and hwptr changes. Also, ASoC SOF driver will have the SPIB support that has the similar requirement for the explicit control of the applptr and hwptr. This patch adds the new PCM hardware info flag, SNDRV_PCM_INFO_EXPLICIT_SYNC. When this flag is set, PCM core disables both the control and the status mmap, which enforces user-space to update via SYNC_PTR ioctl. In that way, drivers can catch the applptr and hwptr update and apply the sync operation if needed. Link: https://lore.kernel.org/r/20210812113818.6479-1-tiwai@suse.de Link: https://lore.kernel.org/r/20210610205326.1176400-1-pierre-louis.bossart@lin… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/uapi/sound/asound.h | 1 + sound/core/pcm_native.c | 9 +++++++++ 2 files changed, 10 insertions(+) diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index d17c061950df..1d84ec9db93b 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -299,6 +299,7 @@ typedef int __bitwise snd_pcm_subformat_t; #define SNDRV_PCM_INFO_HAS_LINK_ABSOLUTE_ATIME 0x02000000 /* report absolute hardware link audio time, not reset on startup */ #define SNDRV_PCM_INFO_HAS_LINK_ESTIMATED_ATIME 0x04000000 /* report estimated link audio time */ #define SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME 0x08000000 /* report synchronized audio/system time */ +#define SNDRV_PCM_INFO_EXPLICIT_SYNC 0x10000000 /* needs explicit sync of pointers and data */ #define SNDRV_PCM_INFO_DRAIN_TRIGGER 0x40000000 /* internal kernel flag - trigger in drain */ #define SNDRV_PCM_INFO_FIFO_IN_FRAMES 0x80000000 /* internal kernel flag - FIFO size is in frames */ diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index dc9fa312fadd..d233cb3b41d8 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3621,6 +3621,12 @@ static int snd_pcm_mmap_control(struct snd_pcm_substream *substream, struct file static bool pcm_status_mmap_allowed(struct snd_pcm_file *pcm_file) { + /* If drivers require the explicit sync (typically for non-coherent + * pages), we have to disable the mmap of status and control data + * to enforce the control via SYNC_PTR ioctl. + */ + if (pcm_file->substream->runtime->hw.info & SNDRV_PCM_INFO_EXPLICIT_SYNC) + return false; /* See pcm_control_mmap_allowed() below. * Since older alsa-lib requires both status and control mmaps to be * coupled, we have to disable the status mmap for old alsa-lib, too. @@ -3635,6 +3641,9 @@ static bool pcm_control_mmap_allowed(struct snd_pcm_file *pcm_file) { if (pcm_file->no_compat_mmap) return false; + /* see above */ + if (pcm_file->substream->runtime->hw.info & SNDRV_PCM_INFO_EXPLICIT_SYNC) + return false; /* Disallow the control mmap when SYNC_APPLPTR flag is set; * it enforces the user-space to fall back to snd_pcm_sync_ptr(), * thus it effectively assures the manual update of appl_ptr. -- 2.26.2

1 1

[PATCH v2] usb-audio: Input source control - digidesign mbox
by Damien Zammit 14 Aug '21

14 Aug '21

This adds a second mixer control to Digidesign Mbox to select between Analog/SPDIF capture. Users will note that selecting the SPDIF input source automatically switches the clock mode to sync to SPDIF, which is a feature of the hardware. (You can change the clock source back to internal if you want to capture from spdif but not sync to its clock although this mode is probably not recommended). Unfortunately, starting the stream resets both modes to Internally clocked and Analog input mode. Signed-off-by: Damien Zammit <damien(a)zamaudio.com> Tested-by: Damien Zammit <damien(a)zamaudio.com> --- sound/usb/mixer_quirks.c | 265 ++++++++++++++++++++++++++++++++++++----------- 1 file changed, 203 insertions(+), 62 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 0a3cb8fd7..a66ce0375 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -594,85 +594,208 @@ static int snd_xonar_u1_controls_create(struct usb_mixer_interface *mixer) &snd_xonar_u1_output_switch, NULL); } +/* Digidesign Mbox 1 helper functions */ + +static int snd_mbox1_is_spdif_synced(struct snd_usb_audio *chip) +{ + unsigned char buff[3]; + int err; + int is_spdif_synced; + + /* Read clock source */ + err = snd_usb_ctl_msg(chip->dev, + usb_rcvctrlpipe(chip->dev, 0), 0x81, + USB_DIR_IN | + USB_TYPE_CLASS | + USB_RECIP_ENDPOINT, 0x100, 0x81, buff, 3); + if (err < 0) + return err; + + /* spdif sync: buff is all zeroes */ + is_spdif_synced = !(buff[0] | buff[1] | buff[2]); + return is_spdif_synced; +} + +static int snd_mbox1_set_clk_source(struct snd_usb_audio *chip, int rate_or_zero) +{ + /* 2 possibilities: Internal -> expects sample rate + * S/PDIF sync -> expects rate = 0 + */ + unsigned char buff[3]; + + buff[0] = (rate_or_zero >> 0) & 0xff; + buff[1] = (rate_or_zero >> 8) & 0xff; + buff[2] = (rate_or_zero >> 16) & 0xff; + + /* Set clock source */ + return snd_usb_ctl_msg(chip->dev, + usb_sndctrlpipe(chip->dev, 0), 0x1, + USB_TYPE_CLASS | + USB_RECIP_ENDPOINT, 0x100, 0x81, buff, 3); +} + +static int snd_mbox1_is_spdif_input(struct snd_usb_audio *chip) +{ + /* Hardware gives 2 possibilities: ANALOG Source -> 0x01 + * S/PDIF Source -> 0x02 + */ + int err; + unsigned char source[1]; + + /* Read input source */ + err = snd_usb_ctl_msg(chip->dev, + usb_rcvctrlpipe(chip->dev, 0), 0x81, + USB_DIR_IN | + USB_TYPE_CLASS | + USB_RECIP_INTERFACE, 0x00, 0x500, source, 1); + if (err < 0) + return err; + + return (source[0] == 2); +} + +static int snd_mbox1_set_input_source(struct snd_usb_audio *chip, int is_spdif) +{ + /* NB: Setting the input source to S/PDIF resets the clock source to S/PDIF + * Hardware expects 2 possibilities: ANALOG Source -> 0x01 + * S/PDIF Source -> 0x02 + */ + unsigned char buff[1]; + + buff[0] = (is_spdif & 1) + 1; + + /* Set input source */ + return snd_usb_ctl_msg(chip->dev, + usb_sndctrlpipe(chip->dev, 0), 0x1, + USB_TYPE_CLASS | + USB_RECIP_INTERFACE, 0x00, 0x500, buff, 1); +} + /* Digidesign Mbox 1 clock source switch (internal/spdif) */ -static int snd_mbox1_switch_get(struct snd_kcontrol *kctl, - struct snd_ctl_elem_value *ucontrol) +static int snd_mbox1_clk_switch_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) { + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kctl); + struct snd_usb_audio *chip = list->mixer->chip; + int err; + + err = snd_usb_lock_shutdown(chip); + if (err < 0) + goto err; + + err = snd_mbox1_is_spdif_synced(chip); + if (err < 0) + goto err; + + kctl->private_value = err; + err = 0; ucontrol->value.enumerated.item[0] = kctl->private_value; - return 0; +err: + snd_usb_unlock_shutdown(chip); + return err; } -static int snd_mbox1_switch_update(struct usb_mixer_interface *mixer, int val) +static int snd_mbox1_clk_switch_update(struct usb_mixer_interface *mixer, int is_spdif_sync) { struct snd_usb_audio *chip = mixer->chip; int err; - unsigned char buff[3]; err = snd_usb_lock_shutdown(chip); if (err < 0) return err; - /* Prepare for magic command to toggle clock source */ - err = snd_usb_ctl_msg(chip->dev, - usb_rcvctrlpipe(chip->dev, 0), 0x81, - USB_DIR_IN | - USB_TYPE_CLASS | - USB_RECIP_INTERFACE, 0x00, 0x500, buff, 1); + err = snd_mbox1_is_spdif_input(chip); if (err < 0) goto err; - err = snd_usb_ctl_msg(chip->dev, - usb_rcvctrlpipe(chip->dev, 0), 0x81, - USB_DIR_IN | - USB_TYPE_CLASS | - USB_RECIP_ENDPOINT, 0x100, 0x81, buff, 3); + + err = snd_mbox1_is_spdif_synced(chip); if (err < 0) goto err; - /* 2 possibilities: Internal -> send sample rate - * S/PDIF sync -> send zeroes - * NB: Sample rate locked to 48kHz on purpose to - * prevent user from resetting the sample rate - * while S/PDIF sync is enabled and confusing - * this configuration. - */ - if (val == 0) { - buff[0] = 0x80; - buff[1] = 0xbb; - buff[2] = 0x00; - } else { - buff[0] = buff[1] = buff[2] = 0x00; - } + /* FIXME: hardcoded sample rate */ + err = snd_mbox1_set_clk_source(chip, is_spdif_sync ? 0 : 48000); + if (err < 0) + goto err; - /* Send the magic command to toggle the clock source */ - err = snd_usb_ctl_msg(chip->dev, - usb_sndctrlpipe(chip->dev, 0), 0x1, - USB_TYPE_CLASS | - USB_RECIP_ENDPOINT, 0x100, 0x81, buff, 3); + err = snd_mbox1_is_spdif_synced(chip); +err: + snd_usb_unlock_shutdown(chip); + return err; +} + +static int snd_mbox1_clk_switch_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kctl); + struct usb_mixer_interface *mixer = list->mixer; + int err; + bool cur_val, new_val; + + cur_val = kctl->private_value; + new_val = ucontrol->value.enumerated.item[0]; + if (cur_val == new_val) + return 0; + + kctl->private_value = new_val; + err = snd_mbox1_clk_switch_update(mixer, new_val); + return err < 0 ? err : 1; +} + +static int snd_mbox1_clk_switch_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const texts[2] = { + "Internal", + "S/PDIF" + }; + + return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts); +} + +static int snd_mbox1_clk_switch_resume(struct usb_mixer_elem_list *list) +{ + return snd_mbox1_clk_switch_update(list->mixer, list->kctl->private_value); +} + +/* Digidesign Mbox 1 input source switch (analog/spdif) */ + +static int snd_mbox1_src_switch_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.enumerated.item[0] = kctl->private_value; + return 0; +} + +static int snd_mbox1_src_switch_update(struct usb_mixer_interface *mixer, int is_spdif_input) +{ + struct snd_usb_audio *chip = mixer->chip; + int err; + + err = snd_usb_lock_shutdown(chip); + if (err < 0) + return err; + + err = snd_mbox1_is_spdif_input(chip); if (err < 0) goto err; - err = snd_usb_ctl_msg(chip->dev, - usb_rcvctrlpipe(chip->dev, 0), 0x81, - USB_DIR_IN | - USB_TYPE_CLASS | - USB_RECIP_ENDPOINT, 0x100, 0x81, buff, 3); + + err = snd_mbox1_set_input_source(chip, is_spdif_input); if (err < 0) goto err; - err = snd_usb_ctl_msg(chip->dev, - usb_rcvctrlpipe(chip->dev, 0), 0x81, - USB_DIR_IN | - USB_TYPE_CLASS | - USB_RECIP_ENDPOINT, 0x100, 0x2, buff, 3); + + err = snd_mbox1_is_spdif_input(chip); if (err < 0) goto err; + err = snd_mbox1_is_spdif_synced(chip); err: snd_usb_unlock_shutdown(chip); return err; } -static int snd_mbox1_switch_put(struct snd_kcontrol *kctl, - struct snd_ctl_elem_value *ucontrol) +static int snd_mbox1_src_switch_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_elem_list *list = snd_kcontrol_chip(kctl); struct usb_mixer_interface *mixer = list->mixer; @@ -685,42 +808,60 @@ static int snd_mbox1_switch_put(struct snd_kcontrol *kctl, return 0; kctl->private_value = new_val; - err = snd_mbox1_switch_update(mixer, new_val); + err = snd_mbox1_src_switch_update(mixer, new_val); return err < 0 ? err : 1; } -static int snd_mbox1_switch_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) +static int snd_mbox1_src_switch_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) { static const char *const texts[2] = { - "Internal", + "Analog", "S/PDIF" }; return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts); } -static int snd_mbox1_switch_resume(struct usb_mixer_elem_list *list) +static int snd_mbox1_src_switch_resume(struct usb_mixer_elem_list *list) { - return snd_mbox1_switch_update(list->mixer, list->kctl->private_value); + return snd_mbox1_src_switch_update(list->mixer, list->kctl->private_value); } -static const struct snd_kcontrol_new snd_mbox1_switch = { +static const struct snd_kcontrol_new snd_mbox1_clk_switch = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Clock Source", .index = 0, .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, - .info = snd_mbox1_switch_info, - .get = snd_mbox1_switch_get, - .put = snd_mbox1_switch_put, + .info = snd_mbox1_clk_switch_info, + .get = snd_mbox1_clk_switch_get, + .put = snd_mbox1_clk_switch_put, + .private_value = 0 +}; + +static const struct snd_kcontrol_new snd_mbox1_src_switch = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Input Source", + .index = 1, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_mbox1_src_switch_info, + .get = snd_mbox1_src_switch_get, + .put = snd_mbox1_src_switch_put, .private_value = 0 }; -static int snd_mbox1_create_sync_switch(struct usb_mixer_interface *mixer) +static int snd_mbox1_controls_create(struct usb_mixer_interface *mixer) { - return add_single_ctl_with_resume(mixer, 0, - snd_mbox1_switch_resume, - &snd_mbox1_switch, NULL); + int err; + err = add_single_ctl_with_resume(mixer, 0, + snd_mbox1_clk_switch_resume, + &snd_mbox1_clk_switch, NULL); + if (err < 0) + return err; + + return add_single_ctl_with_resume(mixer, 1, + snd_mbox1_src_switch_resume, + &snd_mbox1_src_switch, NULL); } /* Native Instruments device quirks */ @@ -3029,7 +3170,7 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) break; case USB_ID(0x0dba, 0x1000): /* Digidesign Mbox 1 */ - err = snd_mbox1_create_sync_switch(mixer); + err = snd_mbox1_controls_create(mixer); break; case USB_ID(0x17cc, 0x1011): /* Traktor Audio 6 */ -- 2.13.1

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